diff --git a/Makefile.rules b/Makefile.rules
index 7b508e6ab2..d39640ee29 100644
--- a/Makefile.rules
+++ b/Makefile.rules
@@ -66,7 +66,7 @@ OPTIMIZE?=-O3
ifneq ($(findstring darwin,$(OSARCH)),)
ifeq ($(shell if test `/usr/bin/sw_vers -productVersion | cut -c4` -gt 5; then echo 6; else echo 0; fi),6)
- # Snow Leopard/Lion has an issue with this optimization flag on large files (like chan_sip)
+ # Snow Leopard/Lion has an issue with this optimization flag on large files
OPTIMIZE+=-fno-inline-functions
endif
endif
diff --git a/README-SERIOUSLY.bestpractices.md b/README-SERIOUSLY.bestpractices.md
index 4344c0e3ca..f021f9d7d8 100644
--- a/README-SERIOUSLY.bestpractices.md
+++ b/README-SERIOUSLY.bestpractices.md
@@ -52,28 +52,28 @@ request.
```INI
[incoming]
exten => _X.,1,Verbose(2,Incoming call to extension ${EXTEN})
-exten => _X.,n,Dial(SIP/${EXTEN})
+exten => _X.,n,Dial(PJSIP/${EXTEN})
exten => _X.,n,Hangup()
```
This dialplan may be utilized to accept calls to extensions, which then dial a
numbered device name configured in one of the channel configuration files (such
-as sip.conf, iax.conf, etc...) (see [Proper Device Naming] for more information
+as pjsip.conf, iax.conf, etc...) (see [Proper Device Naming] for more information
on why this approach is flawed).
The example we've given above looks harmless enough until you take into
consideration that several channel technologies accept characters that could
be utilized in a clever attack. For example, instead of just sending a request
to dial extension 500 (which in our example above would create the string
-SIP/500 and is then used by the Dial() application to place a call), someone
-could potentially send a string like "500&SIP/itsp/14165551212".
+PJSIP/500 and is then used by the Dial() application to place a call), someone
+could potentially send a string like "500&PJSIP/itsp/14165551212".
-The string "500&SIP/itsp/14165551212" would then be contained within the
+The string "500&PJSIP/itsp/14165551212" would then be contained within the
${EXTEN} channel variable, which is then utilized by the Dial() application in
our example, thereby giving you the dialplan line of:
```INI
-exten => _X.,n,Dial(SIP/500&SIP/itsp/14165551212)
+exten => _X.,n,Dial(PJSIP/500&PJSIP/itsp/14165551212)
```
Our example above has now provided someone with a method to place calls out of
@@ -98,7 +98,7 @@ to only accept three digit extensions, we could change our pattern match to
be:
```INI
-exten => _XXX,n,Dial(SIP/${EXTEN})
+exten => _XXX,n,Dial(PJSIP/${EXTEN})
```
In this way, we have minimized our impact because we're not allowing anything
@@ -124,7 +124,7 @@ we will accept to just numbers. Our example would then change to something like:
```INI
[incoming]
exten => _X.,1,Verbose(2,Incoming call to extension ${EXTEN})
-exten => _X.,n,Dial(SIP/${FILTER(0-9,${EXTEN})})
+exten => _X.,n,Dial(PJSIP/${FILTER(0-9,${EXTEN})})
exten => _X.,n,Hangup()
```
@@ -141,7 +141,7 @@ necessary, and to handle error checking in a separate location.
[incoming]
exten => _X.,1,Verbose(2,Incoming call to extension ${EXTEN})
exten => _X.,n,Set(SAFE_EXTEN=${FILTER(0-9,${EXTEN})})
-exten => _X.,n,Dial(SIP/${SAFE_EXTEN})
+exten => _X.,n,Dial(PJSIP/${SAFE_EXTEN})
exten => _X.,n,Hangup()
```
@@ -155,7 +155,7 @@ passed back by FILTER(), and to fail the call if things do not match.
exten => _X.,1,Verbose(2,Incoming call to extension ${EXTEN})
exten => _X.,n,Set(SAFE_EXTEN=${FILTER(0-9,${EXTEN})})
exten => _X.,n,GotoIf($[${EXTEN} != ${SAFE_EXTEN}]?error,1)
-exten => _X.,n,Dial(SIP/${SAFE_EXTEN})
+exten => _X.,n,Dial(PJSIP/${SAFE_EXTEN})
exten => _X.,n,Hangup()
exten => error,1,Verbose(2,Values of EXTEN and SAFE_EXTEN did not match.)
@@ -170,7 +170,7 @@ we're expecting to get a SIP URI for dialing.
```INI
[incoming]
exten => _[0-9a-zA-Z].,1,Verbose(2,Incoming call to extension ${EXTEN})
-exten => _[0-9a-zA-Z].,n,Dial(SIP/${FILTER(.@0-9a-zA-Z,${EXTEN})
+exten => _[0-9a-zA-Z].,n,Dial(PJSIP/${FILTER(.@0-9a-zA-Z,${EXTEN})
exten => _[0-9a-zA-Z].,n,Hangup()
```
@@ -201,13 +201,14 @@ It can also be a security hazard to name your devices with a number, as this can
open you up to brute force attacks. Many of the current exploits deal with
device configurations which utilize a number, and even worse, a password that
matches the devices name. For example, take a look at this poorly created device
-in sip.conf:
+in pjsip.conf:
```INI
[1000]
-type=friend
-context=international_dialing
-secret=1000
+type=auth
+auth_type=userpass
+password=1000
+username=1000
```
As implied by the context, we've permitted a device named 1000 with a password
@@ -223,9 +224,10 @@ Passwords). The following example would be more secure:
```INI
[0004f2040001]
-type=friend
-context=international_dialing
-secret=aE3%B8*$jk^G
+type=auth
+auth_type=userpass
+password=aE3%B8*$jk^G
+username=0004f2040001
```
Then in your dialplan, you would reference the device via the MAC address of the
@@ -323,7 +325,7 @@ the Originate manager command:
```
Action: Originate
-Channel: SIP/foo
+Channel: PJSIP/foo
Exten: s
Context: default
Priority: 1
@@ -340,7 +342,7 @@ circumvent these checks. For example, take the following dialplan:
```INI
exten => s,1,Verbose(Incoming call)
same => n,MixMonitor(foo.wav,,${EXEC_COMMAND})
-same => n,Dial(SIP/bar)
+same => n,Dial(PJSIP/bar)
same => n,Hangup()
```
diff --git a/addons/chan_ooh323.c b/addons/chan_ooh323.c
index 814022c260..4df1859e9b 100644
--- a/addons/chan_ooh323.c
+++ b/addons/chan_ooh323.c
@@ -3248,7 +3248,7 @@ static char *handle_cli_ooh323_show_peer(struct ast_cli_entry *e, int cmd, struc
if (peer->t38support == T38_DISABLED) {
ast_cli(a->fd, "%s\n", "disabled");
} else if (peer->t38support == T38_FAXGW) {
- ast_cli(a->fd, "%s\n", "faxgw/chan_sip compatible");
+ ast_cli(a->fd, "%s\n", "faxgw compatible");
}
if (peer->faxdetect == (FAXDETECT_CNG | FAXDETECT_T38)) {
ast_cli(a->fd,"%-20s%s\n", "FAX Detect:", "Yes");
@@ -3386,7 +3386,7 @@ static char *handle_cli_ooh323_show_user(struct ast_cli_entry *e, int cmd, struc
if (user->t38support == T38_DISABLED) {
ast_cli(a->fd, "%s\n", "disabled");
} else if (user->t38support == T38_FAXGW) {
- ast_cli(a->fd, "%s\n", "faxgw/chan_sip compatible");
+ ast_cli(a->fd, "%s\n", "faxgw compatible");
}
if (user->faxdetect == (FAXDETECT_CNG | FAXDETECT_T38)) {
ast_cli(a->fd,"%-20s%s\n", "FAX Detect:", "Yes");
@@ -3633,7 +3633,7 @@ static char *handle_cli_ooh323_show_config(struct ast_cli_entry *e, int cmd, str
if (gT38Support == T38_DISABLED) {
ast_cli(a->fd, "%s\n", "disabled");
} else if (gT38Support == T38_FAXGW) {
- ast_cli(a->fd, "%s\n", "faxgw/chan_sip compatible");
+ ast_cli(a->fd, "%s\n", "faxgw compatible");
}
if (gFAXdetect == (FAXDETECT_CNG | FAXDETECT_T38)) {
ast_cli(a->fd,"%-20s%s\n", "FAX Detect:", "Yes");
diff --git a/apps/app_dial.c b/apps/app_dial.c
index c3892254b8..bc6eee4584 100644
--- a/apps/app_dial.c
+++ b/apps/app_dial.c
@@ -401,8 +401,6 @@
to send no cause. See the causes.h file for the
full list of valid causes and names.
- NOTE: chan_sip does not support setting the cause on a CANCEL to anything
- other than ANSWERED_ELSEWHERE.
Default: Indicate ringing to the calling party, even if the called party isn't actually ringing. Pass no audio to the calling
diff --git a/apps/app_sendtext.c b/apps/app_sendtext.c
index ee42316965..b224c245d1 100644
--- a/apps/app_sendtext.c
+++ b/apps/app_sendtext.c
@@ -113,8 +113,7 @@
The text encoding and transmission method is completely at the
discretion of the channel driver. chan_pjsip will use in-dialog SIP MESSAGE
- messages always. chan_sip will use T.140 via RTP if a text media type was
- negotiated and in-dialog SIP MESSAGE messages otherwise.
Examples:
diff --git a/channels/Makefile b/channels/Makefile
index 4cbed9c985..2d243d2262 100644
--- a/channels/Makefile
+++ b/channels/Makefile
@@ -26,13 +26,11 @@ endif
$(call MOD_ADD_C,chan_iax2,$(wildcard iax2/*.c))
iax2/parser.o: _ASTCFLAGS+=$(call get_menuselect_cflags,MALLOC_DEBUG)
-$(call MOD_ADD_C,chan_sip,$(wildcard sip/*.c))
$(call MOD_ADD_C,chan_pjsip,$(wildcard pjsip/*.c))
$(call MOD_ADD_C,chan_dahdi,$(wildcard dahdi/*.c) sig_analog.c sig_pri.c sig_ss7.c)
chan_dahdi.o: _ASTCFLAGS+=$(call get_menuselect_cflags,LOTS_OF_SPANS)
chan_unistim.o: _ASTCFLAGS+=$(AST_NO_FORMAT_TRUNCATION)
chan_phone.o: _ASTCFLAGS+=$(AST_NO_FORMAT_TRUNCATION)
-chan_sip.o: _ASTCFLAGS+=$(AST_NO_FORMAT_TRUNCATION)
$(call MOD_ADD_C,console_video.c vgrabbers.c console_board.c)
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
deleted file mode 100644
index bf03c5ff26..0000000000
--- a/channels/chan_sip.c
+++ /dev/null
@@ -1,35946 +0,0 @@
-/*
- * Asterisk -- An open source telephony toolkit.
- *
- * Copyright (C) 1999 - 2012, Digium, Inc.
- *
- * Mark Spencer
- *
- * See http://www.asterisk.org for more information about
- * the Asterisk project. Please do not directly contact
- * any of the maintainers of this project for assistance;
- * the project provides a web site, mailing lists and IRC
- * channels for your use.
- *
- * This program is free software, distributed under the terms of
- * the GNU General Public License Version 2. See the LICENSE file
- * at the top of the source tree.
- */
-
-/*!
- * \file
- * \brief Implementation of Session Initiation Protocol
- *
- * \author Mark Spencer
- *
- * See Also:
- * \arg \ref AstCREDITS
- *
- * Implementation of RFC 3261 - without S/MIME, and experimental TCP and TLS support
- * Configuration file \ref sip.conf "Config_sip"
- *
- * ********** IMPORTANT *
- * \note TCP/TLS support is EXPERIMENTAL and WILL CHANGE. This applies to configuration
- * settings, dialplan commands and dialplans apps/functions
- * See \ref sip_tcp_tls
- *
- *
- * ******** General TODO:s
- * \todo Better support of forking
- * \todo VIA branch tag transaction checking
- * \todo Transaction support
- *
- * ******** Wishlist: Improvements
- * - Support of SIP domains for devices, so that we match on username\@domain in the From: header
- * - Connect registrations with a specific device on the incoming call. It's not done
- * automatically in Asterisk
- *
- * \ingroup channel_drivers
- *
- * \par Overview of the handling of SIP sessions
- * The SIP channel handles several types of SIP sessions, or dialogs,
- * not all of them being "telephone calls".
- * - Incoming calls that will be sent to the PBX core
- * - Outgoing calls, generated by the PBX
- * - SIP subscriptions and notifications of states and voicemail messages
- * - SIP registrations, both inbound and outbound
- * - SIP peer management (peerpoke, OPTIONS)
- * - SIP text messages
- *
- * In the SIP channel, there's a list of active SIP dialogs, which includes
- * all of these when they are active. "sip show channels" in the CLI will
- * show most of these, excluding subscriptions which are shown by
- * "sip show subscriptions"
- *
- * \par incoming packets
- * Incoming packets are received in the monitoring thread, then handled by
- * sipsock_read() for udp only. In tcp, packets are read by the tcp_helper thread.
- * sipsock_read() function parses the packet and matches an existing
- * dialog or starts a new SIP dialog.
- *
- * sipsock_read sends the packet to handle_incoming(), that parses a bit more.
- * If it is a response to an outbound request, the packet is sent to handle_response().
- * If it is a request, handle_incoming() sends it to one of a list of functions
- * depending on the request type - INVITE, OPTIONS, REFER, BYE, CANCEL etc
- * sipsock_read locks the ast_channel if it exists (an active call) and
- * unlocks it after we have processed the SIP message.
- *
- * A new INVITE is sent to handle_request_invite(), that will end up
- * starting a new channel in the PBX, the new channel after that executing
- * in a separate channel thread. This is an incoming "call".
- * When the call is answered, either by a bridged channel or the PBX itself
- * the sip_answer() function is called.
- *
- * The actual media - Video or Audio - is mostly handled by the RTP subsystem
- * in rtp.c
- *
- * \par Outbound calls
- * Outbound calls are set up by the PBX through the sip_request_call()
- * function. After that, they are activated by sip_call().
- *
- * \par Hanging up
- * The PBX issues a hangup on both incoming and outgoing calls through
- * the sip_hangup() function
- */
-
-/*! \li \ref chan_sip.c uses configuration files \ref sip.conf and \ref sip_notify.conf
- * \addtogroup configuration_file
- */
-
-/*! \page sip.conf sip.conf
- * \verbinclude sip.conf.sample
- */
-
-/*! \page sip_notify.conf sip_notify.conf
- * \verbinclude sip_notify.conf.sample
- */
-
-/*!
- * \page sip_tcp_tls SIP TCP and TLS support
- *
- * \par tcpfixes TCP implementation changes needed
- * \todo Fix TCP/TLS handling in dialplan, SRV records, transfers and much more
- * \todo Save TCP/TLS sessions in registry
- * If someone registers a SIPS uri, this forces us to set up a TLS connection back.
- * \todo Add TCP/TLS information to function SIPPEER and CHANNEL function
- * \todo If tcpenable=yes, we must open a TCP socket on the same address as the IP for UDP.
- * The tcpbindaddr config option should only be used to open ADDITIONAL ports
- * So we should propably go back to
- * bindaddr= the default address to bind to. If tcpenable=yes, then bind this to both udp and TCP
- * if tlsenable=yes, open TLS port (provided we also have cert)
- * tcpbindaddr = extra address for additional TCP connections
- * tlsbindaddr = extra address for additional TCP/TLS connections
- * udpbindaddr = extra address for additional UDP connections
- * These three options should take multiple IP/port pairs
- * Note: Since opening additional listen sockets is a *new* feature we do not have today
- * the XXXbindaddr options needs to be disabled until we have support for it
- *
- * \todo re-evaluate the transport= setting in sip.conf. This is right now not well
- * thought of. If a device in sip.conf contacts us via TCP, we should not switch transport,
- * even if udp is the configured first transport.
- *
- * \todo Be prepared for one outbound and another incoming socket per pvt. This applies
- * specially to communication with other peers (proxies).
- * \todo We need to test TCP sessions with SIP proxies and in regards
- * to the SIP outbound specs.
- * \todo ;transport=tls was deprecated in RFC3261 and should not be used at all. See section 26.2.2.
- *
- * \todo If the message is smaller than the given Content-length, the request should get a 400 Bad request
- * message. If it's a response, it should be dropped. (RFC 3261, Section 18.3)
- * \todo Since we have had multidomain support in Asterisk for quite a while, we need to support
- * multiple domains in our TLS implementation, meaning one socket and one cert per domain
- * \todo Selection of transport for a request needs to be done after we've parsed all route headers,
- * also considering outbound proxy options.
- * First request: Outboundproxy, routes, (reg contact or URI. If URI doesn't have port: DNS naptr, srv, AAA)
- * Intermediate requests: Outboundproxy(only when forced), routes, contact/uri
- * DNS naptr support is crucial. A SIP uri might lead to a TLS connection.
- * Also note that due to outbound proxy settings, a SIPS uri might have to be sent on UDP (not to recommend though)
- * \todo Default transports are set to UDP, which cause the wrong behaviour when contacting remote
- * devices directly from the dialplan. UDP is only a fallback if no other method works,
- * in order to be compatible with RFC2543 (SIP/1.0) devices. For transactions that exceed the
- * MTU (like INIVTE with video, audio and RTT) TCP should be preferred.
- *
- * When dialling unconfigured peers (with no port number) or devices in external domains
- * NAPTR records MUST be consulted to find configured transport. If they are not found,
- * SRV records for both TCP and UDP should be checked. If there's a record for TCP, use that.
- * If there's no record for TCP, then use UDP as a last resort. If there's no SRV records,
- * \note this only applies if there's no outbound proxy configured for the session. If an outbound
- * proxy is configured, these procedures might apply for locating the proxy and determining
- * the transport to use for communication with the proxy.
- * \par Other bugs to fix ----
- * __set_address_from_contact(const char *fullcontact, struct sockaddr_in *sin, int tcp)
- * - sets TLS port as default for all TCP connections, unless other port is given in contact.
- * parse_register_contact(struct sip_pvt *pvt, struct sip_peer *peer, struct sip_request *req)
- * - assumes that the contact the UA registers is using the same transport as the REGISTER request, which is
- * a bad guess.
- * - Does not save any information about TCP/TLS connected devices, which is a severe BUG, as discussed on the mailing list.
- * get_destination(struct sip_pvt *p, struct sip_request *oreq)
- * - Doesn't store the information that we got an incoming SIPS request in the channel, so that
- * we can require a secure signalling path OUT of Asterisk (on SIP or IAX2). Possibly, the call should
- * fail on in-secure signalling paths if there's no override in our configuration. At least, provide a
- * channel variable in the dialplan.
- * get_refer_info(struct sip_pvt *transferer, struct sip_request *outgoing_req)
- * - As above, if we have a SIPS: uri in the refer-to header
- * - Does not check transport in refer_to uri.
- */
-
-/*** MODULEINFO
- res_crypto
- res_http_websocket
- no
- deprecated
- chan_pjsip
- 17
- 21
- ***/
-
-/*! \page sip_session_timers SIP Session Timers in Asterisk Chan_sip
-
- The SIP Session-Timers is an extension of the SIP protocol that allows end-points and proxies to
- refresh a session periodically. The sessions are kept alive by sending a RE-INVITE or UPDATE
- request at a negotiated interval. If a session refresh fails then all the entities that support Session-
- Timers clear their internal session state. In addition, UAs generate a BYE request in order to clear
- the state in the proxies and the remote UA (this is done for the benefit of SIP entities in the path
- that do not support Session-Timers).
-
- The Session-Timers can be configured on a system-wide, per-user, or per-peer basis. The peruser/
- per-peer settings override the global settings. The following new parameters have been
- added to the sip.conf file.
- session-timers=["accept", "originate", "refuse"]
- session-expires=[integer]
- session-minse=[integer]
- session-refresher=["uas", "uac"]
-
- The session-timers parameter in sip.conf defines the mode of operation of SIP session-timers feature in
- Asterisk. The Asterisk can be configured in one of the following three modes:
-
- 1. Accept :: In the "accept" mode, the Asterisk server honors
- session-timers requests made by remote end-points. A remote
- end-point can request Asterisk to engage session-timers by either
- sending it an INVITE request with a "Supported: timer" header in
- it or by responding to Asterisk's INVITE with a 200 OK that
- contains Session-Expires: header in it. In this mode, the Asterisk
- server does not request session-timers from remote
- end-points. This is the default mode.
-
- 2. Originate :: In the "originate" mode, the Asterisk server
- requests the remote end-points to activate session-timers in
- addition to honoring such requests made by the remote
- end-points. In order to get as much protection as possible against
- hanging SIP channels due to network or end-point failures,
- Asterisk resends periodic re-INVITEs even if a remote end-point
- does not support the session-timers feature.
-
- 3. Refuse :: In the "refuse" mode, Asterisk acts as if it does not
- support session- timers for inbound or outbound requests. If a
- remote end-point requests session-timers in a dialog, then
- Asterisk ignores that request unless it's noted as a requirement
- (Require: header), in which case the INVITE is rejected with a 420
- Bad Extension response.
-
-*/
-
-#include "asterisk.h"
-
-#include
-#include
-#include
-
-#include "asterisk/network.h"
-#include "asterisk/paths.h" /* need ast_config_AST_SYSTEM_NAME */
-#include "asterisk/lock.h"
-#include "asterisk/config.h"
-#include "asterisk/module.h"
-#include "asterisk/pbx.h"
-#include "asterisk/sched.h"
-#include "asterisk/io.h"
-#include "asterisk/rtp_engine.h"
-#include "asterisk/udptl.h"
-#include "asterisk/acl.h"
-#include "asterisk/manager.h"
-#include "asterisk/callerid.h"
-#include "asterisk/cli.h"
-#include "asterisk/musiconhold.h"
-#include "asterisk/dsp.h"
-#include "asterisk/pickup.h"
-#include "asterisk/parking.h"
-#include "asterisk/srv.h"
-#include "asterisk/astdb.h"
-#include "asterisk/causes.h"
-#include "asterisk/utils.h"
-#include "asterisk/file.h"
-#include "asterisk/astobj2.h"
-#include "asterisk/dnsmgr.h"
-#include "asterisk/devicestate.h"
-#include "asterisk/netsock2.h"
-#include "asterisk/localtime.h"
-#include "asterisk/abstract_jb.h"
-#include "asterisk/threadstorage.h"
-#include "asterisk/translate.h"
-#include "asterisk/ast_version.h"
-#include "asterisk/aoc.h"
-#include "asterisk/message.h"
-#include "sip/include/sip.h"
-#include "sip/include/globals.h"
-#include "sip/include/config_parser.h"
-#include "sip/include/reqresp_parser.h"
-#include "sip/include/sip_utils.h"
-#include "asterisk/sdp_srtp.h"
-#include "asterisk/ccss.h"
-#include "asterisk/xml.h"
-#include "sip/include/dialog.h"
-#include "sip/include/dialplan_functions.h"
-#include "sip/include/security_events.h"
-#include "sip/include/route.h"
-#include "asterisk/sip_api.h"
-#include "asterisk/mwi.h"
-#include "asterisk/bridge.h"
-#include "asterisk/stasis.h"
-#include "asterisk/stasis_endpoints.h"
-#include "asterisk/stasis_system.h"
-#include "asterisk/stasis_channels.h"
-#include "asterisk/features_config.h"
-#include "asterisk/http_websocket.h"
-#include "asterisk/format_cache.h"
-#include "asterisk/linkedlists.h" /* for AST_LIST_NEXT */
-
-/*** DOCUMENTATION
-
-
- Change the dtmfmode for a SIP call.
-
-
-
-
-
-
-
-
- Changes the dtmfmode for a SIP call.
-
-
-
-
- Add a SIP header to the outbound call.
-
-
-
-
- Adds a header to a SIP call placed with DIAL.
- Remember to use the X-header if you are adding non-standard SIP
- headers, like X-Asterisk-Accountcode: . Use this with care.
- Adding the wrong headers may jeopardize the SIP dialog.
- Always returns 0 .
-
-
-
-
- Remove SIP headers previously added with SIPAddHeader
-
-
-
-
- SIPRemoveHeader() allows you to remove headers which were previously
- added with SIPAddHeader(). If no parameter is supplied, all previously added
- headers will be removed. If a parameter is supplied, only the matching headers
- will be removed.
-
- same => n,SIPAddHeader(P-Asserted-Identity: sip:foo@bar)
- same => n,SIPAddHeader(P-Preferred-Identity: sip:bar@foo)
-
-
- same => n,SIPRemoveHeader()
-
-
- same => n,SIPRemoveHeader(P-)
-
-
- same => n,SIPRemoveHeader(P-Asserted-Identity:)
-
- Always returns 0 .
-
-
-
-
- Send a custom INFO frame on specified channels.
-
-
-
-
- SIPSendCustomINFO() allows you to send a custom INFO message on all
- active SIP channels or on channels with the specified User Agent. This
- application is only available if TEST_FRAMEWORK is defined.
-
-
-
-
- Gets the specified SIP header from an incoming INVITE message.
-
-
-
- If not specified, defaults to 1 .
-
-
-
- Since there are several headers (such as Via) which can occur multiple
- times, SIP_HEADER takes an optional second argument to specify which header with
- that name to retrieve. Headers start at offset 1 .
- This function does not access headers from the REFER message if the call
- was transferred. To obtain the REFER headers, set the dialplan variable
- GET_TRANSFERRER_DATA to the prefix of the headers of the
- REFER message that you need to access; for example, X- to
- get all headers starting with X- . The variable must be set
- before a call to the application that starts the channel that may eventually
- transfer back into the dialplan, and must be inherited by that channel, so prefix
- it with the _ or __ when setting (or
- set it in the pre-dial handler executed on the new channel). To get all headers
- of the REFER message, set the value to * . Headers
- are returned in the form of a dialplan hash TRANSFER_DATA, and can be accessed
- with the functions HASHKEYS(TRANSFER_DATA) and, e. g.,
- HASH(TRANSFER_DATA,X-That-Special-Header) .
- Please also note that contents of the SDP (an attachment to the
- SIP request) can't be accessed with this function.
-
-
- [SIP_HEADERS]
-
-
-
-
- Gets the list of SIP header names from an incoming INVITE message.
-
-
-
- If specified, only the headers matching the given prefix are returned.
-
-
-
- Returns a comma-separated list of header names (without values) from the
- INVITE message that originated the current channel. Multiple headers with the
- same name are included in the list only once. The returned list can be iterated
- over using the functions POP() and SIP_HEADER().
- For example, ${SIP_HEADERS(Co)} might return
- Contact,Content-Length,Content-Type . As a practical example,
- you may use ${SIP_HEADERS(X-)} to enumerate optional extended
- headers.
- This function does not access headers from the incoming SIP REFER message;
- see the documentation of the function SIP_HEADER for how to access them.
- Please observe that contents of the SDP (an attachment to the
- SIP request) can't be accessed with this function.
-
-
- [SIP_HEADER]
- [POP]
-
-
-
-
- Gets SIP peer information.
-
-
-
-
-
- (default) The IP address.
-
-
- The port number.
-
-
- The configured mailbox.
-
-
- The configured context.
-
-
- The epoch time of the next expire.
-
-
- Is it dynamic? (yes/no).
-
-
- The configured Caller ID name.
-
-
- The configured Caller ID number.
-
-
- The configured Callgroup.
-
-
- The configured Pickupgroup.
-
-
- The configured Named Callgroup.
-
-
- The configured Named Pickupgroup.
-
-
- The configured codecs.
-
-
- Status (if qualify=yes).
-
-
- Extension activated at registration.
-
-
- Call limit (call-limit).
-
-
- Configured call level for signalling busy.
-
-
- Current amount of calls. Only available if call-limit is set.
-
-
- Default language for peer.
-
-
- Account code for this peer.
-
-
- Current user agent header used by peer.
-
-
- The value used for SIP loop prevention in outbound requests
-
-
- A channel variable configured with setvar for this peer.
-
-
- Preferred codec index number x (beginning with zero).
-
-
-
-
-
-
-
- Checks if domain is a local domain.
-
-
-
-
- This function checks if the domain in the argument is configured
- as a local SIP domain that this Asterisk server is configured to handle.
- Returns the domain name if it is locally handled, otherwise an empty string.
- Check the domain= configuration in sip.conf .
-
-
-
-
- List SIP peers (text format).
-
-
-
-
- Lists SIP peers in text format with details on current status.
- Peerlist will follow as separate events, followed by a final event called
- PeerlistComplete .
-
-
-
-
- show SIP peer (text format).
-
-
-
- The peer name you want to check.
-
-
-
- Show one SIP peer with details on current status.
-
-
-
-
- Qualify SIP peers.
-
-
-
- The peer name you want to qualify.
-
-
-
- Qualify a SIP peer.
-
-
- [SIPQualifyPeerDone]
-
-
-
-
- Show SIP registrations (text format).
-
-
-
-
- Lists all registration requests and status. Registrations will follow as separate
- events followed by a final event called RegistrationsComplete .
-
-
-
-
- Send a SIP notify.
-
-
-
- Peer to receive the notify.
-
-
- At least one variable pair must be specified.
- name =value
-
-
- When specified, SIP notity will be sent as a part of an existing dialog.
-
-
-
- Sends a SIP Notify event.
- All parameters for this event must be specified in the body of this request
- via multiple Variable: name=value sequences.
-
-
-
-
- Show the status of one or all of the sip peers.
-
-
-
- The peer name you want to check.
-
-
-
- Retrieves the status of one or all of the sip peers. If no peer name is specified, status
- for all of the sip peers will be retrieved.
-
-
-
- Specifying a prefix of sip: will send the
- message as a SIP MESSAGE request.
-
-
- The from parameter can be a configured peer name
- or in the form of "display-name" <URI>.
-
-
- Ignored
-
-
-
- Raised when SIPQualifyPeer has finished qualifying the specified peer.
-
-
- The name of the peer.
-
-
- This is only included if an ActionID Header was sent with the action request, in which case it will be that ActionID.
-
-
-
- [SIPqualifypeer]
-
-
-
-
-
- Raised when a SIP session times out.
-
-
- The source of the session timeout.
-
-
-
-
-
-
- ***/
-
-static int log_level = -1;
-
-static int min_expiry = DEFAULT_MIN_EXPIRY; /*!< Minimum accepted registration time */
-static int max_expiry = DEFAULT_MAX_EXPIRY; /*!< Maximum accepted registration time */
-static int default_expiry = DEFAULT_DEFAULT_EXPIRY;
-static int min_subexpiry = DEFAULT_MIN_EXPIRY; /*!< Minimum accepted subscription time */
-static int max_subexpiry = DEFAULT_MAX_EXPIRY; /*!< Maximum accepted subscription time */
-static int mwi_expiry = DEFAULT_MWI_EXPIRY;
-
-static int unauth_sessions = 0;
-static int authlimit = DEFAULT_AUTHLIMIT;
-static int authtimeout = DEFAULT_AUTHTIMEOUT;
-
-/*! \brief Global jitterbuffer configuration - by default, jb is disabled
- * \note Values shown here match the defaults shown in sip.conf.sample */
-static struct ast_jb_conf default_jbconf =
-{
- .flags = 0,
- .max_size = 200,
- .resync_threshold = 1000,
- .impl = "fixed",
- .target_extra = 40,
-};
-static struct ast_jb_conf global_jbconf; /*!< Global jitterbuffer configuration */
-
-static const char config[] = "sip.conf"; /*!< Main configuration file */
-static const char notify_config[] = "sip_notify.conf"; /*!< Configuration file for sending Notify with CLI commands to reconfigure or reboot phones */
-
-/*! \brief Readable descriptions of device states.
- * \note Should be aligned to above table as index */
-static const struct invstate2stringtable {
- const enum invitestates state;
- const char *desc;
-} invitestate2string[] = {
- {INV_NONE, "None" },
- {INV_CALLING, "Calling (Trying)"},
- {INV_PROCEEDING, "Proceeding "},
- {INV_EARLY_MEDIA, "Early media"},
- {INV_COMPLETED, "Completed (done)"},
- {INV_CONFIRMED, "Confirmed (up)"},
- {INV_TERMINATED, "Done"},
- {INV_CANCELLED, "Cancelled"}
-};
-
-/*! \brief Subscription types that we support. We support
- * - dialoginfo updates (really device status, not dialog info as was the original intent of the standard)
- * - SIMPLE presence used for device status
- * - Voicemail notification subscriptions
- */
-static const struct cfsubscription_types {
- enum subscriptiontype type;
- const char * const event;
- const char * const mediatype;
- const char * const text;
-} subscription_types[] = {
- { NONE, "-", "unknown", "unknown" },
- /* RFC 4235: SIP Dialog event package */
- { DIALOG_INFO_XML, "dialog", "application/dialog-info+xml", "dialog-info+xml" },
- { CPIM_PIDF_XML, "presence", "application/cpim-pidf+xml", "cpim-pidf+xml" }, /* RFC 3863 */
- { PIDF_XML, "presence", "application/pidf+xml", "pidf+xml" }, /* RFC 3863 */
- { XPIDF_XML, "presence", "application/xpidf+xml", "xpidf+xml" }, /* Pre-RFC 3863 with MS additions */
- { MWI_NOTIFICATION, "message-summary", "application/simple-message-summary", "mwi" } /* RFC 3842: Mailbox notification */
-};
-
-/*! \brief The core structure to setup dialogs. We parse incoming messages by using
- * structure and then route the messages according to the type.
- *
- * \note Note that sip_methods[i].id == i must hold or the code breaks
- */
-static const struct cfsip_methods {
- enum sipmethod id;
- int need_rtp; /*!< when this is the 'primary' use for a pvt structure, does it need RTP? */
- char * const text;
- enum can_create_dialog can_create;
-} sip_methods[] = {
- { SIP_UNKNOWN, RTP, "-UNKNOWN-",CAN_CREATE_DIALOG },
- { SIP_RESPONSE, NO_RTP, "SIP/2.0", CAN_NOT_CREATE_DIALOG },
- { SIP_REGISTER, NO_RTP, "REGISTER", CAN_CREATE_DIALOG },
- { SIP_OPTIONS, NO_RTP, "OPTIONS", CAN_CREATE_DIALOG },
- { SIP_NOTIFY, NO_RTP, "NOTIFY", CAN_CREATE_DIALOG },
- { SIP_INVITE, RTP, "INVITE", CAN_CREATE_DIALOG },
- { SIP_ACK, NO_RTP, "ACK", CAN_NOT_CREATE_DIALOG },
- { SIP_PRACK, NO_RTP, "PRACK", CAN_NOT_CREATE_DIALOG },
- { SIP_BYE, NO_RTP, "BYE", CAN_NOT_CREATE_DIALOG },
- { SIP_REFER, NO_RTP, "REFER", CAN_CREATE_DIALOG },
- { SIP_SUBSCRIBE, NO_RTP, "SUBSCRIBE",CAN_CREATE_DIALOG },
- { SIP_MESSAGE, NO_RTP, "MESSAGE", CAN_CREATE_DIALOG },
- { SIP_UPDATE, NO_RTP, "UPDATE", CAN_NOT_CREATE_DIALOG },
- { SIP_INFO, NO_RTP, "INFO", CAN_NOT_CREATE_DIALOG },
- { SIP_CANCEL, NO_RTP, "CANCEL", CAN_NOT_CREATE_DIALOG },
- { SIP_PUBLISH, NO_RTP, "PUBLISH", CAN_CREATE_DIALOG },
- { SIP_PING, NO_RTP, "PING", CAN_CREATE_DIALOG_UNSUPPORTED_METHOD }
-};
-
-/*! \brief Diversion header reasons
- *
- * The core defines a bunch of constants used to define
- * redirecting reasons. This provides a translation table
- * between those and the strings which may be present in
- * a SIP Diversion header
- */
-static const struct sip_reasons {
- enum AST_REDIRECTING_REASON code;
- const char *text;
-} sip_reason_table[] = {
- { AST_REDIRECTING_REASON_UNKNOWN, "unknown" },
- { AST_REDIRECTING_REASON_USER_BUSY, "user-busy" },
- { AST_REDIRECTING_REASON_NO_ANSWER, "no-answer" },
- { AST_REDIRECTING_REASON_UNAVAILABLE, "unavailable" },
- { AST_REDIRECTING_REASON_UNCONDITIONAL, "unconditional" },
- { AST_REDIRECTING_REASON_TIME_OF_DAY, "time-of-day" },
- { AST_REDIRECTING_REASON_DO_NOT_DISTURB, "do-not-disturb" },
- { AST_REDIRECTING_REASON_DEFLECTION, "deflection" },
- { AST_REDIRECTING_REASON_FOLLOW_ME, "follow-me" },
- { AST_REDIRECTING_REASON_OUT_OF_ORDER, "out-of-service" },
- { AST_REDIRECTING_REASON_AWAY, "away" },
- { AST_REDIRECTING_REASON_CALL_FWD_DTE, "cf_dte" }, /* Non-standard */
- { AST_REDIRECTING_REASON_SEND_TO_VM, "send_to_vm" }, /* Non-standard */
-};
-
-
-/*! \name DefaultSettings
- Default setttings are used as a channel setting and as a default when
- configuring devices
-*/
-static char default_language[MAX_LANGUAGE]; /*!< Default language setting for new channels */
-static char default_callerid[AST_MAX_EXTENSION]; /*!< Default caller ID for sip messages */
-static char default_mwi_from[80]; /*!< Default caller ID for MWI updates */
-static char default_fromdomain[AST_MAX_EXTENSION]; /*!< Default domain on outbound messages */
-static int default_fromdomainport; /*!< Default domain port on outbound messages */
-static char default_notifymime[AST_MAX_EXTENSION]; /*!< Default MIME media type for MWI notify messages */
-static char default_vmexten[AST_MAX_EXTENSION]; /*!< Default From Username on MWI updates */
-static int default_qualify; /*!< Default Qualify= setting */
-static int default_keepalive; /*!< Default keepalive= setting */
-static char default_mohinterpret[MAX_MUSICCLASS]; /*!< Global setting for moh class to use when put on hold */
-static char default_mohsuggest[MAX_MUSICCLASS]; /*!< Global setting for moh class to suggest when putting
- * a bridged channel on hold */
-static char default_parkinglot[AST_MAX_CONTEXT]; /*!< Parkinglot */
-static char default_engine[256]; /*!< Default RTP engine */
-static int default_maxcallbitrate; /*!< Maximum bitrate for call */
-static char default_zone[MAX_TONEZONE_COUNTRY]; /*!< Default tone zone for channels created from the SIP driver */
-static unsigned int default_transports; /*!< Default Transports (enum ast_transport) that are acceptable */
-static unsigned int default_primary_transport; /*!< Default primary Transport (enum ast_transport) for outbound connections to devices */
-
-static struct sip_settings sip_cfg; /*!< SIP configuration data.
- \note in the future we could have multiple of these (per domain, per device group etc) */
-
-/*!< use this macro when ast_uri_decode is dependent on pedantic checking to be on. */
-#define SIP_PEDANTIC_DECODE(str) \
- if (sip_cfg.pedanticsipchecking && !ast_strlen_zero(str)) { \
- ast_uri_decode(str, ast_uri_sip_user); \
- } \
-
-static unsigned int chan_idx; /*!< used in naming sip channel */
-static int global_match_auth_username; /*!< Match auth username if available instead of From: Default off. */
-
-static int global_relaxdtmf; /*!< Relax DTMF */
-static int global_prematuremediafilter; /*!< Enable/disable premature frames in a call (causing 183 early media) */
-static int global_rtptimeout; /*!< Time out call if no RTP */
-static int global_rtpholdtimeout; /*!< Time out call if no RTP during hold */
-static int global_rtpkeepalive; /*!< Send RTP keepalives */
-static int global_reg_timeout; /*!< Global time between attempts for outbound registrations */
-static int global_regattempts_max; /*!< Registration attempts before giving up */
-static int global_reg_retry_403; /*!< Treat 403 responses to registrations as 401 responses */
-static int global_shrinkcallerid; /*!< enable or disable shrinking of caller id */
-static int global_callcounter; /*!< Enable call counters for all devices. This is currently enabled by setting the peer
- * call-limit to INT_MAX. When we remove the call-limit from the code, we can make it
- * with just a boolean flag in the device structure */
-static unsigned int global_tos_sip; /*!< IP type of service for SIP packets */
-static unsigned int global_tos_audio; /*!< IP type of service for audio RTP packets */
-static unsigned int global_tos_video; /*!< IP type of service for video RTP packets */
-static unsigned int global_tos_text; /*!< IP type of service for text RTP packets */
-static unsigned int global_cos_sip; /*!< 802.1p class of service for SIP packets */
-static unsigned int global_cos_audio; /*!< 802.1p class of service for audio RTP packets */
-static unsigned int global_cos_video; /*!< 802.1p class of service for video RTP packets */
-static unsigned int global_cos_text; /*!< 802.1p class of service for text RTP packets */
-static unsigned int recordhistory; /*!< Record SIP history. Off by default */
-static unsigned int dumphistory; /*!< Dump history to verbose before destroying SIP dialog */
-static char global_useragent[AST_MAX_EXTENSION]; /*!< Useragent for the SIP channel */
-static char global_sdpsession[AST_MAX_EXTENSION]; /*!< SDP session name for the SIP channel */
-static char global_sdpowner[AST_MAX_EXTENSION]; /*!< SDP owner name for the SIP channel */
-static int global_authfailureevents; /*!< Whether we send authentication failure manager events or not. Default no. */
-static int global_t1; /*!< T1 time */
-static int global_t1min; /*!< T1 roundtrip time minimum */
-static int global_timer_b; /*!< Timer B - RFC 3261 Section 17.1.1.2 */
-static unsigned int global_autoframing; /*!< Turn autoframing on or off. */
-static int global_qualifyfreq; /*!< Qualify frequency */
-static int global_qualify_gap; /*!< Time between our group of peer pokes */
-static int global_qualify_peers; /*!< Number of peers to poke at a given time */
-
-static enum st_mode global_st_mode; /*!< Mode of operation for Session-Timers */
-static enum st_refresher_param global_st_refresher; /*!< Session-Timer refresher */
-static int global_min_se; /*!< Lowest threshold for session refresh interval */
-static int global_max_se; /*!< Highest threshold for session refresh interval */
-
-static int global_store_sip_cause; /*!< Whether the MASTER_CHANNEL(HASH(SIP_CAUSE,[chan_name])) var should be set */
-
-static int global_dynamic_exclude_static = 0; /*!< Exclude static peers from contact registrations */
-static unsigned char global_refer_addheaders; /*!< Add extra headers to outgoing REFER */
-
-/*!
- * We use libxml2 in order to parse XML that may appear in the body of a SIP message. Currently,
- * the only usage is for parsing PIDF bodies of incoming PUBLISH requests in the call-completion
- * event package. This variable is set at module load time and may be checked at runtime to determine
- * if XML parsing support was found.
- */
-static int can_parse_xml;
-
-/*! \name Object counters
- *
- * \bug These counters are not handled in a thread-safe way ast_atomic_fetchadd_int()
- * should be used to modify these values.
- *
- * @{
- */
-static int speerobjs = 0; /*!< Static peers */
-static int rpeerobjs = 0; /*!< Realtime peers */
-static int apeerobjs = 0; /*!< Autocreated peer objects */
-/*! @} */
-
-static struct ast_flags global_flags[3] = {{0}}; /*!< global SIP_ flags */
-static unsigned int global_t38_maxdatagram; /*!< global T.38 FaxMaxDatagram override */
-
-static struct stasis_subscription *network_change_sub; /*!< subscription id for network change events */
-static struct stasis_subscription *acl_change_sub; /*!< subscription id for named ACL system change events */
-static int network_change_sched_id = -1;
-
-static char used_context[AST_MAX_CONTEXT]; /*!< name of automatically created context for unloading */
-
-AST_MUTEX_DEFINE_STATIC(netlock);
-
-/*! \brief Protect the monitoring thread, so only one process can kill or start it, and not
- when it's doing something critical. */
-AST_MUTEX_DEFINE_STATIC(monlock);
-
-AST_MUTEX_DEFINE_STATIC(sip_reload_lock);
-
-/*! \brief This is the thread for the monitor which checks for input on the channels
- which are not currently in use. */
-static pthread_t monitor_thread = AST_PTHREADT_NULL;
-
-static int sip_reloading = FALSE; /*!< Flag for avoiding multiple reloads at the same time */
-static enum channelreloadreason sip_reloadreason; /*!< Reason for last reload/load of configuration */
-
-struct ast_sched_context *sched; /*!< The scheduling context */
-static struct io_context *io; /*!< The IO context */
-static int *sipsock_read_id; /*!< ID of IO entry for sipsock FD */
-struct sip_pkt;
-static AST_LIST_HEAD_STATIC(domain_list, domain); /*!< The SIP domain list */
-
-AST_LIST_HEAD_NOLOCK(sip_history_head, sip_history); /*!< history list, entry in sip_pvt */
-
-static enum sip_debug_e sipdebug;
-
-/*! \brief extra debugging for 'text' related events.
- * At the moment this is set together with sip_debug_console.
- * \note It should either go away or be implemented properly.
- */
-static int sipdebug_text;
-
-static const struct _map_x_s referstatusstrings[] = {
- { REFER_IDLE, "" },
- { REFER_SENT, "Request sent" },
- { REFER_RECEIVED, "Request received" },
- { REFER_CONFIRMED, "Confirmed" },
- { REFER_ACCEPTED, "Accepted" },
- { REFER_RINGING, "Target ringing" },
- { REFER_200OK, "Done" },
- { REFER_FAILED, "Failed" },
- { REFER_NOAUTH, "Failed - auth failure" },
- { -1, NULL} /* terminator */
-};
-
-/* --- Hash tables of various objects --------*/
-#ifdef LOW_MEMORY
-static const int HASH_PEER_SIZE = 17;
-static const int HASH_DIALOG_SIZE = 17;
-static const int HASH_REGISTRY_SIZE = 17;
-#else
-static const int HASH_PEER_SIZE = 563; /*!< Size of peer hash table, prime number preferred! */
-static const int HASH_DIALOG_SIZE = 563;
-static const int HASH_REGISTRY_SIZE = 563;
-#endif
-
-static const struct {
- enum ast_cc_service_type service;
- const char *service_string;
-} sip_cc_service_map [] = {
- [AST_CC_NONE] = { AST_CC_NONE, "" },
- [AST_CC_CCBS] = { AST_CC_CCBS, "BS" },
- [AST_CC_CCNR] = { AST_CC_CCNR, "NR" },
- [AST_CC_CCNL] = { AST_CC_CCNL, "NL" },
-};
-
-static const struct {
- enum sip_cc_notify_state state;
- const char *state_string;
-} sip_cc_notify_state_map [] = {
- [CC_QUEUED] = {CC_QUEUED, "cc-state: queued"},
- [CC_READY] = {CC_READY, "cc-state: ready"},
-};
-
-AST_LIST_HEAD_STATIC(epa_static_data_list, epa_backend);
-
-
-/*!
- * Used to create new entity IDs by ESCs.
- */
-static int esc_etag_counter;
-static const int DEFAULT_PUBLISH_EXPIRES = 3600;
-
-#ifdef HAVE_LIBXML2
-static int cc_esc_publish_handler(struct sip_pvt *pvt, struct sip_request *req, struct event_state_compositor *esc, struct sip_esc_entry *esc_entry);
-
-static const struct sip_esc_publish_callbacks cc_esc_publish_callbacks = {
- .initial_handler = cc_esc_publish_handler,
- .modify_handler = cc_esc_publish_handler,
-};
-#endif
-
-/*!
- * \brief The Event State Compositors
- *
- * An Event State Compositor is an entity which
- * accepts PUBLISH requests and acts appropriately
- * based on these requests.
- *
- * The actual event_state_compositor structure is simply
- * an ao2_container of sip_esc_entrys. When an incoming
- * PUBLISH is received, we can match the appropriate sip_esc_entry
- * using the entity ID of the incoming PUBLISH.
- */
-static struct event_state_compositor {
- enum subscriptiontype event;
- const char * name;
- const struct sip_esc_publish_callbacks *callbacks;
- struct ao2_container *compositor;
-} event_state_compositors [] = {
-#ifdef HAVE_LIBXML2
- {CALL_COMPLETION, "call-completion", &cc_esc_publish_callbacks},
-#endif
-};
-
-struct state_notify_data {
- int state;
- struct ao2_container *device_state_info;
- int presence_state;
- const char *presence_subtype;
- const char *presence_message;
-};
-
-
-static const int ESC_MAX_BUCKETS = 37;
-
-/*!
- * \details
- * Here we implement the container for dialogs which are in the
- * dialog_needdestroy state to iterate only through the dialogs
- * unlink them instead of iterate through all dialogs
- */
-struct ao2_container *dialogs_needdestroy;
-
-/*!
- * \details
- * Here we implement the container for dialogs which have rtp
- * traffic and rtptimeout, rtpholdtimeout or rtpkeepalive
- * set. We use this container instead the whole dialog list.
- */
-struct ao2_container *dialogs_rtpcheck;
-
-/*!
- * \details
- * Here we implement the container for dialogs (sip_pvt), defining
- * generic wrapper functions to ease the transition from the current
- * implementation (a single linked list) to a different container.
- * In addition to a reference to the container, we need functions to lock/unlock
- * the container and individual items, and functions to add/remove
- * references to the individual items.
- */
-static struct ao2_container *dialogs;
-#define sip_pvt_lock(x) ao2_lock(x)
-#define sip_pvt_trylock(x) ao2_trylock(x)
-#define sip_pvt_unlock(x) ao2_unlock(x)
-
-/*! \brief The table of TCP threads */
-static struct ao2_container *threadt;
-
-/*! \brief The peer list: Users, Peers and Friends */
-static struct ao2_container *peers;
-static struct ao2_container *peers_by_ip;
-
-/*! \brief A bogus peer, to be used when authentication should fail */
-static AO2_GLOBAL_OBJ_STATIC(g_bogus_peer);
-/*! \brief We can recognize the bogus peer by this invalid MD5 hash */
-#define BOGUS_PEER_MD5SECRET "intentionally_invalid_md5_string"
-
-/*! \brief The register list: Other SIP proxies we register with and receive calls from */
-static struct ao2_container *registry_list;
-
-/*! \brief The MWI subscription list */
-static struct ao2_container *subscription_mwi_list;
-
-static int temp_pvt_init(void *);
-static void temp_pvt_cleanup(void *);
-
-/*! \brief A per-thread temporary pvt structure */
-AST_THREADSTORAGE_CUSTOM(ts_temp_pvt, temp_pvt_init, temp_pvt_cleanup);
-
-/*! \brief A per-thread buffer for transport to string conversion */
-AST_THREADSTORAGE(sip_transport_str_buf);
-
-/*! \brief Size of the SIP transport buffer */
-#define SIP_TRANSPORT_STR_BUFSIZE 128
-
-/*! \brief Authentication container for realm authentication */
-static struct sip_auth_container *authl = NULL;
-/*! \brief Global authentication container protection while adjusting the references. */
-AST_MUTEX_DEFINE_STATIC(authl_lock);
-
-static struct ast_manager_event_blob *session_timeout_to_ami(struct stasis_message *msg);
-STASIS_MESSAGE_TYPE_DEFN_LOCAL(session_timeout_type,
- .to_ami = session_timeout_to_ami,
- );
-
-/* --- Sockets and networking --------------*/
-
-/*! \brief Main socket for UDP SIP communication.
- *
- * sipsock is shared between the SIP manager thread (which handles reload
- * requests), the udp io handler (sipsock_read()) and the user routines that
- * issue udp writes (using __sip_xmit()).
- * The socket is -1 only when opening fails (this is a permanent condition),
- * or when we are handling a reload() that changes its address (this is
- * a transient situation during which we might have a harmless race, see
- * below). Because the conditions for the race to be possible are extremely
- * rare, we don't want to pay the cost of locking on every I/O.
- * Rather, we remember that when the race may occur, communication is
- * bound to fail anyways, so we just live with this event and let
- * the protocol handle this above us.
- */
-static int sipsock = -1;
-
-struct ast_sockaddr bindaddr; /*!< UDP: The address we bind to */
-
-/*! \brief our (internal) default address/port to put in SIP/SDP messages
- * internip is initialized picking a suitable address from one of the
- * interfaces, and the same port number we bind to. It is used as the
- * default address/port in SIP messages, and as the default address
- * (but not port) in SDP messages.
- */
-static struct ast_sockaddr internip;
-
-/*! \brief our external IP address/port for SIP sessions.
- * externaddr.sin_addr is only set when we know we might be behind
- * a NAT, and this is done using a variety of (mutually exclusive)
- * ways from the config file:
- *
- * + with "externaddr = host[:port]" we specify the address/port explicitly.
- * The address is looked up only once when (re)loading the config file;
- *
- * + with "externhost = host[:port]" we do a similar thing, but the
- * hostname is stored in externhost, and the hostname->IP mapping
- * is refreshed every 'externrefresh' seconds;
- *
- * Other variables (externhost, externexpire, externrefresh) are used
- * to support the above functions.
- */
-static struct ast_sockaddr externaddr; /*!< External IP address if we are behind NAT */
-static struct ast_sockaddr media_address; /*!< External RTP IP address if we are behind NAT */
-static struct ast_sockaddr rtpbindaddr; /*!< RTP: The address we bind to */
-
-static char externhost[MAXHOSTNAMELEN]; /*!< External host name */
-static time_t externexpire; /*!< Expiration counter for re-resolving external host name in dynamic DNS */
-static int externrefresh = 10; /*!< Refresh timer for DNS-based external address (dyndns) */
-static uint16_t externtcpport; /*!< external tcp port */
-static uint16_t externtlsport; /*!< external tls port */
-
-/*! \brief List of local networks
- * We store "localnet" addresses from the config file into an access list,
- * marked as 'DENY', so the call to ast_apply_ha() will return
- * AST_SENSE_DENY for 'local' addresses, and AST_SENSE_ALLOW for 'non local'
- * (i.e. presumably public) addresses.
- */
-static struct ast_ha *localaddr; /*!< List of local networks, on the same side of NAT as this Asterisk */
-
-static int ourport_tcp; /*!< The port used for TCP connections */
-static int ourport_tls; /*!< The port used for TCP/TLS connections */
-static struct ast_sockaddr debugaddr;
-
-static struct ast_config *notify_types = NULL; /*!< The list of manual NOTIFY types we know how to send */
-
-/*! some list management macros. */
-
-#define UNLINK(element, head, prev) do { \
- if (prev) \
- (prev)->next = (element)->next; \
- else \
- (head) = (element)->next; \
- } while (0)
-
-struct ao2_container *sip_monitor_instances;
-
-struct show_peers_context;
-
-/*---------------------------- Forward declarations of functions in chan_sip.c */
-/* Note: This is added to help splitting up chan_sip.c into several files
- in coming releases. */
-
-/*--- PBX interface functions */
-static struct ast_channel *sip_request_call(const char *type, struct ast_format_cap *cap, const struct ast_assigned_ids *assignedids, const struct ast_channel *requestor, const char *dest, int *cause);
-static int sip_devicestate(const char *data);
-static int sip_sendtext(struct ast_channel *ast, const char *text);
-static int sip_call(struct ast_channel *ast, const char *dest, int timeout);
-static int sip_sendhtml(struct ast_channel *chan, int subclass, const char *data, int datalen);
-static int sip_hangup(struct ast_channel *ast);
-static int sip_answer(struct ast_channel *ast);
-static struct ast_frame *sip_read(struct ast_channel *ast);
-static int sip_write(struct ast_channel *ast, struct ast_frame *frame);
-static int sip_indicate(struct ast_channel *ast, int condition, const void *data, size_t datalen);
-static int sip_transfer(struct ast_channel *ast, const char *dest);
-static int sip_fixup(struct ast_channel *oldchan, struct ast_channel *newchan);
-static int sip_senddigit_begin(struct ast_channel *ast, char digit);
-static int sip_senddigit_end(struct ast_channel *ast, char digit, unsigned int duration);
-static int sip_setoption(struct ast_channel *chan, int option, void *data, int datalen);
-static int sip_queryoption(struct ast_channel *chan, int option, void *data, int *datalen);
-static const char *sip_get_callid(struct ast_channel *chan);
-
-static int handle_request_do(struct sip_request *req, struct ast_sockaddr *addr);
-static int sip_standard_port(enum ast_transport type, int port);
-static int sip_prepare_socket(struct sip_pvt *p);
-static int get_address_family_filter(unsigned int transport);
-
-/*--- Transmitting responses and requests */
-static int sipsock_read(int *id, int fd, short events, void *ignore);
-static int __sip_xmit(struct sip_pvt *p, struct ast_str *data);
-static int __sip_reliable_xmit(struct sip_pvt *p, uint32_t seqno, int resp, struct ast_str *data, int fatal, int sipmethod);
-static void add_cc_call_info_to_response(struct sip_pvt *p, struct sip_request *resp);
-static int __transmit_response(struct sip_pvt *p, const char *msg, const struct sip_request *req, enum xmittype reliable);
-static int retrans_pkt(const void *data);
-static int transmit_response_using_temp(ast_string_field callid, struct ast_sockaddr *addr, int useglobal_nat, const int intended_method, const struct sip_request *req, const char *msg);
-static int transmit_response(struct sip_pvt *p, const char *msg, const struct sip_request *req);
-static int transmit_response_reliable(struct sip_pvt *p, const char *msg, const struct sip_request *req);
-static int transmit_response_with_date(struct sip_pvt *p, const char *msg, const struct sip_request *req);
-static int transmit_response_with_sdp(struct sip_pvt *p, const char *msg, const struct sip_request *req, enum xmittype reliable, int oldsdp, int rpid);
-static int transmit_response_with_unsupported(struct sip_pvt *p, const char *msg, const struct sip_request *req, const char *unsupported);
-static int transmit_response_with_auth(struct sip_pvt *p, const char *msg, const struct sip_request *req, const char *rand, enum xmittype reliable, const char *header, int stale);
-static int transmit_provisional_response(struct sip_pvt *p, const char *msg, const struct sip_request *req, int with_sdp);
-static int transmit_response_with_allow(struct sip_pvt *p, const char *msg, const struct sip_request *req, enum xmittype reliable);
-static void transmit_fake_auth_response(struct sip_pvt *p, struct sip_request *req, enum xmittype reliable);
-static int transmit_request(struct sip_pvt *p, int sipmethod, uint32_t seqno, enum xmittype reliable, int newbranch);
-static int transmit_request_with_auth(struct sip_pvt *p, int sipmethod, uint32_t seqno, enum xmittype reliable, int newbranch);
-static int transmit_publish(struct sip_epa_entry *epa_entry, enum sip_publish_type publish_type, const char * const explicit_uri);
-static int transmit_invite(struct sip_pvt *p, int sipmethod, int sdp, int init, const char * const explicit_uri);
-static int transmit_reinvite_with_sdp(struct sip_pvt *p, int t38version, int oldsdp);
-static int transmit_info_with_aoc(struct sip_pvt *p, struct ast_aoc_decoded *decoded);
-static int transmit_info_with_digit(struct sip_pvt *p, const char digit, unsigned int duration);
-static int transmit_info_with_vidupdate(struct sip_pvt *p);
-static int transmit_message(struct sip_pvt *p, int init, int auth);
-static int transmit_refer(struct sip_pvt *p, const char *dest);
-static int transmit_notify_with_mwi(struct sip_pvt *p, int newmsgs, int oldmsgs, const char *vmexten);
-static int transmit_notify_with_sipfrag(struct sip_pvt *p, int cseq, char *message, int terminate);
-static int transmit_cc_notify(struct ast_cc_agent *agent, struct sip_pvt *subscription, enum sip_cc_notify_state state);
-static int transmit_register(struct sip_registry *r, int sipmethod, const char *auth, const char *authheader);
-static int send_response(struct sip_pvt *p, struct sip_request *req, enum xmittype reliable, uint32_t seqno);
-static int send_request(struct sip_pvt *p, struct sip_request *req, enum xmittype reliable, uint32_t seqno);
-static void copy_request(struct sip_request *dst, const struct sip_request *src);
-static void receive_message(struct sip_pvt *p, struct sip_request *req, struct ast_sockaddr *addr, const char *e);
-static void parse_moved_contact(struct sip_pvt *p, struct sip_request *req, char **name, char **number, int set_call_forward);
-static int sip_send_mwi_to_peer(struct sip_peer *peer, int cache_only);
-
-/* Misc dialog routines */
-static int __sip_autodestruct(const void *data);
-static int update_call_counter(struct sip_pvt *fup, int event);
-static int auto_congest(const void *arg);
-static struct sip_pvt *__find_call(struct sip_request *req, struct ast_sockaddr *addr, const int intended_method,
- const char *file, int line, const char *func);
-#define find_call(req, addr, intended_method) \
- __find_call(req, addr, intended_method, __FILE__, __LINE__, __PRETTY_FUNCTION__)
-
-static void build_route(struct sip_pvt *p, struct sip_request *req, int backwards, int resp);
-static int build_path(struct sip_pvt *p, struct sip_peer *peer, struct sip_request *req, const char *pathbuf);
-static enum check_auth_result register_verify(struct sip_pvt *p, struct ast_sockaddr *addr,
- struct sip_request *req, const char *uri);
-static int get_sip_pvt_from_replaces(const char *callid, const char *totag, const char *fromtag,
- struct sip_pvt **out_pvt, struct ast_channel **out_chan);
-static void check_pendings(struct sip_pvt *p);
-static void sip_set_owner(struct sip_pvt *p, struct ast_channel *chan);
-
-static void *sip_pickup_thread(void *stuff);
-static int sip_pickup(struct ast_channel *chan);
-
-static int sip_sipredirect(struct sip_pvt *p, const char *dest);
-static int is_method_allowed(unsigned int *allowed_methods, enum sipmethod method);
-
-/*--- Codec handling / SDP */
-static void try_suggested_sip_codec(struct sip_pvt *p);
-static const char *get_sdp_iterate(int* start, struct sip_request *req, const char *name);
-static char get_sdp_line(int *start, int stop, struct sip_request *req, const char **value);
-static int find_sdp(struct sip_request *req);
-static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action, int is_offer);
-static int process_sdp_o(const char *o, struct sip_pvt *p);
-static int process_sdp_c(const char *c, struct ast_sockaddr *addr);
-static int process_sdp_a_sendonly(const char *a, int *sendonly);
-static int process_sdp_a_ice(const char *a, struct sip_pvt *p, struct ast_rtp_instance *instance, int rtcp_mux);
-static int process_sdp_a_rtcp_mux(const char *a, struct sip_pvt *p, int *requested);
-static int process_sdp_a_dtls(const char *a, struct sip_pvt *p, struct ast_rtp_instance *instance);
-static int process_sdp_a_audio(const char *a, struct sip_pvt *p, struct ast_rtp_codecs *newaudiortp, int *last_rtpmap_codec);
-static int process_sdp_a_video(const char *a, struct sip_pvt *p, struct ast_rtp_codecs *newvideortp, int *last_rtpmap_codec);
-static int process_sdp_a_text(const char *a, struct sip_pvt *p, struct ast_rtp_codecs *newtextrtp, char *red_fmtp, int *red_num_gen, int *red_data_pt, int *last_rtpmap_codec);
-static int process_sdp_a_image(const char *a, struct sip_pvt *p);
-static void add_ice_to_sdp(struct ast_rtp_instance *instance, struct ast_str **a_buf);
-static void add_dtls_to_sdp(struct ast_rtp_instance *instance, struct ast_str **a_buf);
-static void start_ice(struct ast_rtp_instance *instance, int offer);
-static void add_codec_to_sdp(const struct sip_pvt *p, struct ast_format *codec,
- struct ast_str **m_buf, struct ast_str **a_buf,
- int debug, int *min_packet_size, int *max_packet_size);
-static void add_noncodec_to_sdp(const struct sip_pvt *p, int format,
- struct ast_str **m_buf, struct ast_str **a_buf,
- int debug);
-static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int oldsdp, int add_audio, int add_t38);
-static void do_setnat(struct sip_pvt *p);
-static void stop_media_flows(struct sip_pvt *p);
-
-/*--- Authentication stuff */
-static int reply_digest(struct sip_pvt *p, struct sip_request *req, char *header, int sipmethod, char *digest, int digest_len);
-static int build_reply_digest(struct sip_pvt *p, int method, char *digest, int digest_len);
-static enum check_auth_result check_auth(struct sip_pvt *p, struct sip_request *req, const char *username,
- const char *secret, const char *md5secret, int sipmethod,
- const char *uri, enum xmittype reliable);
-static enum check_auth_result check_user_full(struct sip_pvt *p, struct sip_request *req,
- int sipmethod, const char *uri, enum xmittype reliable,
- struct ast_sockaddr *addr, struct sip_peer **authpeer);
-static int check_user(struct sip_pvt *p, struct sip_request *req, int sipmethod, const char *uri, enum xmittype reliable, struct ast_sockaddr *addr);
-
-/*--- Domain handling */
-static int check_sip_domain(const char *domain, char *context, size_t len); /* Check if domain is one of our local domains */
-static int add_sip_domain(const char *domain, const enum domain_mode mode, const char *context);
-static void clear_sip_domains(void);
-
-/*--- SIP realm authentication */
-static void add_realm_authentication(struct sip_auth_container **credentials, const char *configuration, int lineno);
-static struct sip_auth *find_realm_authentication(struct sip_auth_container *credentials, const char *realm);
-
-/*--- Misc functions */
-static int check_rtp_timeout(struct sip_pvt *dialog, time_t t);
-static int reload_config(enum channelreloadreason reason);
-static void add_diversion(struct sip_request *req, struct sip_pvt *pvt);
-static int expire_register(const void *data);
-static void *do_monitor(void *data);
-static int restart_monitor(void);
-static void peer_mailboxes_to_str(struct ast_str **mailbox_str, struct sip_peer *peer);
-static struct ast_variable *copy_vars(struct ast_variable *src);
-static int dialog_find_multiple(void *obj, void *arg, int flags);
-static struct ast_channel *sip_pvt_lock_full(struct sip_pvt *pvt);
-/* static int sip_addrcmp(char *name, struct sockaddr_in *sin); Support for peer matching */
-static int sip_refer_alloc(struct sip_pvt *p);
-static void sip_refer_destroy(struct sip_pvt *p);
-static int sip_notify_alloc(struct sip_pvt *p);
-static int do_magic_pickup(struct ast_channel *channel, const char *extension, const char *context);
-static void set_peer_nat(const struct sip_pvt *p, struct sip_peer *peer);
-static void check_for_nat(const struct ast_sockaddr *them, struct sip_pvt *p);
-
-/*--- Device monitoring and Device/extension state/event handling */
-static int extensionstate_update(const char *context, const char *exten, struct state_notify_data *data, struct sip_pvt *p, int force);
-static int cb_extensionstate(const char *context, const char *exten, struct ast_state_cb_info *info, void *data);
-static int sip_poke_noanswer(const void *data);
-static int sip_poke_peer(struct sip_peer *peer, int force);
-static void sip_poke_all_peers(void);
-static void sip_peer_hold(struct sip_pvt *p, int hold);
-static void mwi_event_cb(void *, struct stasis_subscription *, struct stasis_message *);
-static void network_change_stasis_cb(void *data, struct stasis_subscription *sub, struct stasis_message *message);
-static void acl_change_stasis_cb(void *data, struct stasis_subscription *sub, struct stasis_message *message);
-static void sip_keepalive_all_peers(void);
-#define peer_in_destruction(peer) (ao2_ref(peer, 0) == 0)
-
-/*--- Applications, functions, CLI and manager command helpers */
-static const char *sip_nat_mode(const struct sip_pvt *p);
-static char *sip_show_inuse(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
-static char *transfermode2str(enum transfermodes mode) attribute_const;
-static int peer_status(struct sip_peer *peer, char *status, int statuslen);
-static char *sip_show_sched(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
-static char * _sip_show_peers(int fd, int *total, struct mansession *s, const struct message *m, int argc, const char *argv[]);
-static struct sip_peer *_sip_show_peers_one(int fd, struct mansession *s, struct show_peers_context *cont, struct sip_peer *peer);
-static char *sip_show_peers(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
-static char *sip_show_objects(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
-static void print_group(int fd, ast_group_t group, int crlf);
-static void print_named_groups(int fd, struct ast_namedgroups *groups, int crlf);
-static const char *dtmfmode2str(int mode) attribute_const;
-static int str2dtmfmode(const char *str) attribute_unused;
-static const char *insecure2str(int mode) attribute_const;
-static const char *allowoverlap2str(int mode) attribute_const;
-static void cleanup_stale_contexts(char *new, char *old);
-static const char *domain_mode_to_text(const enum domain_mode mode);
-static char *sip_show_domains(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
-static char *_sip_show_peer(int type, int fd, struct mansession *s, const struct message *m, int argc, const char *argv[]);
-static char *sip_show_peer(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
-static char *_sip_qualify_peer(int type, int fd, struct mansession *s, const struct message *m, int argc, const char *argv[]);
-static char *sip_qualify_peer(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
-static char *sip_show_registry(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
-static char *sip_unregister(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
-static char *sip_show_settings(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
-static char *sip_show_mwi(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
-static const char *subscription_type2str(enum subscriptiontype subtype) attribute_pure;
-static const struct cfsubscription_types *find_subscription_type(enum subscriptiontype subtype);
-static char *complete_sip_peer(const char *word, int state, int flags2);
-static char *complete_sip_registered_peer(const char *word, int state, int flags2);
-static char *complete_sip_show_history(const char *line, const char *word, int pos, int state);
-static char *complete_sip_show_peer(const char *line, const char *word, int pos, int state);
-static char *complete_sip_unregister(const char *line, const char *word, int pos, int state);
-static char *complete_sip_notify(const char *line, const char *word, int pos, int state);
-static char *sip_show_channel(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
-static char *sip_show_channelstats(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
-static char *sip_show_history(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
-static char *sip_do_debug_ip(int fd, const char *arg);
-static char *sip_do_debug_peer(int fd, const char *arg);
-static char *sip_do_debug(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
-static char *sip_cli_notify(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
-static char *sip_set_history(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
-static int sip_dtmfmode(struct ast_channel *chan, const char *data);
-static int sip_addheader(struct ast_channel *chan, const char *data);
-static int sip_do_reload(enum channelreloadreason reason);
-static char *sip_reload(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
-static int ast_sockaddr_resolve_first(struct ast_sockaddr *addr,
- const char *name, int flag);
-static int ast_sockaddr_resolve_first_transport(struct ast_sockaddr *addr,
- const char *name, int flag, unsigned int transport);
-
-/*--- Debugging
- Functions for enabling debug per IP or fully, or enabling history logging for
- a SIP dialog
-*/
-static void sip_dump_history(struct sip_pvt *dialog); /* Dump history to debuglog at end of dialog, before destroying data */
-static inline int sip_debug_test_addr(const struct ast_sockaddr *addr);
-static inline int sip_debug_test_pvt(struct sip_pvt *p);
-static void append_history_full(struct sip_pvt *p, const char *fmt, ...);
-static void sip_dump_history(struct sip_pvt *dialog);
-
-/*--- Device object handling */
-static struct sip_peer *build_peer(const char *name, struct ast_variable *v, struct ast_variable *alt, int realtime, int devstate_only);
-static int update_call_counter(struct sip_pvt *fup, int event);
-static void sip_destroy_peer(struct sip_peer *peer);
-static void sip_destroy_peer_fn(void *peer);
-static void set_peer_defaults(struct sip_peer *peer);
-static struct sip_peer *temp_peer(const char *name);
-static void register_peer_exten(struct sip_peer *peer, int onoff);
-static int sip_poke_peer_s(const void *data);
-static enum parse_register_result parse_register_contact(struct sip_pvt *pvt, struct sip_peer *p, struct sip_request *req);
-static void reg_source_db(struct sip_peer *peer);
-static void destroy_association(struct sip_peer *peer);
-static void set_insecure_flags(struct ast_flags *flags, const char *value, int lineno);
-static int handle_common_options(struct ast_flags *flags, struct ast_flags *mask, struct ast_variable *v);
-static void set_socket_transport(struct sip_socket *socket, int transport);
-static int peer_ipcmp_cb_full(void *obj, void *arg, void *data, int flags);
-
-/* Realtime device support */
-static void realtime_update_peer(const char *peername, struct ast_sockaddr *addr, const char *username, const char *fullcontact, const char *useragent, int expirey, unsigned short deprecated_username, int lastms, const char *path);
-static void update_peer(struct sip_peer *p, int expire);
-static struct ast_variable *get_insecure_variable_from_config(struct ast_config *config);
-static const char *get_name_from_variable(const struct ast_variable *var);
-static struct sip_peer *realtime_peer(const char *peername, struct ast_sockaddr *sin, char *callbackexten, int devstate_only, int which_objects);
-static char *sip_prune_realtime(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
-
-/*--- Internal UA client handling (outbound registrations) */
-static void ast_sip_ouraddrfor(const struct ast_sockaddr *them, struct ast_sockaddr *us, struct sip_pvt *p);
-static void sip_registry_destroy(void *reg);
-static int sip_register(const char *value, int lineno);
-static const char *regstate2str(enum sipregistrystate regstate) attribute_const;
-static int __sip_do_register(struct sip_registry *r);
-static int sip_reg_timeout(const void *data);
-static void sip_send_all_registers(void);
-static int sip_reinvite_retry(const void *data);
-
-/*--- Parsing SIP requests and responses */
-static int determine_firstline_parts(struct sip_request *req);
-static const struct cfsubscription_types *find_subscription_type(enum subscriptiontype subtype);
-static const char *gettag(const struct sip_request *req, const char *header, char *tagbuf, int tagbufsize);
-static int find_sip_method(const char *msg);
-static unsigned int parse_allowed_methods(struct sip_request *req);
-static unsigned int set_pvt_allowed_methods(struct sip_pvt *pvt, struct sip_request *req);
-static int parse_request(struct sip_request *req);
-static const char *referstatus2str(enum referstatus rstatus) attribute_pure;
-static int method_match(enum sipmethod id, const char *name);
-static void parse_copy(struct sip_request *dst, const struct sip_request *src);
-static void parse_oli(struct sip_request *req, struct ast_channel *chan);
-static const char *find_alias(const char *name, const char *_default);
-static const char *__get_header(const struct sip_request *req, const char *name, int *start);
-static void lws2sws(struct ast_str *msgbuf);
-static void extract_uri(struct sip_pvt *p, struct sip_request *req);
-static char *remove_uri_parameters(char *uri);
-static int get_refer_info(struct sip_pvt *transferer, struct sip_request *outgoing_req);
-static int get_also_info(struct sip_pvt *p, struct sip_request *oreq);
-static int parse_ok_contact(struct sip_pvt *pvt, struct sip_request *req);
-static int use_reason_header(struct sip_pvt *pvt, struct sip_request *req);
-static int set_address_from_contact(struct sip_pvt *pvt);
-static void check_via(struct sip_pvt *p, const struct sip_request *req);
-static int get_rpid(struct sip_pvt *p, struct sip_request *oreq);
-static int get_rdnis(struct sip_pvt *p, struct sip_request *oreq, char **name, char **number, int *reason, char **reason_str);
-static enum sip_get_dest_result get_destination(struct sip_pvt *p, struct sip_request *oreq, int *cc_recall_core_id);
-static int transmit_state_notify(struct sip_pvt *p, struct state_notify_data *data, int full, int timeout);
-static void update_connectedline(struct sip_pvt *p, const void *data, size_t datalen);
-static void update_redirecting(struct sip_pvt *p, const void *data, size_t datalen);
-static int get_domain(const char *str, char *domain, int len);
-static void get_realm(struct sip_pvt *p, const struct sip_request *req);
-static char *get_content(struct sip_request *req);
-
-/*-- TCP connection handling ---*/
-static void *_sip_tcp_helper_thread(struct ast_tcptls_session_instance *tcptls_session);
-static void *sip_tcp_worker_fn(void *);
-
-/*--- Constructing requests and responses */
-static void initialize_initreq(struct sip_pvt *p, struct sip_request *req);
-static int init_req(struct sip_request *req, int sipmethod, const char *recip);
-static void deinit_req(struct sip_request *req);
-static int reqprep(struct sip_request *req, struct sip_pvt *p, int sipmethod, uint32_t seqno, int newbranch);
-static void initreqprep(struct sip_request *req, struct sip_pvt *p, int sipmethod, const char * const explicit_uri);
-static int init_resp(struct sip_request *resp, const char *msg);
-static inline int resp_needs_contact(const char *msg, enum sipmethod method);
-static int respprep(struct sip_request *resp, struct sip_pvt *p, const char *msg, const struct sip_request *req);
-static const struct ast_sockaddr *sip_real_dst(const struct sip_pvt *p);
-static void build_via(struct sip_pvt *p);
-static int create_addr_from_peer(struct sip_pvt *r, struct sip_peer *peer);
-static int create_addr(struct sip_pvt *dialog, const char *opeer, struct ast_sockaddr *addr, int newdialog);
-static char *generate_random_string(char *buf, size_t size);
-static void build_callid_pvt(struct sip_pvt *pvt);
-static void change_callid_pvt(struct sip_pvt *pvt, const char *callid);
-static void build_callid_registry(struct sip_registry *reg, const struct ast_sockaddr *ourip, const char *fromdomain);
-static void build_localtag_registry(struct sip_registry *reg);
-static void make_our_tag(struct sip_pvt *pvt);
-static int add_header(struct sip_request *req, const char *var, const char *value);
-static int add_max_forwards(struct sip_pvt *dialog, struct sip_request *req);
-static int add_content(struct sip_request *req, const char *line);
-static int finalize_content(struct sip_request *req);
-static void destroy_msg_headers(struct sip_pvt *pvt);
-static int add_text(struct sip_request *req, struct sip_pvt *p);
-static int add_digit(struct sip_request *req, char digit, unsigned int duration, int mode);
-static int add_rpid(struct sip_request *req, struct sip_pvt *p);
-static int add_vidupdate(struct sip_request *req);
-static void add_route(struct sip_request *req, struct sip_route *route, int skip);
-static int copy_header(struct sip_request *req, const struct sip_request *orig, const char *field);
-static int copy_all_header(struct sip_request *req, const struct sip_request *orig, const char *field);
-static int copy_via_headers(struct sip_pvt *p, struct sip_request *req, const struct sip_request *orig, const char *field);
-static void set_destination(struct sip_pvt *p, const char *uri);
-static void add_date(struct sip_request *req);
-static void add_expires(struct sip_request *req, int expires);
-static void build_contact(struct sip_pvt *p, struct sip_request *req, int incoming);
-
-/*------Request handling functions */
-static int handle_incoming(struct sip_pvt *p, struct sip_request *req, struct ast_sockaddr *addr, int *recount, int *nounlock);
-static int handle_request_update(struct sip_pvt *p, struct sip_request *req);
-static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, struct ast_sockaddr *addr, uint32_t seqno, int *recount, const char *e, int *nounlock);
-static int handle_request_refer(struct sip_pvt *p, struct sip_request *req, uint32_t seqno, int *nounlock);
-static int handle_request_bye(struct sip_pvt *p, struct sip_request *req);
-static int handle_request_register(struct sip_pvt *p, struct sip_request *req, struct ast_sockaddr *sin, const char *e);
-static int handle_request_cancel(struct sip_pvt *p, struct sip_request *req);
-static int handle_request_message(struct sip_pvt *p, struct sip_request *req, struct ast_sockaddr *addr, const char *e);
-static int handle_request_subscribe(struct sip_pvt *p, struct sip_request *req, struct ast_sockaddr *addr, uint32_t seqno, const char *e);
-static void handle_request_info(struct sip_pvt *p, struct sip_request *req);
-static int handle_request_options(struct sip_pvt *p, struct sip_request *req, struct ast_sockaddr *addr, const char *e);
-static int handle_invite_replaces(struct sip_pvt *p, struct sip_request *req,
- int *nounlock, struct sip_pvt *replaces_pvt, struct ast_channel *replaces_chan);
-static int handle_request_notify(struct sip_pvt *p, struct sip_request *req, struct ast_sockaddr *addr, uint32_t seqno, const char *e);
-static int local_attended_transfer(struct sip_pvt *transferer, struct ast_channel *transferer_chan, uint32_t seqno, int *nounlock);
-
-/*------Response handling functions */
-static void handle_response_publish(struct sip_pvt *p, int resp, const char *rest, struct sip_request *req, uint32_t seqno);
-static void handle_response_invite(struct sip_pvt *p, int resp, const char *rest, struct sip_request *req, uint32_t seqno);
-static void handle_response_notify(struct sip_pvt *p, int resp, const char *rest, struct sip_request *req, uint32_t seqno);
-static void handle_response_refer(struct sip_pvt *p, int resp, const char *rest, struct sip_request *req, uint32_t seqno);
-static void handle_response_subscribe(struct sip_pvt *p, int resp, const char *rest, struct sip_request *req, uint32_t seqno);
-static int handle_response_register(struct sip_pvt *p, int resp, const char *rest, struct sip_request *req, uint32_t seqno);
-static void handle_response(struct sip_pvt *p, int resp, const char *rest, struct sip_request *req, uint32_t seqno);
-
-/*------ SRTP Support -------- */
-static int process_crypto(struct sip_pvt *p, struct ast_rtp_instance *rtp, struct ast_sdp_srtp **srtp,
- const char *a);
-
-/*------ T38 Support --------- */
-static int transmit_response_with_t38_sdp(struct sip_pvt *p, char *msg, struct sip_request *req, int retrans);
-static void change_t38_state(struct sip_pvt *p, int state);
-
-/*------ Session-Timers functions --------- */
-static void proc_422_rsp(struct sip_pvt *p, struct sip_request *rsp);
-static void stop_session_timer(struct sip_pvt *p);
-static void start_session_timer(struct sip_pvt *p);
-static void restart_session_timer(struct sip_pvt *p);
-static const char *strefresherparam2str(enum st_refresher_param r);
-static int parse_session_expires(const char *p_hdrval, int *const p_interval, enum st_refresher_param *const p_ref);
-static int parse_minse(const char *p_hdrval, int *const p_interval);
-static int st_get_se(struct sip_pvt *, int max);
-static enum st_refresher st_get_refresher(struct sip_pvt *);
-static enum st_mode st_get_mode(struct sip_pvt *, int no_cached);
-static struct sip_st_dlg* sip_st_alloc(struct sip_pvt *const p);
-
-/*------- RTP Glue functions -------- */
-static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance *instance, struct ast_rtp_instance *vinstance, struct ast_rtp_instance *tinstance, const struct ast_format_cap *cap, int nat_active);
-
-/*!--- SIP MWI Subscription support */
-static int sip_subscribe_mwi(const char *value, int lineno);
-static void sip_send_all_mwi_subscriptions(void);
-static int __sip_subscribe_mwi_do(struct sip_subscription_mwi *mwi);
-
-/* Scheduler id start/stop/reschedule functions. */
-static void stop_provisional_keepalive(struct sip_pvt *pvt);
-static void do_stop_session_timer(struct sip_pvt *pvt);
-static void stop_reinvite_retry(struct sip_pvt *pvt);
-static void stop_retrans_pkt(struct sip_pkt *pkt);
-static void stop_t38_abort_timer(struct sip_pvt *pvt);
-
-/*! \brief Definition of this channel for PBX channel registration */
-struct ast_channel_tech sip_tech = {
- .type = "SIP",
- .description = "Session Initiation Protocol (SIP)",
- .properties = AST_CHAN_TP_WANTSJITTER | AST_CHAN_TP_CREATESJITTER,
- .requester = sip_request_call, /* called with chan unlocked */
- .devicestate = sip_devicestate, /* called with chan unlocked (not chan-specific) */
- .call = sip_call, /* called with chan locked */
- .send_html = sip_sendhtml,
- .hangup = sip_hangup, /* called with chan locked */
- .answer = sip_answer, /* called with chan locked */
- .read = sip_read, /* called with chan locked */
- .write = sip_write, /* called with chan locked */
- .write_video = sip_write, /* called with chan locked */
- .write_text = sip_write,
- .indicate = sip_indicate, /* called with chan locked */
- .transfer = sip_transfer, /* called with chan locked */
- .fixup = sip_fixup, /* called with chan locked */
- .send_digit_begin = sip_senddigit_begin, /* called with chan unlocked */
- .send_digit_end = sip_senddigit_end,
- .early_bridge = ast_rtp_instance_early_bridge,
- .send_text = sip_sendtext, /* called with chan locked */
- .func_channel_read = sip_acf_channel_read,
- .setoption = sip_setoption,
- .queryoption = sip_queryoption,
- .get_pvt_uniqueid = sip_get_callid,
-};
-
-/*! \brief This version of the sip channel tech has no send_digit_begin
- * callback so that the core knows that the channel does not want
- * DTMF BEGIN frames.
- * The struct is initialized just before registering the channel driver,
- * and is for use with channels using SIP INFO DTMF.
- */
-struct ast_channel_tech sip_tech_info;
-
-/*------- CC Support -------- */
-static int sip_cc_agent_init(struct ast_cc_agent *agent, struct ast_channel *chan);
-static int sip_cc_agent_start_offer_timer(struct ast_cc_agent *agent);
-static int sip_cc_agent_stop_offer_timer(struct ast_cc_agent *agent);
-static void sip_cc_agent_respond(struct ast_cc_agent *agent, enum ast_cc_agent_response_reason reason);
-static int sip_cc_agent_status_request(struct ast_cc_agent *agent);
-static int sip_cc_agent_start_monitoring(struct ast_cc_agent *agent);
-static int sip_cc_agent_recall(struct ast_cc_agent *agent);
-static void sip_cc_agent_destructor(struct ast_cc_agent *agent);
-
-static struct ast_cc_agent_callbacks sip_cc_agent_callbacks = {
- .type = "SIP",
- .init = sip_cc_agent_init,
- .start_offer_timer = sip_cc_agent_start_offer_timer,
- .stop_offer_timer = sip_cc_agent_stop_offer_timer,
- .respond = sip_cc_agent_respond,
- .status_request = sip_cc_agent_status_request,
- .start_monitoring = sip_cc_agent_start_monitoring,
- .callee_available = sip_cc_agent_recall,
- .destructor = sip_cc_agent_destructor,
-};
-
-/* -------- End of declarations of structures, constants and forward declarations of functions
- Below starts actual code
- ------------------------
-*/
-
-static int sip_epa_register(const struct epa_static_data *static_data)
-{
- struct epa_backend *backend = ast_calloc(1, sizeof(*backend));
-
- if (!backend) {
- return -1;
- }
-
- backend->static_data = static_data;
-
- AST_LIST_LOCK(&epa_static_data_list);
- AST_LIST_INSERT_TAIL(&epa_static_data_list, backend, next);
- AST_LIST_UNLOCK(&epa_static_data_list);
- return 0;
-}
-
-static void sip_epa_unregister_all(void)
-{
- struct epa_backend *backend;
-
- AST_LIST_LOCK(&epa_static_data_list);
- while ((backend = AST_LIST_REMOVE_HEAD(&epa_static_data_list, next))) {
- ast_free(backend);
- }
- AST_LIST_UNLOCK(&epa_static_data_list);
-}
-
-static void cc_handle_publish_error(struct sip_pvt *pvt, const int resp, struct sip_request *req, struct sip_epa_entry *epa_entry);
-
-static void cc_epa_destructor(void *data)
-{
- struct sip_epa_entry *epa_entry = data;
- struct cc_epa_entry *cc_entry = epa_entry->instance_data;
- ast_free(cc_entry);
-}
-
-static const struct epa_static_data cc_epa_static_data = {
- .event = CALL_COMPLETION,
- .name = "call-completion",
- .handle_error = cc_handle_publish_error,
- .destructor = cc_epa_destructor,
-};
-
-static const struct epa_static_data *find_static_data(const char * const event_package)
-{
- const struct epa_backend *backend = NULL;
-
- AST_LIST_LOCK(&epa_static_data_list);
- AST_LIST_TRAVERSE(&epa_static_data_list, backend, next) {
- if (!strcmp(backend->static_data->name, event_package)) {
- break;
- }
- }
- AST_LIST_UNLOCK(&epa_static_data_list);
- return backend ? backend->static_data : NULL;
-}
-
-static struct sip_epa_entry *create_epa_entry (const char * const event_package, const char * const destination)
-{
- struct sip_epa_entry *epa_entry;
- const struct epa_static_data *static_data;
-
- if (!(static_data = find_static_data(event_package))) {
- return NULL;
- }
-
- if (!(epa_entry = ao2_t_alloc(sizeof(*epa_entry), static_data->destructor, "Allocate new EPA entry"))) {
- return NULL;
- }
-
- epa_entry->static_data = static_data;
- ast_copy_string(epa_entry->destination, destination, sizeof(epa_entry->destination));
- return epa_entry;
-}
-static enum ast_cc_service_type service_string_to_service_type(const char * const service_string)
-{
- enum ast_cc_service_type service;
- for (service = AST_CC_CCBS; service <= AST_CC_CCNL; ++service) {
- if (!strcasecmp(service_string, sip_cc_service_map[service].service_string)) {
- return service;
- }
- }
- return AST_CC_NONE;
-}
-
-/* Even state compositors code */
-static void esc_entry_destructor(void *obj)
-{
- struct sip_esc_entry *esc_entry = obj;
- if (esc_entry->sched_id > -1) {
- AST_SCHED_DEL(sched, esc_entry->sched_id);
- }
-}
-
-static int esc_hash_fn(const void *obj, const int flags)
-{
- const struct sip_esc_entry *entry = obj;
- return ast_str_hash(entry->entity_tag);
-}
-
-static int esc_cmp_fn(void *obj, void *arg, int flags)
-{
- struct sip_esc_entry *entry1 = obj;
- struct sip_esc_entry *entry2 = arg;
-
- return (!strcmp(entry1->entity_tag, entry2->entity_tag)) ? (CMP_MATCH | CMP_STOP) : 0;
-}
-
-static struct event_state_compositor *get_esc(const char * const event_package) {
- int i;
- for (i = 0; i < ARRAY_LEN(event_state_compositors); i++) {
- if (!strcasecmp(event_package, event_state_compositors[i].name)) {
- return &event_state_compositors[i];
- }
- }
- return NULL;
-}
-
-static struct sip_esc_entry *get_esc_entry(const char * entity_tag, struct event_state_compositor *esc) {
- struct sip_esc_entry *entry;
- struct sip_esc_entry finder;
-
- ast_copy_string(finder.entity_tag, entity_tag, sizeof(finder.entity_tag));
-
- entry = ao2_find(esc->compositor, &finder, OBJ_POINTER);
-
- return entry;
-}
-
-static int publish_expire(const void *data)
-{
- struct sip_esc_entry *esc_entry = (struct sip_esc_entry *) data;
- struct event_state_compositor *esc = get_esc(esc_entry->event);
-
- ast_assert(esc != NULL);
-
- ao2_unlink(esc->compositor, esc_entry);
- esc_entry->sched_id = -1;
- ao2_ref(esc_entry, -1);
- return 0;
-}
-
-static void create_new_sip_etag(struct sip_esc_entry *esc_entry, int is_linked)
-{
- int new_etag = ast_atomic_fetchadd_int(&esc_etag_counter, +1);
- struct event_state_compositor *esc = get_esc(esc_entry->event);
-
- ast_assert(esc != NULL);
- if (is_linked) {
- ao2_unlink(esc->compositor, esc_entry);
- }
- snprintf(esc_entry->entity_tag, sizeof(esc_entry->entity_tag), "%d", new_etag);
- ao2_link(esc->compositor, esc_entry);
-}
-
-static struct sip_esc_entry *create_esc_entry(struct event_state_compositor *esc, struct sip_request *req, const int expires)
-{
- struct sip_esc_entry *esc_entry;
- int expires_ms;
-
- if (!(esc_entry = ao2_alloc(sizeof(*esc_entry), esc_entry_destructor))) {
- return NULL;
- }
-
- esc_entry->event = esc->name;
-
- expires_ms = expires * 1000;
- /* Bump refcount for scheduler */
- ao2_ref(esc_entry, +1);
- esc_entry->sched_id = ast_sched_add(sched, expires_ms, publish_expire, esc_entry);
- if (esc_entry->sched_id == -1) {
- ao2_ref(esc_entry, -1);
- ao2_ref(esc_entry, -1);
- return NULL;
- }
-
- /* Note: This links the esc_entry into the ESC properly */
- create_new_sip_etag(esc_entry, 0);
-
- return esc_entry;
-}
-
-static int initialize_escs(void)
-{
- int i, res = 0;
- for (i = 0; i < ARRAY_LEN(event_state_compositors); i++) {
- event_state_compositors[i].compositor = ao2_container_alloc_hash(
- AO2_ALLOC_OPT_LOCK_MUTEX, 0, ESC_MAX_BUCKETS, esc_hash_fn, NULL, esc_cmp_fn);
- if (!event_state_compositors[i].compositor) {
- res = -1;
- }
- }
- return res;
-}
-
-static void destroy_escs(void)
-{
- int i;
- for (i = 0; i < ARRAY_LEN(event_state_compositors); i++) {
- ao2_replace(event_state_compositors[i].compositor, NULL);
- }
-}
-
-
-static int find_by_notify_uri_helper(void *obj, void *arg, int flags)
-{
- struct ast_cc_agent *agent = obj;
- struct sip_cc_agent_pvt *agent_pvt = agent->private_data;
- const char *uri = arg;
-
- return !sip_uri_cmp(agent_pvt->notify_uri, uri) ? CMP_MATCH | CMP_STOP : 0;
-}
-
-static struct ast_cc_agent *find_sip_cc_agent_by_notify_uri(const char * const uri)
-{
- struct ast_cc_agent *agent = ast_cc_agent_callback(0, find_by_notify_uri_helper, (char *)uri, "SIP");
- return agent;
-}
-
-static int find_by_subscribe_uri_helper(void *obj, void *arg, int flags)
-{
- struct ast_cc_agent *agent = obj;
- struct sip_cc_agent_pvt *agent_pvt = agent->private_data;
- const char *uri = arg;
-
- return !sip_uri_cmp(agent_pvt->subscribe_uri, uri) ? CMP_MATCH | CMP_STOP : 0;
-}
-
-static struct ast_cc_agent *find_sip_cc_agent_by_subscribe_uri(const char * const uri)
-{
- struct ast_cc_agent *agent = ast_cc_agent_callback(0, find_by_subscribe_uri_helper, (char *)uri, "SIP");
- return agent;
-}
-
-static int find_by_callid_helper(void *obj, void *arg, int flags)
-{
- struct ast_cc_agent *agent = obj;
- struct sip_cc_agent_pvt *agent_pvt = agent->private_data;
- struct sip_pvt *call_pvt = arg;
-
- return !strcmp(agent_pvt->original_callid, call_pvt->callid) ? CMP_MATCH | CMP_STOP : 0;
-}
-
-static struct ast_cc_agent *find_sip_cc_agent_by_original_callid(struct sip_pvt *pvt)
-{
- struct ast_cc_agent *agent = ast_cc_agent_callback(0, find_by_callid_helper, pvt, "SIP");
- return agent;
-}
-
-static int sip_cc_agent_init(struct ast_cc_agent *agent, struct ast_channel *chan)
-{
- struct sip_cc_agent_pvt *agent_pvt = ast_calloc(1, sizeof(*agent_pvt));
- struct sip_pvt *call_pvt = ast_channel_tech_pvt(chan);
-
- if (!agent_pvt) {
- return -1;
- }
-
- ast_assert(!strcmp(ast_channel_tech(chan)->type, "SIP"));
-
- ast_copy_string(agent_pvt->original_callid, call_pvt->callid, sizeof(agent_pvt->original_callid));
- ast_copy_string(agent_pvt->original_exten, call_pvt->exten, sizeof(agent_pvt->original_exten));
- agent_pvt->offer_timer_id = -1;
- agent->private_data = agent_pvt;
- sip_pvt_lock(call_pvt);
- ast_set_flag(&call_pvt->flags[0], SIP_OFFER_CC);
- sip_pvt_unlock(call_pvt);
- return 0;
-}
-
-static int sip_offer_timer_expire(const void *data)
-{
- struct ast_cc_agent *agent = (struct ast_cc_agent *) data;
- struct sip_cc_agent_pvt *agent_pvt = agent->private_data;
-
- agent_pvt->offer_timer_id = -1;
-
- return ast_cc_failed(agent->core_id, "SIP agent %s's offer timer expired", agent->device_name);
-}
-
-static int sip_cc_agent_start_offer_timer(struct ast_cc_agent *agent)
-{
- struct sip_cc_agent_pvt *agent_pvt = agent->private_data;
- int when;
-
- when = ast_get_cc_offer_timer(agent->cc_params) * 1000;
- agent_pvt->offer_timer_id = ast_sched_add(sched, when, sip_offer_timer_expire, agent);
- return 0;
-}
-
-static int sip_cc_agent_stop_offer_timer(struct ast_cc_agent *agent)
-{
- struct sip_cc_agent_pvt *agent_pvt = agent->private_data;
-
- AST_SCHED_DEL(sched, agent_pvt->offer_timer_id);
- return 0;
-}
-
-static void sip_cc_agent_respond(struct ast_cc_agent *agent, enum ast_cc_agent_response_reason reason)
-{
- struct sip_cc_agent_pvt *agent_pvt = agent->private_data;
-
- sip_pvt_lock(agent_pvt->subscribe_pvt);
- ast_set_flag(&agent_pvt->subscribe_pvt->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
- if (reason == AST_CC_AGENT_RESPONSE_SUCCESS || !ast_strlen_zero(agent_pvt->notify_uri)) {
- /* The second half of this if statement may be a bit hard to grasp,
- * so here's an explanation. When a subscription comes into
- * chan_sip, as long as it is not malformed, it will be passed
- * to the CC core. If the core senses an out-of-order state transition,
- * then the core will call this callback with the "reason" set to a
- * failure condition.
- * However, an out-of-order state transition will occur during a resubscription
- * for CC. In such a case, we can see that we have already generated a notify_uri
- * and so we can detect that this isn't a *real* failure. Rather, it is just
- * something the core doesn't recognize as a legitimate SIP state transition.
- * Thus we respond with happiness and flowers.
- */
- transmit_response(agent_pvt->subscribe_pvt, "200 OK", &agent_pvt->subscribe_pvt->initreq);
- transmit_cc_notify(agent, agent_pvt->subscribe_pvt, CC_QUEUED);
- } else {
- transmit_response(agent_pvt->subscribe_pvt, "500 Internal Error", &agent_pvt->subscribe_pvt->initreq);
- }
- sip_pvt_unlock(agent_pvt->subscribe_pvt);
- agent_pvt->is_available = TRUE;
-}
-
-static int sip_cc_agent_status_request(struct ast_cc_agent *agent)
-{
- struct sip_cc_agent_pvt *agent_pvt = agent->private_data;
- enum ast_device_state state = agent_pvt->is_available ? AST_DEVICE_NOT_INUSE : AST_DEVICE_INUSE;
- return ast_cc_agent_status_response(agent->core_id, state);
-}
-
-static int sip_cc_agent_start_monitoring(struct ast_cc_agent *agent)
-{
- /* To start monitoring just means to wait for an incoming PUBLISH
- * to tell us that the caller has become available again. No special
- * action is needed
- */
- return 0;
-}
-
-static int sip_cc_agent_recall(struct ast_cc_agent *agent)
-{
- struct sip_cc_agent_pvt *agent_pvt = agent->private_data;
- /* If we have received a PUBLISH beforehand stating that the caller in question
- * is not available, we can save ourself a bit of effort here and just report
- * the caller as busy
- */
- if (!agent_pvt->is_available) {
- return ast_cc_agent_caller_busy(agent->core_id, "Caller %s is busy, reporting to the core",
- agent->device_name);
- }
- /* Otherwise, we transmit a NOTIFY to the caller and await either
- * a PUBLISH or an INVITE
- */
- sip_pvt_lock(agent_pvt->subscribe_pvt);
- transmit_cc_notify(agent, agent_pvt->subscribe_pvt, CC_READY);
- sip_pvt_unlock(agent_pvt->subscribe_pvt);
- return 0;
-}
-
-static void sip_cc_agent_destructor(struct ast_cc_agent *agent)
-{
- struct sip_cc_agent_pvt *agent_pvt = agent->private_data;
-
- if (!agent_pvt) {
- /* The agent constructor probably failed. */
- return;
- }
-
- sip_cc_agent_stop_offer_timer(agent);
- if (agent_pvt->subscribe_pvt) {
- sip_pvt_lock(agent_pvt->subscribe_pvt);
- if (!ast_test_flag(&agent_pvt->subscribe_pvt->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED)) {
- /* If we haven't sent a 200 OK for the SUBSCRIBE dialog yet, then we need to send a response letting
- * the subscriber know something went wrong
- */
- transmit_response(agent_pvt->subscribe_pvt, "500 Internal Server Error", &agent_pvt->subscribe_pvt->initreq);
- }
- sip_pvt_unlock(agent_pvt->subscribe_pvt);
- agent_pvt->subscribe_pvt = dialog_unref(agent_pvt->subscribe_pvt, "SIP CC agent destructor: Remove ref to subscription");
- }
- ast_free(agent_pvt);
-}
-
-
-static int sip_monitor_instance_hash_fn(const void *obj, const int flags)
-{
- const struct sip_monitor_instance *monitor_instance = obj;
- return monitor_instance->core_id;
-}
-
-static int sip_monitor_instance_cmp_fn(void *obj, void *arg, int flags)
-{
- struct sip_monitor_instance *monitor_instance1 = obj;
- struct sip_monitor_instance *monitor_instance2 = arg;
-
- return monitor_instance1->core_id == monitor_instance2->core_id ? CMP_MATCH | CMP_STOP : 0;
-}
-
-static void sip_monitor_instance_destructor(void *data)
-{
- struct sip_monitor_instance *monitor_instance = data;
- if (monitor_instance->subscription_pvt) {
- sip_pvt_lock(monitor_instance->subscription_pvt);
- monitor_instance->subscription_pvt->expiry = 0;
- transmit_invite(monitor_instance->subscription_pvt, SIP_SUBSCRIBE, FALSE, 0, monitor_instance->subscribe_uri);
- sip_pvt_unlock(monitor_instance->subscription_pvt);
- dialog_unref(monitor_instance->subscription_pvt, "Unref monitor instance ref of subscription pvt");
- }
- if (monitor_instance->suspension_entry) {
- monitor_instance->suspension_entry->body[0] = '\0';
- transmit_publish(monitor_instance->suspension_entry, SIP_PUBLISH_REMOVE ,monitor_instance->notify_uri);
- ao2_t_ref(monitor_instance->suspension_entry, -1, "Decrementing suspension entry refcount in sip_monitor_instance_destructor");
- }
- ast_string_field_free_memory(monitor_instance);
-}
-
-static struct sip_monitor_instance *sip_monitor_instance_init(int core_id, const char * const subscribe_uri, const char * const peername, const char * const device_name)
-{
- struct sip_monitor_instance *monitor_instance = ao2_alloc(sizeof(*monitor_instance), sip_monitor_instance_destructor);
-
- if (!monitor_instance) {
- return NULL;
- }
-
- if (ast_string_field_init(monitor_instance, 256)) {
- ao2_ref(monitor_instance, -1);
- return NULL;
- }
-
- ast_string_field_set(monitor_instance, subscribe_uri, subscribe_uri);
- ast_string_field_set(monitor_instance, peername, peername);
- ast_string_field_set(monitor_instance, device_name, device_name);
- monitor_instance->core_id = core_id;
- ao2_link(sip_monitor_instances, monitor_instance);
- return monitor_instance;
-}
-
-static int find_sip_monitor_instance_by_subscription_pvt(void *obj, void *arg, int flags)
-{
- struct sip_monitor_instance *monitor_instance = obj;
- return monitor_instance->subscription_pvt == arg ? CMP_MATCH | CMP_STOP : 0;
-}
-
-static int find_sip_monitor_instance_by_suspension_entry(void *obj, void *arg, int flags)
-{
- struct sip_monitor_instance *monitor_instance = obj;
- return monitor_instance->suspension_entry == arg ? CMP_MATCH | CMP_STOP : 0;
-}
-
-static int sip_cc_monitor_request_cc(struct ast_cc_monitor *monitor, int *available_timer_id);
-static int sip_cc_monitor_suspend(struct ast_cc_monitor *monitor);
-static int sip_cc_monitor_unsuspend(struct ast_cc_monitor *monitor);
-static int sip_cc_monitor_cancel_available_timer(struct ast_cc_monitor *monitor, int *sched_id);
-static void sip_cc_monitor_destructor(void *private_data);
-
-static struct ast_cc_monitor_callbacks sip_cc_monitor_callbacks = {
- .type = "SIP",
- .request_cc = sip_cc_monitor_request_cc,
- .suspend = sip_cc_monitor_suspend,
- .unsuspend = sip_cc_monitor_unsuspend,
- .cancel_available_timer = sip_cc_monitor_cancel_available_timer,
- .destructor = sip_cc_monitor_destructor,
-};
-
-static int sip_cc_monitor_request_cc(struct ast_cc_monitor *monitor, int *available_timer_id)
-{
- struct sip_monitor_instance *monitor_instance = monitor->private_data;
- enum ast_cc_service_type service = monitor->service_offered;
- int when;
-
- if (!monitor_instance) {
- return -1;
- }
-
- if (!(monitor_instance->subscription_pvt = sip_alloc(NULL, NULL, 0, SIP_SUBSCRIBE, NULL, 0))) {
- return -1;
- }
-
- when = service == AST_CC_CCBS ? ast_get_ccbs_available_timer(monitor->interface->config_params) :
- ast_get_ccnr_available_timer(monitor->interface->config_params);
-
- sip_pvt_lock(monitor_instance->subscription_pvt);
- ast_set_flag(&monitor_instance->subscription_pvt->flags[0], SIP_OUTGOING);
- create_addr(monitor_instance->subscription_pvt, monitor_instance->peername, 0, 1);
- ast_sip_ouraddrfor(&monitor_instance->subscription_pvt->sa, &monitor_instance->subscription_pvt->ourip, monitor_instance->subscription_pvt);
- monitor_instance->subscription_pvt->subscribed = CALL_COMPLETION;
- monitor_instance->subscription_pvt->expiry = when;
-
- transmit_invite(monitor_instance->subscription_pvt, SIP_SUBSCRIBE, FALSE, 2, monitor_instance->subscribe_uri);
- sip_pvt_unlock(monitor_instance->subscription_pvt);
-
- ao2_t_ref(monitor, +1, "Adding a ref to the monitor for the scheduler");
- *available_timer_id = ast_sched_add(sched, when * 1000, ast_cc_available_timer_expire, monitor);
- return 0;
-}
-
-static int construct_pidf_body(enum sip_cc_publish_state state, char *pidf_body, size_t size, const char *presentity)
-{
- struct ast_str *body = ast_str_alloca(size);
- char tuple_id[64];
-
- generate_random_string(tuple_id, sizeof(tuple_id));
-
- /* We'll make this a bare-bones pidf body. In state_notify_build_xml, the PIDF
- * body gets a lot more extra junk that isn't necessary, so we'll leave it out here.
- */
- ast_str_append(&body, 0, "\n");
- /* XXX The entity attribute is currently set to the peer name associated with the
- * dialog. This is because we currently only call this function for call-completion
- * PUBLISH bodies. In such cases, the entity is completely disregarded. For other
- * event packages, it may be crucial to have a proper URI as the presentity so this
- * should be revisited as support is expanded.
- */
- ast_str_append(&body, 0, "\n", presentity);
- ast_str_append(&body, 0, "\n", tuple_id);
- ast_str_append(&body, 0, "%s \n");
- ast_str_append(&body, 0, " \n");
- ast_copy_string(pidf_body, ast_str_buffer(body), size);
- return 0;
-}
-
-static int sip_cc_monitor_suspend(struct ast_cc_monitor *monitor)
-{
- struct sip_monitor_instance *monitor_instance = monitor->private_data;
- enum sip_publish_type publish_type;
- struct cc_epa_entry *cc_entry;
-
- if (!monitor_instance) {
- return -1;
- }
-
- if (!monitor_instance->suspension_entry) {
- /* We haven't yet allocated the suspension entry, so let's give it a shot */
- if (!(monitor_instance->suspension_entry = create_epa_entry("call-completion", monitor_instance->peername))) {
- ast_log(LOG_WARNING, "Unable to allocate sip EPA entry for call-completion\n");
- ao2_ref(monitor_instance, -1);
- return -1;
- }
- if (!(cc_entry = ast_calloc(1, sizeof(*cc_entry)))) {
- ast_log(LOG_WARNING, "Unable to allocate space for instance data of EPA entry for call-completion\n");
- ao2_ref(monitor_instance, -1);
- return -1;
- }
- cc_entry->core_id = monitor->core_id;
- monitor_instance->suspension_entry->instance_data = cc_entry;
- publish_type = SIP_PUBLISH_INITIAL;
- } else {
- publish_type = SIP_PUBLISH_MODIFY;
- cc_entry = monitor_instance->suspension_entry->instance_data;
- }
-
- cc_entry->current_state = CC_CLOSED;
-
- if (ast_strlen_zero(monitor_instance->notify_uri)) {
- /* If we have no set notify_uri, then what this means is that we have
- * not received a NOTIFY from this destination stating that he is
- * currently available.
- *
- * This situation can arise when the core calls the suspend callbacks
- * of multiple destinations. If one of the other destinations aside
- * from this one notified Asterisk that he is available, then there
- * is no reason to take any suspension action on this device. Rather,
- * we should return now and if we receive a NOTIFY while monitoring
- * is still "suspended" then we can immediately respond with the
- * proper PUBLISH to let this endpoint know what is going on.
- */
- return 0;
- }
- construct_pidf_body(CC_CLOSED, monitor_instance->suspension_entry->body, sizeof(monitor_instance->suspension_entry->body), monitor_instance->peername);
- return transmit_publish(monitor_instance->suspension_entry, publish_type, monitor_instance->notify_uri);
-}
-
-static int sip_cc_monitor_unsuspend(struct ast_cc_monitor *monitor)
-{
- struct sip_monitor_instance *monitor_instance = monitor->private_data;
- struct cc_epa_entry *cc_entry;
-
- if (!monitor_instance) {
- return -1;
- }
-
- ast_assert(monitor_instance->suspension_entry != NULL);
-
- cc_entry = monitor_instance->suspension_entry->instance_data;
- cc_entry->current_state = CC_OPEN;
- if (ast_strlen_zero(monitor_instance->notify_uri)) {
- /* This means we are being asked to unsuspend a call leg we never
- * sent a PUBLISH on. As such, there is no reason to send another
- * PUBLISH at this point either. We can just return instead.
- */
- return 0;
- }
- construct_pidf_body(CC_OPEN, monitor_instance->suspension_entry->body, sizeof(monitor_instance->suspension_entry->body), monitor_instance->peername);
- return transmit_publish(monitor_instance->suspension_entry, SIP_PUBLISH_MODIFY, monitor_instance->notify_uri);
-}
-
-static int sip_cc_monitor_cancel_available_timer(struct ast_cc_monitor *monitor, int *sched_id)
-{
- if (*sched_id != -1) {
- AST_SCHED_DEL(sched, *sched_id);
- ao2_t_ref(monitor, -1, "Removing scheduler's reference to the monitor");
- }
- return 0;
-}
-
-static void sip_cc_monitor_destructor(void *private_data)
-{
- struct sip_monitor_instance *monitor_instance = private_data;
- ao2_unlink(sip_monitor_instances, monitor_instance);
- ast_module_unref(ast_module_info->self);
-}
-
-static int sip_get_cc_information(struct sip_request *req, char *subscribe_uri, size_t size, enum ast_cc_service_type *service)
-{
- char *call_info = ast_strdupa(sip_get_header(req, "Call-Info"));
- char *uri;
- char *purpose;
- char *service_str;
- static const char cc_purpose[] = "purpose=call-completion";
- static const int cc_purpose_len = sizeof(cc_purpose) - 1;
-
- if (ast_strlen_zero(call_info)) {
- /* No Call-Info present. Definitely no CC offer */
- return -1;
- }
-
- uri = strsep(&call_info, ";");
-
- while ((purpose = strsep(&call_info, ";"))) {
- if (!strncmp(purpose, cc_purpose, cc_purpose_len)) {
- break;
- }
- }
- if (!purpose) {
- /* We didn't find the appropriate purpose= parameter. Oh well */
- return -1;
- }
-
- /* Okay, call-completion has been offered. Let's figure out what type of service this is */
- while ((service_str = strsep(&call_info, ";"))) {
- if (!strncmp(service_str, "m=", 2)) {
- break;
- }
- }
- if (!service_str) {
- /* So they didn't offer a particular service, We'll just go with CCBS since it really
- * doesn't matter anyway
- */
- service_str = "BS";
- } else {
- /* We already determined that there is an "m=" so no need to check
- * the result of this strsep
- */
- strsep(&service_str, "=");
- }
-
- if ((*service = service_string_to_service_type(service_str)) == AST_CC_NONE) {
- /* Invalid service offered */
- return -1;
- }
-
- ast_copy_string(subscribe_uri, get_in_brackets(uri), size);
-
- return 0;
-}
-
-/*!
- * \brief Determine what, if any, CC has been offered and queue a CC frame if possible
- *
- * After taking care of some formalities to be sure that this call is eligible for CC,
- * we first try to see if we can make use of native CC. We grab the information from
- * the passed-in sip_request (which is always a response to an INVITE). If we can
- * use native CC monitoring for the call, then so be it.
- *
- * If native cc monitoring is not possible or not supported, then we will instead attempt
- * to use generic monitoring. Falling back to generic from a failed attempt at using native
- * monitoring will only work if the monitor policy of the endpoint is "always"
- *
- * \param pvt The current dialog. Contains CC parameters for the endpoint
- * \param req The response to the INVITE we want to inspect
- * \param service The service to use if generic monitoring is to be used. For native
- * monitoring, we get the service from the SIP response itself
- */
-static void sip_handle_cc(struct sip_pvt *pvt, struct sip_request *req, enum ast_cc_service_type service)
-{
- enum ast_cc_monitor_policies monitor_policy = ast_get_cc_monitor_policy(pvt->cc_params);
- int core_id;
- char interface_name[AST_CHANNEL_NAME];
-
- if (monitor_policy == AST_CC_MONITOR_NEVER) {
- /* Don't bother, just return */
- return;
- }
-
- if ((core_id = ast_cc_get_current_core_id(pvt->owner)) == -1) {
- /* For some reason, CC is invalid, so don't try it! */
- return;
- }
-
- ast_channel_get_device_name(pvt->owner, interface_name, sizeof(interface_name));
-
- if (monitor_policy == AST_CC_MONITOR_ALWAYS || monitor_policy == AST_CC_MONITOR_NATIVE) {
- char subscribe_uri[SIPBUFSIZE];
- char device_name[AST_CHANNEL_NAME];
- enum ast_cc_service_type offered_service;
- struct sip_monitor_instance *monitor_instance;
- if (sip_get_cc_information(req, subscribe_uri, sizeof(subscribe_uri), &offered_service)) {
- /* If CC isn't being offered to us, or for some reason the CC offer is
- * not formatted correctly, then it may still be possible to use generic
- * call completion since the monitor policy may be "always"
- */
- goto generic;
- }
- ast_channel_get_device_name(pvt->owner, device_name, sizeof(device_name));
- if (!(monitor_instance = sip_monitor_instance_init(core_id, subscribe_uri, pvt->peername, device_name))) {
- /* Same deal. We can try using generic still */
- goto generic;
- }
- /* We bump the refcount of chan_sip because once we queue this frame, the CC core
- * will have a reference to callbacks in this module. We decrement the module
- * refcount once the monitor destructor is called
- */
- ast_module_ref(ast_module_info->self);
- ast_queue_cc_frame(pvt->owner, "SIP", pvt->dialstring, offered_service, monitor_instance);
- ao2_ref(monitor_instance, -1);
- return;
- }
-
-generic:
- if (monitor_policy == AST_CC_MONITOR_GENERIC || monitor_policy == AST_CC_MONITOR_ALWAYS) {
- ast_queue_cc_frame(pvt->owner, AST_CC_GENERIC_MONITOR_TYPE, interface_name, service, NULL);
- }
-}
-
-/*! \brief Working TLS connection configuration */
-static struct ast_tls_config sip_tls_cfg;
-
-/*! \brief Default TLS connection configuration */
-static struct ast_tls_config default_tls_cfg;
-
-/*! \brief Default DTLS connection configuration */
-static struct ast_rtp_dtls_cfg default_dtls_cfg;
-
-/*! \brief The TCP server definition */
-static struct ast_tcptls_session_args sip_tcp_desc = {
- .accept_fd = -1,
- .master = AST_PTHREADT_NULL,
- .tls_cfg = NULL,
- .poll_timeout = -1,
- .name = "SIP TCP server",
- .accept_fn = ast_tcptls_server_root,
- .worker_fn = sip_tcp_worker_fn,
-};
-
-/*! \brief The TCP/TLS server definition */
-static struct ast_tcptls_session_args sip_tls_desc = {
- .accept_fd = -1,
- .master = AST_PTHREADT_NULL,
- .tls_cfg = &sip_tls_cfg,
- .poll_timeout = -1,
- .name = "SIP TLS server",
- .accept_fn = ast_tcptls_server_root,
- .worker_fn = sip_tcp_worker_fn,
-};
-
-/*! \brief Append to SIP dialog history
- \retval 0 always */
-#define append_history(p, event, fmt , args... ) append_history_full(p, "%-15s " fmt, event, ## args)
-
-/*! \brief map from an integer value to a string.
- * If no match is found, return errorstring
- */
-static const char *map_x_s(const struct _map_x_s *table, int x, const char *errorstring)
-{
- const struct _map_x_s *cur;
-
- for (cur = table; cur->s; cur++) {
- if (cur->x == x) {
- return cur->s;
- }
- }
- return errorstring;
-}
-
-/*! \brief map from a string to an integer value, case insensitive.
- * If no match is found, return errorvalue.
- */
-static int map_s_x(const struct _map_x_s *table, const char *s, int errorvalue)
-{
- const struct _map_x_s *cur;
-
- for (cur = table; cur->s; cur++) {
- if (!strcasecmp(cur->s, s)) {
- return cur->x;
- }
- }
- return errorvalue;
-}
-
-/*!
- * \internal
- * \brief Determine if the given string is a SIP token.
- * \since 13.8.0
- *
- * \param str String to determine if is a SIP token.
- *
- * \note A token is defined by RFC3261 Section 25.1
- *
- * \return Non-zero if the string is a SIP token.
- */
-static int sip_is_token(const char *str)
-{
- int is_token;
-
- if (ast_strlen_zero(str)) {
- /* An empty string is not a token. */
- return 0;
- }
-
- is_token = 1;
- do {
- if (!isalnum(*str)
- && !strchr("-.!%*_+`'~", *str)) {
- /* The character is not allowed in a token. */
- is_token = 0;
- break;
- }
- } while (*++str);
-
- return is_token;
-}
-
-static const char *sip_reason_code_to_str(struct ast_party_redirecting_reason *reason)
-{
- int idx;
- int code;
-
- /* use specific string if given */
- if (!ast_strlen_zero(reason->str)) {
- return reason->str;
- }
-
- code = reason->code;
- for (idx = 0; idx < ARRAY_LEN(sip_reason_table); ++idx) {
- if (code == sip_reason_table[idx].code) {
- return sip_reason_table[idx].text;
- }
- }
-
- return "unknown";
-}
-
-/*!
- * \brief generic function for determining if a correct transport is being
- * used to contact a peer
- *
- * this is done as a macro so that the "tmpl" var can be passed either a
- * sip_request or a sip_peer
- */
-#define check_request_transport(peer, tmpl) ({ \
- int ret = 0; \
- if (peer->socket.type == tmpl->socket.type) \
- ; \
- else if (!(peer->transports & tmpl->socket.type)) {\
- ast_log(LOG_ERROR, \
- "'%s' is not a valid transport for '%s'. we only use '%s'! ending call.\n", \
- sip_get_transport(tmpl->socket.type), peer->name, get_transport_list(peer->transports) \
- ); \
- ret = 1; \
- } else if (peer->socket.type & AST_TRANSPORT_TLS) { \
- ast_log(LOG_WARNING, \
- "peer '%s' HAS NOT USED (OR SWITCHED TO) TLS in favor of '%s' (but this was allowed in sip.conf)!\n", \
- peer->name, sip_get_transport(tmpl->socket.type) \
- ); \
- } else { \
- ast_debug(1, \
- "peer '%s' has contacted us over %s even though we prefer %s.\n", \
- peer->name, sip_get_transport(tmpl->socket.type), sip_get_transport(peer->socket.type) \
- ); \
- }\
- (ret); \
-})
-
-/*! \brief
- * duplicate a list of channel variables, \return the copy.
- */
-static struct ast_variable *copy_vars(struct ast_variable *src)
-{
- struct ast_variable *res = NULL, *tmp, *v = NULL;
-
- for (v = src ; v ; v = v->next) {
- if ((tmp = ast_variable_new(v->name, v->value, v->file))) {
- tmp->next = res;
- res = tmp;
- }
- }
- return res;
-}
-
-static void tcptls_packet_destructor(void *obj)
-{
- struct tcptls_packet *packet = obj;
-
- ast_free(packet->data);
-}
-
-static void sip_tcptls_client_args_destructor(void *obj)
-{
- struct ast_tcptls_session_args *args = obj;
- if (args->tls_cfg) {
- ast_free(args->tls_cfg->certfile);
- ast_free(args->tls_cfg->pvtfile);
- ast_free(args->tls_cfg->cipher);
- ast_free(args->tls_cfg->cafile);
- ast_free(args->tls_cfg->capath);
-
- ast_ssl_teardown(args->tls_cfg);
- }
- ast_free(args->tls_cfg);
- ast_free((char *) args->name);
-}
-
-static void sip_threadinfo_destructor(void *obj)
-{
- struct sip_threadinfo *th = obj;
- struct tcptls_packet *packet;
-
- if (th->alert_pipe[0] > -1) {
- close(th->alert_pipe[0]);
- }
- if (th->alert_pipe[1] > -1) {
- close(th->alert_pipe[1]);
- }
- th->alert_pipe[0] = th->alert_pipe[1] = -1;
-
- while ((packet = AST_LIST_REMOVE_HEAD(&th->packet_q, entry))) {
- ao2_t_ref(packet, -1, "thread destruction, removing packet from frame queue");
- }
-
- if (th->tcptls_session) {
- ao2_t_ref(th->tcptls_session, -1, "remove tcptls_session for sip_threadinfo object");
- }
-}
-
-/*! \brief creates a sip_threadinfo object and links it into the threadt table. */
-static struct sip_threadinfo *sip_threadinfo_create(struct ast_tcptls_session_instance *tcptls_session, int transport)
-{
- struct sip_threadinfo *th;
-
- if (!tcptls_session || !(th = ao2_alloc(sizeof(*th), sip_threadinfo_destructor))) {
- return NULL;
- }
-
- th->alert_pipe[0] = th->alert_pipe[1] = -1;
-
- if (pipe(th->alert_pipe) == -1) {
- ao2_t_ref(th, -1, "Failed to open alert pipe on sip_threadinfo");
- ast_log(LOG_ERROR, "Could not create sip alert pipe in tcptls thread, error %s\n", strerror(errno));
- return NULL;
- }
- ao2_t_ref(tcptls_session, +1, "tcptls_session ref for sip_threadinfo object");
- th->tcptls_session = tcptls_session;
- th->type = transport ? transport : (ast_iostream_get_ssl(tcptls_session->stream) ? AST_TRANSPORT_TLS: AST_TRANSPORT_TCP);
- ao2_t_link(threadt, th, "Adding new tcptls helper thread");
- ao2_t_ref(th, -1, "Decrementing threadinfo ref from alloc, only table ref remains");
- return th;
-}
-
-/*! \brief used to indicate to a tcptls thread that data is ready to be written */
-static int sip_tcptls_write(struct ast_tcptls_session_instance *tcptls_session, const void *buf, size_t len)
-{
- int res = len;
- struct sip_threadinfo *th = NULL;
- struct tcptls_packet *packet = NULL;
- struct sip_threadinfo tmp = {
- .tcptls_session = tcptls_session,
- };
- enum sip_tcptls_alert alert = TCPTLS_ALERT_DATA;
-
- if (!tcptls_session) {
- return XMIT_ERROR;
- }
-
- ao2_lock(tcptls_session);
-
- if (!tcptls_session->stream ||
- !(packet = ao2_alloc(sizeof(*packet), tcptls_packet_destructor)) ||
- !(packet->data = ast_str_create(len))) {
- goto tcptls_write_setup_error;
- }
-
- if (!(th = ao2_t_find(threadt, &tmp, OBJ_POINTER, "ao2_find, getting sip_threadinfo in tcp helper thread"))) {
- ast_log(LOG_ERROR, "Unable to locate tcptls_session helper thread.\n");
- goto tcptls_write_setup_error;
- }
-
- /* goto tcptls_write_error should _NOT_ be used beyond this point */
- ast_str_set(&packet->data, 0, "%s", (char *) buf);
- packet->len = len;
-
- /* alert tcptls thread handler that there is a packet to be sent.
- * must lock the thread info object to guarantee control of the
- * packet queue */
- ao2_lock(th);
- if (write(th->alert_pipe[1], &alert, sizeof(alert)) == -1) {
- ast_log(LOG_ERROR, "write() to alert pipe failed: %s\n", strerror(errno));
- ao2_t_ref(packet, -1, "could not write to alert pipe, remove packet");
- packet = NULL;
- res = XMIT_ERROR;
- } else { /* it is safe to queue the frame after issuing the alert when we hold the threadinfo lock */
- AST_LIST_INSERT_TAIL(&th->packet_q, packet, entry);
- }
- ao2_unlock(th);
-
- ao2_unlock(tcptls_session);
- ao2_t_ref(th, -1, "In sip_tcptls_write, unref threadinfo object after finding it");
- return res;
-
-tcptls_write_setup_error:
- if (th) {
- ao2_t_ref(th, -1, "In sip_tcptls_write, unref threadinfo obj, could not create packet");
- }
- if (packet) {
- ao2_t_ref(packet, -1, "could not allocate packet's data");
- }
- ao2_unlock(tcptls_session);
-
- return XMIT_ERROR;
-}
-
-/*! \brief SIP TCP connection handler */
-static void *sip_tcp_worker_fn(void *data)
-{
- struct ast_tcptls_session_instance *tcptls_session = data;
-
- return _sip_tcp_helper_thread(tcptls_session);
-}
-
-/*! \brief SIP WebSocket connection handler */
-static void sip_websocket_callback(struct ast_websocket *session, struct ast_variable *parameters, struct ast_variable *headers)
-{
- int res;
-
- if (ast_websocket_set_nonblock(session)) {
- goto end;
- }
-
- if (ast_websocket_set_timeout(session, sip_cfg.websocket_write_timeout)) {
- goto end;
- }
-
- while ((res = ast_wait_for_input(ast_websocket_fd(session), -1)) > 0) {
- char *payload;
- uint64_t payload_len;
- enum ast_websocket_opcode opcode;
- int fragmented;
-
- if (ast_websocket_read(session, &payload, &payload_len, &opcode, &fragmented)) {
- /* We err on the side of caution and terminate the session if any error occurs */
- break;
- }
-
- if (opcode == AST_WEBSOCKET_OPCODE_TEXT || opcode == AST_WEBSOCKET_OPCODE_BINARY) {
- struct sip_request req = { 0, };
- char data[payload_len + 1];
-
- if (!(req.data = ast_str_create(payload_len + 1))) {
- goto end;
- }
-
- strncpy(data, payload, payload_len);
- data[payload_len] = '\0';
-
- if (ast_str_set(&req.data, -1, "%s", data) == AST_DYNSTR_BUILD_FAILED) {
- deinit_req(&req);
- goto end;
- }
-
- req.socket.fd = ast_websocket_fd(session);
- set_socket_transport(&req.socket, ast_websocket_is_secure(session) ? AST_TRANSPORT_WSS : AST_TRANSPORT_WS);
- req.socket.ws_session = session;
-
- handle_request_do(&req, ast_websocket_remote_address(session));
- deinit_req(&req);
-
- } else if (opcode == AST_WEBSOCKET_OPCODE_CLOSE) {
- break;
- }
- }
-
-end:
- ast_websocket_unref(session);
-}
-
-/*! \brief Check if the authtimeout has expired.
- * \param start the time when the session started
- *
- * \retval 0 the timeout has expired
- * \retval -1 error
- * \return the number of milliseconds until the timeout will expire
- */
-static int sip_check_authtimeout(time_t start)
-{
- int timeout;
- time_t now;
- if(time(&now) == -1) {
- ast_log(LOG_ERROR, "error executing time(): %s\n", strerror(errno));
- return -1;
- }
-
- timeout = (authtimeout - (now - start)) * 1000;
- if (timeout < 0) {
- /* we have timed out */
- return 0;
- }
-
- return timeout;
-}
-
-/*!
- * \brief Indication of a TCP message's integrity
- */
-enum message_integrity {
- /*!
- * The message has an error in it with
- * regards to its Content-Length header
- */
- MESSAGE_INVALID,
- /*!
- * The message is incomplete
- */
- MESSAGE_FRAGMENT,
- /*!
- * The data contains a complete message
- * plus a fragment of another.
- */
- MESSAGE_FRAGMENT_COMPLETE,
- /*!
- * The message is complete
- */
- MESSAGE_COMPLETE,
-};
-
-/*!
- * \brief
- * Get the content length from an unparsed SIP message
- *
- * \param message The unparsed SIP message headers
- * \return The value of the Content-Length header or -1 if message is invalid
- */
-static int read_raw_content_length(const char *message)
-{
- char *content_length_str;
- int content_length = -1;
-
- struct ast_str *msg_copy;
- char *msg;
-
- /* Using a ast_str because lws2sws takes one of those */
- if (!(msg_copy = ast_str_create(strlen(message) + 1))) {
- return -1;
- }
- ast_str_set(&msg_copy, 0, "%s", message);
-
- if (sip_cfg.pedanticsipchecking) {
- lws2sws(msg_copy);
- }
-
- msg = ast_str_buffer(msg_copy);
-
- /* Let's find a Content-Length header */
- if ((content_length_str = strcasestr(msg, "\nContent-Length:"))) {
- content_length_str += sizeof("\nContent-Length:") - 1;
- } else if ((content_length_str = strcasestr(msg, "\nl:"))) {
- content_length_str += sizeof("\nl:") - 1;
- } else {
- /* RFC 3261 18.3
- * "In the case of stream-oriented transports such as TCP, the Content-
- * Length header field indicates the size of the body. The Content-
- * Length header field MUST be used with stream oriented transports."
- */
- goto done;
- }
-
- /* Double-check that this is a complete header */
- if (!strchr(content_length_str, '\n')) {
- goto done;
- }
-
- if (sscanf(content_length_str, "%30d", &content_length) != 1) {
- content_length = -1;
- }
-
-done:
- ast_free(msg_copy);
- return content_length;
-}
-
-/*!
- * \brief Check that a message received over TCP is a full message
- *
- * This will take the information read in and then determine if
- * 1) The message is a full SIP request
- * 2) The message is a partial SIP request
- * 3) The message contains a full SIP request along with another partial request
- * \param request The resulting request with extra fragments removed.
- * \param overflow If the message contains more than a full request, this is the remainder of the message
- * \return The resulting integrity of the message
- */
-static enum message_integrity check_message_integrity(struct ast_str **request, struct ast_str **overflow)
-{
- char *message = ast_str_buffer(*request);
- char *body;
- int content_length;
- int message_len = ast_str_strlen(*request);
- int body_len;
-
- /* Important pieces to search for in a SIP request are \r\n\r\n. This
- * marks either
- * 1) The division between the headers and body
- * 2) The end of the SIP request
- */
- body = strstr(message, "\r\n\r\n");
- if (!body) {
- /* This is clearly a partial message since we haven't reached an end
- * yet.
- */
- return MESSAGE_FRAGMENT;
- }
- body += sizeof("\r\n\r\n") - 1;
- body_len = message_len - (body - message);
-
- body[-1] = '\0';
- content_length = read_raw_content_length(message);
- body[-1] = '\n';
-
- if (content_length < 0) {
- return MESSAGE_INVALID;
- } else if (content_length == 0) {
- /* We've definitely received an entire message. We need
- * to check if there's also a fragment of another message
- * in addition.
- */
- if (body_len == 0) {
- return MESSAGE_COMPLETE;
- } else {
- ast_str_append(overflow, 0, "%s", body);
- ast_str_truncate(*request, message_len - body_len);
- return MESSAGE_FRAGMENT_COMPLETE;
- }
- }
- /* Positive content length. Let's see what sort of
- * message body we're dealing with.
- */
- if (body_len < content_length) {
- /* We don't have the full message body yet */
- return MESSAGE_FRAGMENT;
- } else if (body_len > content_length) {
- /* We have the full message plus a fragment of a further
- * message
- */
- ast_str_append(overflow, 0, "%s", body + content_length);
- ast_str_truncate(*request, message_len - (body_len - content_length));
- return MESSAGE_FRAGMENT_COMPLETE;
- } else {
- /* Yay! Full message with no extra content */
- return MESSAGE_COMPLETE;
- }
-}
-
-/*!
- * \internal
- * \brief Read SIP request or response from a TCP/TLS connection
- *
- * \param req The request structure to be filled in
- * \param tcptls_session The TCP/TLS connection from which to read
- * \param authenticated 0 means unauthenticated
- * \param start timeout for unauthenticated server sessions
- * \retval -1 Failed to read data
- * \retval 0 Successfully read data
- */
-static int sip_tcptls_read(struct sip_request *req, struct ast_tcptls_session_instance *tcptls_session,
- int authenticated, time_t start)
-{
- enum message_integrity message_integrity = MESSAGE_FRAGMENT;
-
- while (message_integrity == MESSAGE_FRAGMENT) {
- size_t datalen;
-
- if (ast_str_strlen(tcptls_session->overflow_buf) == 0) {
- char readbuf[4097];
- int timeout;
- int res;
- if (!tcptls_session->client && !authenticated) {
- if ((timeout = sip_check_authtimeout(start)) < 0) {
- return -1;
- }
-
- if (timeout == 0) {
- ast_debug(2, "SIP TCP/TLS server timed out\n");
- return -1;
- }
- } else {
- timeout = -1;
- }
- res = ast_wait_for_input(ast_iostream_get_fd(tcptls_session->stream), timeout);
- if (res < 0) {
- ast_debug(2, "SIP TCP/TLS server :: ast_wait_for_input returned %d\n", res);
- return -1;
- } else if (res == 0) {
- ast_debug(2, "SIP TCP/TLS server timed out\n");
- return -1;
- }
-
- res = ast_iostream_read(tcptls_session->stream, readbuf, sizeof(readbuf) - 1);
- if (res < 0) {
- if (errno == EAGAIN || errno == EINTR) {
- continue;
- }
- ast_debug(2, "SIP TCP/TLS server error when receiving data\n");
- return -1;
- } else if (res == 0) {
- ast_debug(2, "SIP TCP/TLS server has shut down\n");
- return -1;
- }
- readbuf[res] = '\0';
- ast_str_append(&req->data, 0, "%s", readbuf);
- } else {
- ast_str_append(&req->data, 0, "%s", ast_str_buffer(tcptls_session->overflow_buf));
- ast_str_reset(tcptls_session->overflow_buf);
- }
-
- datalen = ast_str_strlen(req->data);
- if (datalen > SIP_MAX_PACKET_SIZE) {
- ast_log(LOG_WARNING, "Rejecting TCP/TLS packet from '%s' because way too large: %zu\n",
- ast_sockaddr_stringify(&tcptls_session->remote_address), datalen);
- return -1;
- }
-
- message_integrity = check_message_integrity(&req->data, &tcptls_session->overflow_buf);
- }
-
- return 0;
-}
-
-/*! \brief SIP TCP thread management function
- This function reads from the socket, parses the packet into a request
-*/
-static void *_sip_tcp_helper_thread(struct ast_tcptls_session_instance *tcptls_session)
-{
- int res, timeout = -1, authenticated = 0, flags;
- time_t start;
- struct sip_request req = { 0, } , reqcpy = { 0, };
- struct sip_threadinfo *me = NULL;
- char buf[1024] = "";
- struct pollfd fds[2] = { { 0 }, { 0 }, };
- struct ast_tcptls_session_args *ca = NULL;
-
- /* If this is a server session, then the connection has already been
- * setup. Check if the authlimit has been reached and if not create the
- * threadinfo object so we can access this thread for writing.
- *
- * if this is a client connection more work must be done.
- * 1. We own the parent session args for a client connection. This pointer needs
- * to be held on to so we can decrement it's ref count on thread destruction.
- * 2. The threadinfo object was created before this thread was launched, however
- * it must be found within the threadt table.
- * 3. Last, the tcptls_session must be started.
- */
- if (!tcptls_session->client) {
- if (ast_atomic_fetchadd_int(&unauth_sessions, +1) >= authlimit) {
- /* unauth_sessions is decremented in the cleanup code */
- goto cleanup;
- }
-
- ast_iostream_nonblock(tcptls_session->stream);
- if (!(me = sip_threadinfo_create(tcptls_session, ast_iostream_get_ssl(tcptls_session->stream) ? AST_TRANSPORT_TLS : AST_TRANSPORT_TCP))) {
- goto cleanup;
- }
- me->threadid = pthread_self();
- ao2_t_ref(me, +1, "Adding threadinfo ref for tcp_helper_thread");
- } else {
- struct sip_threadinfo tmp = {
- .tcptls_session = tcptls_session,
- };
-
- if ((!(ca = tcptls_session->parent)) ||
- (!(me = ao2_t_find(threadt, &tmp, OBJ_POINTER, "ao2_find, getting sip_threadinfo in tcp helper thread")))) {
- goto cleanup;
- }
-
- me->threadid = pthread_self();
-
- if (!(tcptls_session = ast_tcptls_client_start(tcptls_session))) {
- goto cleanup;
- }
- }
-
- flags = 1;
- if (setsockopt(ast_iostream_get_fd(tcptls_session->stream), SOL_SOCKET, SO_KEEPALIVE, &flags, sizeof(flags))) {
- ast_log(LOG_ERROR, "error enabling TCP keep-alives on sip socket: %s\n", strerror(errno));
- goto cleanup;
- }
-
- ast_debug(2, "Starting thread for %s server\n", ast_iostream_get_ssl(tcptls_session->stream) ? "TLS" : "TCP");
-
- /* set up pollfd to watch for reads on both the socket and the alert_pipe */
- fds[0].fd = ast_iostream_get_fd(tcptls_session->stream);
- fds[1].fd = me->alert_pipe[0];
- fds[0].events = fds[1].events = POLLIN | POLLPRI;
-
- if (!(req.data = ast_str_create(SIP_MIN_PACKET))) {
- goto cleanup;
- }
- if (!(reqcpy.data = ast_str_create(SIP_MIN_PACKET))) {
- goto cleanup;
- }
-
- if(time(&start) == -1) {
- ast_log(LOG_ERROR, "error executing time(): %s\n", strerror(errno));
- goto cleanup;
- }
-
- /*
- * We cannot let the stream exclusively wait for data to arrive.
- * We have to wake up the task to send outgoing messages.
- */
- ast_iostream_set_exclusive_input(tcptls_session->stream, 0);
-
- ast_iostream_set_timeout_sequence(tcptls_session->stream, ast_tvnow(),
- tcptls_session->client ? -1 : (authtimeout * 1000));
-
- for (;;) {
- struct ast_str *str_save;
-
- if (!tcptls_session->client && req.authenticated && !authenticated) {
- authenticated = 1;
- ast_iostream_set_timeout_disable(tcptls_session->stream);
- ast_atomic_fetchadd_int(&unauth_sessions, -1);
- }
-
- /* calculate the timeout for unauthenticated server sessions */
- if (!tcptls_session->client && !authenticated ) {
- if ((timeout = sip_check_authtimeout(start)) < 0) {
- goto cleanup;
- }
-
- if (timeout == 0) {
- ast_debug(2, "SIP %s server timed out\n", ast_iostream_get_ssl(tcptls_session->stream) ? "TLS": "TCP");
- goto cleanup;
- }
- } else {
- timeout = -1;
- }
-
- if (ast_str_strlen(tcptls_session->overflow_buf) == 0) {
- res = ast_poll(fds, 2, timeout); /* polls for both socket and alert_pipe */
- if (res < 0) {
- ast_debug(2, "SIP %s server :: ast_wait_for_input returned %d\n", ast_iostream_get_ssl(tcptls_session->stream) ? "TLS": "TCP", res);
- goto cleanup;
- } else if (res == 0) {
- /* timeout */
- ast_debug(2, "SIP %s server timed out\n", ast_iostream_get_ssl(tcptls_session->stream) ? "TLS": "TCP");
- goto cleanup;
- }
- }
-
- /*
- * handle the socket event, check for both reads from the socket fd or TCP overflow buffer,
- * and writes from alert_pipe fd.
- */
- if (fds[0].revents || (ast_str_strlen(tcptls_session->overflow_buf) > 0)) { /* there is data on the socket to be read */
- fds[0].revents = 0;
-
- /* clear request structure */
- str_save = req.data;
- memset(&req, 0, sizeof(req));
- req.data = str_save;
- ast_str_reset(req.data);
-
- str_save = reqcpy.data;
- memset(&reqcpy, 0, sizeof(reqcpy));
- reqcpy.data = str_save;
- ast_str_reset(reqcpy.data);
-
- memset(buf, 0, sizeof(buf));
-
- if (ast_iostream_get_ssl(tcptls_session->stream)) {
- set_socket_transport(&req.socket, AST_TRANSPORT_TLS);
- } else {
- set_socket_transport(&req.socket, AST_TRANSPORT_TCP);
- }
- req.socket.fd = ast_iostream_get_fd(tcptls_session->stream);
-
- res = sip_tcptls_read(&req, tcptls_session, authenticated, start);
- if (res < 0) {
- goto cleanup;
- }
-
- req.socket.tcptls_session = tcptls_session;
- req.socket.ws_session = NULL;
- handle_request_do(&req, &tcptls_session->remote_address);
- }
-
- if (fds[1].revents) { /* alert_pipe indicates there is data in the send queue to be sent */
- enum sip_tcptls_alert alert;
- struct tcptls_packet *packet;
-
- fds[1].revents = 0;
-
- if (read(me->alert_pipe[0], &alert, sizeof(alert)) == -1) {
- ast_log(LOG_ERROR, "read() failed: %s\n", strerror(errno));
- goto cleanup;
- }
-
- switch (alert) {
- case TCPTLS_ALERT_STOP:
- goto cleanup;
- case TCPTLS_ALERT_DATA:
- ao2_lock(me);
- if (!(packet = AST_LIST_REMOVE_HEAD(&me->packet_q, entry))) {
- ast_log(LOG_WARNING, "TCPTLS thread alert_pipe indicated packet should be sent, but frame_q is empty\n");
- }
- ao2_unlock(me);
-
- if (packet) {
- if (ast_iostream_write(tcptls_session->stream, ast_str_buffer(packet->data), packet->len) == -1) {
- ast_log(LOG_WARNING, "Failure to write to tcp/tls socket\n");
- }
- ao2_t_ref(packet, -1, "tcptls packet sent, this is no longer needed");
- } else {
- goto cleanup;
- }
- break;
- default:
- ast_log(LOG_ERROR, "Unknown tcptls thread alert '%u'\n", alert);
- goto cleanup;
- }
- }
- }
-
- ast_debug(2, "Shutting down thread for %s server\n", ast_iostream_get_ssl(tcptls_session->stream) ? "TLS" : "TCP");
-
-cleanup:
- if (tcptls_session && !tcptls_session->client && !authenticated) {
- ast_atomic_fetchadd_int(&unauth_sessions, -1);
- }
-
- if (me) {
- ao2_t_unlink(threadt, me, "Removing tcptls helper thread, thread is closing");
- ao2_t_ref(me, -1, "Removing tcp_helper_threads threadinfo ref");
- }
- deinit_req(&reqcpy);
- deinit_req(&req);
-
- /* if client, we own the parent session arguments and must decrement ref */
- if (ca) {
- ao2_t_ref(ca, -1, "closing tcptls thread, getting rid of client tcptls_session arguments");
- }
-
- if (tcptls_session) {
- ao2_lock(tcptls_session);
- ast_tcptls_close_session_file(tcptls_session);
- tcptls_session->parent = NULL;
- ao2_unlock(tcptls_session);
-
- ao2_ref(tcptls_session, -1);
- tcptls_session = NULL;
- }
- return NULL;
-}
-
-static void peer_sched_cleanup(struct sip_peer *peer)
-{
- if (peer->pokeexpire != -1) {
- AST_SCHED_DEL_UNREF(sched, peer->pokeexpire,
- sip_unref_peer(peer, "removing poke peer ref"));
- }
- if (peer->expire != -1) {
- AST_SCHED_DEL_UNREF(sched, peer->expire,
- sip_unref_peer(peer, "remove register expire ref"));
- }
- if (peer->keepalivesend != -1) {
- AST_SCHED_DEL_UNREF(sched, peer->keepalivesend,
- sip_unref_peer(peer, "remove keepalive peer ref"));
- }
-}
-
-typedef enum {
- SIP_PEERS_MARKED,
- SIP_PEERS_ALL,
-} peer_unlink_flag_t;
-
-/* this func is used with ao2_callback to unlink/delete all marked or linked
- peers, depending on arg */
-static int match_and_cleanup_peer_sched(void *peerobj, void *arg, int flags)
-{
- struct sip_peer *peer = peerobj;
- peer_unlink_flag_t which = *(peer_unlink_flag_t *)arg;
-
- if (which == SIP_PEERS_ALL || peer->the_mark) {
- peer_sched_cleanup(peer);
- if (peer->dnsmgr) {
- ast_dnsmgr_release(peer->dnsmgr);
- peer->dnsmgr = NULL;
- sip_unref_peer(peer, "Release peer from dnsmgr");
- }
- return CMP_MATCH;
- }
- return 0;
-}
-
-static void unlink_peers_from_tables(peer_unlink_flag_t flag)
-{
- struct ao2_iterator *peers_iter;
-
- /*
- * We must remove the ref outside of the peers container to prevent
- * a deadlock condition when unsubscribing from stasis while it is
- * invoking a subscription event callback.
- */
- peers_iter = ao2_t_callback(peers, OBJ_UNLINK | OBJ_MULTIPLE,
- match_and_cleanup_peer_sched, &flag, "initiating callback to remove marked peers");
- if (peers_iter) {
- ao2_iterator_destroy(peers_iter);
- }
-
- peers_iter = ao2_t_callback(peers_by_ip, OBJ_UNLINK | OBJ_MULTIPLE,
- match_and_cleanup_peer_sched, &flag, "initiating callback to remove marked peers_by_ip");
- if (peers_iter) {
- ao2_iterator_destroy(peers_iter);
- }
-}
-
-/*! \brief Unlink all marked peers from ao2 containers */
-static void unlink_marked_peers_from_tables(void)
-{
- unlink_peers_from_tables(SIP_PEERS_MARKED);
-}
-
-static void unlink_all_peers_from_tables(void)
-{
- unlink_peers_from_tables(SIP_PEERS_ALL);
-}
-
-/*!
- * \internal
- * \brief maintain proper refcounts for a sip_pvt's outboundproxy
- *
- * This function sets pvt's outboundproxy pointer to the one referenced
- * by the proxy parameter. Because proxy may be a refcounted object, and
- * because pvt's old outboundproxy may also be a refcounted object, we need
- * to maintain the proper refcounts.
- *
- * \param pvt The sip_pvt for which we wish to set the outboundproxy
- * \param proxy The sip_proxy which we will point pvt towards.
- */
-static void ref_proxy(struct sip_pvt *pvt, struct sip_proxy *proxy)
-{
- struct sip_proxy *old_obproxy = pvt->outboundproxy;
- /* The sip_cfg.outboundproxy is statically allocated, and so
- * we don't ever need to adjust refcounts for it
- */
- if (proxy && proxy != &sip_cfg.outboundproxy) {
- ao2_ref(proxy, +1);
- }
- pvt->outboundproxy = proxy;
- if (old_obproxy && old_obproxy != &sip_cfg.outboundproxy) {
- ao2_ref(old_obproxy, -1);
- }
-}
-
-static void do_dialog_unlink_sched_items(struct sip_pvt *dialog)
-{
- struct sip_pkt *cp;
-
- /* remove all current packets in this dialog */
- sip_pvt_lock(dialog);
- while ((cp = dialog->packets)) {
- /* Unlink and destroy the packet object. */
- dialog->packets = dialog->packets->next;
- AST_SCHED_DEL_UNREF(sched, cp->retransid,
- ao2_t_ref(cp, -1, "Stop scheduled packet retransmission"));
- ao2_t_ref(cp, -1, "Packet retransmission list");
- }
- sip_pvt_unlock(dialog);
-
- AST_SCHED_DEL_UNREF(sched, dialog->waitid,
- dialog_unref(dialog, "Stop scheduled waitid"));
-
- AST_SCHED_DEL_UNREF(sched, dialog->initid,
- dialog_unref(dialog, "Stop scheduled initid"));
-
- AST_SCHED_DEL_UNREF(sched, dialog->reinviteid,
- dialog_unref(dialog, "Stop scheduled reinviteid"));
-
- AST_SCHED_DEL_UNREF(sched, dialog->autokillid,
- dialog_unref(dialog, "Stop scheduled autokillid"));
-
- AST_SCHED_DEL_UNREF(sched, dialog->request_queue_sched_id,
- dialog_unref(dialog, "Stop scheduled request_queue_sched_id"));
-
- AST_SCHED_DEL_UNREF(sched, dialog->provisional_keepalive_sched_id,
- dialog_unref(dialog, "Stop scheduled provisional keepalive"));
-
- AST_SCHED_DEL_UNREF(sched, dialog->t38id,
- dialog_unref(dialog, "Stop scheduled t38id"));
-
- if (dialog->stimer) {
- dialog->stimer->st_active = FALSE;
- do_stop_session_timer(dialog);
- }
-}
-
-/* Run by the sched thread. */
-static int __dialog_unlink_sched_items(const void *data)
-{
- struct sip_pvt *dialog = (void *) data;
-
- do_dialog_unlink_sched_items(dialog);
- dialog_unref(dialog, "Stop scheduled items for unlink action");
- return 0;
-}
-
-/*!
- * \brief Unlink a dialog from the dialogs container, as well as any other places
- * that it may be currently stored.
- *
- * \note A reference to the dialog must be held before calling this function, and this
- * function does not release that reference.
- */
-void dialog_unlink_all(struct sip_pvt *dialog)
-{
- struct ast_channel *owner;
-
- dialog_ref(dialog, "Let's bump the count in the unlink so it doesn't accidentally become dead before we are done");
-
- ao2_t_unlink(dialogs, dialog, "unlinking dialog via ao2_unlink");
- ao2_t_unlink(dialogs_needdestroy, dialog, "unlinking dialog_needdestroy via ao2_unlink");
- ao2_t_unlink(dialogs_rtpcheck, dialog, "unlinking dialog_rtpcheck via ao2_unlink");
-
- /* Unlink us from the owner (channel) if we have one */
- owner = sip_pvt_lock_full(dialog);
- if (owner) {
- ast_debug(1, "Detaching from channel %s\n", ast_channel_name(owner));
- ast_channel_tech_pvt_set(owner, dialog_unref(ast_channel_tech_pvt(owner), "resetting channel dialog ptr in unlink_all"));
- ast_channel_unlock(owner);
- ast_channel_unref(owner);
- sip_set_owner(dialog, NULL);
- }
- sip_pvt_unlock(dialog);
-
- if (dialog->registry) {
- if (dialog->registry->call == dialog) {
- dialog->registry->call = dialog_unref(dialog->registry->call, "nulling out the registry's call dialog field in unlink_all");
- }
- ao2_t_replace(dialog->registry, NULL, "delete dialog->registry");
- }
- if (dialog->stateid != -1) {
- ast_extension_state_del(dialog->stateid, cb_extensionstate);
- dialog->stateid = -1;
- }
- /* Remove link from peer to subscription of MWI */
- if (dialog->relatedpeer && dialog->relatedpeer->mwipvt == dialog) {
- dialog->relatedpeer->mwipvt = dialog_unref(dialog->relatedpeer->mwipvt, "delete ->relatedpeer->mwipvt");
- }
- if (dialog->relatedpeer && dialog->relatedpeer->call == dialog) {
- dialog->relatedpeer->call = dialog_unref(dialog->relatedpeer->call, "unset the relatedpeer->call field in tandem with relatedpeer field itself");
- }
-
- dialog_ref(dialog, "Stop scheduled items for unlink action");
- if (ast_sched_add(sched, 0, __dialog_unlink_sched_items, dialog) < 0) {
- /*
- * Uh Oh. Fall back to unscheduling things immediately
- * despite the potential deadlock risk.
- */
- dialog_unref(dialog, "Failed to schedule stop scheduled items for unlink action");
- do_dialog_unlink_sched_items(dialog);
- }
-
- dialog_unref(dialog, "Let's unbump the count in the unlink so the poor pvt can disappear if it is time");
-}
-
-static void append_history_full(struct sip_pvt *p, const char *fmt, ...)
- __attribute__((format(printf, 2, 3)));
-
-
-/*! \brief Convert transfer status to string */
-static const char *referstatus2str(enum referstatus rstatus)
-{
- return map_x_s(referstatusstrings, rstatus, "");
-}
-
-static inline void pvt_set_needdestroy(struct sip_pvt *pvt, const char *reason)
-{
- if (pvt->final_destruction_scheduled) {
- return; /* This is already scheduled for final destruction, let the scheduler take care of it. */
- }
- append_history(pvt, "NeedDestroy", "Setting needdestroy because %s", reason);
- if (!pvt->needdestroy) {
- pvt->needdestroy = 1;
- ao2_t_link(dialogs_needdestroy, pvt, "link pvt into dialogs_needdestroy container");
- }
-}
-
-/*! \brief Initialize the initital request packet in the pvt structure.
- This packet is used for creating replies and future requests in
- a dialog */
-static void initialize_initreq(struct sip_pvt *p, struct sip_request *req)
-{
- if (p->initreq.headers) {
- ast_debug(1, "Initializing already initialized SIP dialog %s (presumably reinvite)\n", p->callid);
- } else {
- ast_debug(1, "Initializing initreq for method %s - callid %s\n", sip_methods[req->method].text, p->callid);
- }
- /* Use this as the basis */
- copy_request(&p->initreq, req);
- parse_request(&p->initreq);
- if (req->debug) {
- ast_verbose("Initreq: %d headers, %d lines\n", p->initreq.headers, p->initreq.lines);
- }
-}
-
-/*! \brief Encapsulate setting of SIP_ALREADYGONE to be able to trace it with debugging */
-static void sip_alreadygone(struct sip_pvt *dialog)
-{
- ast_debug(3, "Setting SIP_ALREADYGONE on dialog %s\n", dialog->callid);
- dialog->alreadygone = 1;
-}
-
-/*! Resolve DNS srv name or host name in a sip_proxy structure */
-static int proxy_update(struct sip_proxy *proxy)
-{
- /* if it's actually an IP address and not a name,
- there's no need for a managed lookup */
- if (!ast_sockaddr_parse(&proxy->ip, proxy->name, 0)) {
- /* Ok, not an IP address, then let's check if it's a domain or host */
- /* XXX Todo - if we have proxy port, don't do SRV */
- proxy->ip.ss.ss_family = get_address_family_filter(AST_TRANSPORT_UDP); /* Filter address family */
- if (ast_get_ip_or_srv(&proxy->ip, proxy->name, sip_cfg.srvlookup ? "_sip._udp" : NULL) < 0) {
- ast_log(LOG_WARNING, "Unable to locate host '%s'\n", proxy->name);
- return FALSE;
- }
-
- }
-
- ast_sockaddr_set_port(&proxy->ip, proxy->port);
-
- proxy->last_dnsupdate = time(NULL);
- return TRUE;
-}
-
-/*! \brief Parse proxy string and return an ao2_alloc'd proxy. If dest is
- * non-NULL, no allocation is performed and dest is used instead.
- * On error NULL is returned. */
-static struct sip_proxy *proxy_from_config(const char *proxy, int sipconf_lineno, struct sip_proxy *dest)
-{
- char *mutable_proxy, *sep, *name;
- int allocated = 0;
-
- if (!dest) {
- dest = ao2_alloc(sizeof(struct sip_proxy), NULL);
- if (!dest) {
- ast_log(LOG_WARNING, "Unable to allocate config storage for proxy\n");
- return NULL;
- }
- allocated = 1;
- }
-
- /* Format is: [transport://]name[:port][,force] */
- mutable_proxy = ast_skip_blanks(ast_strdupa(proxy));
- sep = strchr(mutable_proxy, ',');
- if (sep) {
- *sep++ = '\0';
- dest->force = !strncasecmp(ast_skip_blanks(sep), "force", 5);
- } else {
- dest->force = FALSE;
- }
-
- sip_parse_host(mutable_proxy, sipconf_lineno, &name, &dest->port, &dest->transport);
-
- /* Check that there is a name at all */
- if (ast_strlen_zero(name)) {
- if (allocated) {
- ao2_ref(dest, -1);
- } else {
- dest->name[0] = '\0';
- }
- return NULL;
- }
- ast_copy_string(dest->name, name, sizeof(dest->name));
-
- /* Resolve host immediately */
- proxy_update(dest);
-
- return dest;
-}
-
-/*! \brief converts ascii port to int representation. If no
- * pt buffer is provided or the pt has errors when being converted
- * to an int value, the port provided as the standard is used.
- */
-unsigned int port_str2int(const char *pt, unsigned int standard)
-{
- int port = standard;
- if (ast_strlen_zero(pt) || (sscanf(pt, "%30d", &port) != 1) || (port < 1) || (port > 65535)) {
- port = standard;
- }
-
- return port;
-}
-
-/*! \brief Get default outbound proxy or global proxy */
-static struct sip_proxy *obproxy_get(struct sip_pvt *dialog, struct sip_peer *peer)
-{
- if (dialog && dialog->options && dialog->options->outboundproxy) {
- if (sipdebug) {
- ast_debug(1, "OBPROXY: Applying dialplan set OBproxy to this call\n");
- }
- append_history(dialog, "OBproxy", "Using dialplan obproxy %s", dialog->options->outboundproxy->name);
- return dialog->options->outboundproxy;
- }
- if (peer && peer->outboundproxy) {
- if (sipdebug) {
- ast_debug(1, "OBPROXY: Applying peer OBproxy to this call\n");
- }
- append_history(dialog, "OBproxy", "Using peer obproxy %s", peer->outboundproxy->name);
- return peer->outboundproxy;
- }
- if (sip_cfg.outboundproxy.name[0]) {
- if (sipdebug) {
- ast_debug(1, "OBPROXY: Applying global OBproxy to this call\n");
- }
- append_history(dialog, "OBproxy", "Using global obproxy %s", sip_cfg.outboundproxy.name);
- return &sip_cfg.outboundproxy;
- }
- if (sipdebug) {
- ast_debug(1, "OBPROXY: Not applying OBproxy to this call\n");
- }
- return NULL;
-}
-
-/*! \brief returns true if 'name' (with optional trailing whitespace)
- * matches the sip method 'id'.
- * Strictly speaking, SIP methods are case SENSITIVE, but we do
- * a case-insensitive comparison to be more tolerant.
- * following Jon Postel's rule: Be gentle in what you accept, strict with what you send
- */
-static int method_match(enum sipmethod id, const char *name)
-{
- int len = strlen(sip_methods[id].text);
- int l_name = name ? strlen(name) : 0;
- /* true if the string is long enough, and ends with whitespace, and matches */
- return (l_name >= len && name && name[len] < 33 &&
- !strncasecmp(sip_methods[id].text, name, len));
-}
-
-/*! \brief find_sip_method: Find SIP method from header */
-static int find_sip_method(const char *msg)
-{
- int i, res = 0;
-
- if (ast_strlen_zero(msg)) {
- return 0;
- }
- for (i = 1; i < ARRAY_LEN(sip_methods) && !res; i++) {
- if (method_match(i, msg)) {
- res = sip_methods[i].id;
- }
- }
- return res;
-}
-
-/*! \brief See if we pass debug IP filter */
-static inline int sip_debug_test_addr(const struct ast_sockaddr *addr)
-{
- /* Can't debug if sipdebug is not enabled */
- if (!sipdebug) {
- return 0;
- }
-
- /* A null debug_addr means we'll debug any address */
- if (ast_sockaddr_isnull(&debugaddr)) {
- return 1;
- }
-
- /* If no port was specified for a debug address, just compare the
- * addresses, otherwise compare the address and port
- */
- if (ast_sockaddr_port(&debugaddr)) {
- return !ast_sockaddr_cmp(&debugaddr, addr);
- } else {
- return !ast_sockaddr_cmp_addr(&debugaddr, addr);
- }
-}
-
-/*! \brief The real destination address for a write */
-static const struct ast_sockaddr *sip_real_dst(const struct sip_pvt *p)
-{
- if (p->outboundproxy) {
- return &p->outboundproxy->ip;
- }
-
- return ast_test_flag(&p->flags[0], SIP_NAT_FORCE_RPORT) || ast_test_flag(&p->flags[0], SIP_NAT_RPORT_PRESENT) ? &p->recv : &p->sa;
-}
-
-/*! \brief Display SIP nat mode */
-static const char *sip_nat_mode(const struct sip_pvt *p)
-{
- return ast_test_flag(&p->flags[0], SIP_NAT_FORCE_RPORT) ? "NAT" : "no NAT";
-}
-
-/*! \brief Test PVT for debugging output */
-static inline int sip_debug_test_pvt(struct sip_pvt *p)
-{
- if (!sipdebug) {
- return 0;
- }
- return sip_debug_test_addr(sip_real_dst(p));
-}
-
-/*! \brief Return int representing a bit field of transport types found in const char *transport */
-static int get_transport_str2enum(const char *transport)
-{
- int res = 0;
-
- if (ast_strlen_zero(transport)) {
- return res;
- }
-
- if (!strcasecmp(transport, "udp")) {
- res |= AST_TRANSPORT_UDP;
- }
- if (!strcasecmp(transport, "tcp")) {
- res |= AST_TRANSPORT_TCP;
- }
- if (!strcasecmp(transport, "tls")) {
- res |= AST_TRANSPORT_TLS;
- }
- if (!strcasecmp(transport, "ws")) {
- res |= AST_TRANSPORT_WS;
- }
- if (!strcasecmp(transport, "wss")) {
- res |= AST_TRANSPORT_WSS;
- }
-
- return res;
-}
-
-/*! \brief Return configuration of transports for a device */
-static inline const char *get_transport_list(unsigned int transports)
-{
- char *buf;
-
- if (!transports) {
- return "UNKNOWN";
- }
-
- if (!(buf = ast_threadstorage_get(&sip_transport_str_buf, SIP_TRANSPORT_STR_BUFSIZE))) {
- return "";
- }
-
- memset(buf, 0, SIP_TRANSPORT_STR_BUFSIZE);
-
- if (transports & AST_TRANSPORT_UDP) {
- strncat(buf, "UDP,", SIP_TRANSPORT_STR_BUFSIZE - strlen(buf));
- }
- if (transports & AST_TRANSPORT_TCP) {
- strncat(buf, "TCP,", SIP_TRANSPORT_STR_BUFSIZE - strlen(buf));
- }
- if (transports & AST_TRANSPORT_TLS) {
- strncat(buf, "TLS,", SIP_TRANSPORT_STR_BUFSIZE - strlen(buf));
- }
- if (transports & AST_TRANSPORT_WS) {
- strncat(buf, "WS,", SIP_TRANSPORT_STR_BUFSIZE - strlen(buf));
- }
- if (transports & AST_TRANSPORT_WSS) {
- strncat(buf, "WSS,", SIP_TRANSPORT_STR_BUFSIZE - strlen(buf));
- }
-
- /* Remove the trailing ',' if present */
- if (strlen(buf)) {
- buf[strlen(buf) - 1] = 0;
- }
-
- return buf;
-}
-
-/*! \brief Return transport as string */
-const char *sip_get_transport(enum ast_transport t)
-{
- switch (t) {
- case AST_TRANSPORT_UDP:
- return "UDP";
- case AST_TRANSPORT_TCP:
- return "TCP";
- case AST_TRANSPORT_TLS:
- return "TLS";
- case AST_TRANSPORT_WS:
- case AST_TRANSPORT_WSS:
- return "WS";
- }
-
- return "UNKNOWN";
-}
-
-/*! \brief Return protocol string for srv dns query */
-static inline const char *get_srv_protocol(enum ast_transport t)
-{
- switch (t) {
- case AST_TRANSPORT_UDP:
- return "udp";
- case AST_TRANSPORT_WS:
- return "ws";
- case AST_TRANSPORT_TLS:
- case AST_TRANSPORT_TCP:
- return "tcp";
- case AST_TRANSPORT_WSS:
- return "wss";
- }
-
- return "udp";
-}
-
-/*! \brief Return service string for srv dns query */
-static inline const char *get_srv_service(enum ast_transport t)
-{
- switch (t) {
- case AST_TRANSPORT_TCP:
- case AST_TRANSPORT_UDP:
- case AST_TRANSPORT_WS:
- return "sip";
- case AST_TRANSPORT_TLS:
- case AST_TRANSPORT_WSS:
- return "sips";
- }
- return "sip";
-}
-
-/*! \brief Return transport of dialog.
- \note this is based on a false assumption. We don't always use the
- outbound proxy for all requests in a dialog. It depends on the
- "force" parameter. The FIRST request is always sent to the ob proxy.
- \todo Fix this function to work correctly
-*/
-static inline const char *get_transport_pvt(struct sip_pvt *p)
-{
- if (p->outboundproxy && p->outboundproxy->transport) {
- set_socket_transport(&p->socket, p->outboundproxy->transport);
- }
-
- return sip_get_transport(p->socket.type);
-}
-
-/*!
- * \internal
- * \brief Transmit SIP message
- *
- * \details
- * Sends a SIP request or response on a given socket (in the pvt)
- * \note
- * Called by retrans_pkt, send_request, send_response and __sip_reliable_xmit
- *
- * \return length of transmitted message, XMIT_ERROR on known network failures -1 on other failures.
- */
-static int __sip_xmit(struct sip_pvt *p, struct ast_str *data)
-{
- int res = 0;
- const struct ast_sockaddr *dst = sip_real_dst(p);
-
- ast_debug(2, "Trying to put '%.11s' onto %s socket destined for %s\n", ast_str_buffer(data), get_transport_pvt(p), ast_sockaddr_stringify(dst));
-
- if (sip_prepare_socket(p) < 0) {
- return XMIT_ERROR;
- }
-
- if (p->socket.type == AST_TRANSPORT_UDP) {
- res = ast_sendto(p->socket.fd, ast_str_buffer(data), ast_str_strlen(data), 0, dst);
- } else if (p->socket.tcptls_session) {
- res = sip_tcptls_write(p->socket.tcptls_session, ast_str_buffer(data), ast_str_strlen(data));
- if (res < -1) {
- return res;
- }
- } else if (p->socket.ws_session) {
- if (!(res = ast_websocket_write_string(p->socket.ws_session, ast_str_buffer(data)))) {
- /* The WebSocket API just returns 0 on success and -1 on failure, while this code expects the payload length to be returned */
- res = ast_str_strlen(data);
- }
- } else {
- ast_debug(2, "Socket type is TCP but no tcptls_session is present to write to\n");
- return XMIT_ERROR;
- }
-
- if (res == -1) {
- switch (errno) {
- case EBADF: /* Bad file descriptor - seems like this is generated when the host exist, but doesn't accept the UDP packet */
- case EHOSTUNREACH: /* Host can't be reached */
- case ENETDOWN: /* Interface down */
- case ENETUNREACH: /* Network failure */
- case ECONNREFUSED: /* ICMP port unreachable */
- res = XMIT_ERROR; /* Don't bother with trying to transmit again */
- }
- }
- if (res != ast_str_strlen(data)) {
- ast_log(LOG_WARNING, "sip_xmit of %p (len %zu) to %s returned %d: %s\n", data, ast_str_strlen(data), ast_sockaddr_stringify(dst), res, strerror(errno));
- }
-
- return res;
-}
-
-/*! \brief Build a Via header for a request */
-static void build_via(struct sip_pvt *p)
-{
- /* Work around buggy UNIDEN UIP200 firmware */
- const char *rport = (ast_test_flag(&p->flags[0], SIP_NAT_FORCE_RPORT) || ast_test_flag(&p->flags[0], SIP_NAT_RPORT_PRESENT)) ? ";rport" : "";
-
- /* z9hG4bK is a magic cookie. See RFC 3261 section 8.1.1.7 */
- snprintf(p->via, sizeof(p->via), "SIP/2.0/%s %s;branch=z9hG4bK%08x%s",
- get_transport_pvt(p),
- ast_sockaddr_stringify_remote(&p->ourip),
- (unsigned)p->branch, rport);
-}
-
-/*! \brief NAT fix - decide which IP address to use for Asterisk server?
- *
- * Using the localaddr structure built up with localnet statements in sip.conf
- * apply it to their address to see if we need to substitute our
- * externaddr or can get away with our internal bindaddr
- * 'us' is always overwritten.
- */
-static void ast_sip_ouraddrfor(const struct ast_sockaddr *them, struct ast_sockaddr *us, struct sip_pvt *p)
-{
- struct ast_sockaddr theirs;
-
- /* Set want_remap to non-zero if we want to remap 'us' to an externally
- * reachable IP address and port. This is done if:
- * 1. we have a localaddr list (containing 'internal' addresses marked
- * as 'deny', so ast_apply_ha() will return AST_SENSE_DENY on them,
- * and AST_SENSE_ALLOW on 'external' ones);
- * 2. externaddr is set, so we know what to use as the
- * externally visible address;
- * 3. the remote address, 'them', is external;
- * 4. the address returned by ast_ouraddrfor() is 'internal' (AST_SENSE_DENY
- * when passed to ast_apply_ha() so it does need to be remapped.
- * This fourth condition is checked later.
- */
- int want_remap = 0;
-
- ast_sockaddr_copy(us, &internip); /* starting guess for the internal address */
- /* now ask the system what would it use to talk to 'them' */
- ast_ouraddrfor(them, us);
- ast_sockaddr_copy(&theirs, them);
-
- if (ast_sockaddr_is_ipv6(&theirs) && !ast_sockaddr_is_ipv4_mapped(&theirs)) {
- if (localaddr && !ast_sockaddr_isnull(&externaddr) && !ast_sockaddr_is_any(&bindaddr)) {
- ast_log(LOG_WARNING, "Address remapping activated in sip.conf "
- "but we're using IPv6, which doesn't need it. Please "
- "remove \"localnet\" and/or \"externaddr\" settings.\n");
- }
- } else {
- want_remap = localaddr &&
- !ast_sockaddr_isnull(&externaddr) &&
- ast_apply_ha(localaddr, &theirs) == AST_SENSE_ALLOW ;
- }
-
- if (want_remap &&
- (!sip_cfg.matchexternaddrlocally || !ast_apply_ha(localaddr, us)) ) {
- /* if we used externhost, see if it is time to refresh the info */
- if (externexpire && time(NULL) >= externexpire) {
- if (ast_sockaddr_resolve_first_af(&externaddr, externhost, 0, AST_AF_INET)) {
- ast_log(LOG_NOTICE, "Warning: Re-lookup of '%s' failed!\n", externhost);
- }
- externexpire = time(NULL) + externrefresh;
- }
- if (!ast_sockaddr_isnull(&externaddr)) {
- ast_sockaddr_copy(us, &externaddr);
- switch (p->socket.type) {
- case AST_TRANSPORT_TCP:
- if (!externtcpport && ast_sockaddr_port(&externaddr)) {
- /* for consistency, default to the externaddr port */
- externtcpport = ast_sockaddr_port(&externaddr);
- }
- if (!externtcpport) {
- externtcpport = ast_sockaddr_port(&sip_tcp_desc.local_address);
- }
- if (!externtcpport) {
- externtcpport = STANDARD_SIP_PORT;
- }
- ast_sockaddr_set_port(us, externtcpport);
- break;
- case AST_TRANSPORT_TLS:
- if (!externtlsport) {
- externtlsport = ast_sockaddr_port(&sip_tls_desc.local_address);
- }
- if (!externtlsport) {
- externtlsport = STANDARD_TLS_PORT;
- }
- ast_sockaddr_set_port(us, externtlsport);
- break;
- case AST_TRANSPORT_UDP:
- if (!ast_sockaddr_port(&externaddr)) {
- ast_sockaddr_set_port(us, ast_sockaddr_port(&bindaddr));
- }
- break;
- default:
- break;
- }
- }
- ast_debug(1, "Target address %s is not local, substituting externaddr\n",
- ast_sockaddr_stringify(them));
- } else {
- /* no remapping, but we bind to a specific address, so use it. */
- switch (p->socket.type) {
- case AST_TRANSPORT_TCP:
- if (!ast_sockaddr_isnull(&sip_tcp_desc.local_address)) {
- if (!ast_sockaddr_is_any(&sip_tcp_desc.local_address)) {
- ast_sockaddr_copy(us,
- &sip_tcp_desc.local_address);
- } else {
- ast_sockaddr_set_port(us,
- ast_sockaddr_port(&sip_tcp_desc.local_address));
- }
- break;
- } /* fall through on purpose */
- case AST_TRANSPORT_TLS:
- if (!ast_sockaddr_isnull(&sip_tls_desc.local_address)) {
- if (!ast_sockaddr_is_any(&sip_tls_desc.local_address)) {
- ast_sockaddr_copy(us,
- &sip_tls_desc.local_address);
- } else {
- ast_sockaddr_set_port(us,
- ast_sockaddr_port(&sip_tls_desc.local_address));
- }
- break;
- } /* fall through on purpose */
- case AST_TRANSPORT_UDP:
- /* fall through on purpose */
- default:
- if (!ast_sockaddr_is_any(&bindaddr)) {
- ast_sockaddr_copy(us, &bindaddr);
- }
- if (!ast_sockaddr_port(us)) {
- ast_sockaddr_set_port(us, ast_sockaddr_port(&bindaddr));
- }
- }
- }
- ast_debug(3, "Setting AST_TRANSPORT_%s with address %s\n", sip_get_transport(p->socket.type), ast_sockaddr_stringify(us));
-}
-
-/*! \brief Append to SIP dialog history with arg list */
-static __attribute__((format(printf, 2, 0))) void append_history_va(struct sip_pvt *p, const char *fmt, va_list ap)
-{
- char buf[80], *c = buf; /* max history length */
- struct sip_history *hist;
- int l;
-
- vsnprintf(buf, sizeof(buf), fmt, ap);
- strsep(&c, "\r\n"); /* Trim up everything after \r or \n */
- l = strlen(buf) + 1;
- if (!(hist = ast_calloc(1, sizeof(*hist) + l))) {
- return;
- }
- if (!p->history && !(p->history = ast_calloc(1, sizeof(*p->history)))) {
- ast_free(hist);
- return;
- }
- memcpy(hist->event, buf, l);
- if (p->history_entries == MAX_HISTORY_ENTRIES) {
- struct sip_history *oldest;
- oldest = AST_LIST_REMOVE_HEAD(p->history, list);
- p->history_entries--;
- ast_free(oldest);
- }
- AST_LIST_INSERT_TAIL(p->history, hist, list);
- p->history_entries++;
- if (log_level != -1) {
- ast_log_dynamic_level(log_level, "%s\n", buf);
- }
-}
-
-/*! \brief Append to SIP dialog history with arg list */
-static void append_history_full(struct sip_pvt *p, const char *fmt, ...)
-{
- va_list ap;
-
- if (!p) {
- return;
- }
-
- if (!p->do_history && !recordhistory && !dumphistory) {
- return;
- }
-
- va_start(ap, fmt);
- append_history_va(p, fmt, ap);
- va_end(ap);
-
- return;
-}
-
-/*!
- * \brief Retransmit SIP message if no answer
- *
- * \note Run by the sched thread.
- */
-static int retrans_pkt(const void *data)
-{
- struct sip_pkt *pkt = (struct sip_pkt *) data;
- struct sip_pkt *prev;
- struct sip_pkt *cur;
- struct ast_channel *owner_chan;
- int reschedule = DEFAULT_RETRANS;
- int xmitres = 0;
- /* how many ms until retrans timeout is reached */
- int64_t diff = pkt->retrans_stop_time - ast_tvdiff_ms(ast_tvnow(), pkt->time_sent);
-
- /* Do not retransmit if time out is reached. This will be negative if the time between
- * the first transmission and now is larger than our timeout period. This is a fail safe
- * check in case the scheduler gets behind or the clock is changed. */
- if ((diff <= 0) || (diff > pkt->retrans_stop_time)) {
- pkt->retrans_stop = 1;
- }
-
- /* Lock channel PVT */
- sip_pvt_lock(pkt->owner);
-
- if (!pkt->retrans_stop) {
- pkt->retrans++;
- if (!pkt->timer_t1) { /* Re-schedule using timer_a and timer_t1 */
- if (sipdebug) {
- ast_debug(4, "SIP TIMER: Not rescheduling id #%d:%s (Method %d) (No timer T1)\n",
- pkt->retransid,
- sip_methods[pkt->method].text,
- pkt->method);
- }
- } else {
- int siptimer_a;
-
- if (sipdebug) {
- ast_debug(4, "SIP TIMER: Rescheduling retransmission #%d (%d) %s - %d\n",
- pkt->retransid,
- pkt->retrans,
- sip_methods[pkt->method].text,
- pkt->method);
- }
- if (!pkt->timer_a) {
- pkt->timer_a = 2 ;
- } else {
- pkt->timer_a = 2 * pkt->timer_a;
- }
-
- /* For non-invites, a maximum of 4 secs */
- if (INT_MAX / pkt->timer_a < pkt->timer_t1) {
- /*
- * Uh Oh, we will have an integer overflow.
- * Recalculate previous timeout time instead.
- */
- pkt->timer_a = pkt->timer_a / 2;
- }
- siptimer_a = pkt->timer_t1 * pkt->timer_a; /* Double each time */
- if (pkt->method != SIP_INVITE && siptimer_a > 4000) {
- siptimer_a = 4000;
- }
-
- /* Reschedule re-transmit */
- reschedule = siptimer_a;
- ast_debug(4, "** SIP timers: Rescheduling retransmission %d to %d ms (t1 %d ms (Retrans id #%d)) \n",
- pkt->retrans + 1,
- siptimer_a,
- pkt->timer_t1,
- pkt->retransid);
- }
-
- if (sip_debug_test_pvt(pkt->owner)) {
- const struct ast_sockaddr *dst = sip_real_dst(pkt->owner);
-
- ast_verbose("Retransmitting #%d (%s) to %s:\n%s\n---\n",
- pkt->retrans, sip_nat_mode(pkt->owner),
- ast_sockaddr_stringify(dst),
- ast_str_buffer(pkt->data));
- }
-
- append_history(pkt->owner, "ReTx", "%d %s", reschedule, ast_str_buffer(pkt->data));
- xmitres = __sip_xmit(pkt->owner, pkt->data);
-
- /* If there was no error during the network transmission, schedule the next retransmission,
- * but if the next retransmission is going to be beyond our timeout period, mark the packet's
- * stop_retrans value and set the next retransmit to be the exact time of timeout. This will
- * allow any responses to the packet to be processed before the packet is destroyed on the next
- * call to this function by the scheduler. */
- if (xmitres != XMIT_ERROR) {
- if (reschedule >= diff) {
- pkt->retrans_stop = 1;
- reschedule = diff;
- }
- sip_pvt_unlock(pkt->owner);
- return reschedule;
- }
- }
-
- /* At this point, either the packet's retransmission timed out, or there was a
- * transmission error, either way destroy the scheduler item and this packet. */
-
- pkt->retransid = -1; /* Kill this scheduler item */
-
- if (pkt->method != SIP_OPTIONS && xmitres == 0) {
- if (pkt->is_fatal || sipdebug) { /* Tell us if it's critical or if we're debugging */
- ast_log(LOG_WARNING, "Retransmission timeout reached on transmission %s for seqno %u (%s %s) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions\n"
- "Packet timed out after %dms with no response\n",
- pkt->owner->callid,
- pkt->seqno,
- pkt->is_fatal ? "Critical" : "Non-critical",
- pkt->is_resp ? "Response" : "Request",
- (int) ast_tvdiff_ms(ast_tvnow(), pkt->time_sent));
- }
- } else if (pkt->method == SIP_OPTIONS && sipdebug) {
- ast_log(LOG_WARNING, "Cancelling retransmit of OPTIONs (call id %s) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions\n", pkt->owner->callid);
- }
-
- if (xmitres == XMIT_ERROR) {
- ast_log(LOG_WARNING, "Transmit error :: Cancelling transmission on Call ID %s\n", pkt->owner->callid);
- append_history(pkt->owner, "XmitErr", "%s", pkt->is_fatal ? "(Critical)" : "(Non-critical)");
- } else {
- append_history(pkt->owner, "MaxRetries", "%s", pkt->is_fatal ? "(Critical)" : "(Non-critical)");
- }
-
- sip_pvt_unlock(pkt->owner); /* SIP_PVT, not channel */
- owner_chan = sip_pvt_lock_full(pkt->owner);
-
- if (pkt->is_fatal) {
- if (owner_chan) {
- ast_log(LOG_WARNING, "Hanging up call %s - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).\n", pkt->owner->callid);
-
- if (pkt->is_resp &&
- (pkt->response_code >= 200) &&
- (pkt->response_code < 300) &&
- pkt->owner->pendinginvite &&
- ast_test_flag(&pkt->owner->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED)) {
- /* This is a timeout of the 2XX response to a pending INVITE. In this case terminate the INVITE
- * transaction just as if we received the ACK, but immediately hangup with a BYE (sip_hangup
- * will send the BYE as long as the dialog is not set as "alreadygone")
- * RFC 3261 section 13.3.1.4.
- * "If the server retransmits the 2xx response for 64*T1 seconds without receiving
- * an ACK, the dialog is confirmed, but the session SHOULD be terminated. This is
- * accomplished with a BYE, as described in Section 15." */
- pkt->owner->invitestate = INV_TERMINATED;
- pkt->owner->pendinginvite = 0;
- } else {
- /* there is nothing left to do, mark the dialog as gone */
- sip_alreadygone(pkt->owner);
- }
- if (!ast_channel_hangupcause(owner_chan)) {
- ast_channel_hangupcause_set(owner_chan, AST_CAUSE_NO_USER_RESPONSE);
- }
- ast_queue_hangup_with_cause(owner_chan, AST_CAUSE_NO_USER_RESPONSE);
- } else {
- /* If no channel owner, destroy now */
-
- /* Let the peerpoke system expire packets when the timer expires for poke_noanswer */
- if (pkt->method != SIP_OPTIONS && pkt->method != SIP_REGISTER) {
- pvt_set_needdestroy(pkt->owner, "no response to critical packet");
- sip_alreadygone(pkt->owner);
- append_history(pkt->owner, "DialogKill", "Killing this failed dialog immediately");
- }
- }
- } else if (pkt->owner->pendinginvite == pkt->seqno) {
- ast_log(LOG_WARNING, "Timeout on %s on non-critical invite transaction.\n", pkt->owner->callid);
- pkt->owner->invitestate = INV_TERMINATED;
- pkt->owner->pendinginvite = 0;
- check_pendings(pkt->owner);
- }
-
- if (owner_chan) {
- ast_channel_unlock(owner_chan);
- ast_channel_unref(owner_chan);
- }
-
- if (pkt->method == SIP_BYE) {
- /* We're not getting answers on SIP BYE's. Tear down the call anyway. */
- sip_alreadygone(pkt->owner);
- append_history(pkt->owner, "ByeFailure", "Remote peer doesn't respond to bye. Destroying call anyway.");
- pvt_set_needdestroy(pkt->owner, "no response to BYE");
- }
-
- /* Unlink and destroy the packet object. */
- for (prev = NULL, cur = pkt->owner->packets; cur; prev = cur, cur = cur->next) {
- if (cur == pkt) {
- /* Unlink the node from the list. */
- UNLINK(cur, pkt->owner->packets, prev);
- ao2_t_ref(pkt, -1, "Packet retransmission list (retransmission complete)");
- break;
- }
- }
-
- /*
- * If the object was not in the list then we were in the process of
- * stopping retransmisions while we were sending this retransmission.
- */
-
- sip_pvt_unlock(pkt->owner);
- ao2_t_ref(pkt, -1, "Scheduled packet retransmission complete");
- return 0;
-}
-
-/* Run by the sched thread. */
-static int __stop_retrans_pkt(const void *data)
-{
- struct sip_pkt *pkt = (void *) data;
-
- AST_SCHED_DEL_UNREF(sched, pkt->retransid,
- ao2_t_ref(pkt, -1, "Stop scheduled packet retransmission"));
- ao2_t_ref(pkt, -1, "Stop packet retransmission action");
- return 0;
-}
-
-static void stop_retrans_pkt(struct sip_pkt *pkt)
-{
- ao2_t_ref(pkt, +1, "Stop packet retransmission action");
- if (ast_sched_add(sched, 0, __stop_retrans_pkt, pkt) < 0) {
- /* Uh Oh. Expect bad behavior. */
- ao2_t_ref(pkt, -1, "Failed to schedule stop packet retransmission action");
- }
-}
-
-static void sip_pkt_dtor(void *vdoomed)
-{
- struct sip_pkt *pkt = (void *) vdoomed;
-
- if (pkt->owner) {
- dialog_unref(pkt->owner, "Retransmission packet is being destroyed");
- }
- ast_free(pkt->data);
-}
-
-/*!
- * \internal
- * \brief Transmit packet with retransmits
- * \retval 0 on success
- * \retval -1 on failure to allocate packet.
- */
-static enum sip_result __sip_reliable_xmit(struct sip_pvt *p, uint32_t seqno, int resp, struct ast_str *data, int fatal, int sipmethod)
-{
- struct sip_pkt *pkt = NULL;
- int siptimer_a = DEFAULT_RETRANS;
- int xmitres = 0;
- unsigned respid;
-
- if (sipmethod == SIP_INVITE) {
- /* Note this is a pending invite */
- p->pendinginvite = seqno;
- }
-
- pkt = ao2_alloc_options(sizeof(*pkt), sip_pkt_dtor, AO2_ALLOC_OPT_LOCK_NOLOCK);
- if (!pkt) {
- return AST_FAILURE;
- }
- /* copy data, add a terminator and save length */
- pkt->data = ast_str_create(ast_str_strlen(data));
- if (!pkt->data) {
- ao2_t_ref(pkt, -1, "Failed to initialize");
- return AST_FAILURE;
- }
- ast_str_set(&pkt->data, 0, "%s%s", ast_str_buffer(data), "\0");
- /* copy other parameters from the caller */
- pkt->method = sipmethod;
- pkt->seqno = seqno;
- pkt->is_resp = resp;
- pkt->is_fatal = fatal;
- pkt->owner = dialog_ref(p, "__sip_reliable_xmit: setting pkt->owner");
-
- /* The retransmission list owns a pkt ref */
- pkt->next = p->packets;
- p->packets = pkt; /* Add it to the queue */
-
- if (resp) {
- /* Parse out the response code */
- if (sscanf(ast_str_buffer(pkt->data), "SIP/2.0 %30u", &respid) == 1) {
- pkt->response_code = respid;
- }
- }
- pkt->timer_t1 = p->timer_t1; /* Set SIP timer T1 */
- if (pkt->timer_t1) {
- siptimer_a = pkt->timer_t1;
- }
-
- pkt->time_sent = ast_tvnow(); /* time packet was sent */
- pkt->retrans_stop_time = 64 * (pkt->timer_t1 ? pkt->timer_t1 : DEFAULT_TIMER_T1); /* time in ms after pkt->time_sent to stop retransmission */
-
- if (!(p->socket.type & AST_TRANSPORT_UDP)) {
- /* TCP does not need retransmits as that's built in, but with
- * retrans_stop set, we must give it the full timer_H treatment */
- pkt->retrans_stop = 1;
- siptimer_a = pkt->retrans_stop_time;
- }
-
- /* Schedule retransmission */
- ao2_t_ref(pkt, +1, "Schedule packet retransmission");
- pkt->retransid = ast_sched_add_variable(sched, siptimer_a, retrans_pkt, pkt, 1);
- if (pkt->retransid < 0) {
- ao2_t_ref(pkt, -1, "Failed to schedule packet retransmission");
- }
-
- if (sipdebug) {
- ast_debug(4, "*** SIP TIMER: Initializing retransmit timer on packet: Id #%d\n", pkt->retransid);
- }
-
- xmitres = __sip_xmit(pkt->owner, pkt->data); /* Send packet */
-
- if (xmitres == XMIT_ERROR) { /* Serious network trouble, no need to try again */
- append_history(pkt->owner, "XmitErr", "%s", pkt->is_fatal ? "(Critical)" : "(Non-critical)");
- ast_log(LOG_ERROR, "Serious Network Trouble; __sip_xmit returns error for pkt data\n");
-
- /* Unlink and destroy the packet object. */
- p->packets = pkt->next;
- stop_retrans_pkt(pkt);
- ao2_t_ref(pkt, -1, "Packet retransmission list");
- return AST_FAILURE;
- } else {
- /* This is odd, but since the retrans timer starts at 500ms and the do_monitor thread
- * only wakes up every 1000ms by default, we have to poke the thread here to make
- * sure it successfully detects this must be retransmitted in less time than
- * it usually sleeps for. Otherwise it might not retransmit this packet for 1000ms. */
- if (monitor_thread != AST_PTHREADT_NULL) {
- pthread_kill(monitor_thread, SIGURG);
- }
- return AST_SUCCESS;
- }
-}
-
-/*! \brief Kill a SIP dialog (called only by the scheduler)
- * The scheduler has a reference to this dialog when p->autokillid != -1,
- * and we are called using that reference. So if the event is not
- * rescheduled, we need to call dialog_unref().
- */
-static int __sip_autodestruct(const void *data)
-{
- struct sip_pvt *p = (struct sip_pvt *)data;
- struct ast_channel *owner;
-
- /* If this is a subscription, tell the phone that we got a timeout */
- if (p->subscribed && p->subscribed != MWI_NOTIFICATION && p->subscribed != CALL_COMPLETION) {
- struct state_notify_data data = { 0, };
-
- data.state = AST_EXTENSION_DEACTIVATED;
-
- transmit_state_notify(p, &data, 1, TRUE); /* Send last notification */
- p->subscribed = NONE;
- append_history(p, "Subscribestatus", "timeout");
- ast_debug(3, "Re-scheduled destruction of SIP subscription %s\n", p->callid ? p->callid : "");
- return 10000; /* Reschedule this destruction so that we know that it's gone */
- }
-
- /* If there are packets still waiting for delivery, delay the destruction */
- if (p->packets) {
- if (!p->needdestroy) {
- char method_str[31];
-
- ast_debug(3, "Re-scheduled destruction of SIP call %s\n", p->callid ? p->callid : "");
- append_history(p, "ReliableXmit", "timeout");
- if (sscanf(p->lastmsg, "Tx: %30s", method_str) == 1 || sscanf(p->lastmsg, "Rx: %30s", method_str) == 1) {
- if (p->ongoing_reinvite || method_match(SIP_CANCEL, method_str) || method_match(SIP_BYE, method_str)) {
- pvt_set_needdestroy(p, "autodestruct");
- }
- }
- return 10000;
- } else {
- /* They've had their chance to respond. Time to bail */
- __sip_pretend_ack(p);
- }
- }
-
- /*
- * Lock both the pvt and the channel safely so that we can queue up a frame.
- */
- owner = sip_pvt_lock_full(p);
- if (owner) {
- ast_log(LOG_WARNING,
- "Autodestruct on dialog '%s' with owner %s in place (Method: %s). Rescheduling destruction for 10000 ms\n",
- p->callid, ast_channel_name(owner), sip_methods[p->method].text);
- ast_queue_hangup_with_cause(owner, AST_CAUSE_PROTOCOL_ERROR);
- ast_channel_unlock(owner);
- ast_channel_unref(owner);
- sip_pvt_unlock(p);
- return 10000;
- }
-
- /* Reset schedule ID */
- p->autokillid = -1;
-
- if (p->refer && !p->alreadygone) {
- ast_debug(3, "Finally hanging up channel after transfer: %s\n", p->callid);
- stop_media_flows(p);
- transmit_request_with_auth(p, SIP_BYE, 0, XMIT_RELIABLE, 1);
- append_history(p, "ReferBYE", "Sending BYE on transferer call leg %s", p->callid);
- sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
- sip_pvt_unlock(p);
- } else {
- append_history(p, "AutoDestroy", "%s", p->callid);
- ast_debug(3, "Auto destroying SIP dialog '%s'\n", p->callid);
- sip_pvt_unlock(p);
- dialog_unlink_all(p); /* once it's unlinked and unrefd everywhere, it'll be freed automagically */
- }
-
- dialog_unref(p, "autokillid complete");
-
- return 0;
-}
-
-static void do_cancel_destroy(struct sip_pvt *pvt)
-{
- if (-1 < pvt->autokillid) {
- append_history(pvt, "CancelDestroy", "");
- AST_SCHED_DEL_UNREF(sched, pvt->autokillid,
- dialog_unref(pvt, "Stop scheduled autokillid"));
- }
-}
-
-/* Run by the sched thread. */
-static int __sip_cancel_destroy(const void *data)
-{
- struct sip_pvt *pvt = (void *) data;
-
- sip_pvt_lock(pvt);
- do_cancel_destroy(pvt);
- sip_pvt_unlock(pvt);
- dialog_unref(pvt, "Cancel destroy action");
- return 0;
-}
-
-void sip_cancel_destroy(struct sip_pvt *pvt)
-{
- if (pvt->final_destruction_scheduled) {
- return;
- }
-
- dialog_ref(pvt, "Cancel destroy action");
- if (ast_sched_add(sched, 0, __sip_cancel_destroy, pvt) < 0) {
- /* Uh Oh. Expect bad behavior. */
- dialog_unref(pvt, "Failed to schedule cancel destroy action");
- ast_log(LOG_WARNING, "Unable to cancel SIP destruction. Expect bad things.\n");
- }
-}
-
-struct sip_scheddestroy_data {
- struct sip_pvt *pvt;
- int ms;
-};
-
-/* Run by the sched thread. */
-static int __sip_scheddestroy(const void *data)
-{
- struct sip_scheddestroy_data *sched_data = (void *) data;
- struct sip_pvt *pvt = sched_data->pvt;
- int ms = sched_data->ms;
-
- ast_free(sched_data);
-
- sip_pvt_lock(pvt);
- do_cancel_destroy(pvt);
-
- if (pvt->do_history) {
- append_history(pvt, "SchedDestroy", "%d ms", ms);
- }
-
- dialog_ref(pvt, "Schedule autokillid");
- pvt->autokillid = ast_sched_add(sched, ms, __sip_autodestruct, pvt);
- if (pvt->autokillid < 0) {
- /* Uh Oh. Expect bad behavior. */
- dialog_unref(pvt, "Failed to schedule autokillid");
- }
-
- if (pvt->stimer) {
- stop_session_timer(pvt);
- }
- sip_pvt_unlock(pvt);
- dialog_unref(pvt, "Destroy action");
- return 0;
-}
-
-static int sip_scheddestroy_full(struct sip_pvt *p, int ms)
-{
- struct sip_scheddestroy_data *sched_data;
-
- if (ms < 0) {
- if (p->timer_t1 == 0) {
- p->timer_t1 = global_t1; /* Set timer T1 if not set (RFC 3261) */
- }
- if (p->timer_b == 0) {
- p->timer_b = global_timer_b; /* Set timer B if not set (RFC 3261) */
- }
- ms = p->timer_t1 * 64;
- }
- if (sip_debug_test_pvt(p)) {
- ast_verbose("Scheduling destruction of SIP dialog '%s' in %d ms (Method: %s)\n",
- p->callid, ms, sip_methods[p->method].text);
- }
-
- sched_data = ast_malloc(sizeof(*sched_data));
- if (!sched_data) {
- /* Uh Oh. Expect bad behavior. */
- return -1;
- }
- sched_data->pvt = p;
- sched_data->ms = ms;
- dialog_ref(p, "Destroy action");
- if (ast_sched_add(sched, 0, __sip_scheddestroy, sched_data) < 0) {
- /* Uh Oh. Expect bad behavior. */
- dialog_unref(p, "Failed to schedule destroy action");
- ast_free(sched_data);
- return -1;
- }
- return 0;
-}
-
-void sip_scheddestroy(struct sip_pvt *p, int ms)
-{
- if (p->final_destruction_scheduled) {
- return; /* already set final destruction */
- }
- sip_scheddestroy_full(p, ms);
-}
-
-void sip_scheddestroy_final(struct sip_pvt *p, int ms)
-{
- if (p->final_destruction_scheduled) {
- return; /* already set final destruction */
- }
-
- if (!sip_scheddestroy_full(p, ms)) {
- p->final_destruction_scheduled = 1;
- }
-}
-
-/*! \brief Acknowledges receipt of a packet and stops retransmission
- * called with p locked*/
-int __sip_ack(struct sip_pvt *p, uint32_t seqno, int resp, int sipmethod)
-{
- struct sip_pkt *cur, *prev = NULL;
- const char *msg = "Not Found"; /* used only for debugging */
- int res = FALSE;
-
- /* If we have an outbound proxy for this dialog, then delete it now since
- the rest of the requests in this dialog needs to follow the routing.
- If obforcing is set, we will keep the outbound proxy during the whole
- dialog, regardless of what the SIP rfc says
- */
- if (p->outboundproxy && !p->outboundproxy->force) {
- ref_proxy(p, NULL);
- }
-
- for (cur = p->packets; cur; prev = cur, cur = cur->next) {
- if (cur->seqno != seqno || cur->is_resp != resp) {
- continue;
- }
- if (cur->is_resp || cur->method == sipmethod) {
- res = TRUE;
- msg = "Found";
- if (!resp && (seqno == p->pendinginvite)) {
- ast_debug(1, "Acked pending invite %u\n", p->pendinginvite);
- p->pendinginvite = 0;
- }
- if (cur->retransid > -1) {
- if (sipdebug)
- ast_debug(4, "** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #%d\n", cur->retransid);
- }
-
- /* Unlink and destroy the packet object. */
- UNLINK(cur, p->packets, prev);
- stop_retrans_pkt(cur);
- ao2_t_ref(cur, -1, "Packet retransmission list");
- break;
- }
- }
- ast_debug(1, "Stopping retransmission on '%s' of %s %u: Match %s\n",
- p->callid, resp ? "Response" : "Request", seqno, msg);
- return res;
-}
-
-/*! \brief Pretend to ack all packets
- * called with p locked */
-void __sip_pretend_ack(struct sip_pvt *p)
-{
- struct sip_pkt *cur = NULL;
-
- while (p->packets) {
- int method;
- if (cur == p->packets) {
- ast_log(LOG_WARNING, "Have a packet that doesn't want to give up! %s\n", sip_methods[cur->method].text);
- return;
- }
- cur = p->packets;
- method = (cur->method) ? cur->method : find_sip_method(ast_str_buffer(cur->data));
- __sip_ack(p, cur->seqno, cur->is_resp, method);
- }
-}
-
-/*! \brief Acks receipt of packet, keep it around (used for provisional responses) */
-int __sip_semi_ack(struct sip_pvt *p, uint32_t seqno, int resp, int sipmethod)
-{
- struct sip_pkt *cur;
- int res = FALSE;
-
- for (cur = p->packets; cur; cur = cur->next) {
- if (cur->seqno == seqno && cur->is_resp == resp &&
- (cur->is_resp || method_match(sipmethod, ast_str_buffer(cur->data)))) {
- /* this is our baby */
- if (cur->retransid > -1) {
- if (sipdebug)
- ast_debug(4, "*** SIP TIMER: Cancelling retransmission #%d - %s (got response)\n", cur->retransid, sip_methods[sipmethod].text);
- }
- stop_retrans_pkt(cur);
- res = TRUE;
- break;
- }
- }
- ast_debug(1, "(Provisional) Stopping retransmission (but retaining packet) on '%s' %s %u: %s\n", p->callid, resp ? "Response" : "Request", seqno, res == -1 ? "Not Found" : "Found");
- return res;
-}
-
-
-/*! \brief Copy SIP request, parse it */
-static void parse_copy(struct sip_request *dst, const struct sip_request *src)
-{
- copy_request(dst, src);
- parse_request(dst);
-}
-
-/*! \brief add a blank line if no body */
-static void add_blank(struct sip_request *req)
-{
- if (!req->lines) {
- /* Add extra empty return. add_header() reserves 4 bytes so cannot be truncated */
- ast_str_append(&req->data, 0, "\r\n");
- }
-}
-
-/* Run by the sched thread. */
-static int send_provisional_keepalive_full(struct sip_pvt *pvt, int with_sdp)
-{
- const char *msg = NULL;
- struct ast_channel *chan;
- int res = 0;
-
- chan = sip_pvt_lock_full(pvt);
-
- if (!pvt->last_provisional || !strncasecmp(pvt->last_provisional, "100", 3)) {
- msg = "183 Session Progress";
- }
-
- if (pvt->invitestate < INV_COMPLETED) {
- if (with_sdp) {
- transmit_response_with_sdp(pvt, S_OR(msg, pvt->last_provisional), &pvt->initreq, XMIT_UNRELIABLE, FALSE, FALSE);
- } else {
- transmit_response(pvt, S_OR(msg, pvt->last_provisional), &pvt->initreq);
- }
- res = PROVIS_KEEPALIVE_TIMEOUT;
- } else {
- pvt->provisional_keepalive_sched_id = -1;
- }
-
- sip_pvt_unlock(pvt);
- if (chan) {
- ast_channel_unlock(chan);
- ast_channel_unref(chan);
- }
-
- if (!res) {
- dialog_unref(pvt, "Schedule provisional keepalive complete");
- }
- return res;
-}
-
-/* Run by the sched thread. */
-static int send_provisional_keepalive(const void *data)
-{
- struct sip_pvt *pvt = (struct sip_pvt *) data;
-
- return send_provisional_keepalive_full(pvt, 0);
-}
-
-/* Run by the sched thread. */
-static int send_provisional_keepalive_with_sdp(const void *data)
-{
- struct sip_pvt *pvt = (void *) data;
-
- return send_provisional_keepalive_full(pvt, 1);
-}
-
-/* Run by the sched thread. */
-static int __update_provisional_keepalive_full(struct sip_pvt *pvt, int with_sdp)
-{
- AST_SCHED_DEL_UNREF(sched, pvt->provisional_keepalive_sched_id,
- dialog_unref(pvt, "Stop scheduled provisional keepalive for update"));
-
- sip_pvt_lock(pvt);
- if (pvt->invitestate < INV_COMPLETED) {
- /* Provisional keepalive is still needed. */
- dialog_ref(pvt, "Schedule provisional keepalive");
- pvt->provisional_keepalive_sched_id = ast_sched_add(sched, PROVIS_KEEPALIVE_TIMEOUT,
- with_sdp ? send_provisional_keepalive_with_sdp : send_provisional_keepalive,
- pvt);
- if (pvt->provisional_keepalive_sched_id < 0) {
- dialog_unref(pvt, "Failed to schedule provisional keepalive");
- }
- }
- sip_pvt_unlock(pvt);
-
- dialog_unref(pvt, "Update provisional keepalive action");
- return 0;
-}
-
-/* Run by the sched thread. */
-static int __update_provisional_keepalive(const void *data)
-{
- struct sip_pvt *pvt = (void *) data;
-
- return __update_provisional_keepalive_full(pvt, 0);
-}
-
-/* Run by the sched thread. */
-static int __update_provisional_keepalive_with_sdp(const void *data)
-{
- struct sip_pvt *pvt = (void *) data;
-
- return __update_provisional_keepalive_full(pvt, 1);
-}
-
-static void update_provisional_keepalive(struct sip_pvt *pvt, int with_sdp)
-{
- dialog_ref(pvt, "Update provisional keepalive action");
- if (ast_sched_add(sched, 0,
- with_sdp ? __update_provisional_keepalive_with_sdp : __update_provisional_keepalive,
- pvt) < 0) {
- dialog_unref(pvt, "Failed to schedule update provisional keepalive action");
- }
-}
-
-/* Run by the sched thread. */
-static int __stop_provisional_keepalive(const void *data)
-{
- struct sip_pvt *pvt = (void *) data;
-
- AST_SCHED_DEL_UNREF(sched, pvt->provisional_keepalive_sched_id,
- dialog_unref(pvt, "Stop scheduled provisional keepalive"));
- dialog_unref(pvt, "Stop provisional keepalive action");
- return 0;
-}
-
-static void stop_provisional_keepalive(struct sip_pvt *pvt)
-{
- dialog_ref(pvt, "Stop provisional keepalive action");
- if (ast_sched_add(sched, 0, __stop_provisional_keepalive, pvt) < 0) {
- /* Uh Oh. Expect bad behavior. */
- dialog_unref(pvt, "Failed to schedule stop provisional keepalive action");
- }
-}
-
-static void add_required_respheader(struct sip_request *req)
-{
- struct ast_str *str;
- int i;
-
- if (!req->reqsipoptions) {
- return;
- }
-
- str = ast_str_create(32);
-
- for (i = 0; i < ARRAY_LEN(sip_options); ++i) {
- if (!(req->reqsipoptions & sip_options[i].id)) {
- continue;
- }
- if (ast_str_strlen(str) > 0) {
- ast_str_append(&str, 0, ", ");
- }
- ast_str_append(&str, 0, "%s", sip_options[i].text);
- }
-
- if (ast_str_strlen(str) > 0) {
- add_header(req, "Require", ast_str_buffer(str));
- }
-
- ast_free(str);
-}
-
-/*! \brief Transmit response on SIP request*/
-static int send_response(struct sip_pvt *p, struct sip_request *req, enum xmittype reliable, uint32_t seqno)
-{
- int res;
-
- finalize_content(req);
- add_blank(req);
- if (sip_debug_test_pvt(p)) {
- const struct ast_sockaddr *dst = sip_real_dst(p);
-
- ast_verbose("\n<--- %sTransmitting (%s) to %s --->\n%s\n<------------>\n",
- reliable ? "Reliably " : "", sip_nat_mode(p),
- ast_sockaddr_stringify(dst),
- ast_str_buffer(req->data));
- }
- if (p->do_history) {
- struct sip_request tmp = { .rlpart1 = 0, };
- parse_copy(&tmp, req);
- append_history(p, reliable ? "TxRespRel" : "TxResp", "%s / %s - %s", ast_str_buffer(tmp.data), sip_get_header(&tmp, "CSeq"),
- (tmp.method == SIP_RESPONSE || tmp.method == SIP_UNKNOWN) ? REQ_OFFSET_TO_STR(&tmp, rlpart2) : sip_methods[tmp.method].text);
- deinit_req(&tmp);
- }
-
- /* If we are sending a final response to an INVITE, stop retransmitting provisional responses */
- if (p->initreq.method == SIP_INVITE && reliable == XMIT_CRITICAL) {
- stop_provisional_keepalive(p);
- }
-
- res = (reliable) ?
- __sip_reliable_xmit(p, seqno, 1, req->data, (reliable == XMIT_CRITICAL), req->method) :
- __sip_xmit(p, req->data);
- deinit_req(req);
- if (res > 0) {
- return 0;
- }
- return res;
-}
-
-/*!
- * \internal
- * \brief Send SIP Request to the other part of the dialogue
- * \return see \ref __sip_xmit
- */
-static int send_request(struct sip_pvt *p, struct sip_request *req, enum xmittype reliable, uint32_t seqno)
-{
- int res;
-
- /* If we have an outbound proxy, reset peer address
- Only do this once.
- */
- if (p->outboundproxy) {
- p->sa = p->outboundproxy->ip;
- }
-
- finalize_content(req);
- add_blank(req);
- if (sip_debug_test_pvt(p)) {
- if (ast_test_flag(&p->flags[0], SIP_NAT_FORCE_RPORT)) {
- ast_verbose("%sTransmitting (NAT) to %s:\n%s\n---\n", reliable ? "Reliably " : "", ast_sockaddr_stringify(&p->recv), ast_str_buffer(req->data));
- } else {
- ast_verbose("%sTransmitting (no NAT) to %s:\n%s\n---\n", reliable ? "Reliably " : "", ast_sockaddr_stringify(&p->sa), ast_str_buffer(req->data));
- }
- }
- if (p->do_history) {
- struct sip_request tmp = { .rlpart1 = 0, };
- parse_copy(&tmp, req);
- append_history(p, reliable ? "TxReqRel" : "TxReq", "%s / %s - %s", ast_str_buffer(tmp.data), sip_get_header(&tmp, "CSeq"), sip_methods[tmp.method].text);
- deinit_req(&tmp);
- }
- res = (reliable) ?
- __sip_reliable_xmit(p, seqno, 0, req->data, (reliable == XMIT_CRITICAL), req->method) :
- __sip_xmit(p, req->data);
- deinit_req(req);
- return res;
-}
-
-static void enable_dsp_detect(struct sip_pvt *p)
-{
- int features = 0;
-
- if (p->dsp) {
- return;
- }
-
- if ((ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_INBAND) ||
- (ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_AUTO)) {
- if (p->rtp) {
- ast_rtp_instance_dtmf_mode_set(p->rtp, AST_RTP_DTMF_MODE_INBAND);
- }
- features |= DSP_FEATURE_DIGIT_DETECT;
- }
-
- if (ast_test_flag(&p->flags[1], SIP_PAGE2_FAX_DETECT_CNG)) {
- features |= DSP_FEATURE_FAX_DETECT;
- }
-
- if (!features) {
- return;
- }
-
- if (!(p->dsp = ast_dsp_new())) {
- return;
- }
-
- ast_dsp_set_features(p->dsp, features);
- if (global_relaxdtmf) {
- ast_dsp_set_digitmode(p->dsp, DSP_DIGITMODE_DTMF | DSP_DIGITMODE_RELAXDTMF);
- }
-}
-
-static void disable_dsp_detect(struct sip_pvt *p)
-{
- if (p->dsp) {
- ast_dsp_free(p->dsp);
- p->dsp = NULL;
- }
-}
-
-/*! \brief Set an option on a SIP dialog */
-static int sip_setoption(struct ast_channel *chan, int option, void *data, int datalen)
-{
- int res = -1;
- struct sip_pvt *p = ast_channel_tech_pvt(chan);
-
- if (!p) {
- ast_log(LOG_ERROR, "Attempt to Ref a null pointer. sip private structure is gone!\n");
- return -1;
- }
-
- sip_pvt_lock(p);
-
- switch (option) {
- case AST_OPTION_FORMAT_READ:
- if (p->rtp) {
- res = ast_rtp_instance_set_read_format(p->rtp, *(struct ast_format **) data);
- }
- break;
- case AST_OPTION_FORMAT_WRITE:
- if (p->rtp) {
- res = ast_rtp_instance_set_write_format(p->rtp, *(struct ast_format **) data);
- }
- break;
- case AST_OPTION_DIGIT_DETECT:
- if ((ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_INBAND) ||
- (ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_AUTO)) {
- char *cp = (char *) data;
-
- ast_debug(1, "%sabling digit detection on %s\n", *cp ? "En" : "Dis", ast_channel_name(chan));
- if (*cp) {
- enable_dsp_detect(p);
- } else {
- disable_dsp_detect(p);
- }
- res = 0;
- }
- break;
- case AST_OPTION_SECURE_SIGNALING:
- p->req_secure_signaling = *(unsigned int *) data;
- res = 0;
- break;
- case AST_OPTION_SECURE_MEDIA:
- ast_set2_flag(&p->flags[1], *(unsigned int *) data, SIP_PAGE2_USE_SRTP);
- res = 0;
- break;
- default:
- break;
- }
-
- sip_pvt_unlock(p);
-
- return res;
-}
-
-/*! \brief Query an option on a SIP dialog */
-static int sip_queryoption(struct ast_channel *chan, int option, void *data, int *datalen)
-{
- int res = -1;
- enum ast_t38_state state = T38_STATE_UNAVAILABLE;
- struct sip_pvt *p = (struct sip_pvt *) ast_channel_tech_pvt(chan);
- char *cp;
-
- if (!p) {
- ast_debug(1, "Attempt to Ref a null pointer. Sip private structure is gone!\n");
- return -1;
- }
-
- sip_pvt_lock(p);
-
- switch (option) {
- case AST_OPTION_T38_STATE:
- /* Make sure we got an ast_t38_state enum passed in */
- if (*datalen != sizeof(enum ast_t38_state)) {
- ast_log(LOG_ERROR, "Invalid datalen for AST_OPTION_T38_STATE option. Expected %d, got %d\n", (int)sizeof(enum ast_t38_state), *datalen);
- break;
- }
-
- /* Now if T38 support is enabled we need to look and see what the current state is to get what we want to report back */
- if (ast_test_flag(&p->flags[1], SIP_PAGE2_T38SUPPORT)) {
- switch (p->t38.state) {
- case T38_LOCAL_REINVITE:
- case T38_PEER_REINVITE:
- state = T38_STATE_NEGOTIATING;
- break;
- case T38_ENABLED:
- state = T38_STATE_NEGOTIATED;
- break;
- case T38_REJECTED:
- state = T38_STATE_REJECTED;
- break;
- default:
- state = T38_STATE_UNKNOWN;
- }
- }
-
- *((enum ast_t38_state *) data) = state;
- res = 0;
-
- break;
- case AST_OPTION_DIGIT_DETECT:
- cp = (char *) data;
- *cp = p->dsp ? 1 : 0;
- ast_debug(1, "Reporting digit detection %sabled on %s\n", *cp ? "en" : "dis", ast_channel_name(chan));
- break;
- case AST_OPTION_SECURE_SIGNALING:
- *((unsigned int *) data) = p->req_secure_signaling;
- res = 0;
- break;
- case AST_OPTION_SECURE_MEDIA:
- *((unsigned int *) data) = ast_test_flag(&p->flags[1], SIP_PAGE2_USE_SRTP) ? 1 : 0;
- res = 0;
- break;
- case AST_OPTION_DEVICE_NAME:
- if (p && p->outgoing_call) {
- cp = (char *) data;
- ast_copy_string(cp, p->dialstring, *datalen);
- res = 0;
- }
- /* We purposely break with a return of -1 in the
- * implied else case here
- */
- break;
- default:
- break;
- }
-
- sip_pvt_unlock(p);
-
- return res;
-}
-
-/*! \brief Locate closing quote in a string, skipping escaped quotes.
- * optionally with a limit on the search.
- * start must be past the first quote.
- */
-const char *find_closing_quote(const char *start, const char *lim)
-{
- char last_char = '\0';
- const char *s;
- for (s = start; *s && s != lim; last_char = *s++) {
- if (*s == '"' && last_char != '\\')
- break;
- }
- return s;
-}
-
-/*! \brief Send message with Access-URL header, if this is an HTML URL only! */
-static int sip_sendhtml(struct ast_channel *chan, int subclass, const char *data, int datalen)
-{
- struct sip_pvt *p = ast_channel_tech_pvt(chan);
-
- if (subclass != AST_HTML_URL)
- return -1;
-
- ast_string_field_build(p, url, "<%s>;mode=active", data);
-
- if (sip_debug_test_pvt(p))
- ast_debug(1, "Send URL %s, state = %u!\n", data, ast_channel_state(chan));
-
- switch (ast_channel_state(chan)) {
- case AST_STATE_RING:
- transmit_response(p, "100 Trying", &p->initreq);
- break;
- case AST_STATE_RINGING:
- transmit_response(p, "180 Ringing", &p->initreq);
- break;
- case AST_STATE_UP:
- if (!p->pendinginvite) { /* We are up, and have no outstanding invite */
- transmit_reinvite_with_sdp(p, FALSE, FALSE);
- } else if (!ast_test_flag(&p->flags[0], SIP_PENDINGBYE)) {
- ast_set_flag(&p->flags[0], SIP_NEEDREINVITE);
- }
- break;
- default:
- ast_log(LOG_WARNING, "Don't know how to send URI when state is %u!\n", ast_channel_state(chan));
- }
-
- return 0;
-}
-
-/*! \brief Deliver SIP call ID for the call */
-static const char *sip_get_callid(struct ast_channel *chan)
-{
- return ast_channel_tech_pvt(chan) ? ((struct sip_pvt *) ast_channel_tech_pvt(chan))->callid : "";
-}
-
-/*!
- * \internal
- * \brief Send SIP MESSAGE text within a call
- * \note Called from PBX core sendtext() application
- */
-static int sip_sendtext(struct ast_channel *ast, const char *text)
-{
- struct sip_pvt *dialog = ast_channel_tech_pvt(ast);
- int debug;
-
- if (!dialog) {
- return -1;
- }
- /* NOT ast_strlen_zero, because a zero-length message is specifically
- * allowed by RFC 3428 (See section 10, Examples) */
- if (!text) {
- return 0;
- }
- if(!is_method_allowed(&dialog->allowed_methods, SIP_MESSAGE)) {
- ast_debug(2, "Trying to send MESSAGE to device that does not support it.\n");
- return 0;
- }
-
- debug = sip_debug_test_pvt(dialog);
- if (debug) {
- ast_verbose("Sending text %s on %s\n", text, ast_channel_name(ast));
- }
-
- /* Setup to send text message */
- sip_pvt_lock(dialog);
- destroy_msg_headers(dialog);
- ast_string_field_set(dialog, msg_body, text);
- transmit_message(dialog, 0, 0);
- sip_pvt_unlock(dialog);
- return 0;
-}
-
-/*! \brief Update peer object in realtime storage
- If the Asterisk system name is set in asterisk.conf, we will use
- that name and store that in the "regserver" field in the sippeers
- table to facilitate multi-server setups.
-*/
-static void realtime_update_peer(const char *peername, struct ast_sockaddr *addr, const char *defaultuser, const char *fullcontact, const char *useragent, int expirey, unsigned short deprecated_username, int lastms, const char *path)
-{
- char port[10];
- char ipaddr[INET6_ADDRSTRLEN];
- char regseconds[20];
- char *tablename = NULL;
- char str_lastms[20];
-
- const char *sysname = ast_config_AST_SYSTEM_NAME;
- char *syslabel = NULL;
-
- time_t nowtime = time(NULL) + expirey;
- const char *fc = fullcontact ? "fullcontact" : NULL;
-
- int realtimeregs = ast_check_realtime("sipregs");
-
- tablename = realtimeregs ? "sipregs" : "sippeers";
-
- snprintf(str_lastms, sizeof(str_lastms), "%d", lastms);
- snprintf(regseconds, sizeof(regseconds), "%d", (int)nowtime); /* Expiration time */
- ast_copy_string(ipaddr, ast_sockaddr_isnull(addr) ? "" : ast_sockaddr_stringify_addr(addr), sizeof(ipaddr));
- ast_copy_string(port, ast_sockaddr_port(addr) ? ast_sockaddr_stringify_port(addr) : "", sizeof(port));
-
- if (ast_strlen_zero(sysname)) { /* No system name, disable this */
- sysname = NULL;
- } else if (sip_cfg.rtsave_sysname) {
- syslabel = "regserver";
- }
-
- /* XXX IMPORTANT: Anytime you add a new parameter to be updated, you
- * must also add it to contrib/scripts/asterisk.ldap-schema,
- * contrib/scripts/asterisk.ldif,
- * and to configs/res_ldap.conf.sample as described in
- * bugs 15156 and 15895
- */
-
- /* This is ugly, we need something better ;-) */
- if (sip_cfg.rtsave_path) {
- if (fc) {
- ast_update_realtime(tablename, "name", peername, "ipaddr", ipaddr,
- "port", port, "regseconds", regseconds,
- deprecated_username ? "username" : "defaultuser", defaultuser,
- "useragent", useragent, "lastms", str_lastms,
- "path", path, /* Path data can be NULL */
- fc, fullcontact, syslabel, sysname, SENTINEL); /* note fc and syslabel _can_ be NULL */
- } else {
- ast_update_realtime(tablename, "name", peername, "ipaddr", ipaddr,
- "port", port, "regseconds", regseconds,
- "useragent", useragent, "lastms", str_lastms,
- deprecated_username ? "username" : "defaultuser", defaultuser,
- "path", path, /* Path data can be NULL */
- syslabel, sysname, SENTINEL); /* note syslabel _can_ be NULL */
- }
- } else {
- if (fc) {
- ast_update_realtime(tablename, "name", peername, "ipaddr", ipaddr,
- "port", port, "regseconds", regseconds,
- deprecated_username ? "username" : "defaultuser", defaultuser,
- "useragent", useragent, "lastms", str_lastms,
- fc, fullcontact, syslabel, sysname, SENTINEL); /* note fc and syslabel _can_ be NULL */
- } else {
- ast_update_realtime(tablename, "name", peername, "ipaddr", ipaddr,
- "port", port, "regseconds", regseconds,
- "useragent", useragent, "lastms", str_lastms,
- deprecated_username ? "username" : "defaultuser", defaultuser,
- syslabel, sysname, SENTINEL); /* note syslabel _can_ be NULL */
- }
- }
-}
-
-/*! \brief Automatically add peer extension to dial plan */
-static void register_peer_exten(struct sip_peer *peer, int onoff)
-{
- char multi[256];
- char *stringp, *ext, *context;
- struct pbx_find_info q = { .stacklen = 0 };
-
- /* XXX note that sip_cfg.regcontext is both a global 'enable' flag and
- * the name of the global regexten context, if not specified
- * individually.
- */
- if (ast_strlen_zero(sip_cfg.regcontext))
- return;
-
- ast_copy_string(multi, S_OR(peer->regexten, peer->name), sizeof(multi));
- stringp = multi;
- while ((ext = strsep(&stringp, "&"))) {
- if ((context = strchr(ext, '@'))) {
- *context++ = '\0'; /* split ext@context */
- if (!ast_context_find(context)) {
- ast_log(LOG_WARNING, "Context %s must exist in regcontext= in sip.conf!\n", context);
- continue;
- }
- } else {
- context = sip_cfg.regcontext;
- }
- if (onoff) {
- if (!ast_exists_extension(NULL, context, ext, 1, NULL)) {
- ast_add_extension(context, 1, ext, 1, NULL, NULL, "Noop",
- ast_strdup(peer->name), ast_free_ptr, "SIP");
- }
- } else if (pbx_find_extension(NULL, NULL, &q, context, ext, 1, NULL, "", E_MATCH)) {
- ast_context_remove_extension(context, ext, 1, NULL);
- }
- }
-}
-
-/*! Destroy mailbox subscriptions */
-static void destroy_mailbox(struct sip_mailbox *mailbox)
-{
- if (mailbox->event_sub) {
- mailbox->event_sub = ast_mwi_unsubscribe_and_join(mailbox->event_sub);
- }
- ast_free(mailbox);
-}
-
-#define REMOVE_MAILBOX_WITH_LOCKED_PEER(__peer) \
-({\
- struct sip_mailbox *__mailbox;\
- ao2_lock(__peer);\
- __mailbox = AST_LIST_REMOVE_HEAD(&(__peer->mailboxes), entry);\
- ao2_unlock(__peer);\
- __mailbox;\
-})
-
-/*! Destroy all peer-related mailbox subscriptions */
-static void clear_peer_mailboxes(struct sip_peer *peer)
-{
- struct sip_mailbox *mailbox;
-
- /* Lock the peer while accessing/updating the linked list but NOT while destroying the mailbox */
- while ((mailbox = REMOVE_MAILBOX_WITH_LOCKED_PEER(peer))) {
- destroy_mailbox(mailbox);
- }
-}
-
-static void sip_destroy_peer_fn(void *peer)
-{
- sip_destroy_peer(peer);
-}
-
-/*! \brief Destroy peer object from memory */
-static void sip_destroy_peer(struct sip_peer *peer)
-{
- ast_debug(3, "Destroying SIP peer %s\n", peer->name);
-
- /*
- * Remove any mailbox event subscriptions for this peer before
- * we destroy anything. An event subscription callback may be
- * happening right now.
- */
- clear_peer_mailboxes(peer);
-
- if (peer->outboundproxy) {
- ao2_ref(peer->outboundproxy, -1);
- peer->outboundproxy = NULL;
- }
-
- /* Delete it, it needs to disappear */
- if (peer->call) {
- dialog_unlink_all(peer->call);
- peer->call = dialog_unref(peer->call, "peer->call is being unset");
- }
-
- if (peer->mwipvt) { /* We have an active subscription, delete it */
- dialog_unlink_all(peer->mwipvt);
- peer->mwipvt = dialog_unref(peer->mwipvt, "unreffing peer->mwipvt");
- }
-
- if (peer->chanvars) {
- ast_variables_destroy(peer->chanvars);
- peer->chanvars = NULL;
- }
- sip_route_clear(&peer->path);
-
- register_peer_exten(peer, FALSE);
- ast_free_acl_list(peer->acl);
- ast_free_acl_list(peer->contactacl);
- ast_free_acl_list(peer->directmediaacl);
- if (peer->selfdestruct)
- ast_atomic_fetchadd_int(&apeerobjs, -1);
- else if (!ast_test_flag(&global_flags[1], SIP_PAGE2_RTCACHEFRIENDS) && peer->is_realtime) {
- ast_atomic_fetchadd_int(&rpeerobjs, -1);
- ast_debug(3, "-REALTIME- peer Destroyed. Name: %s. Realtime Peer objects: %d\n", peer->name, rpeerobjs);
- } else
- ast_atomic_fetchadd_int(&speerobjs, -1);
- if (peer->auth) {
- ao2_t_ref(peer->auth, -1, "Removing peer authentication");
- peer->auth = NULL;
- }
-
- if (peer->socket.tcptls_session) {
- ao2_ref(peer->socket.tcptls_session, -1);
- peer->socket.tcptls_session = NULL;
- } else if (peer->socket.ws_session) {
- ast_websocket_unref(peer->socket.ws_session);
- peer->socket.ws_session = NULL;
- }
-
- peer->named_callgroups = ast_unref_namedgroups(peer->named_callgroups);
- peer->named_pickupgroups = ast_unref_namedgroups(peer->named_pickupgroups);
-
- ast_cc_config_params_destroy(peer->cc_params);
-
- ast_string_field_free_memory(peer);
-
- ao2_cleanup(peer->caps);
-
- ast_rtp_dtls_cfg_free(&peer->dtls_cfg);
-
- ast_endpoint_shutdown(peer->endpoint);
- peer->endpoint = NULL;
-}
-
-/*! \brief Update peer data in database (if used) */
-static void update_peer(struct sip_peer *p, int expire)
-{
- int rtcachefriends = ast_test_flag(&p->flags[1], SIP_PAGE2_RTCACHEFRIENDS);
- if (sip_cfg.peer_rtupdate && (p->is_realtime || rtcachefriends)) {
- struct ast_str *r = sip_route_list(&p->path, 0, 0);
- if (r) {
- realtime_update_peer(p->name, &p->addr, p->username,
- p->fullcontact, p->useragent, expire, p->deprecated_username,
- p->lastms, ast_str_buffer(r));
- ast_free(r);
- }
- }
-}
-
-static struct ast_variable *get_insecure_variable_from_config(struct ast_config *cfg)
-{
- struct ast_variable *var = NULL;
- struct ast_flags flags = {0};
- char *cat = NULL;
- const char *insecure;
- while ((cat = ast_category_browse(cfg, cat))) {
- insecure = ast_variable_retrieve(cfg, cat, "insecure");
- set_insecure_flags(&flags, insecure, -1);
- if (ast_test_flag(&flags, SIP_INSECURE_PORT)) {
- var = ast_category_root(cfg, cat);
- break;
- }
- }
- return var;
-}
-
-static struct ast_variable *get_insecure_variable_from_sippeers(const char *column, const char *value)
-{
- struct ast_config *peerlist;
- struct ast_variable *var = NULL;
- if ((peerlist = ast_load_realtime_multientry("sippeers", column, value, "insecure LIKE", "%port%", SENTINEL))) {
- if ((var = get_insecure_variable_from_config(peerlist))) {
- /* Must clone, because var will get freed along with
- * peerlist. */
- var = ast_variables_dup(var);
- }
- ast_config_destroy(peerlist);
- }
- return var;
-}
-
-/* Yes.. the only column that makes sense to pass is "ipaddr", but for
- * consistency's sake, we require the column name to be passed. As extra
- * argument, we take a pointer to var. We already got the info, so we better
- * return it and save the caller a query. If return value is nonzero, then *var
- * is nonzero too (and the other way around). */
-static struct ast_variable *get_insecure_variable_from_sipregs(const char *column, const char *value, struct ast_variable **var)
-{
- struct ast_variable *varregs = NULL;
- struct ast_config *regs, *peers;
- char *regscat;
- const char *regname;
-
- if (!(regs = ast_load_realtime_multientry("sipregs", column, value, SENTINEL))) {
- return NULL;
- }
-
- /* Load *all* peers that are probably insecure=port */
- if (!(peers = ast_load_realtime_multientry("sippeers", "insecure LIKE", "%port%", SENTINEL))) {
- ast_config_destroy(regs);
- return NULL;
- }
-
- /* Loop over the sipregs that match IP address and attempt to find an
- * insecure=port match to it in sippeers. */
- regscat = NULL;
- while ((regscat = ast_category_browse(regs, regscat)) && (regname = ast_variable_retrieve(regs, regscat, "name"))) {
- char *peerscat;
- const char *peername;
-
- peerscat = NULL;
- while ((peerscat = ast_category_browse(peers, peerscat)) && (peername = ast_variable_retrieve(peers, peerscat, "name"))) {
- if (!strcasecmp(regname, peername)) {
- /* Ensure that it really is insecure=port and
- * not something else. */
- const char *insecure = ast_variable_retrieve(peers, peerscat, "insecure");
- struct ast_flags flags = {0};
- set_insecure_flags(&flags, insecure, -1);
- if (ast_test_flag(&flags, SIP_INSECURE_PORT)) {
- /* ENOMEM checks till the bitter end. */
- if ((varregs = ast_variables_dup(ast_category_root(regs, regscat)))) {
- if (!(*var = ast_variables_dup(ast_category_root(peers, peerscat)))) {
- ast_variables_destroy(varregs);
- varregs = NULL;
- }
- }
- goto done;
- }
- }
- }
- }
-
-done:
- ast_config_destroy(regs);
- ast_config_destroy(peers);
- return varregs;
-}
-
-static const char *get_name_from_variable(const struct ast_variable *var)
-{
- /* Don't expect this to return non-NULL. Both NULL and empty
- * values can cause the option to get removed from the variable
- * list. This is called on ast_variables gotten from both
- * ast_load_realtime and ast_load_realtime_multientry.
- * - ast_load_realtime removes options with empty values
- * - ast_load_realtime_multientry does not!
- * For consistent behaviour, we check for the empty name and
- * return NULL instead. */
- const struct ast_variable *tmp;
- for (tmp = var; tmp; tmp = tmp->next) {
- if (!strcasecmp(tmp->name, "name")) {
- if (!ast_strlen_zero(tmp->value)) {
- return tmp->value;
- }
- break;
- }
- }
- return NULL;
-}
-
-/* If varregs is NULL, we don't use sipregs.
- * Using empty if-bodies instead of goto's while avoiding unnecessary indents */
-static int realtime_peer_by_name(const char *const *name, struct ast_sockaddr *addr, const char *ipaddr, struct ast_variable **var, struct ast_variable **varregs)
-{
- /* Peer by name and host=dynamic */
- if ((*var = ast_load_realtime("sippeers", "name", *name, "host", "dynamic", SENTINEL))) {
- ;
- /* Peer by name and host=IP */
- } else if (addr && !(*var = ast_load_realtime("sippeers", "name", *name, "host", ipaddr, SENTINEL))) {
- ;
- /* Peer by name and host=HOSTNAME */
- } else if ((*var = ast_load_realtime("sippeers", "name", *name, SENTINEL))) {
- /*!\note
- * If this one loaded something, then we need to ensure that the host
- * field matched. The only reason why we can't have this as a criteria
- * is because we only have the IP address and the host field might be
- * set as a name (and the reverse PTR might not match).
- */
- if (addr) {
- struct ast_variable *tmp;
- for (tmp = *var; tmp; tmp = tmp->next) {
- if (!strcasecmp(tmp->name, "host")) {
- struct ast_sockaddr *addrs = NULL;
-
- if (ast_sockaddr_resolve(&addrs,
- tmp->value,
- PARSE_PORT_FORBID,
- get_address_family_filter(AST_TRANSPORT_UDP)) <= 0 ||
- ast_sockaddr_cmp(&addrs[0], addr)) {
- /* No match */
- ast_variables_destroy(*var);
- *var = NULL;
- }
- ast_free(addrs);
- break;
- }
- }
- }
- }
-
- /* Did we find anything? */
- if (*var) {
- if (varregs) {
- *varregs = ast_load_realtime("sipregs", "name", *name, SENTINEL);
- }
- return 1;
- }
- return 0;
-}
-
-/* Another little helper function for backwards compatibility: this
- * checks/fetches the sippeer that belongs to the sipreg. If none is
- * found, we free the sipreg and return false. This way we can do the
- * check inside the if-condition below. In the old code, not finding
- * the sippeer also had it continue look for another match, so we do
- * the same. */
-static struct ast_variable *realtime_peer_get_sippeer_helper(const char **name, struct ast_variable **varregs) {
- struct ast_variable *var = NULL;
- const char *old_name = *name;
- *name = get_name_from_variable(*varregs);
- if (!*name || !(var = ast_load_realtime("sippeers", "name", *name, SENTINEL))) {
- if (!*name) {
- ast_log(LOG_WARNING, "Found sipreg but it has no name\n");
- }
- ast_variables_destroy(*varregs);
- *varregs = NULL;
- *name = old_name;
- }
- return var;
-}
-
-/* If varregs is NULL, we don't use sipregs. If we return true, then *name is
- * set. Using empty if-bodies instead of goto's while avoiding unnecessary
- * indents. */
-static int realtime_peer_by_addr(const char **name, struct ast_sockaddr *addr, const char *ipaddr, const char *callbackexten, struct ast_variable **var, struct ast_variable **varregs)
-{
- char portstring[6]; /* up to 5 digits plus null terminator */
- ast_copy_string(portstring, ast_sockaddr_stringify_port(addr), sizeof(portstring));
-
- /* We're not finding this peer by this name anymore. Reset it. */
- *name = NULL;
-
- /* First check for fixed IP hosts with matching callbackextensions, if specified */
- if (!ast_strlen_zero(callbackexten) && (*var = ast_load_realtime("sippeers", "host", ipaddr, "port", portstring, "callbackextension", callbackexten, SENTINEL))) {
- ;
- /* Check for fixed IP hosts */
- } else if ((*var = ast_load_realtime("sippeers", "host", ipaddr, "port", portstring, SENTINEL))) {
- ;
- /* Check for registered hosts (in sipregs) */
- } else if (varregs && (*varregs = ast_load_realtime("sipregs", "ipaddr", ipaddr, "port", portstring, SENTINEL)) &&
- (*var = realtime_peer_get_sippeer_helper(name, varregs))) {
- ;
- /* Check for registered hosts (in sippeers) */
- } else if (!varregs && (*var = ast_load_realtime("sippeers", "ipaddr", ipaddr, "port", portstring, SENTINEL))) {
- ;
- /* We couldn't match on ipaddress and port, so we need to check if port is insecure */
- } else if ((*var = get_insecure_variable_from_sippeers("host", ipaddr))) {
- ;
- /* Same as above, but try the IP address field (in sipregs)
- * Observe that it fetches the name/var at the same time, without the
- * realtime_peer_get_sippeer_helper. Also note that it is quite inefficient.
- * Avoid sipregs if possible. */
- } else if (varregs && (*varregs = get_insecure_variable_from_sipregs("ipaddr", ipaddr, var))) {
- ;
- /* Same as above, but try the IP address field (in sippeers) */
- } else if (!varregs && (*var = get_insecure_variable_from_sippeers("ipaddr", ipaddr))) {
- ;
- }
-
- /* Nothing found? */
- if (!*var) {
- return 0;
- }
-
- /* Check peer name. It must not be empty. There may exist a
- * different match that does have a name, but it's too late for
- * that now. */
- if (!*name && !(*name = get_name_from_variable(*var))) {
- ast_log(LOG_WARNING, "Found peer for IP %s but it has no name\n", ipaddr);
- ast_variables_destroy(*var);
- *var = NULL;
- if (varregs && *varregs) {
- ast_variables_destroy(*varregs);
- *varregs = NULL;
- }
- return 0;
- }
-
- /* Make sure varregs is populated if var is. The inverse,
- * ensuring that var is set when varregs is, is taken
- * care of by realtime_peer_get_sippeer_helper(). */
- if (varregs && !*varregs) {
- *varregs = ast_load_realtime("sipregs", "name", *name, SENTINEL);
- }
- return 1;
-}
-
-static int register_realtime_peers_with_callbackextens(void)
-{
- struct ast_config *cfg;
- char *cat = NULL;
-
- if (!(ast_check_realtime("sippeers"))) {
- return 0;
- }
-
- /* This is hacky. We want name to be the cat, so it is the first property */
- if (!(cfg = ast_load_realtime_multientry("sippeers", "name LIKE", "%", "callbackextension LIKE", "%", SENTINEL))) {
- return -1;
- }
-
- while ((cat = ast_category_browse(cfg, cat))) {
- struct sip_peer *peer;
- struct ast_variable *var = ast_category_root(cfg, cat);
-
- if (!(peer = build_peer(cat, var, NULL, TRUE, FALSE))) {
- continue;
- }
- ast_log(LOG_NOTICE, "Created realtime peer '%s' for registration\n", peer->name);
-
- peer->is_realtime = 1;
- sip_unref_peer(peer, "register_realtime_peers: Done registering releasing");
- }
-
- ast_config_destroy(cfg);
-
- return 0;
-}
-
-/*! \brief realtime_peer: Get peer from realtime storage
- * Checks the "sippeers" realtime family from extconfig.conf
- * Checks the "sipregs" realtime family from extconfig.conf if it's configured.
- * This returns a pointer to a peer and because we use build_peer, we can rest
- * assured that the refcount is bumped.
- *
- * \note This is never called with both newpeername and addr at the same time.
- * If you do, be prepared to get a peer with a different name than newpeername.
- */
-static struct sip_peer *realtime_peer(const char *newpeername, struct ast_sockaddr *addr, char *callbackexten, int devstate_only, int which_objects)
-{
- struct sip_peer *peer = NULL;
- struct ast_variable *var = NULL;
- struct ast_variable *varregs = NULL;
- char ipaddr[INET6_ADDRSTRLEN];
- int realtimeregs = ast_check_realtime("sipregs");
-
- if (addr) {
- ast_copy_string(ipaddr, ast_sockaddr_stringify_addr(addr), sizeof(ipaddr));
- } else {
- ipaddr[0] = '\0';
- }
-
- if (newpeername && realtime_peer_by_name(&newpeername, addr, ipaddr, &var, realtimeregs ? &varregs : NULL)) {
- ;
- } else if (addr && realtime_peer_by_addr(&newpeername, addr, ipaddr, callbackexten, &var, realtimeregs ? &varregs : NULL)) {
- ;
- } else {
- return NULL;
- }
-
- /* If we're looking for users, don't return peers (although this check
- * should probably be done in realtime_peer_by_* instead...) */
- if (which_objects == FINDUSERS) {
- struct ast_variable *tmp;
- for (tmp = var; tmp; tmp = tmp->next) {
- if (!strcasecmp(tmp->name, "type") && (!strcasecmp(tmp->value, "peer"))) {
- goto cleanup;
- }
- }
- }
-
- /* Peer found in realtime, now build it in memory */
- peer = build_peer(newpeername, var, varregs, TRUE, devstate_only);
- if (!peer) {
- goto cleanup;
- }
-
- /* Previous versions of Asterisk did not require the type field to be
- * set for real time peers. This statement preserves that behavior. */
- if (peer->type == 0) {
- if (which_objects == FINDUSERS) {
- peer->type = SIP_TYPE_USER;
- } else if (which_objects == FINDPEERS) {
- peer->type = SIP_TYPE_PEER;
- } else {
- peer->type = SIP_TYPE_PEER | SIP_TYPE_USER;
- }
- }
-
- ast_debug(3, "-REALTIME- loading peer from database to memory. Name: %s. Peer objects: %d\n", peer->name, rpeerobjs);
-
- if (ast_test_flag(&global_flags[1], SIP_PAGE2_RTCACHEFRIENDS) && !devstate_only) {
- /* Cache peer */
- ast_copy_flags(&peer->flags[1], &global_flags[1], SIP_PAGE2_RTAUTOCLEAR|SIP_PAGE2_RTCACHEFRIENDS);
- if (ast_test_flag(&global_flags[1], SIP_PAGE2_RTAUTOCLEAR)) {
- AST_SCHED_REPLACE_UNREF(peer->expire, sched, sip_cfg.rtautoclear * 1000, expire_register, peer,
- sip_unref_peer(_data, "remove registration ref"),
- sip_unref_peer(peer, "remove registration ref"),
- sip_ref_peer(peer, "add registration ref"));
- }
- ao2_t_link(peers, peer, "link peer into peers table");
- if (!ast_sockaddr_isnull(&peer->addr)) {
- ao2_t_link(peers_by_ip, peer, "link peer into peers_by_ip table");
- }
- }
- peer->is_realtime = 1;
-
-cleanup:
- ast_variables_destroy(var);
- ast_variables_destroy(varregs);
- return peer;
-}
-
-/* Function to assist finding peers by name only */
-static int find_by_name(void *obj, void *arg, void *data, int flags)
-{
- struct sip_peer *search = obj, *match = arg;
- int *which_objects = data;
-
- /* Usernames in SIP uri's are case sensitive. Domains are not */
- if (strcmp(search->name, match->name)) {
- return 0;
- }
-
- switch (*which_objects) {
- case FINDUSERS:
- if (!(search->type & SIP_TYPE_USER)) {
- return 0;
- }
- break;
- case FINDPEERS:
- if (!(search->type & SIP_TYPE_PEER)) {
- return 0;
- }
- break;
- case FINDALLDEVICES:
- break;
- }
-
- return CMP_MATCH | CMP_STOP;
-}
-
-static struct sip_peer *sip_find_peer_full(const char *peer, struct ast_sockaddr *addr, char *callbackexten, int realtime, int which_objects, int devstate_only, int transport)
-{
- struct sip_peer *p = NULL;
- struct sip_peer tmp_peer;
-
- if (peer) {
- ast_copy_string(tmp_peer.name, peer, sizeof(tmp_peer.name));
- p = ao2_t_callback_data(peers, OBJ_POINTER, find_by_name, &tmp_peer, &which_objects, "ao2_find in peers table");
- } else if (addr) { /* search by addr? */
- ast_sockaddr_copy(&tmp_peer.addr, addr);
- tmp_peer.flags[0].flags = 0;
- tmp_peer.transports = transport;
- p = ao2_t_callback_data(peers_by_ip, OBJ_POINTER, peer_ipcmp_cb_full, &tmp_peer, callbackexten, "ao2_find in peers_by_ip table");
- if (!p) {
- ast_set_flag(&tmp_peer.flags[0], SIP_INSECURE_PORT);
- p = ao2_t_callback_data(peers_by_ip, OBJ_POINTER, peer_ipcmp_cb_full, &tmp_peer, callbackexten, "ao2_find in peers_by_ip table 2");
- if (p) {
- return p;
- }
- }
- }
-
- if (!p && (realtime || devstate_only)) {
- /* realtime_peer will return a peer with matching callbackexten if possible, otherwise one matching
- * without the callbackexten */
- p = realtime_peer(peer, addr, callbackexten, devstate_only, which_objects);
- if (p) {
- switch (which_objects) {
- case FINDUSERS:
- if (!(p->type & SIP_TYPE_USER)) {
- sip_unref_peer(p, "Wrong type of realtime SIP endpoint");
- return NULL;
- }
- break;
- case FINDPEERS:
- if (!(p->type & SIP_TYPE_PEER)) {
- sip_unref_peer(p, "Wrong type of realtime SIP endpoint");
- return NULL;
- }
- break;
- case FINDALLDEVICES:
- break;
- }
- }
- }
-
- return p;
-}
-
-/*!
- * \brief Locate device by name or ip address
- * \param peer, addr, realtime, devstate_only, transport
- * \param which_objects Define which objects should be matched when doing a lookup
- * by name. Valid options are FINDUSERS, FINDPEERS, or FINDALLDEVICES.
- * Note that this option is not used at all when doing a lookup by IP.
- *
- * This is used on find matching device on name or ip/port.
- * If the device was declared as type=peer, we don't match on peer name on incoming INVITEs.
- *
- * \note Avoid using this function in new functions if there is a way to avoid it,
- * since it might cause a database lookup.
- */
-struct sip_peer *sip_find_peer(const char *peer, struct ast_sockaddr *addr, int realtime, int which_objects, int devstate_only, int transport)
-{
- return sip_find_peer_full(peer, addr, NULL, realtime, which_objects, devstate_only, transport);
-}
-
-static struct sip_peer *sip_find_peer_by_ip_and_exten(struct ast_sockaddr *addr, char *callbackexten, int transport)
-{
- return sip_find_peer_full(NULL, addr, callbackexten, TRUE, FINDPEERS, FALSE, transport);
-}
-
-/*! \brief Set nat mode on the various data sockets */
-static void do_setnat(struct sip_pvt *p)
-{
- const char *mode;
- int natflags;
-
- natflags = ast_test_flag(&p->flags[1], SIP_PAGE2_SYMMETRICRTP);
- mode = natflags ? "On" : "Off";
-
- if (p->rtp) {
- ast_debug(1, "Setting NAT on RTP to %s\n", mode);
- ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_NAT, natflags);
- }
- if (p->vrtp) {
- ast_debug(1, "Setting NAT on VRTP to %s\n", mode);
- ast_rtp_instance_set_prop(p->vrtp, AST_RTP_PROPERTY_NAT, natflags);
- }
- if (p->udptl) {
- ast_debug(1, "Setting NAT on UDPTL to %s\n", mode);
- ast_udptl_setnat(p->udptl, natflags);
- }
- if (p->trtp) {
- ast_debug(1, "Setting NAT on TRTP to %s\n", mode);
- ast_rtp_instance_set_prop(p->trtp, AST_RTP_PROPERTY_NAT, natflags);
- }
-}
-
-/*! \brief Change the T38 state on a SIP dialog */
-static void change_t38_state(struct sip_pvt *p, int state)
-{
- int old = p->t38.state;
- struct ast_channel *chan = p->owner;
- struct ast_control_t38_parameters parameters = { .request_response = 0 };
-
- /* Don't bother changing if we are already in the state wanted */
- if (old == state)
- return;
-
- p->t38.state = state;
- ast_debug(2, "T38 state changed to %u on channel %s\n", p->t38.state, chan ? ast_channel_name(chan) : "");
-
- /* If no channel was provided we can't send off a control frame */
- if (!chan)
- return;
-
- /* Given the state requested and old state determine what control frame we want to queue up */
- switch (state) {
- case T38_PEER_REINVITE:
- parameters = p->t38.their_parms;
- parameters.max_ifp = ast_udptl_get_far_max_ifp(p->udptl);
- parameters.request_response = AST_T38_REQUEST_NEGOTIATE;
- ast_udptl_set_tag(p->udptl, "%s", ast_channel_name(chan));
- break;
- case T38_ENABLED:
- parameters = p->t38.their_parms;
- parameters.max_ifp = ast_udptl_get_far_max_ifp(p->udptl);
- parameters.request_response = AST_T38_NEGOTIATED;
- ast_udptl_set_tag(p->udptl, "%s", ast_channel_name(chan));
- break;
- case T38_REJECTED:
- case T38_DISABLED:
- if (old == T38_ENABLED) {
- parameters.request_response = AST_T38_TERMINATED;
- } else if (old == T38_LOCAL_REINVITE) {
- parameters.request_response = AST_T38_REFUSED;
- }
- break;
- case T38_LOCAL_REINVITE:
- /* wait until we get a peer response before responding to local reinvite */
- break;
- }
-
- /* Woot we got a message, create a control frame and send it on! */
- if (parameters.request_response)
- ast_queue_control_data(chan, AST_CONTROL_T38_PARAMETERS, ¶meters, sizeof(parameters));
-}
-
-/*! \brief Set the global T38 capabilities on a SIP dialog structure */
-static void set_t38_capabilities(struct sip_pvt *p)
-{
- if (p->udptl) {
- if (ast_test_flag(&p->flags[1], SIP_PAGE2_T38SUPPORT) == SIP_PAGE2_T38SUPPORT_UDPTL_REDUNDANCY) {
- ast_udptl_set_error_correction_scheme(p->udptl, UDPTL_ERROR_CORRECTION_REDUNDANCY);
- } else if (ast_test_flag(&p->flags[1], SIP_PAGE2_T38SUPPORT) == SIP_PAGE2_T38SUPPORT_UDPTL_FEC) {
- ast_udptl_set_error_correction_scheme(p->udptl, UDPTL_ERROR_CORRECTION_FEC);
- } else if (ast_test_flag(&p->flags[1], SIP_PAGE2_T38SUPPORT) == SIP_PAGE2_T38SUPPORT_UDPTL) {
- ast_udptl_set_error_correction_scheme(p->udptl, UDPTL_ERROR_CORRECTION_NONE);
- }
- }
-}
-
-static void copy_socket_data(struct sip_socket *to_sock, const struct sip_socket *from_sock)
-{
- if (to_sock->tcptls_session) {
- ao2_ref(to_sock->tcptls_session, -1);
- to_sock->tcptls_session = NULL;
- } else if (to_sock->ws_session) {
- ast_websocket_unref(to_sock->ws_session);
- to_sock->ws_session = NULL;
- }
-
- if (from_sock->tcptls_session) {
- ao2_ref(from_sock->tcptls_session, +1);
- } else if (from_sock->ws_session) {
- ast_websocket_ref(from_sock->ws_session);
- }
-
- *to_sock = *from_sock;
-}
-
-/*! Cleanup the RTP and SRTP portions of a dialog
- *
- * \note This procedure excludes vsrtp as it is initialized differently.
- */
-static void dialog_clean_rtp(struct sip_pvt *p)
-{
- if (p->rtp) {
- ast_rtp_instance_destroy(p->rtp);
- p->rtp = NULL;
- }
-
- if (p->vrtp) {
- ast_rtp_instance_destroy(p->vrtp);
- p->vrtp = NULL;
- }
-
- if (p->trtp) {
- ast_rtp_instance_destroy(p->trtp);
- p->trtp = NULL;
- }
-
- if (p->srtp) {
- ast_sdp_srtp_destroy(p->srtp);
- p->srtp = NULL;
- }
-
- if (p->tsrtp) {
- ast_sdp_srtp_destroy(p->tsrtp);
- p->tsrtp = NULL;
- }
-}
-
-/*! \brief Initialize DTLS-SRTP support on an RTP instance */
-static int dialog_initialize_dtls_srtp(const struct sip_pvt *dialog, struct ast_rtp_instance *rtp, struct ast_sdp_srtp **srtp)
-{
- struct ast_rtp_engine_dtls *dtls;
-
- if (!dialog->dtls_cfg.enabled) {
- return 0;
- }
-
- if (!ast_rtp_engine_srtp_is_registered()) {
- ast_log(LOG_ERROR, "No SRTP module loaded, can't setup SRTP session.\n");
- return -1;
- }
-
- if (!(dtls = ast_rtp_instance_get_dtls(rtp))) {
- ast_log(LOG_ERROR, "No DTLS-SRTP support present on engine for RTP instance '%p', was it compiled with support for it?\n",
- rtp);
- return -1;
- }
-
- if (dtls->set_configuration(rtp, &dialog->dtls_cfg)) {
- ast_log(LOG_ERROR, "Attempted to set an invalid DTLS-SRTP configuration on RTP instance '%p'\n",
- rtp);
- return -1;
- }
-
- if (!(*srtp = ast_sdp_srtp_alloc())) {
- ast_log(LOG_ERROR, "Failed to create required SRTP structure on RTP instance '%p'\n",
- rtp);
- return -1;
- }
-
- return 0;
-}
-
-/*! \brief Initialize RTP portion of a dialog
- * \retval -1 on failure.
- * \retval 0 on success.
- */
-static int dialog_initialize_rtp(struct sip_pvt *dialog)
-{
- struct ast_sockaddr bindaddr_tmp;
- struct ast_rtp_engine_ice *ice;
-
- if (!sip_methods[dialog->method].need_rtp) {
- return 0;
- }
-
- if (!ast_sockaddr_isnull(&rtpbindaddr)) {
- ast_sockaddr_copy(&bindaddr_tmp, &rtpbindaddr);
- } else {
- ast_sockaddr_copy(&bindaddr_tmp, &bindaddr);
- }
-
- /* Make sure previous RTP instances/FD's do not leak */
- dialog_clean_rtp(dialog);
-
- if (!(dialog->rtp = ast_rtp_instance_new(dialog->engine, sched, &bindaddr_tmp, NULL))) {
- return -1;
- }
-
- if (!ast_test_flag(&dialog->flags[2], SIP_PAGE3_ICE_SUPPORT) && (ice = ast_rtp_instance_get_ice(dialog->rtp))) {
- ice->stop(dialog->rtp);
- }
-
- if (dialog_initialize_dtls_srtp(dialog, dialog->rtp, &dialog->srtp)) {
- return -1;
- }
-
- if (ast_test_flag(&dialog->flags[1], SIP_PAGE2_VIDEOSUPPORT_ALWAYS) ||
- (ast_test_flag(&dialog->flags[1], SIP_PAGE2_VIDEOSUPPORT) && (ast_format_cap_has_type(dialog->caps, AST_MEDIA_TYPE_VIDEO)))) {
- if (!(dialog->vrtp = ast_rtp_instance_new(dialog->engine, sched, &bindaddr_tmp, NULL))) {
- return -1;
- }
-
- if (!ast_test_flag(&dialog->flags[2], SIP_PAGE3_ICE_SUPPORT) && (ice = ast_rtp_instance_get_ice(dialog->vrtp))) {
- ice->stop(dialog->vrtp);
- }
-
- if (dialog_initialize_dtls_srtp(dialog, dialog->vrtp, &dialog->vsrtp)) {
- return -1;
- }
-
- ast_rtp_instance_set_timeout(dialog->vrtp, dialog->rtptimeout);
- ast_rtp_instance_set_hold_timeout(dialog->vrtp, dialog->rtpholdtimeout);
- ast_rtp_instance_set_keepalive(dialog->vrtp, dialog->rtpkeepalive);
-
- ast_rtp_instance_set_prop(dialog->vrtp, AST_RTP_PROPERTY_RTCP, AST_RTP_INSTANCE_RTCP_STANDARD);
- ast_rtp_instance_set_qos(dialog->vrtp, global_tos_video, global_cos_video, "SIP VIDEO");
- }
-
- if (ast_test_flag(&dialog->flags[1], SIP_PAGE2_TEXTSUPPORT)) {
- if (!(dialog->trtp = ast_rtp_instance_new(dialog->engine, sched, &bindaddr_tmp, NULL))) {
- return -1;
- }
-
- if (!ast_test_flag(&dialog->flags[2], SIP_PAGE3_ICE_SUPPORT) && (ice = ast_rtp_instance_get_ice(dialog->trtp))) {
- ice->stop(dialog->trtp);
- }
-
- if (dialog_initialize_dtls_srtp(dialog, dialog->trtp, &dialog->tsrtp)) {
- return -1;
- }
-
- /* Do not timeout text as its not constant*/
- ast_rtp_instance_set_keepalive(dialog->trtp, dialog->rtpkeepalive);
-
- ast_rtp_instance_set_prop(dialog->trtp, AST_RTP_PROPERTY_RTCP, AST_RTP_INSTANCE_RTCP_STANDARD);
- }
-
- ast_rtp_instance_set_timeout(dialog->rtp, dialog->rtptimeout);
- ast_rtp_instance_set_hold_timeout(dialog->rtp, dialog->rtpholdtimeout);
- ast_rtp_instance_set_keepalive(dialog->rtp, dialog->rtpkeepalive);
-
- ast_rtp_instance_set_prop(dialog->rtp, AST_RTP_PROPERTY_RTCP, AST_RTP_INSTANCE_RTCP_STANDARD);
- ast_rtp_instance_set_prop(dialog->rtp, AST_RTP_PROPERTY_DTMF, ast_test_flag(&dialog->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833);
- ast_rtp_instance_set_prop(dialog->rtp, AST_RTP_PROPERTY_DTMF_COMPENSATE, ast_test_flag(&dialog->flags[1], SIP_PAGE2_RFC2833_COMPENSATE));
-
- ast_rtp_instance_set_qos(dialog->rtp, global_tos_audio, global_cos_audio, "SIP RTP");
-
- do_setnat(dialog);
-
- return 0;
-}
-
-static int __set_address_from_contact(const char *fullcontact, struct ast_sockaddr *addr, int tcp);
-
-/*! \brief Create address structure from peer reference.
- * This function copies data from peer to the dialog, so we don't have to look up the peer
- * again from memory or database during the life time of the dialog.
- *
- * \retval -1 on error.
- * \retval 0 on success.
- *
- */
-static int create_addr_from_peer(struct sip_pvt *dialog, struct sip_peer *peer)
-{
- struct sip_auth_container *credentials;
-
- /* this checks that the dialog is contacting the peer on a valid
- * transport type based on the peers transport configuration,
- * otherwise, this function bails out */
- if (dialog->socket.type && check_request_transport(peer, dialog))
- return -1;
- copy_socket_data(&dialog->socket, &peer->socket);
-
- if (!(ast_sockaddr_isnull(&peer->addr) && ast_sockaddr_isnull(&peer->defaddr)) &&
- (!peer->maxms || ((peer->lastms >= 0) && (peer->lastms <= peer->maxms)))) {
- dialog->sa = ast_sockaddr_isnull(&peer->addr) ? peer->defaddr : peer->addr;
- dialog->recv = dialog->sa;
- } else
- return -1;
-
- /* XXX TODO: get flags directly from peer only as they are needed using dialog->relatedpeer */
- ast_copy_flags(&dialog->flags[0], &peer->flags[0], SIP_FLAGS_TO_COPY);
- ast_copy_flags(&dialog->flags[1], &peer->flags[1], SIP_PAGE2_FLAGS_TO_COPY);
- ast_copy_flags(&dialog->flags[2], &peer->flags[2], SIP_PAGE3_FLAGS_TO_COPY);
- /* Take the peer's caps */
- if (peer->caps) {
- ast_format_cap_remove_by_type(dialog->caps, AST_MEDIA_TYPE_UNKNOWN);
- ast_format_cap_append_from_cap(dialog->caps, peer->caps, AST_MEDIA_TYPE_UNKNOWN);
- }
- dialog->amaflags = peer->amaflags;
-
- ast_string_field_set(dialog, engine, peer->engine);
-
- ast_rtp_dtls_cfg_copy(&peer->dtls_cfg, &dialog->dtls_cfg);
-
- dialog->rtptimeout = peer->rtptimeout;
- dialog->rtpholdtimeout = peer->rtpholdtimeout;
- dialog->rtpkeepalive = peer->rtpkeepalive;
- sip_route_copy(&dialog->route, &peer->path);
- if (!sip_route_empty(&dialog->route)) {
- /* Parse SIP URI of first route-set hop and use it as target address */
- __set_address_from_contact(sip_route_first_uri(&dialog->route), &dialog->sa, dialog->socket.type == AST_TRANSPORT_TLS ? 1 : 0);
- }
-
- if (dialog_initialize_rtp(dialog)) {
- return -1;
- }
-
- if (dialog->rtp) { /* Audio */
- ast_rtp_instance_set_prop(dialog->rtp, AST_RTP_PROPERTY_DTMF, ast_test_flag(&dialog->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833);
- ast_rtp_instance_set_prop(dialog->rtp, AST_RTP_PROPERTY_DTMF_COMPENSATE, ast_test_flag(&dialog->flags[1], SIP_PAGE2_RFC2833_COMPENSATE));
- /* Set Frame packetization */
- dialog->autoframing = peer->autoframing;
- ast_rtp_codecs_set_framing(ast_rtp_instance_get_codecs(dialog->rtp), ast_format_cap_get_framing(dialog->caps));
- }
-
- /* XXX TODO: get fields directly from peer only as they are needed using dialog->relatedpeer */
- ast_string_field_set(dialog, peername, peer->name);
- ast_string_field_set(dialog, authname, peer->username);
- ast_string_field_set(dialog, username, peer->username);
- ast_string_field_set(dialog, peersecret, peer->secret);
- ast_string_field_set(dialog, peermd5secret, peer->md5secret);
- ast_string_field_set(dialog, mohsuggest, peer->mohsuggest);
- ast_string_field_set(dialog, mohinterpret, peer->mohinterpret);
- ast_string_field_set(dialog, tohost, peer->tohost);
- ast_string_field_set(dialog, fullcontact, peer->fullcontact);
- ast_string_field_set(dialog, accountcode, peer->accountcode);
- ast_string_field_set(dialog, context, peer->context);
- ast_string_field_set(dialog, cid_num, peer->cid_num);
- ast_string_field_set(dialog, cid_name, peer->cid_name);
- ast_string_field_set(dialog, cid_tag, peer->cid_tag);
- ast_string_field_set(dialog, mwi_from, peer->mwi_from);
- if (!ast_strlen_zero(peer->parkinglot)) {
- ast_string_field_set(dialog, parkinglot, peer->parkinglot);
- }
- ast_string_field_set(dialog, engine, peer->engine);
- ref_proxy(dialog, obproxy_get(dialog, peer));
- dialog->callgroup = peer->callgroup;
- dialog->pickupgroup = peer->pickupgroup;
- ast_unref_namedgroups(dialog->named_callgroups);
- dialog->named_callgroups = ast_ref_namedgroups(peer->named_callgroups);
- ast_unref_namedgroups(dialog->named_pickupgroups);
- dialog->named_pickupgroups = ast_ref_namedgroups(peer->named_pickupgroups);
- ast_copy_string(dialog->zone, peer->zone, sizeof(dialog->zone));
- dialog->allowtransfer = peer->allowtransfer;
- dialog->jointnoncodeccapability = dialog->noncodeccapability;
-
- /* Update dialog authorization credentials */
- ao2_lock(peer);
- credentials = peer->auth;
- if (credentials) {
- ao2_t_ref(credentials, +1, "Ref peer auth for dialog");
- }
- ao2_unlock(peer);
- ao2_lock(dialog);
- if (dialog->peerauth) {
- ao2_t_ref(dialog->peerauth, -1, "Unref old dialog peer auth");
- }
- dialog->peerauth = credentials;
- ao2_unlock(dialog);
-
- dialog->maxcallbitrate = peer->maxcallbitrate;
- dialog->disallowed_methods = peer->disallowed_methods;
- ast_cc_copy_config_params(dialog->cc_params, peer->cc_params);
- if (ast_strlen_zero(dialog->tohost))
- ast_string_field_set(dialog, tohost, ast_sockaddr_stringify_host_remote(&dialog->sa));
- if (!ast_strlen_zero(peer->fromdomain)) {
- ast_string_field_set(dialog, fromdomain, peer->fromdomain);
- if (!dialog->initreq.headers) {
- char *new_callid;
- char *tmpcall = ast_strdupa(dialog->callid);
- /* this sure looks to me like we are going to change the callid on this dialog!! */
- new_callid = strchr(tmpcall, '@');
- if (new_callid) {
- int callid_size;
-
- *new_callid = '\0';
-
- /* Change the dialog callid. */
- callid_size = strlen(tmpcall) + strlen(peer->fromdomain) + 2;
- new_callid = ast_alloca(callid_size);
- snprintf(new_callid, callid_size, "%s@%s", tmpcall, peer->fromdomain);
- change_callid_pvt(dialog, new_callid);
- }
- }
- }
- if (!ast_strlen_zero(peer->fromuser)) {
- ast_string_field_set(dialog, fromuser, peer->fromuser);
- }
- if (!ast_strlen_zero(peer->language)) {
- ast_string_field_set(dialog, language, peer->language);
- }
- /* Set timer T1 to RTT for this peer (if known by qualify=) */
- /* Minimum is settable or default to 100 ms */
- /* If there is a maxms and lastms from a qualify use that over a manual T1
- value. Otherwise, use the peer's T1 value. */
- if (peer->maxms && peer->lastms) {
- dialog->timer_t1 = peer->lastms < global_t1min ? global_t1min : peer->lastms;
- } else {
- dialog->timer_t1 = peer->timer_t1;
- }
-
- /* Set timer B to control transaction timeouts, the peer setting is the default and overrides
- the known timer */
- if (peer->timer_b) {
- dialog->timer_b = peer->timer_b;
- } else {
- dialog->timer_b = 64 * dialog->timer_t1;
- }
-
- if ((ast_test_flag(&dialog->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833) ||
- (ast_test_flag(&dialog->flags[0], SIP_DTMF) == SIP_DTMF_AUTO)) {
- dialog->noncodeccapability |= AST_RTP_DTMF;
- } else {
- dialog->noncodeccapability &= ~AST_RTP_DTMF;
- }
-
- dialog->directmediaacl = ast_duplicate_acl_list(peer->directmediaacl);
-
- if (peer->call_limit) {
- ast_set_flag(&dialog->flags[0], SIP_CALL_LIMIT);
- }
- if (!dialog->portinuri) {
- dialog->portinuri = peer->portinuri;
- }
- dialog->chanvars = copy_vars(peer->chanvars);
- if (peer->fromdomainport) {
- dialog->fromdomainport = peer->fromdomainport;
- }
- dialog->callingpres = peer->callingpres;
-
- return 0;
-}
-
-/*! \brief The default sip port for the given transport */
-static inline int default_sip_port(enum ast_transport type)
-{
- return type == AST_TRANSPORT_TLS ? STANDARD_TLS_PORT : STANDARD_SIP_PORT;
-}
-
-/*! \brief create address structure from device name
- * Or, if peer not found, find it in the global DNS
- * returns TRUE (-1) on failure, FALSE on success */
-static int create_addr(struct sip_pvt *dialog, const char *opeer, struct ast_sockaddr *addr, int newdialog)
-{
- struct sip_peer *peer;
- char *peername, *peername2, *hostn;
- char host[MAXHOSTNAMELEN];
- char service[MAXHOSTNAMELEN];
- int srv_ret = 0;
- int tportno;
-
- AST_DECLARE_APP_ARGS(hostport,
- AST_APP_ARG(host);
- AST_APP_ARG(port);
- );
-
- peername = ast_strdupa(opeer);
- peername2 = ast_strdupa(opeer);
- AST_NONSTANDARD_RAW_ARGS(hostport, peername2, ':');
-
- if (hostport.port)
- dialog->portinuri = 1;
-
- dialog->timer_t1 = global_t1; /* Default SIP retransmission timer T1 (RFC 3261) */
- dialog->timer_b = global_timer_b; /* Default SIP transaction timer B (RFC 3261) */
- peer = sip_find_peer(peername, NULL, TRUE, FINDPEERS, FALSE, 0);
-
- if (peer) {
- int res;
- if (newdialog) {
- set_socket_transport(&dialog->socket, 0);
- }
- res = create_addr_from_peer(dialog, peer);
- dialog->relatedpeer = sip_ref_peer(peer, "create_addr: setting dialog's relatedpeer pointer");
- sip_unref_peer(peer, "create_addr: unref peer from sip_find_peer hashtab lookup");
- return res;
- } else if (ast_check_digits(peername)) {
- /* Although an IPv4 hostname *could* be represented as a 32-bit integer, it is uncommon and
- * it makes dialing SIP/${EXTEN} for a peer that isn't defined resolve to an IP that is
- * almost certainly not intended. It is much better to just reject purely numeric hostnames */
- ast_log(LOG_WARNING, "Purely numeric hostname (%s), and not a peer--rejecting!\n", peername);
- return -1;
- } else {
- dialog->rtptimeout = global_rtptimeout;
- dialog->rtpholdtimeout = global_rtpholdtimeout;
- dialog->rtpkeepalive = global_rtpkeepalive;
- if (dialog_initialize_rtp(dialog)) {
- return -1;
- }
- }
-
- ast_string_field_set(dialog, tohost, hostport.host);
- dialog->allowed_methods &= ~sip_cfg.disallowed_methods;
-
- /* Get the outbound proxy information */
- ref_proxy(dialog, obproxy_get(dialog, NULL));
-
- if (addr) {
- /* This address should be updated using dnsmgr */
- ast_sockaddr_copy(&dialog->sa, addr);
- } else {
-
- /* Let's see if we can find the host in DNS. First try DNS SRV records,
- then hostname lookup */
- /*! \todo Fix this function. When we ask for SRV, we should check all transports
- In the future, we should first check NAPTR to find out transport preference
- */
- hostn = peername;
- /* Section 4.2 of RFC 3263 specifies that if a port number is specified, then
- * an A record lookup should be used instead of SRV.
- */
- if (!hostport.port && sip_cfg.srvlookup) {
- snprintf(service, sizeof(service), "_%s._%s.%s",
- get_srv_service(dialog->socket.type),
- get_srv_protocol(dialog->socket.type), peername);
- if ((srv_ret = ast_get_srv(NULL, host, sizeof(host), &tportno,
- service)) > 0) {
- hostn = host;
- }
- }
-
- if (ast_sockaddr_resolve_first_transport(&dialog->sa, hostn, 0, dialog->socket.type ? dialog->socket.type : AST_TRANSPORT_UDP)) {
- ast_log(LOG_WARNING, "No such host: %s\n", peername);
- return -1;
- }
-
- if (srv_ret > 0) {
- ast_sockaddr_set_port(&dialog->sa, tportno);
- }
- }
-
- if (!dialog->socket.type) {
- set_socket_transport(&dialog->socket, AST_TRANSPORT_UDP);
- }
-
- if (!ast_sockaddr_port(&dialog->sa)) {
- ast_sockaddr_set_port(&dialog->sa, default_sip_port(dialog->socket.type));
- }
- ast_sockaddr_copy(&dialog->recv, &dialog->sa);
- return 0;
-}
-
-/*! \brief Scheduled congestion on a call.
- * Only called by the scheduler, must return the reference when done.
- */
-static int auto_congest(const void *arg)
-{
- struct sip_pvt *p = (struct sip_pvt *)arg;
-
- sip_pvt_lock(p);
- p->initid = -1; /* event gone, will not be rescheduled */
- if (p->owner) {
- /* XXX fails on possible deadlock */
- if (!ast_channel_trylock(p->owner)) {
- append_history(p, "Cong", "Auto-congesting (timer)");
- ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
- ast_channel_unlock(p->owner);
- }
-
- /* Give the channel a chance to act before we proceed with destruction */
- sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
- }
- sip_pvt_unlock(p);
- dialog_unref(p, "unreffing arg passed into auto_congest callback (p->initid)");
- return 0;
-}
-
-
-/*! \brief Initiate SIP call from PBX
- * used from the dial() application */
-static int sip_call(struct ast_channel *ast, const char *dest, int timeout)
-{
- int res;
- struct sip_pvt *p = ast_channel_tech_pvt(ast); /* chan is locked, so the reference cannot go away */
- struct varshead *headp;
- struct ast_var_t *current;
- const char *referer = NULL; /* SIP referrer */
- int cc_core_id;
- char uri[SIPBUFSIZE] = "";
-
- if ((ast_channel_state(ast) != AST_STATE_DOWN) && (ast_channel_state(ast) != AST_STATE_RESERVED)) {
- ast_log(LOG_WARNING, "sip_call called on %s, neither down nor reserved\n", ast_channel_name(ast));
- return -1;
- }
-
- if (ast_cc_is_recall(ast, &cc_core_id, "SIP")) {
- char device_name[AST_CHANNEL_NAME];
- struct ast_cc_monitor *recall_monitor;
- struct sip_monitor_instance *monitor_instance;
- ast_channel_get_device_name(ast, device_name, sizeof(device_name));
- if ((recall_monitor = ast_cc_get_monitor_by_recall_core_id(cc_core_id, device_name))) {
- monitor_instance = recall_monitor->private_data;
- ast_copy_string(uri, monitor_instance->notify_uri, sizeof(uri));
- ao2_t_ref(recall_monitor, -1, "Got the URI we need so unreffing monitor");
- }
- }
-
- /* Check whether there is vxml_url, distinctive ring variables */
- headp = ast_channel_varshead(ast);
- AST_LIST_TRAVERSE(headp, current, entries) {
- /* Check whether there is a VXML_URL variable */
- if (!p->options->vxml_url && !strcmp(ast_var_name(current), "VXML_URL")) {
- p->options->vxml_url = ast_var_value(current);
- } else if (!p->options->uri_options && !strcmp(ast_var_name(current), "SIP_URI_OPTIONS")) {
- p->options->uri_options = ast_var_value(current);
- } else if (!p->options->addsipheaders && !strncmp(ast_var_name(current), "SIPADDHEADER", strlen("SIPADDHEADER"))) {
- /* Check whether there is a variable with a name starting with SIPADDHEADER */
- p->options->addsipheaders = 1;
- } else if (!strcmp(ast_var_name(current), "SIPFROMDOMAIN")) {
- ast_string_field_set(p, fromdomain, ast_var_value(current));
- } else if (!strcmp(ast_var_name(current), "SIPTRANSFER")) {
- /* This is a transferred call */
- p->options->transfer = 1;
- } else if (!strcmp(ast_var_name(current), "SIPTRANSFER_REFERER")) {
- /* This is the referrer */
- referer = ast_var_value(current);
- } else if (!strcmp(ast_var_name(current), "SIPTRANSFER_REPLACES")) {
- /* We're replacing a call. */
- p->options->replaces = ast_var_value(current);
- } else if (!strcmp(ast_var_name(current), "SIP_MAX_FORWARDS")) {
- if (sscanf(ast_var_value(current), "%30d", &(p->maxforwards)) != 1) {
- ast_log(LOG_WARNING, "The SIP_MAX_FORWARDS channel variable is not a valid integer.\n");
- }
- }
- }
-
- /* Check to see if we should try to force encryption */
- if (p->req_secure_signaling && p->socket.type != AST_TRANSPORT_TLS) {
- ast_log(LOG_WARNING, "Encrypted signaling is required\n");
- ast_channel_hangupcause_set(ast, AST_CAUSE_BEARERCAPABILITY_NOTAVAIL);
- return -1;
- }
-
- if (ast_test_flag(&p->flags[1], SIP_PAGE2_USE_SRTP)) {
- if (ast_test_flag(&p->flags[0], SIP_REINVITE)) {
- ast_debug(1, "Direct media not possible when using SRTP, ignoring canreinvite setting\n");
- ast_clear_flag(&p->flags[0], SIP_REINVITE);
- }
-
- if (p->rtp && !p->srtp && !(p->srtp = ast_sdp_srtp_alloc())) {
- ast_log(LOG_WARNING, "SRTP audio setup failed\n");
- return -1;
- }
-
- if (p->vrtp && !p->vsrtp && !(p->vsrtp = ast_sdp_srtp_alloc())) {
- ast_log(LOG_WARNING, "SRTP video setup failed\n");
- return -1;
- }
-
- if (p->trtp && !p->tsrtp && !(p->tsrtp = ast_sdp_srtp_alloc())) {
- ast_log(LOG_WARNING, "SRTP text setup failed\n");
- return -1;
- }
- }
-
- res = 0;
- ast_set_flag(&p->flags[0], SIP_OUTGOING);
-
- /* T.38 re-INVITE FAX detection should never be done for outgoing calls,
- * so ensure it is disabled.
- */
- ast_clear_flag(&p->flags[1], SIP_PAGE2_FAX_DETECT_T38);
-
- if (p->options->transfer) {
- char buf[SIPBUFSIZE / 2];
-
- if (referer) {
- if (sipdebug)
- ast_debug(3, "Call for %s transferred by %s\n", p->username, referer);
- snprintf(buf, sizeof(buf)-1, "-> %s (via %s)", p->cid_name, referer);
- } else
- snprintf(buf, sizeof(buf)-1, "-> %s", p->cid_name);
- ast_string_field_set(p, cid_name, buf);
- }
- ast_debug(1, "Outgoing Call for %s\n", p->username);
-
- res = update_call_counter(p, INC_CALL_RINGING);
-
- if (res == -1) {
- ast_channel_hangupcause_set(ast, AST_CAUSE_USER_BUSY);
- return res;
- }
- p->callingpres = ast_party_id_presentation(&ast_channel_caller(ast)->id);
- ast_rtp_instance_available_formats(p->rtp, p->caps, p->prefcaps, p->jointcaps);
- p->jointnoncodeccapability = p->noncodeccapability;
-
- /* If there are no formats left to offer, punt */
- if (ast_format_cap_empty(p->jointcaps)) {
- ast_log(LOG_WARNING, "No format found to offer. Cancelling call to %s\n", p->username);
- res = -1;
- /* If audio was requested (prefcaps) and the [peer] section contains
- * audio (caps) the user expects audio. In that case, if jointcaps
- * contain no audio, punt. Furthermore, this check allows the [peer]
- * section to have no audio. In that case, the user expects no audio
- * and we can pass. Finally, this check allows the requester not to
- * offer any audio. In that case, the call is expected to have no audio
- * and we can pass, as well.
- */
- } else if ((ast_format_cap_empty(p->caps) || ast_format_cap_has_type(p->caps, AST_MEDIA_TYPE_AUDIO)) &&
- (ast_format_cap_empty(p->prefcaps) || ast_format_cap_has_type(p->prefcaps, AST_MEDIA_TYPE_AUDIO)) &&
- !ast_format_cap_has_type(p->jointcaps, AST_MEDIA_TYPE_AUDIO)) {
- ast_log(LOG_WARNING, "No audio format found to offer. Cancelling call to %s\n", p->username);
- res = -1;
- } else {
- int xmitres;
- struct ast_party_connected_line connected;
- struct ast_set_party_connected_line update_connected;
-
- sip_pvt_lock(p);
-
- /* Supply initial connected line information if available. */
- memset(&update_connected, 0, sizeof(update_connected));
- ast_party_connected_line_init(&connected);
- if (!ast_strlen_zero(p->cid_num)
- || (p->callingpres & AST_PRES_RESTRICTION) != AST_PRES_ALLOWED) {
- update_connected.id.number = 1;
- connected.id.number.valid = 1;
- connected.id.number.str = (char *) p->cid_num;
- connected.id.number.presentation = p->callingpres;
- }
- if (!ast_strlen_zero(p->cid_name)
- || (p->callingpres & AST_PRES_RESTRICTION) != AST_PRES_ALLOWED) {
- update_connected.id.name = 1;
- connected.id.name.valid = 1;
- connected.id.name.str = (char *) p->cid_name;
- connected.id.name.presentation = p->callingpres;
- }
- if (update_connected.id.number || update_connected.id.name) {
- /* Invalidate any earlier private connected id representation */
- ast_set_party_id_all(&update_connected.priv);
-
- connected.id.tag = (char *) p->cid_tag;
- connected.source = AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER;
- ast_channel_queue_connected_line_update(ast, &connected, &update_connected);
- }
-
- xmitres = transmit_invite(p, SIP_INVITE, 1, 2, uri);
- if (xmitres == XMIT_ERROR) {
- sip_pvt_unlock(p);
- return -1;
- }
- p->invitestate = INV_CALLING;
-
- /* Initialize auto-congest time */
- AST_SCHED_REPLACE_UNREF(p->initid, sched, p->timer_b, auto_congest, p,
- dialog_unref(_data, "dialog ptr dec when SCHED_REPLACE del op succeeded"),
- dialog_unref(p, "dialog ptr dec when SCHED_REPLACE add failed"),
- dialog_ref(p, "dialog ptr inc when SCHED_REPLACE add succeeded") );
- sip_pvt_unlock(p);
- }
- return res;
-}
-
-/*! \brief Destroy registry object
- Objects created with the register= statement in static configuration */
-static void sip_registry_destroy(void *obj)
-{
- struct sip_registry *reg = obj;
- /* Really delete */
- ast_debug(3, "Destroying registry entry for %s@%s\n", reg->username, reg->hostname);
-
- if (reg->call) {
- /* Clear registry before destroying to ensure
- we don't get reentered trying to grab the registry lock */
- ao2_t_replace(reg->call->registry, NULL, "destroy reg->call->registry");
- ast_debug(3, "Destroying active SIP dialog for registry %s@%s\n", reg->username, reg->hostname);
- dialog_unlink_all(reg->call);
- reg->call = dialog_unref(reg->call, "unref reg->call");
- /* reg->call = sip_destroy(reg->call); */
- }
-
- ast_string_field_free_memory(reg);
-}
-
-/*! \brief Destroy MWI subscription object */
-static void sip_subscribe_mwi_destroy(void *data)
-{
- struct sip_subscription_mwi *mwi = data;
-
- if (mwi->call) {
- mwi->call->mwi = NULL;
- mwi->call = dialog_unref(mwi->call, "sip_subscription_mwi destruction");
- }
-
- ast_string_field_free_memory(mwi);
-}
-
-/*! \brief Destroy SDP media offer list */
-static void offered_media_list_destroy(struct sip_pvt *p)
-{
- struct offered_media *offer;
- while ((offer = AST_LIST_REMOVE_HEAD(&p->offered_media, next))) {
- ast_free(offer->decline_m_line);
- ast_free(offer);
- }
-}
-
-/*! \brief ao2 destructor for SIP dialog structure */
-static void sip_pvt_dtor(void *vdoomed)
-{
- struct sip_pvt *p = vdoomed;
- struct sip_request *req;
-
- ast_debug(3, "Destroying SIP dialog %s\n", p->callid);
-
- /* Destroy Session-Timers if allocated */
- ast_free(p->stimer);
- p->stimer = NULL;
-
- if (sip_debug_test_pvt(p))
- ast_verbose("Really destroying SIP dialog '%s' Method: %s\n", p->callid, sip_methods[p->method].text);
-
- if (ast_test_flag(&p->flags[0], SIP_INC_COUNT) || ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD)) {
- update_call_counter(p, DEC_CALL_LIMIT);
- ast_debug(2, "This call did not properly clean up call limits. Call ID %s\n", p->callid);
- }
-
- /* Unlink us from the owner if we have one */
- if (p->owner) {
- ast_channel_lock(p->owner);
- ast_debug(1, "Detaching from %s\n", ast_channel_name(p->owner));
- ast_channel_tech_pvt_set(p->owner, NULL);
- /* Make sure that the channel knows its backend is going away */
- ast_channel_softhangup_internal_flag_add(p->owner, AST_SOFTHANGUP_DEV);
- ast_channel_unlock(p->owner);
- /* Give the channel a chance to react before deallocation */
- usleep(1);
- }
-
- /* Remove link from peer to subscription of MWI */
- if (p->relatedpeer && p->relatedpeer->mwipvt == p)
- p->relatedpeer->mwipvt = dialog_unref(p->relatedpeer->mwipvt, "delete ->relatedpeer->mwipvt");
- if (p->relatedpeer && p->relatedpeer->call == p)
- p->relatedpeer->call = dialog_unref(p->relatedpeer->call, "unset the relatedpeer->call field in tandem with relatedpeer field itself");
-
- if (p->relatedpeer)
- p->relatedpeer = sip_unref_peer(p->relatedpeer,"unsetting a dialog relatedpeer field in sip_destroy");
-
- if (p->registry) {
- if (p->registry->call == p)
- p->registry->call = dialog_unref(p->registry->call, "nulling out the registry's call dialog field in unlink_all");
- ao2_t_replace(p->registry, NULL, "delete p->registry");
- }
-
- if (p->mwi) {
- p->mwi->call = NULL;
- p->mwi = NULL;
- }
-
- if (dumphistory)
- sip_dump_history(p);
-
- if (p->options) {
- if (p->options->outboundproxy) {
- ao2_ref(p->options->outboundproxy, -1);
- }
- ast_free(p->options);
- p->options = NULL;
- }
-
- if (p->outboundproxy) {
- ref_proxy(p, NULL);
- }
-
- if (p->notify) {
- ast_variables_destroy(p->notify->headers);
- ast_free(p->notify->content);
- ast_free(p->notify);
- p->notify = NULL;
- }
-
- /* Free RTP and SRTP instances */
- dialog_clean_rtp(p);
-
- if (p->udptl) {
- ast_udptl_destroy(p->udptl);
- p->udptl = NULL;
- }
- sip_refer_destroy(p);
- sip_route_clear(&p->route);
- deinit_req(&p->initreq);
-
- /* Clear history */
- if (p->history) {
- struct sip_history *hist;
- while ( (hist = AST_LIST_REMOVE_HEAD(p->history, list)) ) {
- ast_free(hist);
- p->history_entries--;
- }
- ast_free(p->history);
- p->history = NULL;
- }
-
- while ((req = AST_LIST_REMOVE_HEAD(&p->request_queue, next))) {
- ast_free(req);
- }
-
- offered_media_list_destroy(p);
-
- if (p->chanvars) {
- ast_variables_destroy(p->chanvars);
- p->chanvars = NULL;
- }
-
- destroy_msg_headers(p);
-
- if (p->vsrtp) {
- ast_sdp_srtp_destroy(p->vsrtp);
- p->vsrtp = NULL;
- }
-
- if (p->directmediaacl) {
- p->directmediaacl = ast_free_acl_list(p->directmediaacl);
- }
-
- ast_string_field_free_memory(p);
-
- ast_cc_config_params_destroy(p->cc_params);
- p->cc_params = NULL;
-
- if (p->epa_entry) {
- ao2_ref(p->epa_entry, -1);
- p->epa_entry = NULL;
- }
-
- if (p->socket.tcptls_session) {
- ao2_ref(p->socket.tcptls_session, -1);
- p->socket.tcptls_session = NULL;
- } else if (p->socket.ws_session) {
- ast_websocket_unref(p->socket.ws_session);
- p->socket.ws_session = NULL;
- }
-
- if (p->peerauth) {
- ao2_t_ref(p->peerauth, -1, "Removing active peer authentication");
- p->peerauth = NULL;
- }
-
- p->named_callgroups = ast_unref_namedgroups(p->named_callgroups);
- p->named_pickupgroups = ast_unref_namedgroups(p->named_pickupgroups);
-
- ao2_cleanup(p->caps);
- ao2_cleanup(p->jointcaps);
- ao2_cleanup(p->peercaps);
- ao2_cleanup(p->redircaps);
- ao2_cleanup(p->prefcaps);
-
- ast_rtp_dtls_cfg_free(&p->dtls_cfg);
-
- if (p->last_device_state_info) {
- ao2_ref(p->last_device_state_info, -1);
- p->last_device_state_info = NULL;
- }
-}
-
-/*! \brief update_call_counter: Handle call_limit for SIP devices
- * Setting a call-limit will cause calls above the limit not to be accepted.
- *
- * Remember that for a type=friend, there's one limit for the user and
- * another for the peer, not a combined call limit.
- * This will cause unexpected behaviour in subscriptions, since a "friend"
- * is *two* devices in Asterisk, not one.
- *
- * Thought: For realtime, we should probably update storage with inuse counter...
- *
- * \retval 0 if call is ok (no call limit, below threshold).
- * \retval -1 on rejection of call.
- *
- */
-static int update_call_counter(struct sip_pvt *fup, int event)
-{
- char name[256];
- int *inuse = NULL, *call_limit = NULL, *ringing = NULL;
- int outgoing = fup->outgoing_call;
- struct sip_peer *p = NULL;
-
- ast_debug(3, "Updating call counter for %s call\n", outgoing ? "outgoing" : "incoming");
-
-
- /* Test if we need to check call limits, in order to avoid
- realtime lookups if we do not need it */
- if (!ast_test_flag(&fup->flags[0], SIP_CALL_LIMIT) && !ast_test_flag(&fup->flags[1], SIP_PAGE2_CALL_ONHOLD))
- return 0;
-
- ast_copy_string(name, fup->username, sizeof(name));
-
- /* Check the list of devices */
- if (fup->relatedpeer) {
- p = sip_ref_peer(fup->relatedpeer, "ref related peer for update_call_counter");
- inuse = &p->inuse;
- call_limit = &p->call_limit;
- ringing = &p->ringing;
- ast_copy_string(name, fup->peername, sizeof(name));
- }
- if (!p) {
- ast_debug(2, "%s is not a local device, no call limit\n", name);
- return 0;
- }
-
- switch(event) {
- /* incoming and outgoing affects the inuse counter */
- case DEC_CALL_LIMIT:
- /* Decrement inuse count if applicable */
- if (inuse) {
- sip_pvt_lock(fup);
- ao2_lock(p);
- if (*inuse > 0) {
- if (ast_test_flag(&fup->flags[0], SIP_INC_COUNT)) {
- (*inuse)--;
- ast_clear_flag(&fup->flags[0], SIP_INC_COUNT);
- }
- } else {
- *inuse = 0;
- }
- ao2_unlock(p);
- sip_pvt_unlock(fup);
- }
-
- /* Decrement ringing count if applicable */
- if (ringing) {
- sip_pvt_lock(fup);
- ao2_lock(p);
- if (*ringing > 0) {
- if (ast_test_flag(&fup->flags[0], SIP_INC_RINGING)) {
- (*ringing)--;
- ast_clear_flag(&fup->flags[0], SIP_INC_RINGING);
- }
- } else {
- *ringing = 0;
- }
- ao2_unlock(p);
- sip_pvt_unlock(fup);
- }
-
- /* Decrement onhold count if applicable */
- sip_pvt_lock(fup);
- ao2_lock(p);
- if (ast_test_flag(&fup->flags[1], SIP_PAGE2_CALL_ONHOLD) && sip_cfg.notifyhold) {
- ast_clear_flag(&fup->flags[1], SIP_PAGE2_CALL_ONHOLD);
- ao2_unlock(p);
- sip_pvt_unlock(fup);
- sip_peer_hold(fup, FALSE);
- } else {
- ao2_unlock(p);
- sip_pvt_unlock(fup);
- }
- if (sipdebug)
- ast_debug(2, "Call %s %s '%s' removed from call limit %d\n", outgoing ? "to" : "from", "peer", name, *call_limit);
- break;
-
- case INC_CALL_RINGING:
- case INC_CALL_LIMIT:
- /* If call limit is active and we have reached the limit, reject the call */
- if (*call_limit > 0 ) {
- if (*inuse >= *call_limit) {
- ast_log(LOG_NOTICE, "Call %s %s '%s' rejected due to usage limit of %d\n", outgoing ? "to" : "from", "peer", name, *call_limit);
- sip_unref_peer(p, "update_call_counter: unref peer p, call limit exceeded");
- return -1;
- }
- }
- if (ringing && (event == INC_CALL_RINGING)) {
- sip_pvt_lock(fup);
- ao2_lock(p);
- if (!ast_test_flag(&fup->flags[0], SIP_INC_RINGING)) {
- (*ringing)++;
- ast_set_flag(&fup->flags[0], SIP_INC_RINGING);
- }
- ao2_unlock(p);
- sip_pvt_unlock(fup);
- }
- if (inuse) {
- sip_pvt_lock(fup);
- ao2_lock(p);
- if (!ast_test_flag(&fup->flags[0], SIP_INC_COUNT)) {
- (*inuse)++;
- ast_set_flag(&fup->flags[0], SIP_INC_COUNT);
- }
- ao2_unlock(p);
- sip_pvt_unlock(fup);
- }
- if (sipdebug) {
- ast_debug(2, "Call %s %s '%s' is %d out of %d\n", outgoing ? "to" : "from", "peer", name, *inuse, *call_limit);
- }
- break;
-
- case DEC_CALL_RINGING:
- if (ringing) {
- sip_pvt_lock(fup);
- ao2_lock(p);
- if (ast_test_flag(&fup->flags[0], SIP_INC_RINGING)) {
- if (*ringing > 0) {
- (*ringing)--;
- }
- ast_clear_flag(&fup->flags[0], SIP_INC_RINGING);
- }
- ao2_unlock(p);
- sip_pvt_unlock(fup);
- }
- break;
-
- default:
- ast_log(LOG_ERROR, "update_call_counter(%s, %d) called with no event!\n", name, event);
- }
-
- ast_devstate_changed(AST_DEVICE_UNKNOWN, AST_DEVSTATE_CACHABLE, "SIP/%s", p->name);
- sip_unref_peer(p, "update_call_counter: sip_unref_peer from call counter");
-
- return 0;
-}
-
-/*! \brief Convert SIP hangup causes to Asterisk hangup causes */
-int hangup_sip2cause(int cause)
-{
- /* Possible values taken from causes.h */
-
- switch(cause) {
- case 401: /* Unauthorized */
- return AST_CAUSE_CALL_REJECTED;
- case 403: /* Not found */
- return AST_CAUSE_CALL_REJECTED;
- case 404: /* Not found */
- return AST_CAUSE_UNALLOCATED;
- case 405: /* Method not allowed */
- return AST_CAUSE_INTERWORKING;
- case 407: /* Proxy authentication required */
- return AST_CAUSE_CALL_REJECTED;
- case 408: /* No reaction */
- return AST_CAUSE_NO_USER_RESPONSE;
- case 409: /* Conflict */
- return AST_CAUSE_NORMAL_TEMPORARY_FAILURE;
- case 410: /* Gone */
- return AST_CAUSE_NUMBER_CHANGED;
- case 411: /* Length required */
- return AST_CAUSE_INTERWORKING;
- case 413: /* Request entity too large */
- return AST_CAUSE_INTERWORKING;
- case 414: /* Request URI too large */
- return AST_CAUSE_INTERWORKING;
- case 415: /* Unsupported media type */
- return AST_CAUSE_INTERWORKING;
- case 420: /* Bad extension */
- return AST_CAUSE_NO_ROUTE_DESTINATION;
- case 480: /* No answer */
- return AST_CAUSE_NO_ANSWER;
- case 481: /* No answer */
- return AST_CAUSE_INTERWORKING;
- case 482: /* Loop detected */
- return AST_CAUSE_INTERWORKING;
- case 483: /* Too many hops */
- return AST_CAUSE_NO_ANSWER;
- case 484: /* Address incomplete */
- return AST_CAUSE_INVALID_NUMBER_FORMAT;
- case 485: /* Ambiguous */
- return AST_CAUSE_UNALLOCATED;
- case 486: /* Busy everywhere */
- return AST_CAUSE_BUSY;
- case 487: /* Request terminated */
- return AST_CAUSE_INTERWORKING;
- case 488: /* No codecs approved */
- return AST_CAUSE_BEARERCAPABILITY_NOTAVAIL;
- case 491: /* Request pending */
- return AST_CAUSE_INTERWORKING;
- case 493: /* Undecipherable */
- return AST_CAUSE_INTERWORKING;
- case 500: /* Server internal failure */
- return AST_CAUSE_FAILURE;
- case 501: /* Call rejected */
- return AST_CAUSE_FACILITY_REJECTED;
- case 502:
- return AST_CAUSE_DESTINATION_OUT_OF_ORDER;
- case 503: /* Service unavailable */
- return AST_CAUSE_CONGESTION;
- case 504: /* Gateway timeout */
- return AST_CAUSE_RECOVERY_ON_TIMER_EXPIRE;
- case 505: /* SIP version not supported */
- return AST_CAUSE_INTERWORKING;
- case 600: /* Busy everywhere */
- return AST_CAUSE_USER_BUSY;
- case 603: /* Decline */
- return AST_CAUSE_CALL_REJECTED;
- case 604: /* Does not exist anywhere */
- return AST_CAUSE_UNALLOCATED;
- case 606: /* Not acceptable */
- return AST_CAUSE_BEARERCAPABILITY_NOTAVAIL;
- default:
- if (cause < 500 && cause >= 400) {
- /* 4xx class error that is unknown - someting wrong with our request */
- return AST_CAUSE_INTERWORKING;
- } else if (cause < 600 && cause >= 500) {
- /* 5xx class error - problem in the remote end */
- return AST_CAUSE_CONGESTION;
- } else if (cause < 700 && cause >= 600) {
- /* 6xx - global errors in the 4xx class */
- return AST_CAUSE_INTERWORKING;
- }
- return AST_CAUSE_NORMAL;
- }
- /* Never reached */
- return 0;
-}
-
-/*! \brief Convert Asterisk hangup causes to SIP codes
-\verbatim
- Possible values from causes.h
- AST_CAUSE_NOTDEFINED AST_CAUSE_NORMAL AST_CAUSE_BUSY
- AST_CAUSE_FAILURE AST_CAUSE_CONGESTION AST_CAUSE_UNALLOCATED
-
- In addition to these, a lot of PRI codes is defined in causes.h
- ...should we take care of them too ?
-
- Quote RFC 3398
-
- ISUP Cause value SIP response
- ---------------- ------------
- 1 unallocated number 404 Not Found
- 2 no route to network 404 Not found
- 3 no route to destination 404 Not found
- 16 normal call clearing --- (*)
- 17 user busy 486 Busy here
- 18 no user responding 408 Request Timeout
- 19 no answer from the user 480 Temporarily unavailable
- 20 subscriber absent 480 Temporarily unavailable
- 21 call rejected 403 Forbidden (+)
- 22 number changed (w/o diagnostic) 410 Gone
- 22 number changed (w/ diagnostic) 301 Moved Permanently
- 23 redirection to new destination 410 Gone
- 26 non-selected user clearing 404 Not Found (=)
- 27 destination out of order 502 Bad Gateway
- 28 address incomplete 484 Address incomplete
- 29 facility rejected 501 Not implemented
- 31 normal unspecified 480 Temporarily unavailable
-\endverbatim
-*/
-const char *hangup_cause2sip(int cause)
-{
- switch (cause) {
- case AST_CAUSE_UNALLOCATED: /* 1 */
- case AST_CAUSE_NO_ROUTE_DESTINATION: /* 3 IAX2: Can't find extension in context */
- case AST_CAUSE_NO_ROUTE_TRANSIT_NET: /* 2 */
- return "404 Not Found";
- case AST_CAUSE_CONGESTION: /* 34 */
- case AST_CAUSE_SWITCH_CONGESTION: /* 42 */
- return "503 Service Unavailable";
- case AST_CAUSE_NO_USER_RESPONSE: /* 18 */
- return "408 Request Timeout";
- case AST_CAUSE_NO_ANSWER: /* 19 */
- case AST_CAUSE_UNREGISTERED: /* 20 */
- return "480 Temporarily unavailable";
- case AST_CAUSE_CALL_REJECTED: /* 21 */
- return "403 Forbidden";
- case AST_CAUSE_NUMBER_CHANGED: /* 22 */
- return "410 Gone";
- case AST_CAUSE_NORMAL_UNSPECIFIED: /* 31 */
- return "480 Temporarily unavailable";
- case AST_CAUSE_INVALID_NUMBER_FORMAT:
- return "484 Address incomplete";
- case AST_CAUSE_USER_BUSY:
- return "486 Busy here";
- case AST_CAUSE_FAILURE:
- return "500 Server internal failure";
- case AST_CAUSE_FACILITY_REJECTED: /* 29 */
- return "501 Not Implemented";
- case AST_CAUSE_CHAN_NOT_IMPLEMENTED:
- return "503 Service Unavailable";
- /* Used in chan_iax2 */
- case AST_CAUSE_DESTINATION_OUT_OF_ORDER:
- return "502 Bad Gateway";
- case AST_CAUSE_BEARERCAPABILITY_NOTAVAIL: /* Can't find codec to connect to host */
- return "488 Not Acceptable Here";
- case AST_CAUSE_INTERWORKING: /* Unspecified Interworking issues */
- return "500 Network error";
-
- case AST_CAUSE_NOTDEFINED:
- default:
- ast_debug(1, "AST hangup cause %d (no match found in SIP)\n", cause);
- return NULL;
- }
-
- /* Never reached */
- return 0;
-}
-
-/* Run by the sched thread. */
-static int reinvite_timeout(const void *data)
-{
- struct sip_pvt *dialog = (struct sip_pvt *) data;
- struct ast_channel *owner;
-
- owner = sip_pvt_lock_full(dialog);
- dialog->reinviteid = -1;
- check_pendings(dialog);
- if (owner) {
- ast_channel_unlock(owner);
- ast_channel_unref(owner);
- }
- sip_pvt_unlock(dialog);
- dialog_unref(dialog, "reinviteid complete");
- return 0;
-}
-
-/* Run by the sched thread. */
-static int __stop_reinviteid(const void *data)
-{
- struct sip_pvt *pvt = (void *) data;
-
- AST_SCHED_DEL_UNREF(sched, pvt->reinviteid,
- dialog_unref(pvt, "Stop scheduled reinviteid"));
- dialog_unref(pvt, "Stop reinviteid action");
- return 0;
-}
-
-static void stop_reinviteid(struct sip_pvt *pvt)
-{
- dialog_ref(pvt, "Stop reinviteid action");
- if (ast_sched_add(sched, 0, __stop_reinviteid, pvt) < 0) {
- /* Uh Oh. Expect bad behavior. */
- dialog_unref(pvt, "Failed to schedule stop reinviteid action");
- }
-}
-
-/*! \brief sip_hangup: Hangup SIP call
- * Part of PBX interface, called from ast_hangup */
-static int sip_hangup(struct ast_channel *ast)
-{
- struct sip_pvt *p = ast_channel_tech_pvt(ast);
- int needcancel = FALSE;
- int needdestroy = 0;
- struct ast_channel *oldowner = ast;
-
- if (!p) {
- ast_debug(1, "Asked to hangup channel that was not connected\n");
- return 0;
- }
- if (ast_channel_hangupcause(ast) == AST_CAUSE_ANSWERED_ELSEWHERE) {
- ast_debug(1, "This call was answered elsewhere\n");
- append_history(p, "Cancel", "Call answered elsewhere");
- p->answered_elsewhere = TRUE;
- }
-
- /* Store hangupcause locally in PVT so we still have it before disconnect */
- if (p->owner)
- p->hangupcause = ast_channel_hangupcause(p->owner);
-
- if (ast_test_flag(&p->flags[0], SIP_DEFER_BYE_ON_TRANSFER)) {
- if (ast_test_flag(&p->flags[0], SIP_INC_COUNT) || ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD)) {
- if (sipdebug)
- ast_debug(1, "update_call_counter(%s) - decrement call limit counter on hangup\n", p->username);
- update_call_counter(p, DEC_CALL_LIMIT);
- }
- ast_debug(4, "SIP Transfer: Not hanging up right now... Rescheduling hangup for %s.\n", p->callid);
- sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
- ast_clear_flag(&p->flags[0], SIP_DEFER_BYE_ON_TRANSFER); /* Really hang up next time */
- if (p->owner) {
- sip_pvt_lock(p);
- oldowner = p->owner;
- sip_set_owner(p, NULL); /* Owner will be gone after we return, so take it away */
- sip_pvt_unlock(p);
- ast_channel_tech_pvt_set(oldowner, dialog_unref(ast_channel_tech_pvt(oldowner), "unref oldowner->tech_pvt"));
- }
- ast_module_unref(ast_module_info->self);
- return 0;
- }
-
- ast_debug(1, "Hangup call %s, SIP callid %s\n", ast_channel_name(ast), p->callid);
-
- sip_pvt_lock(p);
- if (ast_test_flag(&p->flags[0], SIP_INC_COUNT) || ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD)) {
- if (sipdebug)
- ast_debug(1, "update_call_counter(%s) - decrement call limit counter on hangup\n", p->username);
- update_call_counter(p, DEC_CALL_LIMIT);
- }
-
- /* Determine how to disconnect */
- if (p->owner != ast) {
- ast_log(LOG_WARNING, "Huh? We aren't the owner? Can't hangup call.\n");
- sip_pvt_unlock(p);
- return 0;
- }
- /* If the call is not UP, we need to send CANCEL instead of BYE */
- /* In case of re-invites, the call might be UP even though we have an incomplete invite transaction */
- if (p->invitestate < INV_COMPLETED && ast_channel_state(p->owner) != AST_STATE_UP) {
- needcancel = TRUE;
- ast_debug(4, "Hanging up channel in state %s (not UP)\n", ast_state2str(ast_channel_state(ast)));
- }
-
- stop_media_flows(p); /* Immediately stop RTP, VRTP and UDPTL as applicable */
-
- append_history(p, needcancel ? "Cancel" : "Hangup", "Cause %s", ast_cause2str(p->hangupcause));
-
- /* Disconnect */
- disable_dsp_detect(p);
-
- sip_set_owner(p, NULL);
- ast_channel_tech_pvt_set(ast, NULL);
-
- ast_module_unref(ast_module_info->self);
- /* Do not destroy this pvt until we have timeout or
- get an answer to the BYE or INVITE/CANCEL
- If we get no answer during retransmit period, drop the call anyway.
- (Sorry, mother-in-law, you can't deny a hangup by sending
- 603 declined to BYE...)
- */
- if (p->alreadygone)
- needdestroy = 1; /* Set destroy flag at end of this function */
- else if (p->invitestate != INV_CALLING)
- sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
-
- /* Start the process if it's not already started */
- if (!p->alreadygone && p->initreq.data && ast_str_strlen(p->initreq.data)) {
- if (needcancel) { /* Outgoing call, not up */
- if (ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
- /* if we can't send right now, mark it pending */
- if (p->invitestate == INV_CALLING) {
- /* We can't send anything in CALLING state */
- ast_set_flag(&p->flags[0], SIP_PENDINGBYE);
- /* Do we need a timer here if we don't hear from them at all? Yes we do or else we will get hung dialogs and those are no fun. */
- sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
- append_history(p, "DELAY", "Not sending cancel, waiting for timeout");
- } else {
- struct sip_pkt *cur;
-
- for (cur = p->packets; cur; cur = cur->next) {
- __sip_semi_ack(p, cur->seqno, cur->is_resp, cur->method ? cur->method : find_sip_method(ast_str_buffer(cur->data)));
- }
- p->invitestate = INV_CANCELLED;
- /* Send a new request: CANCEL */
- transmit_request(p, SIP_CANCEL, p->lastinvite, XMIT_RELIABLE, FALSE);
- /* Actually don't destroy us yet, wait for the 487 on our original
- INVITE, but do set an autodestruct just in case we never get it. */
- needdestroy = 0;
- sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
- }
- } else { /* Incoming call, not up */
- const char *res;
-
- stop_provisional_keepalive(p);
- if (p->hangupcause && (res = hangup_cause2sip(p->hangupcause)))
- transmit_response_reliable(p, res, &p->initreq);
- else
- transmit_response_reliable(p, "603 Declined", &p->initreq);
- p->invitestate = INV_TERMINATED;
- }
- } else { /* Call is in UP state, send BYE */
- if (p->stimer) {
- stop_session_timer(p);
- }
-
- if (!p->pendinginvite) {
- char *quality;
- char quality_buf[AST_MAX_USER_FIELD];
-
- if (p->rtp) {
- struct ast_rtp_instance *p_rtp;
-
- p_rtp = p->rtp;
- ao2_ref(p_rtp, +1);
- ast_channel_unlock(oldowner);
- sip_pvt_unlock(p);
- ast_rtp_instance_set_stats_vars(oldowner, p_rtp);
- ao2_ref(p_rtp, -1);
- ast_channel_lock(oldowner);
- sip_pvt_lock(p);
- }
-
- /*
- * The channel variables are set below just to get the AMI
- * VarSet event because the channel is being hungup.
- */
- if (p->rtp || p->vrtp || p->trtp) {
- ast_channel_stage_snapshot(oldowner);
- }
- if (p->rtp && (quality = ast_rtp_instance_get_quality(p->rtp, AST_RTP_INSTANCE_STAT_FIELD_QUALITY, quality_buf, sizeof(quality_buf)))) {
- if (p->do_history) {
- append_history(p, "RTCPaudio", "Quality:%s", quality);
- }
- pbx_builtin_setvar_helper(oldowner, "RTPAUDIOQOS", quality);
- }
- if (p->vrtp && (quality = ast_rtp_instance_get_quality(p->vrtp, AST_RTP_INSTANCE_STAT_FIELD_QUALITY, quality_buf, sizeof(quality_buf)))) {
- if (p->do_history) {
- append_history(p, "RTCPvideo", "Quality:%s", quality);
- }
- pbx_builtin_setvar_helper(oldowner, "RTPVIDEOQOS", quality);
- }
- if (p->trtp && (quality = ast_rtp_instance_get_quality(p->trtp, AST_RTP_INSTANCE_STAT_FIELD_QUALITY, quality_buf, sizeof(quality_buf)))) {
- if (p->do_history) {
- append_history(p, "RTCPtext", "Quality:%s", quality);
- }
- pbx_builtin_setvar_helper(oldowner, "RTPTEXTQOS", quality);
- }
- if (p->rtp || p->vrtp || p->trtp) {
- ast_channel_stage_snapshot_done(oldowner);
- }
-
- /* Send a hangup */
- if (ast_channel_state(oldowner) == AST_STATE_UP) {
- transmit_request_with_auth(p, SIP_BYE, 0, XMIT_RELIABLE, 1);
- }
-
- } else {
- /* Note we will need a BYE when this all settles out
- but we can't send one while we have "INVITE" outstanding. */
- ast_set_flag(&p->flags[0], SIP_PENDINGBYE);
- ast_clear_flag(&p->flags[0], SIP_NEEDREINVITE);
- stop_reinvite_retry(p);
- sip_cancel_destroy(p);
-
- /* If we have an ongoing reinvite, there is a chance that we have gotten a provisional
- * response, but something weird has happened and we will never receive a final response.
- * So, just in case, check for pending actions after a bit of time to trigger the pending
- * bye that we are setting above */
- if (p->ongoing_reinvite && p->reinviteid < 0) {
- p->reinviteid = ast_sched_add(sched, 32 * p->timer_t1,
- reinvite_timeout, dialog_ref(p, "Schedule reinviteid"));
- if (p->reinviteid < 0) {
- /* Uh Oh. Expect bad behavior. */
- dialog_unref(p, "Failed to schedule reinviteid");
- }
- }
- }
- }
- }
- if (needdestroy) {
- pvt_set_needdestroy(p, "hangup");
- }
- sip_pvt_unlock(p);
- dialog_unref(p, "unref ast->tech_pvt");
- return 0;
-}
-
-/*! \brief Try setting the codecs suggested by the SIP_CODEC channel variable */
-static void try_suggested_sip_codec(struct sip_pvt *p)
-{
- const char *codec_list;
- char *codec_list_copy;
- struct ast_format_cap *original_jointcaps;
- char *codec;
- int first_codec = 1;
-
- char *strtok_ptr;
-
- if (p->outgoing_call) {
- codec_list = pbx_builtin_getvar_helper(p->owner, "SIP_CODEC_OUTBOUND");
- } else if (!(codec_list = pbx_builtin_getvar_helper(p->owner, "SIP_CODEC_INBOUND"))) {
- codec_list = pbx_builtin_getvar_helper(p->owner, "SIP_CODEC");
- }
-
- if (ast_strlen_zero(codec_list)) {
- return;
- }
-
- codec_list_copy = ast_strdupa(codec_list);
-
- original_jointcaps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
- if (!original_jointcaps) {
- return;
- }
- ast_format_cap_append_from_cap(original_jointcaps, p->jointcaps, AST_MEDIA_TYPE_UNKNOWN);
-
- for (codec = strtok_r(codec_list_copy, ",", &strtok_ptr); codec; codec = strtok_r(NULL, ",", &strtok_ptr)) {
- struct ast_format *fmt;
-
- codec = ast_strip(codec);
-
- fmt = ast_format_cache_get(codec);
- if (!fmt) {
- ast_log(AST_LOG_NOTICE, "Ignoring ${SIP_CODEC*} variable because of unrecognized/not configured codec %s (check allow/disallow in sip.conf)\n", codec);
- continue;
- }
- if (ast_format_cap_iscompatible_format(original_jointcaps, fmt) != AST_FORMAT_CMP_NOT_EQUAL) {
- if (first_codec) {
- ast_verb(4, "Set codec to '%s' for this call because of ${SIP_CODEC*} variable\n", codec);
- ast_format_cap_remove_by_type(p->jointcaps, AST_MEDIA_TYPE_UNKNOWN);
- ast_format_cap_append(p->jointcaps, fmt, 0);
- ast_format_cap_remove_by_type(p->caps, AST_MEDIA_TYPE_UNKNOWN);
- ast_format_cap_append(p->caps, fmt, 0);
- first_codec = 0;
- } else {
- ast_verb(4, "Add codec to '%s' for this call because of ${SIP_CODEC*} variable\n", codec);
- /* Add the format to the capabilities structure */
- ast_format_cap_append(p->jointcaps, fmt, 0);
- ast_format_cap_append(p->caps, fmt, 0);
- }
- } else {
- ast_log(AST_LOG_NOTICE, "Ignoring ${SIP_CODEC*} variable because it is not shared by both ends: %s\n", codec);
- }
-
- ao2_ref(fmt, -1);
- }
-
- /* The original joint formats may have contained negotiated parameters (fmtp)
- * like the Opus Codec or iLBC 20. The cached formats contain the default
- * parameters, which could be different than the negotiated (joint) result. */
- ast_format_cap_replace_from_cap(p->jointcaps, original_jointcaps, AST_MEDIA_TYPE_UNKNOWN);
-
- ao2_ref(original_jointcaps, -1);
- return;
- }
-
-
-/*! \brief sip_answer: Answer SIP call , send 200 OK on Invite
- * Part of PBX interface */
-static int sip_answer(struct ast_channel *ast)
-{
- int res = 0;
- struct sip_pvt *p = ast_channel_tech_pvt(ast);
- int oldsdp = FALSE;
-
- if (!p) {
- ast_debug(1, "Asked to answer channel %s without tech pvt; ignoring\n",
- ast_channel_name(ast));
- return res;
- }
- sip_pvt_lock(p);
- if (ast_channel_state(ast) != AST_STATE_UP) {
- try_suggested_sip_codec(p);
-
- if (ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT)) {
- oldsdp = TRUE;
- }
-
- ast_setstate(ast, AST_STATE_UP);
- ast_debug(1, "SIP answering channel: %s\n", ast_channel_name(ast));
- ast_rtp_instance_update_source(p->rtp);
- res = transmit_response_with_sdp(p, "200 OK", &p->initreq, XMIT_CRITICAL, oldsdp, TRUE);
- ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
- /* RFC says the session timer starts counting on 200,
- * not on INVITE. */
- if (p->stimer) {
- restart_session_timer(p);
- }
- }
- sip_pvt_unlock(p);
- return res;
-}
-
-/*! \brief Send frame to media channel (rtp) */
-static int sip_write(struct ast_channel *ast, struct ast_frame *frame)
-{
- struct sip_pvt *p = ast_channel_tech_pvt(ast);
- int res = 0;
-
- switch (frame->frametype) {
- case AST_FRAME_VOICE:
- if (ast_format_cap_iscompatible_format(ast_channel_nativeformats(ast), frame->subclass.format) == AST_FORMAT_CMP_NOT_EQUAL) {
- struct ast_str *codec_buf = ast_str_alloca(AST_FORMAT_CAP_NAMES_LEN);
- ast_log(LOG_WARNING, "Asked to transmit frame type %s, while native formats is %s read/write = %s/%s\n",
- ast_format_get_name(frame->subclass.format),
- ast_format_cap_get_names(ast_channel_nativeformats(ast), &codec_buf),
- ast_format_get_name(ast_channel_readformat(ast)),
- ast_format_get_name(ast_channel_writeformat(ast)));
- return 0;
- }
- if (p) {
- sip_pvt_lock(p);
- if (p->t38.state == T38_ENABLED) {
- /* drop frame, can't sent VOICE frames while in T.38 mode */
- sip_pvt_unlock(p);
- break;
- } else if (p->rtp) {
- /* If channel is not up, activate early media session */
- if ((ast_channel_state(ast) != AST_STATE_UP) &&
- !ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT) &&
- !ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
- ast_rtp_instance_update_source(p->rtp);
- if (!global_prematuremediafilter) {
- p->invitestate = INV_EARLY_MEDIA;
- transmit_provisional_response(p, "183 Session Progress", &p->initreq, TRUE);
- ast_set_flag(&p->flags[0], SIP_PROGRESS_SENT);
- }
- }
- if (p->invitestate > INV_EARLY_MEDIA || (p->invitestate == INV_EARLY_MEDIA &&
- ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT))) {
- p->lastrtptx = time(NULL);
- res = ast_rtp_instance_write(p->rtp, frame);
- }
- }
- sip_pvt_unlock(p);
- }
- break;
- case AST_FRAME_VIDEO:
- if (p) {
- sip_pvt_lock(p);
- if (p->vrtp) {
- /* Activate video early media */
- if ((ast_channel_state(ast) != AST_STATE_UP) &&
- !ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT) &&
- !ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
- p->invitestate = INV_EARLY_MEDIA;
- transmit_provisional_response(p, "183 Session Progress", &p->initreq, TRUE);
- ast_set_flag(&p->flags[0], SIP_PROGRESS_SENT);
- }
- if (p->invitestate > INV_EARLY_MEDIA || (p->invitestate == INV_EARLY_MEDIA &&
- ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT))) {
- p->lastrtptx = time(NULL);
- res = ast_rtp_instance_write(p->vrtp, frame);
- }
- }
- sip_pvt_unlock(p);
- }
- break;
- case AST_FRAME_TEXT:
- if (p) {
- sip_pvt_lock(p);
- if (p->red) {
- ast_rtp_red_buffer(p->trtp, frame);
- } else {
- if (p->trtp) {
- /* Activate text early media */
- if ((ast_channel_state(ast) != AST_STATE_UP) &&
- !ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT) &&
- !ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
- p->invitestate = INV_EARLY_MEDIA;
- transmit_provisional_response(p, "183 Session Progress", &p->initreq, TRUE);
- ast_set_flag(&p->flags[0], SIP_PROGRESS_SENT);
- }
- if (p->invitestate > INV_EARLY_MEDIA || (p->invitestate == INV_EARLY_MEDIA &&
- ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT))) {
- p->lastrtptx = time(NULL);
- res = ast_rtp_instance_write(p->trtp, frame);
- }
- }
- }
- sip_pvt_unlock(p);
- }
- break;
- case AST_FRAME_IMAGE:
- return 0;
- break;
- case AST_FRAME_MODEM:
- if (p) {
- sip_pvt_lock(p);
- /* UDPTL requires two-way communication, so early media is not needed here.
- we simply forget the frames if we get modem frames before the bridge is up.
- Fax will re-transmit.
- */
- if ((ast_channel_state(ast) == AST_STATE_UP) &&
- p->udptl &&
- (p->t38.state == T38_ENABLED)) {
- res = ast_udptl_write(p->udptl, frame);
- }
- sip_pvt_unlock(p);
- }
- break;
- default:
- ast_log(LOG_WARNING, "Can't send %u type frames with SIP write\n", frame->frametype);
- return 0;
- }
-
- return res;
-}
-
-/*! \brief sip_fixup: Fix up a channel: If a channel is consumed, this is called.
- Basically update any ->owner links */
-static int sip_fixup(struct ast_channel *oldchan, struct ast_channel *newchan)
-{
- int ret = -1;
- struct sip_pvt *p;
-
- if (newchan && ast_test_flag(ast_channel_flags(newchan), AST_FLAG_ZOMBIE))
- ast_debug(1, "New channel is zombie\n");
- if (oldchan && ast_test_flag(ast_channel_flags(oldchan), AST_FLAG_ZOMBIE))
- ast_debug(1, "Old channel is zombie\n");
-
- if (!newchan || !ast_channel_tech_pvt(newchan)) {
- if (!newchan)
- ast_log(LOG_WARNING, "No new channel! Fixup of %s failed.\n", ast_channel_name(oldchan));
- else
- ast_log(LOG_WARNING, "No SIP tech_pvt! Fixup of %s failed.\n", ast_channel_name(oldchan));
- return -1;
- }
- p = ast_channel_tech_pvt(newchan);
-
- sip_pvt_lock(p);
- append_history(p, "Masq", "Old channel: %s\n", ast_channel_name(oldchan));
- append_history(p, "Masq (cont)", "...new owner: %s\n", ast_channel_name(newchan));
- if (p->owner != oldchan)
- ast_log(LOG_WARNING, "old channel wasn't %p but was %p\n", oldchan, p->owner);
- else {
- sip_set_owner(p, newchan);
- /* Re-invite RTP back to Asterisk. Needed if channel is masqueraded out of a native
- RTP bridge (i.e., RTP not going through Asterisk): RTP bridge code might not be
- able to do this if the masquerade happens before the bridge breaks (e.g., AMI
- redirect of both channels). Note that a channel can not be masqueraded *into*
- a native bridge. So there is no danger that this breaks a native bridge that
- should stay up. */
- sip_set_rtp_peer(newchan, NULL, NULL, NULL, NULL, 0);
- ret = 0;
- }
- ast_debug(3, "SIP Fixup: New owner for dialogue %s: %s (Old parent: %s)\n", p->callid, ast_channel_name(p->owner), ast_channel_name(oldchan));
-
- sip_pvt_unlock(p);
- return ret;
-}
-
-static int sip_senddigit_begin(struct ast_channel *ast, char digit)
-{
- struct sip_pvt *p = ast_channel_tech_pvt(ast);
- int res = 0;
-
- if (!p) {
- ast_debug(1, "Asked to begin DTMF digit on channel %s with no pvt; ignoring\n",
- ast_channel_name(ast));
- return res;
- }
-
- sip_pvt_lock(p);
- switch (ast_test_flag(&p->flags[0], SIP_DTMF)) {
- case SIP_DTMF_INBAND:
- res = -1; /* Tell Asterisk to generate inband indications */
- break;
- case SIP_DTMF_RFC2833:
- if (p->rtp)
- ast_rtp_instance_dtmf_begin(p->rtp, digit);
- break;
- default:
- break;
- }
- sip_pvt_unlock(p);
-
- return res;
-}
-
-/*! \brief Send DTMF character on SIP channel
- within one call, we're able to transmit in many methods simultaneously */
-static int sip_senddigit_end(struct ast_channel *ast, char digit, unsigned int duration)
-{
- struct sip_pvt *p = ast_channel_tech_pvt(ast);
- int res = 0;
-
- if (!p) {
- ast_debug(1, "Asked to end DTMF digit on channel %s with no pvt; ignoring\n",
- ast_channel_name(ast));
- return res;
- }
-
- sip_pvt_lock(p);
- switch (ast_test_flag(&p->flags[0], SIP_DTMF)) {
- case SIP_DTMF_INFO:
- case SIP_DTMF_SHORTINFO:
- transmit_info_with_digit(p, digit, duration);
- break;
- case SIP_DTMF_RFC2833:
- if (p->rtp)
- ast_rtp_instance_dtmf_end_with_duration(p->rtp, digit, duration);
- break;
- case SIP_DTMF_INBAND:
- res = -1; /* Tell Asterisk to stop inband indications */
- break;
- }
- sip_pvt_unlock(p);
-
- return res;
-}
-
-/*! \brief Transfer SIP call */
-static int sip_transfer(struct ast_channel *ast, const char *dest)
-{
- struct sip_pvt *p = ast_channel_tech_pvt(ast);
- int res;
-
- if (!p) {
- ast_debug(1, "Asked to transfer channel %s with no pvt; ignoring\n",
- ast_channel_name(ast));
- return -1;
- }
-
- if (dest == NULL) /* functions below do not take a NULL */
- dest = "";
- sip_pvt_lock(p);
- if (ast_channel_state(ast) == AST_STATE_RING)
- res = sip_sipredirect(p, dest);
- else
- res = transmit_refer(p, dest);
- sip_pvt_unlock(p);
- return res;
-}
-
-/*! \brief Helper function which updates T.38 capability information and triggers a reinvite */
-static int interpret_t38_parameters(struct sip_pvt *p, const struct ast_control_t38_parameters *parameters)
-{
- int res = 0;
-
- if (!ast_test_flag(&p->flags[1], SIP_PAGE2_T38SUPPORT) || !p->udptl) {
- return -1;
- }
- switch (parameters->request_response) {
- case AST_T38_NEGOTIATED:
- case AST_T38_REQUEST_NEGOTIATE: /* Request T38 */
- /* Negotiation can not take place without a valid max_ifp value. */
- if (!parameters->max_ifp) {
- if (p->t38.state == T38_PEER_REINVITE) {
- stop_t38_abort_timer(p);
- transmit_response_reliable(p, "488 Not acceptable here", &p->initreq);
- }
- change_t38_state(p, T38_REJECTED);
- break;
- } else if (p->t38.state == T38_PEER_REINVITE) {
- stop_t38_abort_timer(p);
- p->t38.our_parms = *parameters;
- /* modify our parameters to conform to the peer's parameters,
- * based on the rules in the ITU T.38 recommendation
- */
- if (!p->t38.their_parms.fill_bit_removal) {
- p->t38.our_parms.fill_bit_removal = FALSE;
- }
- if (!p->t38.their_parms.transcoding_mmr) {
- p->t38.our_parms.transcoding_mmr = FALSE;
- }
- if (!p->t38.their_parms.transcoding_jbig) {
- p->t38.our_parms.transcoding_jbig = FALSE;
- }
- p->t38.our_parms.version = MIN(p->t38.our_parms.version, p->t38.their_parms.version);
- p->t38.our_parms.rate_management = p->t38.their_parms.rate_management;
- ast_udptl_set_local_max_ifp(p->udptl, p->t38.our_parms.max_ifp);
- change_t38_state(p, T38_ENABLED);
- transmit_response_with_t38_sdp(p, "200 OK", &p->initreq, XMIT_CRITICAL);
- } else if ((p->t38.state != T38_ENABLED) || ((p->t38.state == T38_ENABLED) &&
- (parameters->request_response == AST_T38_REQUEST_NEGOTIATE))) {
- p->t38.our_parms = *parameters;
- ast_udptl_set_local_max_ifp(p->udptl, p->t38.our_parms.max_ifp);
- change_t38_state(p, T38_LOCAL_REINVITE);
- if (!p->pendinginvite) {
- transmit_reinvite_with_sdp(p, TRUE, FALSE);
- } else if (!ast_test_flag(&p->flags[0], SIP_PENDINGBYE)) {
- ast_set_flag(&p->flags[0], SIP_NEEDREINVITE);
- }
- }
- break;
- case AST_T38_TERMINATED:
- case AST_T38_REFUSED:
- case AST_T38_REQUEST_TERMINATE: /* Shutdown T38 */
- if (p->t38.state == T38_PEER_REINVITE) {
- stop_t38_abort_timer(p);
- change_t38_state(p, T38_REJECTED);
- transmit_response_reliable(p, "488 Not acceptable here", &p->initreq);
- } else if (p->t38.state == T38_ENABLED) {
- change_t38_state(p, T38_DISABLED);
- transmit_reinvite_with_sdp(p, FALSE, FALSE);
- }
- break;
- case AST_T38_REQUEST_PARMS: { /* Application wants remote's parameters re-sent */
- struct ast_control_t38_parameters parameters = p->t38.their_parms;
-
- if (p->t38.state == T38_PEER_REINVITE) {
- stop_t38_abort_timer(p);
- parameters.max_ifp = ast_udptl_get_far_max_ifp(p->udptl);
- parameters.request_response = AST_T38_REQUEST_NEGOTIATE;
- if (p->owner) {
- ast_queue_control_data(p->owner, AST_CONTROL_T38_PARAMETERS, ¶meters, sizeof(parameters));
- }
- /* we need to return a positive value here, so that applications that
- * send this request can determine conclusively whether it was accepted or not...
- * older versions of chan_sip would just silently accept it and return zero.
- */
- res = AST_T38_REQUEST_PARMS;
- }
- break;
- }
- default:
- res = -1;
- break;
- }
-
- return res;
-}
-
-enum sip_media_fds {
- SIP_AUDIO_RTP_FD,
- SIP_AUDIO_RTCP_FD,
- SIP_VIDEO_RTP_FD,
- SIP_VIDEO_RTCP_FD,
- SIP_TEXT_RTP_FD,
- SIP_UDPTL_FD,
-};
-
-/*!
- * \internal
- * \brief Create and initialize UDPTL for the specified dialog
- *
- * \param p SIP private structure to create UDPTL object for
- * \pre p is locked
- * \pre p->owner is locked
- *
- * \note In the case of failure, SIP_PAGE2_T38SUPPORT is cleared on p
- *
- * \return 0 on success, any other value on failure
- */
-static int initialize_udptl(struct sip_pvt *p)
-{
- int natflags = ast_test_flag(&p->flags[1], SIP_PAGE2_SYMMETRICRTP);
-
- if (!ast_test_flag(&p->flags[1], SIP_PAGE2_T38SUPPORT)) {
- return 1;
- }
-
- /* If we've already initialized T38, don't take any further action */
- if (p->udptl) {
- return 0;
- }
-
- /* T38 can be supported by this dialog, create it and set the derived properties */
- if ((p->udptl = ast_udptl_new_with_bindaddr(sched, io, 0, &bindaddr))) {
- if (p->owner) {
- ast_channel_set_fd(p->owner, SIP_UDPTL_FD, ast_udptl_fd(p->udptl));
- }
-
- ast_udptl_setqos(p->udptl, global_tos_audio, global_cos_audio);
- p->t38_maxdatagram = p->relatedpeer ? p->relatedpeer->t38_maxdatagram : global_t38_maxdatagram;
- set_t38_capabilities(p);
-
- ast_debug(1, "Setting NAT on UDPTL to %s\n", natflags ? "On" : "Off");
- ast_udptl_setnat(p->udptl, natflags);
- } else {
- ast_log(AST_LOG_WARNING, "UDPTL creation failed - disabling T38 for this dialog\n");
- ast_clear_flag(&p->flags[1], SIP_PAGE2_T38SUPPORT);
- return 1;
- }
-
- return 0;
-}
-
-
-static int sipinfo_send(
- struct ast_channel *chan,
- struct ast_variable *headers,
- const char *content_type,
- const char *content,
- const char *useragent_filter)
-{
- struct sip_pvt *p;
- struct ast_variable *var;
- struct sip_request req;
- int res = -1;
-
- ast_channel_lock(chan);
-
- if (ast_channel_tech(chan) != &sip_tech) {
- ast_log(LOG_WARNING, "Attempted to send a custom INFO on a non-SIP channel %s\n", ast_channel_name(chan));
- ast_channel_unlock(chan);
- return res;
- }
-
- p = ast_channel_tech_pvt(chan);
- sip_pvt_lock(p);
-
- if (!(ast_strlen_zero(useragent_filter))) {
- int match = (strstr(p->useragent, useragent_filter)) ? 1 : 0;
- if (!match) {
- goto cleanup;
- }
- }
-
- reqprep(&req, p, SIP_INFO, 0, 1);
- for (var = headers; var; var = var->next) {
- add_header(&req, var->name, var->value);
- }
- if (!ast_strlen_zero(content) && !ast_strlen_zero(content_type)) {
- add_header(&req, "Content-Type", content_type);
- add_content(&req, content);
- }
-
- res = send_request(p, &req, XMIT_RELIABLE, p->ocseq);
-
-cleanup:
- sip_pvt_unlock(p);
- ast_channel_unlock(chan);
- return res;
-}
-/*! \brief Play indication to user
- * With SIP a lot of indications is sent as messages, letting the device play
- the indication - busy signal, congestion etc
- \return -1 to force ast_indicate to send indication in audio, 0 if SIP can handle the indication by sending a message
-*/
-static int sip_indicate(struct ast_channel *ast, int condition, const void *data, size_t datalen)
-{
- struct sip_pvt *p = ast_channel_tech_pvt(ast);
- int res = 0;
-
- if (!p) {
- ast_debug(1, "Asked to indicate condition on channel %s with no pvt; ignoring\n",
- ast_channel_name(ast));
- return res;
- }
-
- sip_pvt_lock(p);
- switch(condition) {
- case AST_CONTROL_RINGING:
- if (ast_channel_state(ast) == AST_STATE_RING) {
- p->invitestate = INV_EARLY_MEDIA;
- if (!ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT) ||
- (ast_test_flag(&p->flags[0], SIP_PROG_INBAND) == SIP_PROG_INBAND_NEVER)) {
- /* Send 180 ringing if out-of-band seems reasonable */
- transmit_provisional_response(p, "180 Ringing", &p->initreq, 0);
- ast_set_flag(&p->flags[0], SIP_RINGING);
- if (ast_test_flag(&p->flags[0], SIP_PROG_INBAND) != SIP_PROG_INBAND_YES)
- break;
- } else {
- /* Well, if it's not reasonable, just send in-band */
- }
- }
- res = -1;
- break;
- case AST_CONTROL_BUSY:
- if (ast_channel_state(ast) != AST_STATE_UP) {
- transmit_response_reliable(p, "486 Busy Here", &p->initreq);
- p->invitestate = INV_COMPLETED;
- sip_alreadygone(p);
- ast_softhangup_nolock(ast, AST_SOFTHANGUP_DEV);
- break;
- }
- res = -1;
- break;
- case AST_CONTROL_CONGESTION:
- if (ast_channel_state(ast) != AST_STATE_UP) {
- transmit_response_reliable(p, "503 Service Unavailable", &p->initreq);
- p->invitestate = INV_COMPLETED;
- sip_alreadygone(p);
- ast_softhangup_nolock(ast, AST_SOFTHANGUP_DEV);
- break;
- }
- res = -1;
- break;
- case AST_CONTROL_INCOMPLETE:
- if (ast_channel_state(ast) != AST_STATE_UP) {
- switch (ast_test_flag(&p->flags[1], SIP_PAGE2_ALLOWOVERLAP)) {
- case SIP_PAGE2_ALLOWOVERLAP_YES:
- transmit_response_reliable(p, "484 Address Incomplete", &p->initreq);
- p->invitestate = INV_COMPLETED;
- sip_alreadygone(p);
- ast_softhangup_nolock(ast, AST_SOFTHANGUP_DEV);
- break;
- case SIP_PAGE2_ALLOWOVERLAP_DTMF:
- /* Just wait for inband DTMF digits */
- break;
- default:
- /* it actually means no support for overlap */
- transmit_response_reliable(p, "404 Not Found", &p->initreq);
- p->invitestate = INV_COMPLETED;
- sip_alreadygone(p);
- ast_softhangup_nolock(ast, AST_SOFTHANGUP_DEV);
- break;
- }
- }
- break;
- case AST_CONTROL_PROCEEDING:
- if ((ast_channel_state(ast) != AST_STATE_UP) &&
- !ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT) &&
- !ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
- transmit_response(p, "100 Trying", &p->initreq);
- p->invitestate = INV_PROCEEDING;
- break;
- }
- res = -1;
- break;
- case AST_CONTROL_PROGRESS:
- if ((ast_channel_state(ast) != AST_STATE_UP) &&
- !ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT) &&
- !ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
- p->invitestate = INV_EARLY_MEDIA;
- /* SIP_PROG_INBAND_NEVER means sending 180 ringing in place of a 183 */
- if (ast_test_flag(&p->flags[0], SIP_PROG_INBAND) != SIP_PROG_INBAND_NEVER) {
- transmit_provisional_response(p, "183 Session Progress", &p->initreq, TRUE);
- ast_set_flag(&p->flags[0], SIP_PROGRESS_SENT);
- } else if (ast_channel_state(ast) == AST_STATE_RING && !ast_test_flag(&p->flags[0], SIP_RINGING)) {
- transmit_provisional_response(p, "180 Ringing", &p->initreq, 0);
- ast_set_flag(&p->flags[0], SIP_RINGING);
- }
- break;
- }
- res = -1;
- break;
- case AST_CONTROL_HOLD:
- ast_rtp_instance_update_source(p->rtp);
- ast_moh_start(ast, data, p->mohinterpret);
- break;
- case AST_CONTROL_UNHOLD:
- ast_rtp_instance_update_source(p->rtp);
- ast_moh_stop(ast);
- break;
- case AST_CONTROL_VIDUPDATE: /* Request a video frame update */
- if (p->vrtp && !p->novideo) {
- /* FIXME: Only use this for VP8. Additional work would have to be done to
- * fully support other video codecs */
- if (ast_format_cap_iscompatible_format(ast_channel_nativeformats(ast), ast_format_vp8) != AST_FORMAT_CMP_NOT_EQUAL) {
- /* FIXME Fake RTP write, this will be sent as an RTCP packet. Ideally the
- * RTP engine would provide a way to externally write/schedule RTCP
- * packets */
- struct ast_frame fr;
- fr.frametype = AST_FRAME_CONTROL;
- fr.subclass.integer = AST_CONTROL_VIDUPDATE;
- res = ast_rtp_instance_write(p->vrtp, &fr);
- } else {
- transmit_info_with_vidupdate(p);
- }
- } else {
- res = -1;
- }
- break;
- case AST_CONTROL_T38_PARAMETERS:
- res = -1;
- if (datalen != sizeof(struct ast_control_t38_parameters)) {
- ast_log(LOG_ERROR, "Invalid datalen for AST_CONTROL_T38_PARAMETERS. Expected %d, got %d\n", (int) sizeof(struct ast_control_t38_parameters), (int) datalen);
- } else {
- const struct ast_control_t38_parameters *parameters = data;
- if (!initialize_udptl(p)) {
- res = interpret_t38_parameters(p, parameters);
- }
- }
- break;
- case AST_CONTROL_SRCUPDATE:
- ast_rtp_instance_update_source(p->rtp);
- break;
- case AST_CONTROL_SRCCHANGE:
- ast_rtp_instance_change_source(p->rtp);
- break;
- case AST_CONTROL_CONNECTED_LINE:
- update_connectedline(p, data, datalen);
- break;
- case AST_CONTROL_REDIRECTING:
- update_redirecting(p, data, datalen);
- break;
- case AST_CONTROL_AOC:
- {
- struct ast_aoc_decoded *decoded = ast_aoc_decode((struct ast_aoc_encoded *) data, datalen, ast);
- if (!decoded) {
- ast_log(LOG_ERROR, "Error decoding indicated AOC data\n");
- res = -1;
- break;
- }
- switch (ast_aoc_get_msg_type(decoded)) {
- case AST_AOC_REQUEST:
- if (ast_aoc_get_termination_request(decoded)) {
- /* TODO, once there is a way to get AOC-E on hangup, attempt that here
- * before hanging up the channel.*/
-
- /* The other side has already initiated the hangup. This frame
- * just says they are waiting to get AOC-E before completely tearing
- * the call down. Since SIP does not support this at the moment go
- * ahead and terminate the call here to avoid an unnecessary timeout. */
- ast_debug(1, "AOC-E termination request received on %s. This is not yet supported on sip. Continue with hangup \n", ast_channel_name(p->owner));
- ast_softhangup_nolock(p->owner, AST_SOFTHANGUP_DEV);
- }
- break;
- case AST_AOC_D:
- case AST_AOC_E:
- if (ast_test_flag(&p->flags[2], SIP_PAGE3_SNOM_AOC)) {
- transmit_info_with_aoc(p, decoded);
- }
- break;
- case AST_AOC_S: /* S not supported yet */
- default:
- break;
- }
- ast_aoc_destroy_decoded(decoded);
- }
- break;
- case AST_CONTROL_UPDATE_RTP_PEER: /* Absorb this since it is handled by the bridge */
- break;
- case AST_CONTROL_FLASH: /* We don't currently handle AST_CONTROL_FLASH here, but it is expected, so we don't need to warn either. */
- res = -1;
- break;
- case AST_CONTROL_PVT_CAUSE_CODE: /* these should be handled by the code in channel.c */
- case AST_CONTROL_MASQUERADE_NOTIFY:
- case -1:
- res = -1;
- break;
- default:
- ast_log(LOG_WARNING, "Don't know how to indicate condition %d\n", condition);
- res = -1;
- break;
- }
- sip_pvt_unlock(p);
- return res;
-}
-
-/*!
- * \brief Initiate a call in the SIP channel
- *
- * \note called from sip_request_call (calls from the pbx ) for
- * outbound channels and from handle_request_invite for inbound
- * channels
- *
- * \pre i is locked
- *
- * \return New ast_channel locked.
- */
-static struct ast_channel *sip_new(struct sip_pvt *i, int state, const char *title, const struct ast_assigned_ids *assignedids, const struct ast_channel *requestor, ast_callid callid)
-{
- struct ast_format_cap *caps;
- struct ast_channel *tmp;
- struct ast_variable *v = NULL;
- struct ast_format *fmt;
- struct ast_format_cap *what = NULL; /* SHALLOW COPY DO NOT DESTROY! */
- struct ast_str *codec_buf = ast_str_alloca(AST_FORMAT_CAP_NAMES_LEN);
- int needvideo = 0;
- int needtext = 0;
- char *exten;
-
- caps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
- if (!caps) {
- return NULL;
- }
-
- {
- const char *my_name; /* pick a good name */
-
- if (title) {
- my_name = title;
- } else {
- my_name = ast_strdupa(i->fromdomain);
- }
-
- /* Don't hold a sip pvt lock while we allocate a channel */
- sip_pvt_unlock(i);
-
- if (i->relatedpeer && i->relatedpeer->endpoint) {
- tmp = ast_channel_alloc_with_endpoint(1, state, i->cid_num, i->cid_name, i->accountcode, i->exten, i->context, assignedids, requestor, i->amaflags, i->relatedpeer->endpoint, "SIP/%s-%08x", my_name, (unsigned)ast_atomic_fetchadd_int((int *)&chan_idx, +1));
- } else {
- tmp = ast_channel_alloc(1, state, i->cid_num, i->cid_name, i->accountcode, i->exten, i->context, assignedids, requestor, i->amaflags, "SIP/%s-%08x", my_name, (unsigned)ast_atomic_fetchadd_int((int *)&chan_idx, +1));
- }
- }
- if (!tmp) {
- ast_log(LOG_WARNING, "Unable to allocate AST channel structure for SIP channel\n");
- ao2_ref(caps, -1);
- sip_pvt_lock(i);
- return NULL;
- }
-
- ast_channel_stage_snapshot(tmp);
-
- /* If we sent in a callid, bind it to the channel. */
- if (callid) {
- ast_channel_callid_set(tmp, callid);
- }
-
- sip_pvt_lock(i);
- ast_channel_cc_params_init(tmp, i->cc_params);
- ast_channel_caller(tmp)->id.tag = ast_strdup(i->cid_tag);
-
- ast_channel_tech_set(tmp, (ast_test_flag(&i->flags[0], SIP_DTMF) == SIP_DTMF_INFO || ast_test_flag(&i->flags[0], SIP_DTMF) == SIP_DTMF_SHORTINFO) ? &sip_tech_info : &sip_tech);
-
- /* Select our native format based on codec preference until we receive
- something from another device to the contrary. */
- if (ast_format_cap_count(i->jointcaps)) { /* The joint capabilities of us and peer */
- what = i->jointcaps;
- } else if (ast_format_cap_count(i->caps)) { /* Our configured capability for this peer */
- what = i->caps;
- } else {
- what = sip_cfg.caps;
- }
-
- /* Set the native formats */
- ast_format_cap_append_from_cap(caps, what, AST_MEDIA_TYPE_UNKNOWN);
- /* Use only the preferred audio format, which is stored at the '0' index */
- fmt = ast_format_cap_get_best_by_type(what, AST_MEDIA_TYPE_AUDIO); /* get the best audio format */
- if (fmt) {
- int framing;
-
- ast_format_cap_remove_by_type(caps, AST_MEDIA_TYPE_AUDIO); /* remove only the other audio formats */
- framing = ast_format_cap_get_format_framing(what, fmt);
- ast_format_cap_append(caps, fmt, framing); /* add our best choice back */
- } else {
- /* If we don't have an audio format, try to get something */
- fmt = ast_format_cap_get_format(caps, 0);
- if (!fmt) {
- ast_log(LOG_WARNING, "No compatible formats could be found for %s\n", ast_channel_name(tmp));
- ao2_ref(caps, -1);
- ast_channel_stage_snapshot_done(tmp);
- ast_channel_unlock(tmp);
- ast_hangup(tmp);
- return NULL;
- }
- }
- ast_channel_nativeformats_set(tmp, caps);
- ao2_ref(caps, -1);
-
- ast_debug(3, "*** Our native formats are %s \n", ast_format_cap_get_names(ast_channel_nativeformats(tmp), &codec_buf));
- ast_debug(3, "*** Joint capabilities are %s \n", ast_format_cap_get_names(i->jointcaps, &codec_buf));
- ast_debug(3, "*** Our capabilities are %s \n", ast_format_cap_get_names(i->caps, &codec_buf));
- ast_debug(3, "*** AST_CODEC_CHOOSE formats are %s \n", ast_format_get_name(fmt));
- if (ast_format_cap_count(i->prefcaps)) {
- ast_debug(3, "*** Our preferred formats from the incoming channel are %s \n", ast_format_cap_get_names(i->prefcaps, &codec_buf));
- }
-
- /* If we have a prefcodec setting, we have an inbound channel that set a
- preferred format for this call. Otherwise, we check the jointcapability
- We also check for vrtp. If it's not there, we are not allowed do any video anyway.
- */
- if (i->vrtp) {
- if (ast_test_flag(&i->flags[1], SIP_PAGE2_VIDEOSUPPORT_ALWAYS))
- needvideo = 1;
- else if (ast_format_cap_count(i->prefcaps))
- needvideo = ast_format_cap_has_type(i->prefcaps, AST_MEDIA_TYPE_VIDEO); /* Outbound call */
- else
- needvideo = ast_format_cap_has_type(i->jointcaps, AST_MEDIA_TYPE_VIDEO); /* Inbound call */
-
- if (!needvideo) {
- ast_rtp_instance_destroy(i->vrtp);
- i->vrtp = NULL;
- }
- }
-
- if (i->trtp) {
- if (ast_format_cap_count(i->prefcaps))
- needtext = ast_format_cap_has_type(i->prefcaps, AST_MEDIA_TYPE_TEXT); /* Outbound call */
- else
- needtext = ast_format_cap_has_type(i->jointcaps, AST_MEDIA_TYPE_TEXT); /* Inbound call */
- }
-
- if (needvideo) {
- ast_debug(3, "This channel can handle video! HOLLYWOOD next!\n");
- } else {
- ast_debug(3, "This channel will not be able to handle video.\n");
- }
-
- enable_dsp_detect(i);
-
- if ((ast_test_flag(&i->flags[0], SIP_DTMF) == SIP_DTMF_INBAND) ||
- (ast_test_flag(&i->flags[0], SIP_DTMF) == SIP_DTMF_AUTO)) {
- if (i->rtp) {
- ast_rtp_instance_dtmf_mode_set(i->rtp, AST_RTP_DTMF_MODE_INBAND);
- }
- } else if (ast_test_flag(&i->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833) {
- if (i->rtp) {
- ast_rtp_instance_dtmf_mode_set(i->rtp, AST_RTP_DTMF_MODE_RFC2833);
- }
- }
-
- /* Set file descriptors for audio, video, and realtime text. Since
- * UDPTL is created as needed in the lifetime of a dialog, its file
- * descriptor is set in initialize_udptl */
- if (i->rtp) {
- ast_channel_set_fd(tmp, SIP_AUDIO_RTP_FD, ast_rtp_instance_fd(i->rtp, 0));
- if (ast_test_flag(&i->flags[2], SIP_PAGE3_RTCP_MUX)) {
- ast_channel_set_fd(tmp, SIP_AUDIO_RTCP_FD, -1);
- } else {
- ast_channel_set_fd(tmp, SIP_AUDIO_RTCP_FD, ast_rtp_instance_fd(i->rtp, 1));
- }
- ast_rtp_instance_set_write_format(i->rtp, fmt);
- ast_rtp_instance_set_read_format(i->rtp, fmt);
- }
- if (needvideo && i->vrtp) {
- ast_channel_set_fd(tmp, SIP_VIDEO_RTP_FD, ast_rtp_instance_fd(i->vrtp, 0));
- if (ast_test_flag(&i->flags[2], SIP_PAGE3_RTCP_MUX)) {
- ast_channel_set_fd(tmp, SIP_VIDEO_RTCP_FD, -1);
- } else {
- ast_channel_set_fd(tmp, SIP_VIDEO_RTCP_FD, ast_rtp_instance_fd(i->vrtp, 1));
- }
- }
- if (needtext && i->trtp) {
- ast_channel_set_fd(tmp, SIP_TEXT_RTP_FD, ast_rtp_instance_fd(i->trtp, 0));
- }
- if (i->udptl) {
- ast_channel_set_fd(tmp, SIP_UDPTL_FD, ast_udptl_fd(i->udptl));
- }
-
- if (state == AST_STATE_RING) {
- ast_channel_rings_set(tmp, 1);
- }
- ast_channel_adsicpe_set(tmp, AST_ADSI_UNAVAILABLE);
-
- ast_channel_set_writeformat(tmp, fmt);
- ast_channel_set_rawwriteformat(tmp, fmt);
-
- ast_channel_set_readformat(tmp, fmt);
- ast_channel_set_rawreadformat(tmp, fmt);
-
- ao2_ref(fmt, -1);
-
- ast_channel_tech_pvt_set(tmp, dialog_ref(i, "sip_new: set chan->tech_pvt to i"));
-
- ast_channel_callgroup_set(tmp, i->callgroup);
- ast_channel_pickupgroup_set(tmp, i->pickupgroup);
-
- ast_channel_named_callgroups_set(tmp, i->named_callgroups);
- ast_channel_named_pickupgroups_set(tmp, i->named_pickupgroups);
-
- ast_channel_caller(tmp)->id.name.presentation = i->callingpres;
- ast_channel_caller(tmp)->id.number.presentation = i->callingpres;
- if (!ast_strlen_zero(i->parkinglot)) {
- ast_channel_parkinglot_set(tmp, i->parkinglot);
- }
- if (!ast_strlen_zero(i->accountcode)) {
- ast_channel_accountcode_set(tmp, i->accountcode);
- }
- if (i->amaflags) {
- ast_channel_amaflags_set(tmp, i->amaflags);
- }
- if (!ast_strlen_zero(i->language)) {
- ast_channel_language_set(tmp, i->language);
- }
- if (!ast_strlen_zero(i->zone)) {
- struct ast_tone_zone *zone;
- if (!(zone = ast_get_indication_zone(i->zone))) {
- ast_log(LOG_ERROR, "Unknown country code '%s' for tonezone. Check indications.conf for available country codes.\n", i->zone);
- }
- ast_channel_zone_set(tmp, zone);
- }
- sip_set_owner(i, tmp);
- ast_module_ref(ast_module_info->self);
- ast_channel_context_set(tmp, i->context);
- /*Since it is valid to have extensions in the dialplan that have unescaped characters in them
- * we should decode the uri before storing it in the channel, but leave it encoded in the sip_pvt
- * structure so that there aren't issues when forming URI's
- */
- exten = ast_strdupa(i->exten);
- sip_pvt_unlock(i);
- ast_channel_unlock(tmp);
- if (!ast_exists_extension(NULL, i->context, i->exten, 1, i->cid_num)) {
- ast_uri_decode(exten, ast_uri_sip_user);
- }
- ast_channel_lock(tmp);
- sip_pvt_lock(i);
- ast_channel_exten_set(tmp, exten);
-
- /* Don't use ast_set_callerid() here because it will
- * generate an unnecessary NewCallerID event */
- if (!ast_strlen_zero(i->cid_num)) {
- ast_channel_caller(tmp)->ani.number.valid = 1;
- ast_channel_caller(tmp)->ani.number.str = ast_strdup(i->cid_num);
- }
- if (!ast_strlen_zero(i->rdnis)) {
- ast_channel_redirecting(tmp)->from.number.valid = 1;
- ast_channel_redirecting(tmp)->from.number.str = ast_strdup(i->rdnis);
- }
-
- if (!ast_strlen_zero(i->exten) && strcmp(i->exten, "s")) {
- ast_channel_dialed(tmp)->number.str = ast_strdup(i->exten);
- }
-
- ast_channel_priority_set(tmp, 1);
- if (!ast_strlen_zero(i->uri)) {
- pbx_builtin_setvar_helper(tmp, "SIPURI", i->uri);
- }
- if (!ast_strlen_zero(i->domain)) {
- pbx_builtin_setvar_helper(tmp, "SIPDOMAIN", i->domain);
- }
- if (!ast_strlen_zero(i->tel_phone_context)) {
- pbx_builtin_setvar_helper(tmp, "SIPURIPHONECONTEXT", i->tel_phone_context);
- }
- if (!ast_strlen_zero(i->callid)) {
- pbx_builtin_setvar_helper(tmp, "SIPCALLID", i->callid);
- }
- if (i->rtp) {
- ast_jb_configure(tmp, &global_jbconf);
- }
-
- if (!i->relatedpeer) {
- ast_set_flag(ast_channel_flags(tmp), AST_FLAG_DISABLE_DEVSTATE_CACHE);
- }
- /* Set channel variables for this call from configuration */
- for (v = i->chanvars ; v ; v = v->next) {
- char valuebuf[1024];
- pbx_builtin_setvar_helper(tmp, v->name, ast_get_encoded_str(v->value, valuebuf, sizeof(valuebuf)));
- }
-
- if (i->do_history) {
- append_history(i, "NewChan", "Channel %s - from %s", ast_channel_name(tmp), i->callid);
- }
-
- ast_channel_stage_snapshot_done(tmp);
-
- return tmp;
-}
-
-/*! \brief Lookup 'name' in the SDP starting
- * at the 'start' line. Returns the matching line, and 'start'
- * is updated with the next line number.
- */
-static const char *get_sdp_iterate(int *start, struct sip_request *req, const char *name)
-{
- int len = strlen(name);
- const char *line;
-
- while (*start < (req->sdp_start + req->sdp_count)) {
- line = REQ_OFFSET_TO_STR(req, line[(*start)++]);
- if (!strncasecmp(line, name, len) && line[len] == '=') {
- return ast_skip_blanks(line + len + 1);
- }
- }
-
- /* if the line was not found, ensure that *start points past the SDP */
- (*start)++;
-
- return "";
-}
-
-/*! \brief Fetches the next valid SDP line between the 'start' line
- * (inclusive) and the 'stop' line (exclusive). Returns the type
- * ('a', 'c', ...) and matching line in reference 'start' is updated
- * with the next line number.
- */
-static char get_sdp_line(int *start, int stop, struct sip_request *req, const char **value)
-{
- char type = '\0';
- const char *line = NULL;
-
- if (stop > (req->sdp_start + req->sdp_count)) {
- stop = req->sdp_start + req->sdp_count;
- }
-
- while (*start < stop) {
- line = REQ_OFFSET_TO_STR(req, line[(*start)++]);
- if (line[1] == '=') {
- type = line[0];
- *value = ast_skip_blanks(line + 2);
- break;
- }
- }
-
- return type;
-}
-
-/*! \brief Get a specific line from the message content */
-static char *get_content_line(struct sip_request *req, char *name, char delimiter)
-{
- int i;
- int len = strlen(name);
- const char *line;
-
- for (i = 0; i < req->lines; i++) {
- line = REQ_OFFSET_TO_STR(req, line[i]);
- if (!strncasecmp(line, name, len) && line[len] == delimiter) {
- return ast_skip_blanks(line + len + 1);
- }
- }
-
- return "";
-}
-
-/*! \brief Structure for conversion between compressed SIP and "normal" SIP headers */
-struct cfalias {
- const char *fullname;
- const char *shortname;
-};
-static const struct cfalias aliases[] = {
- { "Content-Type", "c" },
- { "Content-Encoding", "e" },
- { "From", "f" },
- { "Call-ID", "i" },
- { "Contact", "m" },
- { "Content-Length", "l" },
- { "Subject", "s" },
- { "To", "t" },
- { "Supported", "k" },
- { "Refer-To", "r" },
- { "Referred-By", "b" },
- { "Allow-Events", "u" },
- { "Event", "o" },
- { "Via", "v" },
- { "Accept-Contact", "a" },
- { "Reject-Contact", "j" },
- { "Request-Disposition", "d" },
- { "Session-Expires", "x" },
- { "Identity", "y" },
- { "Identity-Info", "n" },
-};
-
-/*! \brief Find compressed SIP alias */
-static const char *find_alias(const char *name, const char *_default)
-{
- int x;
-
- for (x = 0; x < ARRAY_LEN(aliases); x++) {
- if (!strcasecmp(aliases[x].fullname, name))
- return aliases[x].shortname;
- }
-
- return _default;
-}
-
-/*! \brief Find full SIP alias */
-static const char *find_full_alias(const char *name, const char *_default)
-{
- int x;
-
- if (strlen(name) == 1) {
- /* We have a short header name to convert. */
- for (x = 0; x < ARRAY_LEN(aliases); ++x) {
- if (!strcasecmp(aliases[x].shortname, name))
- return aliases[x].fullname;
- }
- }
-
- return _default;
-}
-
-static const char *__get_header(const struct sip_request *req, const char *name, int *start)
-{
- /*
- * Technically you can place arbitrary whitespace both before and after the ':' in
- * a header, although RFC3261 clearly says you shouldn't before, and place just
- * one afterwards. If you shouldn't do it, what absolute idiot decided it was
- * a good idea to say you can do it, and if you can do it, why in the hell would.
- * you say you shouldn't.
- */
- const char *sname = find_alias(name, NULL);
- int x, len = strlen(name), slen = (sname ? 1 : 0);
- for (x = *start; x < req->headers; x++) {
- const char *header = REQ_OFFSET_TO_STR(req, header[x]);
- int smatch = 0, match = !strncasecmp(header, name, len);
- if (slen) {
- smatch = !strncasecmp(header, sname, slen);
- }
- if (match || smatch) {
- /* skip name */
- const char *r = header + (match ? len : slen );
- /* HCOLON has optional SP/HTAB; skip past those */
- while (*r == ' ' || *r == '\t') {
- ++r;
- }
- if (*r == ':') {
- *start = x+1;
- return ast_skip_blanks(r+1);
- }
- }
- }
-
- /* Don't return NULL, so sip_get_header is always a valid pointer */
- return "";
-}
-
-/*! \brief Get header from SIP request
- \return Always return something, so don't check for NULL because it won't happen :-)
-*/
-const char *sip_get_header(const struct sip_request *req, const char *name)
-{
- int start = 0;
- return __get_header(req, name, &start);
-}
-
-
-AST_THREADSTORAGE(sip_content_buf);
-
-/*! \brief Get message body content */
-static char *get_content(struct sip_request *req)
-{
- struct ast_str *str;
- int i;
-
- if (!(str = ast_str_thread_get(&sip_content_buf, 128))) {
- return NULL;
- }
-
- ast_str_reset(str);
-
- for (i = 0; i < req->lines; i++) {
- if (ast_str_append(&str, 0, "%s\n", REQ_OFFSET_TO_STR(req, line[i])) < 0) {
- return NULL;
- }
- }
-
- return ast_str_buffer(str);
-}
-
-/*! \brief Read RTP from network */
-static struct ast_frame *sip_rtp_read(struct ast_channel *ast, struct sip_pvt *p, int *faxdetect)
-{
- /* Retrieve audio/etc from channel. Assumes p->lock is already held. */
- struct ast_frame *f;
-
- if (!p->rtp) {
- /* We have no RTP allocated for this channel */
- return &ast_null_frame;
- }
-
- switch(ast_channel_fdno(ast)) {
- case 0:
- f = ast_rtp_instance_read(p->rtp, 0); /* RTP Audio */
- break;
- case 1:
- f = ast_rtp_instance_read(p->rtp, 1); /* RTCP Control Channel */
- break;
- case 2:
- f = ast_rtp_instance_read(p->vrtp, 0); /* RTP Video */
- break;
- case 3:
- f = ast_rtp_instance_read(p->vrtp, 1); /* RTCP Control Channel for video */
- break;
- case 4:
- f = ast_rtp_instance_read(p->trtp, 0); /* RTP Text */
- if (sipdebug_text) {
- struct ast_str *out = ast_str_create(f->datalen * 4 + 6);
- int i;
- unsigned char* arr = f->data.ptr;
- do {
- if (!out) {
- break;
- }
- for (i = 0; i < f->datalen; i++) {
- ast_str_append(&out, 0, "%c", (arr[i] > ' ' && arr[i] < '}') ? arr[i] : '.');
- }
- ast_str_append(&out, 0, " -> ");
- for (i = 0; i < f->datalen; i++) {
- ast_str_append(&out, 0, "%02hhX ", arr[i]);
- }
- ast_verb(0, "%s\n", ast_str_buffer(out));
- ast_free(out);
- } while (0);
- }
- break;
- case 5:
- f = ast_udptl_read(p->udptl); /* UDPTL for T.38 */
- break;
- default:
- f = &ast_null_frame;
- }
- /* Don't forward RFC2833 if we're not supposed to */
- if (f && (f->frametype == AST_FRAME_DTMF_BEGIN || f->frametype == AST_FRAME_DTMF_END) &&
- (ast_test_flag(&p->flags[0], SIP_DTMF) != SIP_DTMF_RFC2833)) {
- ast_debug(1, "Ignoring DTMF (%c) RTP frame because dtmfmode is not RFC2833\n", f->subclass.integer);
- ast_frfree(f);
- return &ast_null_frame;
- }
-
- /* We already hold the channel lock */
- if (!p->owner || (f && f->frametype != AST_FRAME_VOICE)) {
- return f;
- }
-
- if (f && ast_format_cap_iscompatible_format(ast_channel_nativeformats(p->owner), f->subclass.format) == AST_FORMAT_CMP_NOT_EQUAL) {
- struct ast_format_cap *caps;
-
- if (ast_format_cap_iscompatible_format(p->jointcaps, f->subclass.format) == AST_FORMAT_CMP_NOT_EQUAL) {
- ast_debug(1, "Bogus frame of format '%s' received from '%s'!\n",
- ast_format_get_name(f->subclass.format), ast_channel_name(p->owner));
- ast_frfree(f);
- return &ast_null_frame;
- }
- ast_debug(1, "Oooh, format changed to %s\n",
- ast_format_get_name(f->subclass.format));
-
- caps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
- if (caps) {
- ast_format_cap_append_from_cap(caps, ast_channel_nativeformats(p->owner), AST_MEDIA_TYPE_UNKNOWN);
- ast_format_cap_remove_by_type(caps, AST_MEDIA_TYPE_AUDIO);
- ast_format_cap_append(caps, f->subclass.format, 0);
- ast_channel_nativeformats_set(p->owner, caps);
- ao2_ref(caps, -1);
- }
- ast_set_read_format(p->owner, ast_channel_readformat(p->owner));
- ast_set_write_format(p->owner, ast_channel_writeformat(p->owner));
- }
-
- if (f && p->dsp) {
- f = ast_dsp_process(p->owner, p->dsp, f);
- if (f && f->frametype == AST_FRAME_DTMF) {
- if (f->subclass.integer == 'f') {
- ast_debug(1, "Fax CNG detected on %s\n", ast_channel_name(ast));
- *faxdetect = 1;
- /* If we only needed this DSP for fax detection purposes we can just drop it now */
- if (ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_INBAND) {
- ast_dsp_set_features(p->dsp, DSP_FEATURE_DIGIT_DETECT);
- } else {
- ast_dsp_free(p->dsp);
- p->dsp = NULL;
- }
- } else {
- ast_debug(1, "* Detected inband DTMF '%c'\n", f->subclass.integer);
- }
- }
- }
-
- return f;
-}
-
-/*! \brief Read SIP RTP from channel */
-static struct ast_frame *sip_read(struct ast_channel *ast)
-{
- struct ast_frame *fr;
- struct sip_pvt *p = ast_channel_tech_pvt(ast);
- int faxdetected = FALSE;
-
- sip_pvt_lock(p);
- fr = sip_rtp_read(ast, p, &faxdetected);
- p->lastrtprx = time(NULL);
-
- /* If we detect a CNG tone and fax detection is enabled then send us off to the fax extension */
- if (faxdetected && ast_test_flag(&p->flags[1], SIP_PAGE2_FAX_DETECT_CNG)) {
- if (strcmp(ast_channel_exten(ast), "fax")) {
- const char *target_context = S_OR(ast_channel_macrocontext(ast), ast_channel_context(ast));
- /*
- * We need to unlock 'ast' here because
- * ast_exists_extension has the potential to start and
- * stop an autoservice on the channel. Such action is
- * prone to deadlock if the channel is locked.
- *
- * ast_async_goto() has its own restriction on not holding
- * the channel lock.
- */
- sip_pvt_unlock(p);
- ast_channel_unlock(ast);
- ast_frfree(fr);
- fr = &ast_null_frame;
- if (ast_exists_extension(ast, target_context, "fax", 1,
- S_COR(ast_channel_caller(ast)->id.number.valid, ast_channel_caller(ast)->id.number.str, NULL))) {
- ast_verb(2, "Redirecting '%s' to fax extension due to CNG detection\n", ast_channel_name(ast));
- pbx_builtin_setvar_helper(ast, "FAXEXTEN", ast_channel_exten(ast));
- if (ast_async_goto(ast, target_context, "fax", 1)) {
- ast_log(LOG_NOTICE, "Failed to async goto '%s' into fax of '%s'\n", ast_channel_name(ast), target_context);
- }
- } else {
- ast_log(LOG_NOTICE, "FAX CNG detected but no fax extension\n");
- }
- ast_channel_lock(ast);
- sip_pvt_lock(p);
- }
- }
-
- /* Only allow audio through if they sent progress with SDP, or if the channel is actually answered */
- if (fr && fr->frametype == AST_FRAME_VOICE && p->invitestate != INV_EARLY_MEDIA && ast_channel_state(ast) != AST_STATE_UP) {
- ast_frfree(fr);
- fr = &ast_null_frame;
- }
-
- sip_pvt_unlock(p);
-
- return fr;
-}
-
-
-/*! \brief Generate 32 byte random string for callid's etc */
-static char *generate_random_string(char *buf, size_t size)
-{
- long val[4];
- int x;
-
- for (x=0; x<4; x++)
- val[x] = ast_random();
- snprintf(buf, size, "%08lx%08lx%08lx%08lx", (unsigned long)val[0], (unsigned long)val[1], (unsigned long)val[2], (unsigned long)val[3]);
-
- return buf;
-}
-
-static char *generate_uri(struct sip_pvt *pvt, char *buf, size_t size)
-{
- struct ast_str *uri = ast_str_alloca(size);
- ast_str_set(&uri, 0, "%s", pvt->socket.type == AST_TRANSPORT_TLS ? "sips:" : "sip:");
- /* Here would be a great place to generate a UUID, but for now we'll
- * use the handy random string generation function we already have
- */
- ast_str_append(&uri, 0, "%s", generate_random_string(buf, size));
- ast_str_append(&uri, 0, "@%s", ast_sockaddr_stringify_remote(&pvt->ourip));
- ast_copy_string(buf, ast_str_buffer(uri), size);
- return buf;
-}
-
-/*!
- * \brief Build SIP Call-ID value for a non-REGISTER transaction
- *
- * \note The passed in pvt must not be in a dialogs container
- * since this function changes the hash key used by the
- * container.
- */
-static void build_callid_pvt(struct sip_pvt *pvt)
-{
- char buf[33];
- const char *host = S_OR(pvt->fromdomain, ast_sockaddr_stringify_remote(&pvt->ourip));
-
- ast_string_field_build(pvt, callid, "%s@%s", generate_random_string(buf, sizeof(buf)), host);
-}
-
-/*! \brief Unlink the given object from the container and return TRUE if it was in the container. */
-#define CONTAINER_UNLINK(container, obj, tag) \
- ({ \
- int found = 0; \
- typeof((obj)) __removed_obj; \
- __removed_obj = ao2_t_callback((container), \
- OBJ_UNLINK | OBJ_POINTER, ao2_match_by_addr, (obj), (tag)); \
- if (__removed_obj) { \
- ao2_ref(__removed_obj, -1); \
- found = 1; \
- } \
- found; \
- })
-
-/*!
- * \internal
- * \brief Safely change the callid of the given SIP dialog.
- *
- * \param pvt SIP private structure to change callid
- * \param callid Specified new callid to use. NULL if generate new callid.
- */
-static void change_callid_pvt(struct sip_pvt *pvt, const char *callid)
-{
- int in_dialog_container;
- int in_rtp_container;
- char *oldid = ast_strdupa(pvt->callid);
-
- ao2_lock(dialogs);
- ao2_lock(dialogs_rtpcheck);
- in_dialog_container = CONTAINER_UNLINK(dialogs, pvt,
- "About to change the callid -- remove the old name");
- in_rtp_container = CONTAINER_UNLINK(dialogs_rtpcheck, pvt,
- "About to change the callid -- remove the old name");
- if (callid) {
- ast_string_field_set(pvt, callid, callid);
- } else {
- build_callid_pvt(pvt);
- }
- if (in_dialog_container) {
- ao2_t_link(dialogs, pvt, "New dialog callid -- inserted back into table");
- }
- if (in_rtp_container) {
- ao2_t_link(dialogs_rtpcheck, pvt, "New dialog callid -- inserted back into table");
- }
- ao2_unlock(dialogs_rtpcheck);
- ao2_unlock(dialogs);
-
- if (strcmp(oldid, pvt->callid)) {
- ast_debug(1, "SIP call-id changed from '%s' to '%s'\n", oldid, pvt->callid);
- }
-}
-
-/*! \brief Build SIP Call-ID value for a REGISTER transaction */
-static void build_callid_registry(struct sip_registry *reg, const struct ast_sockaddr *ourip, const char *fromdomain)
-{
- char buf[33];
-
- const char *host = S_OR(fromdomain, ast_sockaddr_stringify_host_remote(ourip));
-
- ast_string_field_build(reg, callid, "%s@%s", generate_random_string(buf, sizeof(buf)), host);
-}
-
-/*! \brief Build SIP From tag value for REGISTER */
-static void build_localtag_registry(struct sip_registry *reg)
-{
- ast_string_field_build(reg, localtag, "as%08lx", (unsigned long)ast_random());
-}
-
-/*! \brief Make our SIP dialog tag */
-static void make_our_tag(struct sip_pvt *pvt)
-{
- ast_string_field_build(pvt, tag, "as%08lx", (unsigned long)ast_random());
-}
-
-/*! \brief Allocate Session-Timers struct w/in dialog */
-static struct sip_st_dlg* sip_st_alloc(struct sip_pvt *const p)
-{
- struct sip_st_dlg *stp;
-
- if (p->stimer) {
- ast_log(LOG_ERROR, "Session-Timer struct already allocated\n");
- return p->stimer;
- }
-
- if (!(stp = ast_calloc(1, sizeof(struct sip_st_dlg)))) {
- return NULL;
- }
- stp->st_schedid = -1; /* Session-Timers ast_sched scheduler id */
-
- p->stimer = stp;
-
- return p->stimer;
-}
-
-static void sip_pvt_callid_set(struct sip_pvt *pvt, ast_callid callid)
-{
- pvt->logger_callid = callid;
-}
-
-/*! \brief Allocate sip_pvt structure, set defaults and link in the container.
- * Returns a reference to the object so whoever uses it later must
- * remember to release the reference.
- */
-struct sip_pvt *__sip_alloc(ast_string_field callid, struct ast_sockaddr *addr,
- int useglobal_nat, const int intended_method, struct sip_request *req, ast_callid logger_callid,
- const char *file, int line, const char *func)
-{
- struct sip_pvt *p;
-
- p = __ao2_alloc(sizeof(*p), sip_pvt_dtor,
- AO2_ALLOC_OPT_LOCK_MUTEX, "allocate a dialog(pvt) struct",
- file, line, func);
- if (!p) {
- return NULL;
- }
-
- if (ast_string_field_init(p, 512)) {
- ao2_t_ref(p, -1, "failed to string_field_init, drop p");
- return NULL;
- }
-
- if (!(p->cc_params = ast_cc_config_params_init())) {
- ao2_t_ref(p, -1, "Yuck, couldn't allocate cc_params struct. Get rid o' p");
- return NULL;
- }
-
- if (logger_callid) {
- sip_pvt_callid_set(p, logger_callid);
- }
-
- p->caps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
- p->jointcaps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
- p->peercaps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
- p->redircaps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
- p->prefcaps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
-
- if (!p->caps|| !p->jointcaps || !p->peercaps || !p->redircaps || !p->prefcaps) {
- ao2_cleanup(p->caps);
- ao2_cleanup(p->jointcaps);
- ao2_cleanup(p->peercaps);
- ao2_cleanup(p->redircaps);
- ao2_cleanup(p->prefcaps);
- ao2_t_ref(p, -1, "Yuck, couldn't allocate format capabilities. Get rid o' p");
- return NULL;
- }
-
-
- /* If this dialog is created as a result of a request or response, lets store
- * some information about it in the dialog. */
- if (req) {
- struct sip_via *via;
- const char *cseq = sip_get_header(req, "Cseq");
- uint32_t seqno;
-
- /* get branch parameter from initial Request that started this dialog */
- via = parse_via(sip_get_header(req, "Via"));
- if (via) {
- /* only store the branch if it begins with the magic prefix "z9hG4bK", otherwise
- * it is not useful to us to have it */
- if (!ast_strlen_zero(via->branch) && !strncasecmp(via->branch, "z9hG4bK", 7)) {
- ast_string_field_set(p, initviabranch, via->branch);
- ast_string_field_set(p, initviasentby, via->sent_by);
- }
- free_via(via);
- }
-
- /* Store initial incoming cseq. An error in sscanf here is ignored. There is no approperiate
- * except not storing the number. CSeq validation must take place before dialog creation in find_call */
- if (!ast_strlen_zero(cseq) && (sscanf(cseq, "%30u", &seqno) == 1)) {
- p->init_icseq = seqno;
- }
- /* Later in ast_sip_ouraddrfor we need this to choose the right ip and port for the specific transport */
- set_socket_transport(&p->socket, req->socket.type);
- } else {
- set_socket_transport(&p->socket, AST_TRANSPORT_UDP);
- }
-
- p->socket.fd = -1;
- p->method = intended_method;
- p->initid = -1;
- p->waitid = -1;
- p->reinviteid = -1;
- p->autokillid = -1;
- p->request_queue_sched_id = -1;
- p->provisional_keepalive_sched_id = -1;
- p->t38id = -1;
- p->subscribed = NONE;
- p->stateid = -1;
- p->sessionversion_remote = -1;
- p->session_modify = TRUE;
- p->stimer = NULL;
- ast_copy_string(p->zone, default_zone, sizeof(p->zone));
- p->maxforwards = sip_cfg.default_max_forwards;
-
- if (intended_method != SIP_OPTIONS) { /* Peerpoke has it's own system */
- p->timer_t1 = global_t1; /* Default SIP retransmission timer T1 (RFC 3261) */
- p->timer_b = global_timer_b; /* Default SIP transaction timer B (RFC 3261) */
- }
-
- if (!addr) {
- p->ourip = internip;
- } else {
- ast_sockaddr_copy(&p->sa, addr);
- ast_sip_ouraddrfor(&p->sa, &p->ourip, p);
- }
-
- /* Copy global flags to this PVT at setup. */
- ast_copy_flags(&p->flags[0], &global_flags[0], SIP_FLAGS_TO_COPY);
- ast_copy_flags(&p->flags[1], &global_flags[1], SIP_PAGE2_FLAGS_TO_COPY);
- ast_copy_flags(&p->flags[2], &global_flags[2], SIP_PAGE3_FLAGS_TO_COPY);
-
- p->do_history = recordhistory;
-
- p->branch = ast_random();
- make_our_tag(p);
- p->ocseq = INITIAL_CSEQ;
- p->allowed_methods = UINT_MAX;
-
- if (sip_methods[intended_method].need_rtp) {
- p->maxcallbitrate = default_maxcallbitrate;
- p->autoframing = global_autoframing;
- }
-
- if (useglobal_nat && addr) {
- /* Setup NAT structure according to global settings if we have an address */
- ast_sockaddr_copy(&p->recv, addr);
- check_via(p, req);
- do_setnat(p);
- }
-
- if (p->method != SIP_REGISTER) {
- ast_string_field_set(p, fromdomain, default_fromdomain);
- p->fromdomainport = default_fromdomainport;
- }
- build_via(p);
- if (!callid)
- build_callid_pvt(p);
- else
- ast_string_field_set(p, callid, callid);
- /* Assign default music on hold class */
- ast_string_field_set(p, mohinterpret, default_mohinterpret);
- ast_string_field_set(p, mohsuggest, default_mohsuggest);
- ast_format_cap_append_from_cap(p->caps, sip_cfg.caps, AST_MEDIA_TYPE_UNKNOWN);
- p->allowtransfer = sip_cfg.allowtransfer;
- if ((ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833) ||
- (ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_AUTO)) {
- p->noncodeccapability |= AST_RTP_DTMF;
- }
- ast_string_field_set(p, context, sip_cfg.default_context);
- ast_string_field_set(p, parkinglot, default_parkinglot);
- ast_string_field_set(p, engine, default_engine);
-
- AST_LIST_HEAD_INIT_NOLOCK(&p->request_queue);
- AST_LIST_HEAD_INIT_NOLOCK(&p->offered_media);
-
- /* Add to active dialog list */
-
- ao2_t_link(dialogs, p, "link pvt into dialogs table");
-
- ast_debug(1, "Allocating new SIP dialog for %s - %s (%s)\n", callid ? callid : p->callid, sip_methods[intended_method].text, p->rtp ? "With RTP" : "No RTP");
- return p;
-}
-
-/*!
- * \brief Process the Via header according to RFC 3261 section 18.2.2.
- * \param p a sip_pvt structure that will be modified according to the received
- * header
- * \param req a sip request with a Via header to process
- *
- * This function will update the destination of the response according to the
- * Via header in the request and RFC 3261 section 18.2.2. We do not have a
- * transport layer so we ignore certain values like the 'received' param (we
- * set the destination address to the address the request came from in the
- * respprep() function).
- *
- * \retval -1 error
- * \retval 0 success
- */
-static int process_via(struct sip_pvt *p, const struct sip_request *req)
-{
- struct sip_via *via = parse_via(sip_get_header(req, "Via"));
-
- if (!via) {
- ast_log(LOG_ERROR, "error processing via header\n");
- return -1;
- }
-
- if (via->maddr) {
- if (ast_sockaddr_resolve_first_transport(&p->sa, via->maddr, PARSE_PORT_FORBID, p->socket.type)) {
- ast_log(LOG_WARNING, "Can't find address for maddr '%s'\n", via->maddr);
- ast_log(LOG_ERROR, "error processing via header\n");
- free_via(via);
- return -1;
- }
-
- if (ast_sockaddr_is_ipv4_multicast(&p->sa)) {
- setsockopt(sipsock, IPPROTO_IP, IP_MULTICAST_TTL, &via->ttl, sizeof(via->ttl));
- }
- }
-
- ast_sockaddr_set_port(&p->sa, via->port ? via->port : STANDARD_SIP_PORT);
-
- free_via(via);
- return 0;
-}
-
-/*! \brief arguments used for Request/Response to matching */
-struct match_req_args {
- int method;
- const char *callid;
- const char *totag;
- const char *fromtag;
- uint32_t seqno;
-
- /* Set if this method is a Response */
- int respid;
-
- /* Set if the method is a Request */
- const char *ruri;
- const char *viabranch;
- const char *viasentby;
-
- /* Set this if the Authentication header is present in the Request. */
- int authentication_present;
-};
-
-enum match_req_res {
- SIP_REQ_MATCH,
- SIP_REQ_NOT_MATCH,
- SIP_REQ_LOOP_DETECTED, /* multiple incoming requests with same call-id but different branch parameters have been detected */
- SIP_REQ_FORKED, /* An outgoing request has been forked as result of receiving two differing 200ok responses. */
-};
-
-/*!
- * \brief Match a incoming Request/Response to a dialog
- *
- * \retval enum match_req_res indicating if the dialog matches the arg
- */
-static enum match_req_res match_req_to_dialog(struct sip_pvt *sip_pvt_ptr, struct match_req_args *arg)
-{
- const char *init_ruri = NULL;
- if (sip_pvt_ptr->initreq.headers) {
- init_ruri = REQ_OFFSET_TO_STR(&sip_pvt_ptr->initreq, rlpart2);
- }
-
- /*
- * Match Tags and call-id to Dialog
- */
- if (!ast_strlen_zero(arg->callid) && strcmp(sip_pvt_ptr->callid, arg->callid)) {
- /* call-id does not match. */
- return SIP_REQ_NOT_MATCH;
- }
- if (arg->method == SIP_RESPONSE) {
- /* Verify fromtag of response matches the tag we gave them. */
- if (strcmp(arg->fromtag, sip_pvt_ptr->tag)) {
- /* fromtag from response does not match our tag */
- return SIP_REQ_NOT_MATCH;
- }
-
- /* Verify totag if we have one stored for this dialog, but never be strict about this for
- * a response until the dialog is established */
- if (!ast_strlen_zero(sip_pvt_ptr->theirtag) && ast_test_flag(&sip_pvt_ptr->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED)) {
- if (ast_strlen_zero(arg->totag)) {
- /* missing totag when they already gave us one earlier */
- return SIP_REQ_NOT_MATCH;
- }
- /* compare the totag of response with the tag we have stored for them */
- if (strcmp(arg->totag, sip_pvt_ptr->theirtag)) {
- /* totag did not match what we had stored for them. */
- char invite_branch[32] = { 0, };
- if (sip_pvt_ptr->invite_branch) {
- snprintf(invite_branch, sizeof(invite_branch), "z9hG4bK%08x", (unsigned)sip_pvt_ptr->invite_branch);
- }
- /* Forked Request Detection
- *
- * If this is a 200ok response and the totags do not match, this
- * might be a forked response to an outgoing Request. Detection of
- * a forked response must meet the criteria below.
- *
- * 1. must be a 2xx Response
- * 2. call-d equal to call-id of Request. this is done earlier
- * 3. from-tag equal to from-tag of Request. this is done earlier
- * 4. branch parameter equal to branch of inital Request
- * 5. to-tag _NOT_ equal to previous 2xx response that already established the dialog.
- */
- if ((arg->respid == 200) &&
- !ast_strlen_zero(invite_branch) &&
- !ast_strlen_zero(arg->viabranch) &&
- !strcmp(invite_branch, arg->viabranch)) {
- return SIP_REQ_FORKED;
- }
-
- /* The totag did not match the one we had stored, and this is not a Forked Request. */
- return SIP_REQ_NOT_MATCH;
- }
- }
- } else {
- /* Verify the fromtag of Request matches the tag they provided earlier.
- * If this is a Request with authentication credentials, forget their old
- * tag as it is not valid after the 401 or 407 response. */
- if (!arg->authentication_present && strcmp(arg->fromtag, sip_pvt_ptr->theirtag)) {
- /* their tag does not match the one was have stored for them */
- return SIP_REQ_NOT_MATCH;
- }
- /* Verify if totag is present in Request, that it matches what we gave them as our tag earlier */
- if (!ast_strlen_zero(arg->totag) && (strcmp(arg->totag, sip_pvt_ptr->tag))) {
- /* totag from Request does not match our tag */
- return SIP_REQ_NOT_MATCH;
- }
- }
-
- /*
- * Compare incoming request against initial transaction.
- *
- * This is a best effort attempt at distinguishing forked requests from
- * our initial transaction. If all the elements are NOT in place to evaluate
- * this, this block is ignored and the dialog match is made regardless.
- * Once the totag is established after the dialog is confirmed, this is not necessary.
- *
- * CRITERIA required for initial transaction matching.
- *
- * 1. Is a Request
- * 2. Callid and theirtag match (this is done in the dialog matching block)
- * 3. totag is NOT present
- * 4. CSeq matchs our initial transaction's cseq number
- * 5. pvt has init via branch parameter stored
- */
- if ((arg->method != SIP_RESPONSE) && /* must be a Request */
- ast_strlen_zero(arg->totag) && /* must not have a totag */
- (sip_pvt_ptr->init_icseq == arg->seqno) && /* the cseq must be the same as this dialogs initial cseq */
- !ast_strlen_zero(sip_pvt_ptr->initviabranch) && /* The dialog must have started with a RFC3261 compliant branch tag */
- init_ruri) { /* the dialog must have an initial request uri associated with it */
- /* This Request matches all the criteria required for Loop/Merge detection.
- * Now we must go down the path of comparing VIA's and RURIs. */
- if (ast_strlen_zero(arg->viabranch) ||
- strcmp(arg->viabranch, sip_pvt_ptr->initviabranch) ||
- ast_strlen_zero(arg->viasentby) ||
- strcmp(arg->viasentby, sip_pvt_ptr->initviasentby)) {
- /* At this point, this request does not match this Dialog.*/
-
- /* if methods are different this is just a mismatch */
- if ((sip_pvt_ptr->method != arg->method)) {
- return SIP_REQ_NOT_MATCH;
- }
-
- /* If RUIs are different, this is a forked request to a separate URI.
- * Returning a mismatch allows this Request to be processed separately. */
- if (sip_uri_cmp(init_ruri, arg->ruri)) {
- /* not a match, request uris are different */
- return SIP_REQ_NOT_MATCH;
- }
-
- /* Loop/Merge Detected
- *
- * ---Current Matches to Initial Request---
- * request uri
- * Call-id
- * their-tag
- * no totag present
- * method
- * cseq
- *
- * --- Does not Match Initial Request ---
- * Top Via
- *
- * Without the same Via, this can not match our initial transaction for this dialog,
- * but given that this Request matches everything else associated with that initial
- * Request this is most certainly a Forked request in which we have already received
- * part of the fork.
- */
- return SIP_REQ_LOOP_DETECTED;
- }
- } /* end of Request Via check */
-
- /* Match Authentication Request.
- *
- * A Request with an Authentication header must come back with the
- * same Request URI. Otherwise it is not a match.
- */
- if ((arg->method != SIP_RESPONSE) && /* Must be a Request type to even begin checking this */
- ast_strlen_zero(arg->totag) && /* no totag is present to match */
- arg->authentication_present && /* Authentication header is present in Request */
- sip_uri_cmp(init_ruri, arg->ruri)) { /* Compare the Request URI of both the last Request and this new one */
-
- /* Authentication was provided, but the Request URI did not match the last one on this dialog. */
- return SIP_REQ_NOT_MATCH;
- }
-
- return SIP_REQ_MATCH;
-}
-
-/*! \brief This function creates a dialog to handle a forked request. This dialog
- * exists only to properly terminiate the forked request immediately.
- */
-static void forked_invite_init(struct sip_request *req, const char *new_theirtag, struct sip_pvt *original, struct ast_sockaddr *addr)
-{
- struct sip_pvt *p;
- const char *callid;
- ast_callid logger_callid;
-
- sip_pvt_lock(original);
- callid = ast_strdupa(original->callid);
- logger_callid = original->logger_callid;
- sip_pvt_unlock(original);
-
- p = sip_alloc(callid, addr, 1, SIP_INVITE, req, logger_callid);
- if (!p) {
- return; /* alloc error */
- }
-
- /* Lock p and original private structures. */
- sip_pvt_lock(p);
- while (sip_pvt_trylock(original)) {
- /* Can't use DEADLOCK_AVOIDANCE since p is an ao2 object */
- sip_pvt_unlock(p);
- sched_yield();
- sip_pvt_lock(p);
- }
-
- p->invitestate = INV_TERMINATED;
- p->ocseq = original->ocseq;
- p->branch = original->branch;
-
- memcpy(&p->flags, &original->flags, sizeof(p->flags));
- copy_request(&p->initreq, &original->initreq);
- ast_string_field_set(p, theirtag, new_theirtag);
- ast_string_field_set(p, tag, original->tag);
- ast_string_field_set(p, uri, original->uri);
- ast_string_field_set(p, our_contact, original->our_contact);
- ast_string_field_set(p, fullcontact, original->fullcontact);
-
- sip_pvt_unlock(original);
-
- parse_ok_contact(p, req);
- build_route(p, req, 1, 0);
-
- transmit_request(p, SIP_ACK, p->ocseq, XMIT_UNRELIABLE, TRUE);
- transmit_request(p, SIP_BYE, 0, XMIT_RELIABLE, TRUE);
-
- pvt_set_needdestroy(p, "forked request"); /* this dialog will terminate once the BYE is responed to or times out. */
- sip_pvt_unlock(p);
- dialog_unref(p, "setup forked invite termination");
-}
-
-/*! \internal
- *
- * \brief Locks both pvt and pvt owner if owner is present.
- *
- * \note This function gives a ref to pvt->owner if it is present and locked.
- * This reference must be decremented after pvt->owner is unlocked.
- *
- * \note This function will never give you up,
- * \note This function will never let you down.
- * \note This function will run around and desert you.
- *
- * \pre pvt is not locked
- * \post pvt is locked
- * \post pvt->owner is locked and its reference count is increased (if pvt->owner is not NULL)
- *
- * \return a pointer to the locked and reffed pvt->owner channel if it exists.
- */
-static struct ast_channel *sip_pvt_lock_full(struct sip_pvt *pvt)
-{
- struct ast_channel *chan;
-
- /* Locking is simple when it is done right. If you see a deadlock resulting
- * in this function, it is not this function's fault, Your problem exists elsewhere.
- * This function is perfect... seriously. */
- for (;;) {
- /* First, get the channel and grab a reference to it */
- sip_pvt_lock(pvt);
- chan = pvt->owner;
- if (chan) {
- /* The channel can not go away while we hold the pvt lock.
- * Give the channel a ref so it will not go away after we let
- * the pvt lock go. */
- ast_channel_ref(chan);
- } else {
- /* no channel, return pvt locked */
- return NULL;
- }
-
- /* We had to hold the pvt lock while getting a ref to the owner channel
- * but now we have to let this lock go in order to preserve proper
- * locking order when grabbing the channel lock */
- sip_pvt_unlock(pvt);
-
- /* Look, no deadlock avoidance, hooray! */
- ast_channel_lock(chan);
- sip_pvt_lock(pvt);
-
- if (pvt->owner == chan) {
- /* done */
- break;
- }
-
- /* If the owner changed while everything was unlocked, no problem,
- * just start over and everthing will work. This is rare, do not be
- * confused by this loop and think this it is an expensive operation.
- * The majority of the calls to this function will never involve multiple
- * executions of this loop. */
- ast_channel_unlock(chan);
- ast_channel_unref(chan);
- sip_pvt_unlock(pvt);
- }
-
- /* If owner exists, it is locked and reffed */
- return pvt->owner;
-}
-
-/*! \brief Set the owning channel on the \ref sip_pvt object */
-static void sip_set_owner(struct sip_pvt *p, struct ast_channel *chan)
-{
- p->owner = chan;
- if (p->rtp) {
- ast_rtp_instance_set_channel_id(p->rtp, p->owner ? ast_channel_uniqueid(p->owner) : "");
- }
- if (p->vrtp) {
- ast_rtp_instance_set_channel_id(p->vrtp, p->owner ? ast_channel_uniqueid(p->owner) : "");
- }
- if (p->trtp) {
- ast_rtp_instance_set_channel_id(p->trtp, p->owner ? ast_channel_uniqueid(p->owner) : "");
- }
-}
-
-/*! \brief find or create a dialog structure for an incoming SIP message.
- * Connect incoming SIP message to current dialog or create new dialog structure
- * Returns a reference to the sip_pvt object, remember to give it back once done.
- * Called by handle_request_do
- */
-static struct sip_pvt *__find_call(struct sip_request *req, struct ast_sockaddr *addr, const int intended_method,
- const char *file, int line, const char *func)
-{
- char totag[128];
- char fromtag[128];
- const char *callid = sip_get_header(req, "Call-ID");
- const char *from = sip_get_header(req, "From");
- const char *to = sip_get_header(req, "To");
- const char *cseq = sip_get_header(req, "Cseq");
- struct sip_pvt *sip_pvt_ptr;
- uint32_t seqno;
- /* Call-ID, to, from and Cseq are required by RFC 3261. (Max-forwards and via too - ignored now) */
- /* sip_get_header always returns non-NULL so we must use ast_strlen_zero() */
- if (ast_strlen_zero(callid) || ast_strlen_zero(to) ||
- ast_strlen_zero(from) || ast_strlen_zero(cseq) ||
- (sscanf(cseq, "%30u", &seqno) != 1)) {
-
- /* RFC 3261 section 24.4.1. Send a 400 Bad Request if the request is malformed. */
- if (intended_method != SIP_RESPONSE && intended_method != SIP_ACK) {
- transmit_response_using_temp(callid, addr, 1, intended_method,
- req, "400 Bad Request");
- }
- return NULL; /* Invalid packet */
- }
-
- if (sip_cfg.pedanticsipchecking) {
- /* In principle Call-ID's uniquely identify a call, but with a forking SIP proxy
- we need more to identify a branch - so we have to check branch, from
- and to tags to identify a call leg.
- For Asterisk to behave correctly, you need to turn on pedanticsipchecking
- in sip.conf
- */
- if (gettag(req, "To", totag, sizeof(totag)))
- req->has_to_tag = 1; /* Used in handle_request/response */
- gettag(req, "From", fromtag, sizeof(fromtag));
-
- ast_debug(5, "= Looking for Call ID: %s (Checking %s) --From tag %s --To-tag %s \n", callid, req->method==SIP_RESPONSE ? "To" : "From", fromtag, totag);
-
- /* All messages must always have From: tag */
- if (ast_strlen_zero(fromtag)) {
- ast_debug(5, "%s request has no from tag, dropping callid: %s from: %s\n", sip_methods[req->method].text , callid, from );
- return NULL;
- }
- /* reject requests that must always have a To: tag */
- if (ast_strlen_zero(totag) && (req->method == SIP_ACK || req->method == SIP_BYE || req->method == SIP_INFO )) {
- if (req->method != SIP_ACK) {
- transmit_response_using_temp(callid, addr, 1, intended_method, req, "481 Call leg/transaction does not exist");
- }
- ast_debug(5, "%s must have a to tag. dropping callid: %s from: %s\n", sip_methods[req->method].text , callid, from );
- return NULL;
- }
- }
-
- /* match on callid only for REGISTERs */
- if (!sip_cfg.pedanticsipchecking || req->method == SIP_REGISTER) {
- struct sip_pvt tmp_dialog = {
- .callid = callid,
- };
- sip_pvt_ptr = __ao2_find(dialogs, &tmp_dialog, OBJ_POINTER,
- "find_call in dialogs", file, line, func);
- if (sip_pvt_ptr) { /* well, if we don't find it-- what IS in there? */
- /* Found the call */
- return sip_pvt_ptr;
- }
- } else { /* in pedantic mode! -- do the fancy search */
- struct sip_pvt tmp_dialog = {
- .callid = callid,
- };
- /* if a Outbound forked Request is detected, this pvt will point
- * to the dialog the Request is forking off of. */
- struct sip_pvt *fork_pvt = NULL;
- struct match_req_args args = { 0, };
- int found;
- struct ao2_iterator *iterator = __ao2_callback(dialogs,
- OBJ_POINTER | OBJ_MULTIPLE,
- dialog_find_multiple,
- &tmp_dialog,
- "pedantic ao2_find in dialogs",
- file, line, func);
- struct sip_via *via = NULL;
-
- args.method = req->method;
- args.callid = NULL; /* we already matched this. */
- args.totag = totag;
- args.fromtag = fromtag;
- args.seqno = seqno;
- /* get via header information. */
- args.ruri = REQ_OFFSET_TO_STR(req, rlpart2);
- via = parse_via(sip_get_header(req, "Via"));
- if (via) {
- args.viasentby = via->sent_by;
- args.viabranch = via->branch;
- }
- /* determine if this is a Request with authentication credentials. */
- if (!ast_strlen_zero(sip_get_header(req, "Authorization")) ||
- !ast_strlen_zero(sip_get_header(req, "Proxy-Authorization"))) {
- args.authentication_present = 1;
- }
- /* if it is a response, get the response code */
- if (req->method == SIP_RESPONSE) {
- const char* e = ast_skip_blanks(REQ_OFFSET_TO_STR(req, rlpart2));
- int respid;
- if (!ast_strlen_zero(e) && (sscanf(e, "%30d", &respid) == 1)) {
- args.respid = respid;
- }
- }
-
- /* Iterate a list of dialogs already matched by Call-id */
- while (iterator && (sip_pvt_ptr = ao2_iterator_next(iterator))) {
- sip_pvt_lock(sip_pvt_ptr);
- found = match_req_to_dialog(sip_pvt_ptr, &args);
- sip_pvt_unlock(sip_pvt_ptr);
-
- switch (found) {
- case SIP_REQ_MATCH:
- sip_pvt_lock(sip_pvt_ptr);
- if (args.method != SIP_RESPONSE && args.authentication_present
- && strcmp(args.fromtag, sip_pvt_ptr->theirtag)) {
- /* If we have a request that uses athentication and the fromtag is
- * different from that in the original call dialog, update the
- * fromtag in the saved call dialog */
- ast_string_field_set(sip_pvt_ptr, theirtag, args.fromtag);
- }
- sip_pvt_unlock(sip_pvt_ptr);
- ao2_iterator_destroy(iterator);
- dialog_unref(fork_pvt, "unref fork_pvt");
- free_via(via);
- return sip_pvt_ptr; /* return pvt with ref */
- case SIP_REQ_LOOP_DETECTED:
- /* This is likely a forked Request that somehow resulted in us receiving multiple parts of the fork.
- * RFC 3261 section 8.2.2.2, Indicate that we want to merge requests by sending a 482 response. */
- transmit_response_using_temp(callid, addr, 1, intended_method, req, "482 (Loop Detected)");
- __ao2_ref(sip_pvt_ptr, -1, "pvt did not match incoming SIP msg, unref from search.",
- file, line, func);
- ao2_iterator_destroy(iterator);
- dialog_unref(fork_pvt, "unref fork_pvt");
- free_via(via);
- return NULL;
- case SIP_REQ_FORKED:
- dialog_unref(fork_pvt, "throwing way pvt to fork off of.");
- fork_pvt = dialog_ref(sip_pvt_ptr, "this pvt has a forked request, save this off to copy information into new dialog\n");
- /* fall through */
- case SIP_REQ_NOT_MATCH:
- default:
- __ao2_ref(sip_pvt_ptr, -1, "pvt did not match incoming SIP msg, unref from search",
- file, line, func);
- break;
- }
- }
- if (iterator) {
- ao2_iterator_destroy(iterator);
- }
-
- /* Handle any possible forked requests. This must be done only after transaction matching is complete. */
- if (fork_pvt) {
- /* XXX right now we only support handling forked INVITE Requests. Any other
- * forked request type must be added here. */
- if (fork_pvt->method == SIP_INVITE) {
- forked_invite_init(req, args.totag, fork_pvt, addr);
- dialog_unref(fork_pvt, "throwing way old forked pvt");
- free_via(via);
- return NULL;
- }
- fork_pvt = dialog_unref(fork_pvt, "throwing way pvt to fork off of");
- }
-
- free_via(via);
- } /* end of pedantic mode Request/Reponse to Dialog matching */
-
- /* See if the method is capable of creating a dialog */
- if (sip_methods[intended_method].can_create == CAN_CREATE_DIALOG) {
- struct sip_pvt *p = NULL;
- ast_callid logger_callid = 0;
-
- if (intended_method == SIP_INVITE) {
- logger_callid = ast_create_callid();
- }
-
- /* Ok, time to create a new SIP dialog object, a pvt */
- if (!(p = sip_alloc(callid, addr, 1, intended_method, req, logger_callid))) {
- /* We have a memory or file/socket error (can't allocate RTP sockets or something) so we're not
- getting a dialog from sip_alloc.
-
- Without a dialog we can't retransmit and handle ACKs and all that, but at least
- send an error message.
-
- Sorry, we apologize for the inconvenience
- */
- transmit_response_using_temp(callid, addr, 1, intended_method, req, "500 Server internal error");
- ast_debug(4, "Failed allocating SIP dialog, sending 500 Server internal error and giving up\n");
- }
- return p; /* can be NULL */
- } else if( sip_methods[intended_method].can_create == CAN_CREATE_DIALOG_UNSUPPORTED_METHOD) {
- /* A method we do not support, let's take it on the volley */
- transmit_response_using_temp(callid, addr, 1, intended_method, req, "501 Method Not Implemented");
- ast_debug(2, "Got a request with unsupported SIP method.\n");
- } else if (intended_method != SIP_RESPONSE && intended_method != SIP_ACK) {
- /* This is a request outside of a dialog that we don't know about */
- transmit_response_using_temp(callid, addr, 1, intended_method, req, "481 Call leg/transaction does not exist");
- ast_debug(2, "That's odd... Got a request in unknown dialog. Callid %s\n", callid ? callid : "");
- }
- /* We do not respond to responses for dialogs that we don't know about, we just drop
- the session quickly */
- if (intended_method == SIP_RESPONSE)
- ast_debug(2, "That's odd... Got a response on a call we don't know about. Callid %s\n", callid ? callid : "");
-
- return NULL;
-}
-
-/*! \brief create sip_registry object from register=> line in sip.conf and link into reg container */
-static int sip_register(const char *value, int lineno)
-{
- struct sip_registry *reg;
-
- reg = ao2_t_find(registry_list, value, OBJ_SEARCH_KEY, "check for existing registry");
- if (reg) {
- ao2_t_ref(reg, -1, "throw away found registry");
- return 0;
- }
-
- if (!(reg = ao2_t_alloc(sizeof(*reg), sip_registry_destroy, "allocate a registry struct"))) {
- ast_log(LOG_ERROR, "Out of memory. Can't allocate SIP registry entry\n");
- return -1;
- }
-
- reg->expire = -1;
- reg->timeout = -1;
-
- if (ast_string_field_init(reg, 256)) {
- ao2_t_ref(reg, -1, "failed to string_field_init, drop reg");
- return -1;
- }
-
- ast_string_field_set(reg, configvalue, value);
- if (sip_parse_register_line(reg, default_expiry, value, lineno)) {
- ao2_t_ref(reg, -1, "failure to parse, unref the reg pointer");
- return -1;
- }
-
- /* set default expiry if necessary */
- if (reg->refresh && !reg->expiry && !reg->configured_expiry) {
- reg->refresh = reg->expiry = reg->configured_expiry = default_expiry;
- }
-
- ao2_t_link(registry_list, reg, "link reg to registry_list");
- ao2_t_ref(reg, -1, "unref the reg pointer");
-
- return 0;
-}
-
-/*! \brief Parse mwi=> line in sip.conf and add to list */
-static int sip_subscribe_mwi(const char *value, int lineno)
-{
- struct sip_subscription_mwi *mwi;
- int portnum = 0;
- enum ast_transport transport = AST_TRANSPORT_UDP;
- char buf[256] = "";
- char *username = NULL, *hostname = NULL, *secret = NULL, *authuser = NULL, *porta = NULL, *mailbox = NULL;
-
- if (!value) {
- return -1;
- }
-
- ast_copy_string(buf, value, sizeof(buf));
-
- username = buf;
-
- if ((hostname = strrchr(buf, '@'))) {
- *hostname++ = '\0';
- } else {
- return -1;
- }
-
- if ((secret = strchr(username, ':'))) {
- *secret++ = '\0';
- if ((authuser = strchr(secret, ':'))) {
- *authuser++ = '\0';
- }
- }
-
- if ((mailbox = strchr(hostname, '/'))) {
- *mailbox++ = '\0';
- }
-
- if (ast_strlen_zero(username) || ast_strlen_zero(hostname) || ast_strlen_zero(mailbox)) {
- ast_log(LOG_WARNING, "Format for MWI subscription is user[:secret[:authuser]]@host[:port]/mailbox at line %d\n", lineno);
- return -1;
- }
-
- if ((porta = strchr(hostname, ':'))) {
- *porta++ = '\0';
- if (!(portnum = atoi(porta))) {
- ast_log(LOG_WARNING, "%s is not a valid port number at line %d\n", porta, lineno);
- return -1;
- }
- }
-
- if (!(mwi = ao2_t_alloc(sizeof(*mwi), sip_subscribe_mwi_destroy, "allocate an mwi struct"))) {
- return -1;
- }
-
- mwi->resub = -1;
-
- if (ast_string_field_init(mwi, 256)) {
- ao2_t_ref(mwi, -1, "failed to string_field_init, drop mwi");
- return -1;
- }
-
- ast_string_field_set(mwi, username, username);
- if (secret) {
- ast_string_field_set(mwi, secret, secret);
- }
- if (authuser) {
- ast_string_field_set(mwi, authuser, authuser);
- }
- ast_string_field_set(mwi, hostname, hostname);
- ast_string_field_set(mwi, mailbox, mailbox);
- mwi->portno = portnum;
- mwi->transport = transport;
-
- ao2_t_link(subscription_mwi_list, mwi, "link new mwi object");
- ao2_t_ref(mwi, -1, "unref to match ao2_t_alloc");
-
- return 0;
-}
-
-static void mark_method_allowed(unsigned int *allowed_methods, enum sipmethod method)
-{
- (*allowed_methods) |= (1 << method);
-}
-
-static void mark_method_unallowed(unsigned int *allowed_methods, enum sipmethod method)
-{
- (*allowed_methods) &= ~(1 << method);
-}
-
-/*! \brief Check if method is allowed for a device or a dialog */
-static int is_method_allowed(unsigned int *allowed_methods, enum sipmethod method)
-{
- return ((*allowed_methods) >> method) & 1;
-}
-
-static void mark_parsed_methods(unsigned int *methods, char *methods_str)
-{
- char *method;
- for (method = strsep(&methods_str, ","); !ast_strlen_zero(method); method = strsep(&methods_str, ",")) {
- int id = find_sip_method(ast_skip_blanks(method));
- if (id == SIP_UNKNOWN) {
- continue;
- }
- mark_method_allowed(methods, id);
- }
-}
-/*!
- * \brief parse the Allow header to see what methods the endpoint we
- * are communicating with allows.
- *
- * We parse the allow header on incoming Registrations and save the
- * result to the SIP peer that is registering. When the registration
- * expires, we clear what we know about the peer's allowed methods.
- * When the peer re-registers, we once again parse to see if the
- * list of allowed methods has changed.
- *
- * For peers that do not register, we parse the first message we receive
- * during a call to see what is allowed, and save the information
- * for the duration of the call.
- * \param req The SIP request we are parsing
- * \retval The methods allowed
- */
-static unsigned int parse_allowed_methods(struct sip_request *req)
-{
- char *allow = ast_strdupa(sip_get_header(req, "Allow"));
- unsigned int allowed_methods = SIP_UNKNOWN;
-
- if (ast_strlen_zero(allow)) {
- /* I have witnessed that REGISTER requests from Polycom phones do not
- * place the phone's allowed methods in an Allow header. Instead, they place the
- * allowed methods in a methods= parameter in the Contact header.
- */
- char *contact = ast_strdupa(sip_get_header(req, "Contact"));
- char *methods = strstr(contact, ";methods=");
-
- if (ast_strlen_zero(methods)) {
- /* RFC 3261 states:
- *
- * "The absence of an Allow header field MUST NOT be
- * interpreted to mean that the UA sending the message supports no
- * methods. Rather, it implies that the UA is not providing any
- * information on what methods it supports."
- *
- * For simplicity, we'll assume that the peer allows all known
- * SIP methods if they have no Allow header. We can then clear out the necessary
- * bits if the peer lets us know that we have sent an unsupported method.
- */
- return UINT_MAX;
- }
- allow = ast_strip_quoted(methods + 9, "\"", "\"");
- }
- mark_parsed_methods(&allowed_methods, allow);
- return allowed_methods;
-}
-
-/*! A wrapper for parse_allowed_methods geared toward sip_pvts
- *
- * This function, in addition to setting the allowed methods for a sip_pvt
- * also will take into account the setting of the SIP_PAGE2_RPID_UPDATE flag.
- *
- * \param pvt The sip_pvt we are setting the allowed_methods for
- * \param req The request which we are parsing
- * \retval The methods alloweded by the sip_pvt
- */
-static unsigned int set_pvt_allowed_methods(struct sip_pvt *pvt, struct sip_request *req)
-{
- pvt->allowed_methods = parse_allowed_methods(req);
-
- if (ast_test_flag(&pvt->flags[1], SIP_PAGE2_RPID_UPDATE)) {
- mark_method_allowed(&pvt->allowed_methods, SIP_UPDATE);
- }
- pvt->allowed_methods &= ~(pvt->disallowed_methods);
-
- return pvt->allowed_methods;
-}
-
-/*! \brief Parse multiline SIP headers into one header
- This is enabled if pedanticsipchecking is enabled */
-static void lws2sws(struct ast_str *data)
-{
- char *msgbuf = ast_str_buffer(data);
- int len = ast_str_strlen(data);
- int h = 0, t = 0;
- int lws = 0;
- int just_read_eol = 0;
- int done_with_headers = 0;
-
- while (h < len) {
- /* Eliminate all CRs */
- if (msgbuf[h] == '\r') {
- h++;
- continue;
- }
- /* Check for end-of-line */
- if (msgbuf[h] == '\n') {
- if (just_read_eol) {
- done_with_headers = 1;
- } else {
- just_read_eol = 1;
- }
- /* Check for end-of-message */
- if (h + 1 == len)
- break;
- /* Check for a continuation line */
- if (!done_with_headers
- && (msgbuf[h + 1] == ' ' || msgbuf[h + 1] == '\t')) {
- /* Merge continuation line */
- h++;
- continue;
- }
- /* Propagate LF and start new line */
- msgbuf[t++] = msgbuf[h++];
- lws = 0;
- continue;
- } else {
- just_read_eol = 0;
- }
- if (!done_with_headers
- && (msgbuf[h] == ' ' || msgbuf[h] == '\t')) {
- if (lws) {
- h++;
- continue;
- }
- msgbuf[t++] = msgbuf[h++];
- lws = 1;
- continue;
- }
- msgbuf[t++] = msgbuf[h++];
- if (lws)
- lws = 0;
- }
- msgbuf[t] = '\0';
- ast_str_update(data);
-}
-
-/*! \brief Parse a SIP message
- \note this function is used both on incoming and outgoing packets
-*/
-static int parse_request(struct sip_request *req)
-{
- char *c = ast_str_buffer(req->data);
- ptrdiff_t *dst = req->header;
- int i = 0;
- unsigned int lim = SIP_MAX_HEADERS - 1;
- unsigned int skipping_headers = 0;
- ptrdiff_t current_header_offset = 0;
- char *previous_header = "";
-
- req->header[0] = 0;
- req->headers = -1; /* mark that we are working on the header */
- for (; *c; c++) {
- if (*c == '\r') { /* remove \r */
- *c = '\0';
- } else if (*c == '\n') { /* end of this line */
- *c = '\0';
- current_header_offset = (c + 1) - ast_str_buffer(req->data);
- previous_header = ast_str_buffer(req->data) + dst[i];
- if (skipping_headers) {
- /* check to see if this line is blank; if so, turn off
- the skipping flag, so the next line will be processed
- as a body line */
- if (ast_strlen_zero(previous_header)) {
- skipping_headers = 0;
- }
- dst[i] = current_header_offset; /* record start of next line */
- continue;
- }
- if (sipdebug) {
- ast_debug(4, "%7s %2d [%3d]: %s\n",
- req->headers < 0 ? "Header" : "Body",
- i, (int) strlen(previous_header), previous_header);
- }
- if (ast_strlen_zero(previous_header) && req->headers < 0) {
- req->headers = i; /* record number of header lines */
- dst = req->line; /* start working on the body */
- i = 0;
- lim = SIP_MAX_LINES - 1;
- } else { /* move to next line, check for overflows */
- if (i++ == lim) {
- /* if we're processing headers, then skip any remaining
- headers and move on to processing the body, otherwise
- we're done */
- if (req->headers != -1) {
- break;
- } else {
- req->headers = i;
- dst = req->line;
- i = 0;
- lim = SIP_MAX_LINES - 1;
- skipping_headers = 1;
- }
- }
- }
- dst[i] = current_header_offset; /* record start of next line */
- }
- }
-
- /* Check for last header or body line without CRLF. The RFC for SDP requires CRLF,
- but since some devices send without, we'll be generous in what we accept. However,
- if we've already reached the maximum number of lines for portion of the message
- we were parsing, we can't accept any more, so just ignore it.
- */
- previous_header = ast_str_buffer(req->data) + dst[i];
- if ((i < lim) && !ast_strlen_zero(previous_header)) {
- if (sipdebug) {
- ast_debug(4, "%7s %2d [%3d]: %s\n",
- req->headers < 0 ? "Header" : "Body",
- i, (int) strlen(previous_header), previous_header );
- }
- i++;
- }
-
- /* update count of header or body lines */
- if (req->headers >= 0) { /* we are in the body */
- req->lines = i;
- } else { /* no body */
- req->headers = i;
- req->lines = 0;
- /* req->data->used will be a NULL byte */
- req->line[0] = ast_str_strlen(req->data);
- }
-
- if (*c) {
- ast_log(LOG_WARNING, "Too many lines, skipping <%s>\n", c);
- }
-
- /* Split up the first line parts */
- return determine_firstline_parts(req);
-}
-
-/*!
- \brief Determine whether a SIP message contains an SDP in its body
- \param req the SIP request to process
- \retval 1 if SDP found.
- \retval 0 if not found.
-
- Also updates req->sdp_start and req->sdp_count to indicate where the SDP
- lives in the message body.
-*/
-static int find_sdp(struct sip_request *req)
-{
- const char *content_type;
- const char *content_length;
- const char *search;
- char *boundary;
- unsigned int x;
- int boundaryisquoted = FALSE;
- int found_application_sdp = FALSE;
- int found_end_of_headers = FALSE;
-
- content_length = sip_get_header(req, "Content-Length");
-
- if (!ast_strlen_zero(content_length)) {
- if (sscanf(content_length, "%30u", &x) != 1) {
- ast_log(LOG_WARNING, "Invalid Content-Length: %s\n", content_length);
- return 0;
- }
-
- /* Content-Length of zero means there can't possibly be an
- SDP here, even if the Content-Type says there is */
- if (x == 0)
- return 0;
- }
-
- content_type = sip_get_header(req, "Content-Type");
-
- /* if the body contains only SDP, this is easy */
- if (!strncasecmp(content_type, "application/sdp", 15)) {
- req->sdp_start = 0;
- req->sdp_count = req->lines;
- return req->lines ? 1 : 0;
- }
-
- /* if it's not multipart/mixed, there cannot be an SDP */
- if (strncasecmp(content_type, "multipart/mixed", 15))
- return 0;
-
- /* if there is no boundary marker, it's invalid */
- if ((search = strcasestr(content_type, ";boundary=")))
- search += 10;
- else if ((search = strcasestr(content_type, "; boundary=")))
- search += 11;
- else
- return 0;
-
- if (ast_strlen_zero(search))
- return 0;
-
- /* If the boundary is quoted with ", remove quote */
- if (*search == '\"') {
- search++;
- boundaryisquoted = TRUE;
- }
-
- /* make a duplicate of the string, with two extra characters
- at the beginning */
- boundary = ast_strdupa(search - 2);
- boundary[0] = boundary[1] = '-';
- /* Remove final quote */
- if (boundaryisquoted)
- boundary[strlen(boundary) - 1] = '\0';
-
- /* search for the boundary marker, the empty line delimiting headers from
- sdp part and the end boundry if it exists */
-
- for (x = 0; x < (req->lines); x++) {
- const char *line = REQ_OFFSET_TO_STR(req, line[x]);
- if (!strncasecmp(line, boundary, strlen(boundary))){
- if (found_application_sdp && found_end_of_headers) {
- req->sdp_count = (x - 1) - req->sdp_start;
- return 1;
- }
- found_application_sdp = FALSE;
- }
- if (!strcasecmp(line, "Content-Type: application/sdp"))
- found_application_sdp = TRUE;
-
- if (ast_strlen_zero(line)) {
- if (found_application_sdp && !found_end_of_headers){
- req->sdp_start = x;
- found_end_of_headers = TRUE;
- }
- }
- }
- if (found_application_sdp && found_end_of_headers) {
- req->sdp_count = x - req->sdp_start;
- return TRUE;
- }
- return FALSE;
-}
-
-/*! \brief Change hold state for a call */
-static void change_hold_state(struct sip_pvt *dialog, struct sip_request *req, int holdstate, int sendonly)
-{
- if (sip_cfg.notifyhold && (!holdstate || !ast_test_flag(&dialog->flags[1], SIP_PAGE2_CALL_ONHOLD))) {
- sip_peer_hold(dialog, holdstate);
- }
- append_history(dialog, holdstate ? "Hold" : "Unhold", "%s", ast_str_buffer(req->data));
- if (!holdstate) { /* Put off remote hold */
- ast_clear_flag(&dialog->flags[1], SIP_PAGE2_CALL_ONHOLD); /* Clear both flags */
- return;
- }
- /* No address for RTP, we're on hold */
-
- /* Ensure hold flags are cleared so that overlapping flags do not conflict */
- ast_clear_flag(&dialog->flags[1], SIP_PAGE2_CALL_ONHOLD);
-
- if (sendonly == 1) /* One directional hold (sendonly/recvonly) */
- ast_set_flag(&dialog->flags[1], SIP_PAGE2_CALL_ONHOLD_ONEDIR);
- else if (sendonly == 2) /* Inactive stream */
- ast_set_flag(&dialog->flags[1], SIP_PAGE2_CALL_ONHOLD_INACTIVE);
- else
- ast_set_flag(&dialog->flags[1], SIP_PAGE2_CALL_ONHOLD_ACTIVE);
- return;
-}
-
-/*! \internal
- * \brief Returns whether or not the address is null or ANY / unspecified (0.0.0.0 or ::)
- * \retval TRUE if the address is null or any
- * \retval FALSE if the address it not null or any
- * \note In some circumstances, calls should be placed on hold if either of these conditions exist.
- */
-static int sockaddr_is_null_or_any(const struct ast_sockaddr *addr)
-{
- return ast_sockaddr_isnull(addr) || ast_sockaddr_is_any(addr);
-}
-
-/*! \brief Check the media stream list to see if the given type already exists */
-static int has_media_stream(struct sip_pvt *p, enum media_type m)
-{
- struct offered_media *offer = NULL;
- AST_LIST_TRAVERSE(&p->offered_media, offer, next) {
- if (m == offer->type) {
- return 1;
- }
- }
- return 0;
-}
-
-static void configure_rtcp(struct sip_pvt *p, struct ast_rtp_instance *instance, int which, int remote_rtcp_mux)
-{
- int local_rtcp_mux = ast_test_flag(&p->flags[2], SIP_PAGE3_RTCP_MUX);
- int fd = -1;
-
- if (local_rtcp_mux && remote_rtcp_mux) {
- ast_rtp_instance_set_prop(instance, AST_RTP_PROPERTY_RTCP, AST_RTP_INSTANCE_RTCP_MUX);
- } else {
- ast_rtp_instance_set_prop(instance, AST_RTP_PROPERTY_RTCP, AST_RTP_INSTANCE_RTCP_STANDARD);
- fd = ast_rtp_instance_fd(instance, 1);
- }
-
- if (p->owner) {
- ast_channel_set_fd(p->owner, which, fd);
- }
-}
-
-static void set_ice_components(struct sip_pvt *p, struct ast_rtp_instance *instance, int remote_rtcp_mux)
-{
- struct ast_rtp_engine_ice *ice;
- int local_rtcp_mux = ast_test_flag(&p->flags[2], SIP_PAGE3_RTCP_MUX);
-
- ice = ast_rtp_instance_get_ice(instance);
- if (!ice) {
- return;
- }
-
- if (local_rtcp_mux && remote_rtcp_mux) {
- /* We both support RTCP mux. Only one ICE component necessary */
- ice->change_components(instance, 1);
- } else {
- /* They either don't support RTCP mux or we don't know if they do yet. */
- ice->change_components(instance, 2);
- }
-}
-
-static int has_media_level_attribute(int start, struct sip_request *req, const char *attr)
-{
- int next = start;
- char type;
- const char *value;
-
- /* We don't care about the return result here */
- get_sdp_iterate(&next, req, "m");
-
- while ((type = get_sdp_line(&start, next, req, &value)) != '\0') {
- if (type == 'a' && !strcasecmp(value, attr)) {
- return 1;
- }
- }
-
- return 0;
-}
-
-/*! \brief Process SIP SDP offer, select formats and activate media channels
- If offer is rejected, we will not change any properties of the call
- Return 0 on success, a negative value on errors.
- Must be called after find_sdp().
-*/
-static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action, int is_offer)
-{
- int res = 0;
-
- /* Iterators for SDP parsing */
- int start = req->sdp_start;
- int next = start;
- int iterator = start;
-
- /* Temporary vars for SDP parsing */
- char type = '\0';
- const char *value = NULL;
- const char *m = NULL; /* SDP media offer */
- const char *nextm = NULL;
- int len = -1;
- struct offered_media *offer;
-
- /* Host information */
- struct ast_sockaddr sessionsa;
- struct ast_sockaddr audiosa;
- struct ast_sockaddr videosa;
- struct ast_sockaddr textsa;
- struct ast_sockaddr imagesa;
- struct ast_sockaddr *sa = NULL; /*!< RTP audio destination IP address */
- struct ast_sockaddr *vsa = NULL; /*!< RTP video destination IP address */
- struct ast_sockaddr *tsa = NULL; /*!< RTP text destination IP address */
- struct ast_sockaddr *isa = NULL; /*!< UDPTL image destination IP address */
- int portno = -1; /*!< RTP audio destination port number */
- int vportno = -1; /*!< RTP video destination port number */
- int tportno = -1; /*!< RTP text destination port number */
- int udptlportno = -1; /*!< UDPTL image destination port number */
-
- /* Peer capability is the capability in the SDP, non codec is RFC2833 DTMF (101) */
- struct ast_format_cap *peercapability = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
- struct ast_format_cap *vpeercapability = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
- struct ast_format_cap *tpeercapability = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
-
- int peernoncodeccapability = 0, vpeernoncodeccapability = 0, tpeernoncodeccapability = 0;
-
- struct ast_rtp_codecs newaudiortp = AST_RTP_CODECS_NULL_INIT;
- struct ast_rtp_codecs newvideortp = AST_RTP_CODECS_NULL_INIT;
- struct ast_rtp_codecs newtextrtp = AST_RTP_CODECS_NULL_INIT;
- struct ast_format_cap *newjointcapability = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT); /* Negotiated capability */
- struct ast_format_cap *newpeercapability = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
- int newnoncodeccapability;
-
- const char *codecs;
- unsigned int codec;
-
- /* SRTP */
- int secure_audio = FALSE;
- int secure_video = FALSE;
-
- /* RTCP Multiplexing */
- int remote_rtcp_mux_audio = FALSE;
- int remote_rtcp_mux_video = FALSE;
-
- /* Others */
- int sendonly = -1;
- unsigned int numberofports;
- int last_rtpmap_codec = 0;
- int red_data_pt[10]; /* For T.140 RED */
- int red_num_gen = 0; /* For T.140 RED */
- char red_fmtp[100] = "empty"; /* For T.140 RED */
- int debug = sip_debug_test_pvt(p);
-
- /* START UNKNOWN */
- struct ast_str *codec_buf = ast_str_alloca(AST_FORMAT_CAP_NAMES_LEN);
- struct ast_format *tmp_fmt;
- /* END UNKNOWN */
-
- /* Initial check */
- if (!p->rtp) {
- ast_log(LOG_ERROR, "Got SDP but have no RTP session allocated.\n");
- res = -1;
- goto process_sdp_cleanup;
- }
- if (!peercapability || !vpeercapability || !tpeercapability || !newpeercapability || !newjointcapability) {
- res = -1;
- goto process_sdp_cleanup;
- }
-
- if (ast_rtp_codecs_payloads_initialize(&newaudiortp) || ast_rtp_codecs_payloads_initialize(&newvideortp) ||
- ast_rtp_codecs_payloads_initialize(&newtextrtp)) {
- res = -1;
- goto process_sdp_cleanup;
- }
-
- /* Update our last rtprx when we receive an SDP, too */
- p->lastrtprx = p->lastrtptx = time(NULL); /* XXX why both ? */
-
- offered_media_list_destroy(p);
-
- /* Scan for the first media stream (m=) line to limit scanning of globals */
- nextm = get_sdp_iterate(&next, req, "m");
- if (ast_strlen_zero(nextm)) {
- ast_log(LOG_WARNING, "Insufficient information for SDP (m= not found)\n");
- res = -1;
- goto process_sdp_cleanup;
- }
-
- /* Scan session level SDP parameters (lines before first media stream) */
- while ((type = get_sdp_line(&iterator, next - 1, req, &value)) != '\0') {
- int processed = FALSE;
- switch (type) {
- case 'o':
- /* If we end up receiving SDP that doesn't actually modify the session we don't want to treat this as a fatal
- * error. We just want to ignore the SDP and let the rest of the packet be handled as normal.
- */
- if (!process_sdp_o(value, p)) {
- res = (p->session_modify == FALSE) ? 0 : -1;
- goto process_sdp_cleanup;
- }
- processed = TRUE;
- break;
- case 'c':
- if (process_sdp_c(value, &sessionsa)) {
- processed = TRUE;
- sa = &sessionsa;
- vsa = sa;
- tsa = sa;
- isa = sa;
- }
- break;
- case 'a':
- if (process_sdp_a_sendonly(value, &sendonly)) {
- processed = TRUE;
- }
- else if (process_sdp_a_audio(value, p, &newaudiortp, &last_rtpmap_codec))
- processed = TRUE;
- else if (process_sdp_a_video(value, p, &newvideortp, &last_rtpmap_codec))
- processed = TRUE;
- else if (process_sdp_a_text(value, p, &newtextrtp, red_fmtp, &red_num_gen, red_data_pt, &last_rtpmap_codec))
- processed = TRUE;
- else if (process_sdp_a_image(value, p))
- processed = TRUE;
-
- if (process_sdp_a_ice(value, p, p->rtp, 0)) {
- processed = TRUE;
- }
- if (process_sdp_a_ice(value, p, p->vrtp, 0)) {
- processed = TRUE;
- }
- if (process_sdp_a_ice(value, p, p->trtp, 0)) {
- processed = TRUE;
- }
-
- if (process_sdp_a_dtls(value, p, p->rtp)) {
- processed = TRUE;
- if (p->srtp) {
- ast_set_flag(p->srtp, AST_SRTP_CRYPTO_OFFER_OK);
- }
- }
- if (process_sdp_a_dtls(value, p, p->vrtp)) {
- processed = TRUE;
- if (p->vsrtp) {
- ast_set_flag(p->vsrtp, AST_SRTP_CRYPTO_OFFER_OK);
- }
- }
- if (process_sdp_a_dtls(value, p, p->trtp)) {
- processed = TRUE;
- if (p->tsrtp) {
- ast_set_flag(p->tsrtp, AST_SRTP_CRYPTO_OFFER_OK);
- }
- }
-
- break;
- }
-
- ast_debug(3, "Processing session-level SDP %c=%s... %s\n", type, value, (processed == TRUE)? "OK." : "UNSUPPORTED OR FAILED.");
- }
-
- /* default: novideo and notext set */
- p->novideo = TRUE;
- p->notext = TRUE;
-
- /* Scan media stream (m=) specific parameters loop */
- while (!ast_strlen_zero(nextm)) {
- int audio = FALSE;
- int video = FALSE;
- int image = FALSE;
- int text = FALSE;
- int processed_crypto = FALSE;
- int rtcp_mux_offered = 0;
- char protocol[18] = {0,};
- unsigned int x;
- struct ast_rtp_engine_dtls *dtls;
-
- numberofports = 0;
- len = -1;
- start = next;
- m = nextm;
- iterator = next;
- nextm = get_sdp_iterate(&next, req, "m");
-
- if (!(offer = ast_calloc(1, sizeof(*offer)))) {
- ast_log(LOG_WARNING, "Failed to allocate memory for SDP offer list\n");
- res = -1;
- goto process_sdp_cleanup;
- }
- AST_LIST_INSERT_TAIL(&p->offered_media, offer, next);
- offer->type = SDP_UNKNOWN;
-
- /* We need to check for this ahead of time */
- rtcp_mux_offered = has_media_level_attribute(iterator, req, "rtcp-mux");
-
- /* Check for 'audio' media offer */
- if (p->rtp && strncmp(m, "audio ", 6) == 0) {
- if ((sscanf(m, "audio %30u/%30u %17s %n", &x, &numberofports, protocol, &len) == 3 && len > 0) ||
- (sscanf(m, "audio %30u %17s %n", &x, protocol, &len) == 2 && len > 0)) {
- codecs = m + len;
- /* produce zero-port m-line since it may be needed later
- * length is "m=audio 0 " + protocol + " " + codecs + "\r\n\0" */
- if (!(offer->decline_m_line = ast_malloc(10 + strlen(protocol) + 1 + strlen(codecs) + 3))) {
- ast_log(LOG_WARNING, "Failed to allocate memory for SDP offer declination\n");
- res = -1;
- goto process_sdp_cleanup;
- }
- /* guaranteed to be exactly the right length */
- sprintf(offer->decline_m_line, "m=audio 0 %s %s\r\n", protocol, codecs);
-
- if (x == 0) {
- ast_debug(1, "Ignoring audio media offer because port number is zero\n");
- continue;
- }
-
- if (has_media_stream(p, SDP_AUDIO)) {
- ast_log(LOG_WARNING, "Declining non-primary audio stream: %s\n", m);
- continue;
- }
-
- /* Check number of ports offered for stream */
- if (numberofports > 1) {
- ast_log(LOG_WARNING, "%u ports offered for audio media, not supported by Asterisk. Will try anyway...\n", numberofports);
- }
-
- if ((!strcmp(protocol, "RTP/SAVPF") || !strcmp(protocol, "UDP/TLS/RTP/SAVPF")) && !ast_test_flag(&p->flags[2], SIP_PAGE3_USE_AVPF)) {
- if (req->method != SIP_RESPONSE) {
- ast_log(LOG_NOTICE, "Received SAVPF profle in audio offer but AVPF is not enabled, enabling: %s\n", m);
- secure_audio = 1;
- ast_set_flag(&p->flags[2], SIP_PAGE3_USE_AVPF);
- }
- else {
-
- ast_log(LOG_WARNING, "Received SAVPF profle in audio answer but AVPF is not enabled: %s\n", m);
- continue;
- }
- } else if ((!strcmp(protocol, "RTP/SAVP") || !strcmp(protocol, "UDP/TLS/RTP/SAVP")) && ast_test_flag(&p->flags[2], SIP_PAGE3_USE_AVPF)) {
- if (req->method != SIP_RESPONSE) {
- ast_log(LOG_NOTICE, "Received SAVP profle in audio offer but AVPF is enabled, disabling: %s\n", m);
- secure_audio = 1;
- ast_clear_flag(&p->flags[2], SIP_PAGE3_USE_AVPF);
- }
- else {
- ast_log(LOG_WARNING, "Received SAVP profile in audio offer but AVPF is enabled: %s\n", m);
- continue;
- }
- } else if (!strcmp(protocol, "UDP/TLS/RTP/SAVP") || !strcmp(protocol, "UDP/TLS/RTP/SAVPF")) {
- secure_audio = 1;
-
- processed_crypto = 1;
- if (p->srtp) {
- ast_set_flag(p->srtp, AST_SRTP_CRYPTO_OFFER_OK);
- }
- } else if (!strcmp(protocol, "RTP/SAVP") || !strcmp(protocol, "RTP/SAVPF")) {
- secure_audio = 1;
- } else if (!strcmp(protocol, "RTP/AVPF") && !ast_test_flag(&p->flags[2], SIP_PAGE3_USE_AVPF)) {
- if (req->method != SIP_RESPONSE) {
- ast_log(LOG_NOTICE, "Received AVPF profile in audio offer but AVPF is not enabled, enabling: %s\n", m);
- ast_set_flag(&p->flags[2], SIP_PAGE3_USE_AVPF);
- }
- else {
- ast_log(LOG_WARNING, "Received AVP profile in audio answer but AVPF is enabled: %s\n", m);
- continue;
- }
- } else if (!strcmp(protocol, "RTP/AVP") && ast_test_flag(&p->flags[2], SIP_PAGE3_USE_AVPF)) {
- if (req->method != SIP_RESPONSE) {
- ast_log(LOG_NOTICE, "Received AVP profile in audio answer but AVPF is enabled, disabling: %s\n", m);
- ast_clear_flag(&p->flags[2], SIP_PAGE3_USE_AVPF);
- }
- else {
- ast_log(LOG_WARNING, "Received AVP profile in audio answer but AVPF is enabled: %s\n", m);
- continue;
- }
- } else if ((!strcmp(protocol, "UDP/TLS/RTP/SAVP") || !strcmp(protocol, "UDP/TLS/RTP/SAVPF")) &&
- (!(dtls = ast_rtp_instance_get_dtls(p->rtp)) || !dtls->active(p->rtp))) {
- ast_log(LOG_WARNING, "Received UDP/TLS in audio offer but DTLS is not enabled: %s\n", m);
- continue;
- } else if (strcmp(protocol, "RTP/AVP") && strcmp(protocol, "RTP/AVPF")) {
- ast_log(LOG_WARNING, "Unknown RTP profile in audio offer: %s\n", m);
- continue;
- }
-
- audio = TRUE;
- offer->type = SDP_AUDIO;
- portno = x;
-
- /* Scan through the RTP payload types specified in a "m=" line: */
- for (; !ast_strlen_zero(codecs); codecs = ast_skip_blanks(codecs + len)) {
- if (sscanf(codecs, "%30u%n", &codec, &len) != 1) {
- ast_log(LOG_WARNING, "Invalid syntax in RTP audio format list: %s\n", codecs);
- res = -1;
- goto process_sdp_cleanup;
- }
- if (debug) {
- ast_verbose("Found RTP audio format %u\n", codec);
- }
-
- ast_rtp_codecs_payloads_set_m_type(&newaudiortp, NULL, codec);
- }
- } else {
- ast_log(LOG_WARNING, "Rejecting audio media offer due to invalid or unsupported syntax: %s\n", m);
- res = -1;
- goto process_sdp_cleanup;
- }
- }
- /* Check for 'video' media offer */
- else if (p->vrtp && strncmp(m, "video ", 6) == 0) {
- if ((sscanf(m, "video %30u/%30u %17s %n", &x, &numberofports, protocol, &len) == 3 && len > 0) ||
- (sscanf(m, "video %30u %17s %n", &x, protocol, &len) == 2 && len > 0)) {
- codecs = m + len;
- /* produce zero-port m-line since it may be needed later
- * length is "m=video 0 " + protocol + " " + codecs + "\r\n\0" */
- if (!(offer->decline_m_line = ast_malloc(10 + strlen(protocol) + 1 + strlen(codecs) + 3))) {
- ast_log(LOG_WARNING, "Failed to allocate memory for SDP offer declination\n");
- res = -1;
- goto process_sdp_cleanup;
- }
- /* guaranteed to be exactly the right length */
- sprintf(offer->decline_m_line, "m=video 0 %s %s\r\n", protocol, codecs);
-
- if (x == 0) {
- ast_debug(1, "Ignoring video stream offer because port number is zero\n");
- continue;
- }
-
- /* Check number of ports offered for stream */
- if (numberofports > 1) {
- ast_log(LOG_WARNING, "%u ports offered for video stream, not supported by Asterisk. Will try anyway...\n", numberofports);
- }
-
- if (has_media_stream(p, SDP_VIDEO)) {
- ast_log(LOG_WARNING, "Declining non-primary video stream: %s\n", m);
- continue;
- }
-
- if ((!strcmp(protocol, "RTP/SAVPF") || !strcmp(protocol, "UDP/TLS/RTP/SAVPF")) && !ast_test_flag(&p->flags[2], SIP_PAGE3_USE_AVPF)) {
- ast_log(LOG_WARNING, "Received SAVPF profle in video offer but AVPF is not enabled: %s\n", m);
- continue;
- } else if ((!strcmp(protocol, "RTP/SAVP") || !strcmp(protocol, "UDP/TLS/RTP/SAVP")) && ast_test_flag(&p->flags[2], SIP_PAGE3_USE_AVPF)) {
- ast_log(LOG_WARNING, "Received SAVP profile in video offer but AVPF is enabled: %s\n", m);
- continue;
- } else if (!strcmp(protocol, "UDP/TLS/RTP/SAVP") || !strcmp(protocol, "UDP/TLS/RTP/SAVPF")) {
- secure_video = 1;
-
- processed_crypto = 1;
- if (p->vsrtp || (p->vsrtp = ast_sdp_srtp_alloc())) {
- ast_set_flag(p->vsrtp, AST_SRTP_CRYPTO_OFFER_OK);
- }
- } else if (!strcmp(protocol, "RTP/SAVP") || !strcmp(protocol, "RTP/SAVPF")) {
- secure_video = 1;
- } else if (!strcmp(protocol, "RTP/AVPF") && !ast_test_flag(&p->flags[2], SIP_PAGE3_USE_AVPF)) {
- ast_log(LOG_WARNING, "Received AVPF profile in video offer but AVPF is not enabled: %s\n", m);
- continue;
- } else if (!strcmp(protocol, "RTP/AVP") && ast_test_flag(&p->flags[2], SIP_PAGE3_USE_AVPF)) {
- ast_log(LOG_WARNING, "Received AVP profile in video offer but AVPF is enabled: %s\n", m);
- continue;
- } else if (strcmp(protocol, "RTP/AVP") && strcmp(protocol, "RTP/AVPF")) {
- ast_log(LOG_WARNING, "Unknown RTP profile in video offer: %s\n", m);
- continue;
- }
-
- video = TRUE;
- p->novideo = FALSE;
- offer->type = SDP_VIDEO;
- vportno = x;
-
- /* Scan through the RTP payload types specified in a "m=" line: */
- for (; !ast_strlen_zero(codecs); codecs = ast_skip_blanks(codecs + len)) {
- if (sscanf(codecs, "%30u%n", &codec, &len) != 1) {
- ast_log(LOG_WARNING, "Invalid syntax in RTP video format list: %s\n", codecs);
- res = -1;
- goto process_sdp_cleanup;
- }
- if (debug) {
- ast_verbose("Found RTP video format %u\n", codec);
- }
- ast_rtp_codecs_payloads_set_m_type(&newvideortp, NULL, codec);
- }
- } else {
- ast_log(LOG_WARNING, "Rejecting video media offer due to invalid or unsupported syntax: %s\n", m);
- res = -1;
- goto process_sdp_cleanup;
- }
- }
- /* Check for 'text' media offer */
- else if (p->trtp && strncmp(m, "text ", 5) == 0) {
- if ((sscanf(m, "text %30u/%30u %17s %n", &x, &numberofports, protocol, &len) == 3 && len > 0) ||
- (sscanf(m, "text %30u %17s %n", &x, protocol, &len) == 2 && len > 0)) {
- codecs = m + len;
- /* produce zero-port m-line since it may be needed later
- * length is "m=text 0 " + protocol + " " + codecs + "\r\n\0" */
- if (!(offer->decline_m_line = ast_malloc(9 + strlen(protocol) + 1 + strlen(codecs) + 3))) {
- ast_log(LOG_WARNING, "Failed to allocate memory for SDP offer declination\n");
- res = -1;
- goto process_sdp_cleanup;
- }
- /* guaranteed to be exactly the right length */
- sprintf(offer->decline_m_line, "m=text 0 %s %s\r\n", protocol, codecs);
-
- if (x == 0) {
- ast_debug(1, "Ignoring text stream offer because port number is zero\n");
- continue;
- }
-
- /* Check number of ports offered for stream */
- if (numberofports > 1) {
- ast_log(LOG_WARNING, "%u ports offered for text stream, not supported by Asterisk. Will try anyway...\n", numberofports);
- }
-
- if (!strcmp(protocol, "RTP/AVPF") && !ast_test_flag(&p->flags[2], SIP_PAGE3_USE_AVPF)) {
- ast_log(LOG_WARNING, "Received AVPF profile in text offer but AVPF is not enabled: %s\n", m);
- continue;
- } else if (!strcmp(protocol, "RTP/AVP") && ast_test_flag(&p->flags[2], SIP_PAGE3_USE_AVPF)) {
- ast_log(LOG_WARNING, "Received AVP profile in text offer but AVPF is enabled: %s\n", m);
- continue;
- } else if (strcmp(protocol, "RTP/AVP") && strcmp(protocol, "RTP/AVPF")) {
- ast_log(LOG_WARNING, "Unknown RTP profile in text offer: %s\n", m);
- continue;
- }
-
- if (has_media_stream(p, SDP_TEXT)) {
- ast_log(LOG_WARNING, "Declining non-primary text stream: %s\n", m);
- continue;
- }
-
- text = TRUE;
- p->notext = FALSE;
- offer->type = SDP_TEXT;
- tportno = x;
-
- /* Scan through the RTP payload types specified in a "m=" line: */
- for (; !ast_strlen_zero(codecs); codecs = ast_skip_blanks(codecs + len)) {
- if (sscanf(codecs, "%30u%n", &codec, &len) != 1) {
- ast_log(LOG_WARNING, "Invalid syntax in RTP video format list: %s\n", codecs);
- res = -1;
- goto process_sdp_cleanup;
- }
- if (debug) {
- ast_verbose("Found RTP text format %u\n", codec);
- }
- ast_rtp_codecs_payloads_set_m_type(&newtextrtp, NULL, codec);
- }
- } else {
- ast_log(LOG_WARNING, "Rejecting text stream offer due to invalid or unsupported syntax: %s\n", m);
- res = -1;
- goto process_sdp_cleanup;
- }
- }
- /* Check for 'image' media offer */
- else if (strncmp(m, "image ", 6) == 0) {
- if (((sscanf(m, "image %30u udptl t38%n", &x, &len) == 1 && len > 0) ||
- (sscanf(m, "image %30u UDPTL t38%n", &x, &len) == 1 && len > 0))) {
- /* produce zero-port m-line since it may be needed later
- * length is "m=image 0 udptl t38" + "\r\n\0" */
- if (!(offer->decline_m_line = ast_malloc(22))) {
- ast_log(LOG_WARNING, "Failed to allocate memory for SDP offer declination\n");
- res = -1;
- goto process_sdp_cleanup;
- }
- /* guaranteed to be exactly the right length */
- strcpy(offer->decline_m_line, "m=image 0 udptl t38\r\n");
-
- if (x == 0) {
- ast_debug(1, "Ignoring image stream offer because port number is zero\n");
- continue;
- }
-
- if (initialize_udptl(p)) {
- ast_log(LOG_WARNING, "Failed to initialize UDPTL, declining image stream\n");
- continue;
- }
-
- if (has_media_stream(p, SDP_IMAGE)) {
- ast_log(LOG_WARNING, "Declining non-primary image stream: %s\n", m);
- continue;
- }
-
- image = TRUE;
- if (debug) {
- ast_verbose("Got T.38 offer in SDP in dialog %s\n", p->callid);
- }
-
- offer->type = SDP_IMAGE;
- udptlportno = x;
-
- if (p->t38.state != T38_ENABLED) {
- memset(&p->t38.their_parms, 0, sizeof(p->t38.their_parms));
-
- /* default EC to none, the remote end should
- * respond with the EC they want to use */
- ast_udptl_set_error_correction_scheme(p->udptl, UDPTL_ERROR_CORRECTION_NONE);
- }
- } else if (sscanf(m, "image %30u %17s t38%n", &x, protocol, &len) == 2 && len > 0) {
- ast_log(LOG_WARNING, "Declining image stream due to unsupported transport: %s\n", m);
- /* produce zero-port m-line since this is guaranteed to be declined
- * length is "m=image 0 strlen(protocol) t38" + "\r\n\0" */
- if (!(offer->decline_m_line = ast_malloc(10 + strlen(protocol) + 7))) {
- ast_log(LOG_WARNING, "Failed to allocate memory for SDP offer declination\n");
- res = -1;
- goto process_sdp_cleanup;
- }
- /* guaranteed to be exactly the right length */
- sprintf(offer->decline_m_line, "m=image 0 %s t38\r\n", protocol);
- continue;
- } else {
- ast_log(LOG_WARNING, "Rejecting image media offer due to invalid or unsupported syntax: %s\n", m);
- res = -1;
- goto process_sdp_cleanup;
- }
- } else {
- char type[20] = {0,};
- if ((sscanf(m, "%19s %30u/%30u %n", type, &x, &numberofports, &len) == 3 && len > 0) ||
- (sscanf(m, "%19s %30u %n", type, &x, &len) == 2 && len > 0)) {
- /* produce zero-port m-line since it may be needed later
- * length is "m=" + type + " 0 " + remainder + "\r\n\0" */
- if (!(offer->decline_m_line = ast_malloc(2 + strlen(type) + 3 + strlen(m + len) + 3))) {
- ast_log(LOG_WARNING, "Failed to allocate memory for SDP offer declination\n");
- res = -1;
- goto process_sdp_cleanup;
- }
- /* guaranteed to be long enough */
- sprintf(offer->decline_m_line, "m=%s 0 %s\r\n", type, m + len);
- continue;
- } else {
- ast_log(LOG_WARNING, "Unsupported top-level media type in offer: %s\n", m);
- res = -1;
- goto process_sdp_cleanup;
- }
- }
-
- /* Media stream specific parameters */
- while ((type = get_sdp_line(&iterator, next - 1, req, &value)) != '\0') {
- int processed = FALSE;
-
- switch (type) {
- case 'c':
- if (audio) {
- if (process_sdp_c(value, &audiosa)) {
- processed = TRUE;
- sa = &audiosa;
- }
- } else if (video) {
- if (process_sdp_c(value, &videosa)) {
- processed = TRUE;
- vsa = &videosa;
- }
- } else if (text) {
- if (process_sdp_c(value, &textsa)) {
- processed = TRUE;
- tsa = &textsa;
- }
- } else if (image) {
- if (process_sdp_c(value, &imagesa)) {
- processed = TRUE;
- isa = &imagesa;
- }
- }
- break;
- case 'a':
- /* Audio specific scanning */
- if (audio) {
- if (process_sdp_a_ice(value, p, p->rtp, rtcp_mux_offered)) {
- processed = TRUE;
- } else if (process_sdp_a_dtls(value, p, p->rtp)) {
- processed_crypto = TRUE;
- processed = TRUE;
- if (p->srtp) {
- ast_set_flag(p->srtp, AST_SRTP_CRYPTO_OFFER_OK);
- }
- } else if (process_sdp_a_sendonly(value, &sendonly)) {
- processed = TRUE;
- } else if (!processed_crypto && process_crypto(p, p->rtp, &p->srtp, value)) {
- processed_crypto = TRUE;
- processed = TRUE;
- if (secure_audio == FALSE) {
- ast_log(AST_LOG_NOTICE, "Processed audio crypto attribute without SAVP specified; accepting anyway\n");
- secure_audio = TRUE;
- }
- } else if (process_sdp_a_audio(value, p, &newaudiortp, &last_rtpmap_codec)) {
- processed = TRUE;
- } else if (process_sdp_a_rtcp_mux(value, p, &remote_rtcp_mux_audio)) {
- processed = TRUE;
- }
- }
- /* Video specific scanning */
- else if (video) {
- if (process_sdp_a_ice(value, p, p->vrtp, rtcp_mux_offered)) {
- processed = TRUE;
- } else if (process_sdp_a_dtls(value, p, p->vrtp)) {
- processed_crypto = TRUE;
- processed = TRUE;
- if (p->vsrtp) {
- ast_set_flag(p->vsrtp, AST_SRTP_CRYPTO_OFFER_OK);
- }
- } else if (!processed_crypto && process_crypto(p, p->vrtp, &p->vsrtp, value)) {
- processed_crypto = TRUE;
- processed = TRUE;
- if (secure_video == FALSE) {
- ast_log(AST_LOG_NOTICE, "Processed video crypto attribute without SAVP specified; accepting anyway\n");
- secure_video = TRUE;
- }
- } else if (process_sdp_a_video(value, p, &newvideortp, &last_rtpmap_codec)) {
- processed = TRUE;
- } else if (process_sdp_a_rtcp_mux(value, p, &remote_rtcp_mux_video)) {
- processed = TRUE;
- }
- }
- /* Text (T.140) specific scanning */
- else if (text) {
- if (process_sdp_a_ice(value, p, p->trtp, rtcp_mux_offered)) {
- processed = TRUE;
- } else if (process_sdp_a_text(value, p, &newtextrtp, red_fmtp, &red_num_gen, red_data_pt, &last_rtpmap_codec)) {
- processed = TRUE;
- } else if (!processed_crypto && process_crypto(p, p->trtp, &p->tsrtp, value)) {
- processed_crypto = TRUE;
- processed = TRUE;
- }
- }
- /* Image (T.38 FAX) specific scanning */
- else if (image) {
- if (process_sdp_a_image(value, p))
- processed = TRUE;
- }
- break;
- }
-
- ast_debug(3, "Processing media-level (%s) SDP %c=%s... %s\n",
- (audio == TRUE)? "audio" : (video == TRUE)? "video" : (text == TRUE)? "text" : "image",
- type, value,
- (processed == TRUE)? "OK." : "UNSUPPORTED OR FAILED.");
- }
-
- /* Ensure crypto lines are provided where necessary */
- if (audio && secure_audio && !processed_crypto) {
- ast_log(LOG_WARNING, "Rejecting secure audio stream without encryption details: %s\n", m);
- res = -1;
- goto process_sdp_cleanup;
- } else if (video && secure_video && !processed_crypto) {
- ast_log(LOG_WARNING, "Rejecting secure video stream without encryption details: %s\n", m);
- res = -1;
- goto process_sdp_cleanup;
- }
- }
-
- /* Sanity checks */
- if (!sa && !vsa && !tsa && !isa) {
- ast_log(LOG_WARNING, "Insufficient information in SDP (c=)...\n");
- res = -1;
- goto process_sdp_cleanup;
- }
-
- if ((portno == -1) &&
- (vportno == -1) &&
- (tportno == -1) &&
- (udptlportno == -1)) {
- ast_log(LOG_WARNING, "Failing due to no acceptable offer found\n");
- res = -1;
- goto process_sdp_cleanup;
- }
-
- if (p->srtp && p->udptl && udptlportno != -1) {
- ast_debug(1, "Terminating SRTP due to T.38 UDPTL\n");
- ast_sdp_srtp_destroy(p->srtp);
- p->srtp = NULL;
- }
-
- if (secure_audio && !(p->srtp && (ast_test_flag(p->srtp, AST_SRTP_CRYPTO_OFFER_OK)))) {
- ast_log(LOG_WARNING, "Can't provide secure audio requested in SDP offer\n");
- res = -1;
- goto process_sdp_cleanup;
- }
-
- if (!secure_audio && p->srtp) {
- ast_log(LOG_WARNING, "Failed to receive SDP offer/answer with required SRTP crypto attributes for audio\n");
- res = -1;
- goto process_sdp_cleanup;
- }
-
- if (secure_video && !(p->vsrtp && (ast_test_flag(p->vsrtp, AST_SRTP_CRYPTO_OFFER_OK)))) {
- ast_log(LOG_WARNING, "Can't provide secure video requested in SDP offer\n");
- res = -1;
- goto process_sdp_cleanup;
- }
-
- if (!p->novideo && !secure_video && p->vsrtp) {
- ast_log(LOG_WARNING, "Failed to receive SDP offer/answer with required SRTP crypto attributes for video\n");
- res = -1;
- goto process_sdp_cleanup;
- }
-
- if (!(secure_audio || secure_video || (p->udptl && udptlportno != -1)) && ast_test_flag(&p->flags[1], SIP_PAGE2_USE_SRTP)) {
- ast_log(LOG_WARNING, "Matched device setup to use SRTP, but request was not!\n");
- res = -1;
- goto process_sdp_cleanup;
- }
-
- if (udptlportno == -1) {
- change_t38_state(p, T38_DISABLED);
- }
-
- if (is_offer) {
- /*
- * Setup rx payload type mapping to prefer the mapping
- * from the peer that the RFC says we SHOULD use.
- */
- ast_rtp_codecs_payloads_xover(&newaudiortp, &newaudiortp, NULL);
- ast_rtp_codecs_payloads_xover(&newvideortp, &newvideortp, NULL);
- ast_rtp_codecs_payloads_xover(&newtextrtp, &newtextrtp, NULL);
- }
-
- /* Now gather all of the codecs that we are asked for: */
- ast_rtp_codecs_payload_formats(&newaudiortp, peercapability, &peernoncodeccapability);
- ast_rtp_codecs_payload_formats(&newvideortp, vpeercapability, &vpeernoncodeccapability);
- ast_rtp_codecs_payload_formats(&newtextrtp, tpeercapability, &tpeernoncodeccapability);
-
- ast_format_cap_append_from_cap(newpeercapability, peercapability, AST_MEDIA_TYPE_AUDIO);
- ast_format_cap_append_from_cap(newpeercapability, vpeercapability, AST_MEDIA_TYPE_VIDEO);
- ast_format_cap_append_from_cap(newpeercapability, tpeercapability, AST_MEDIA_TYPE_TEXT);
-
- ast_format_cap_get_compatible(p->caps, newpeercapability, newjointcapability);
- if (!ast_format_cap_count(newjointcapability) && udptlportno == -1) {
- ast_log(LOG_NOTICE, "No compatible codecs, not accepting this offer!\n");
- /* Do NOT Change current setting */
- res = -1;
- goto process_sdp_cleanup;
- }
-
- newnoncodeccapability = p->noncodeccapability & peernoncodeccapability;
-
- if (debug) {
- /* shame on whoever coded this.... */
- struct ast_str *cap_buf = ast_str_alloca(AST_FORMAT_CAP_NAMES_LEN);
- struct ast_str *peer_buf = ast_str_alloca(AST_FORMAT_CAP_NAMES_LEN);
- struct ast_str *vpeer_buf = ast_str_alloca(AST_FORMAT_CAP_NAMES_LEN);
- struct ast_str *tpeer_buf = ast_str_alloca(AST_FORMAT_CAP_NAMES_LEN);
- struct ast_str *joint_buf = ast_str_alloca(AST_FORMAT_CAP_NAMES_LEN);
- struct ast_str *s1 = ast_str_alloca(SIPBUFSIZE);
- struct ast_str *s2 = ast_str_alloca(SIPBUFSIZE);
- struct ast_str *s3 = ast_str_alloca(SIPBUFSIZE);
-
- ast_verbose("Capabilities: us - %s, peer - audio=%s/video=%s/text=%s, combined - %s\n",
- ast_format_cap_get_names(p->caps, &cap_buf),
- ast_format_cap_get_names(peercapability, &peer_buf),
- ast_format_cap_get_names(vpeercapability, &vpeer_buf),
- ast_format_cap_get_names(tpeercapability, &tpeer_buf),
- ast_format_cap_get_names(newjointcapability, &joint_buf));
-
- ast_verbose("Non-codec capabilities (dtmf): us - %s, peer - %s, combined - %s\n",
- ast_rtp_lookup_mime_multiple2(s1, NULL, p->noncodeccapability, 0, 0),
- ast_rtp_lookup_mime_multiple2(s2, NULL, peernoncodeccapability, 0, 0),
- ast_rtp_lookup_mime_multiple2(s3, NULL, newnoncodeccapability, 0, 0));
- }
-
- /* When UDPTL is negotiated it is expected that there are no compatible codecs as audio or
- * video is not being transported, thus we continue in this function further up if that is
- * the case. If we receive an SDP answer containing both a UDPTL stream and another media
- * stream however we need to check again to ensure that there is at least one joint codec
- * instead of assuming there is one.
- */
- if ((portno != -1 || vportno != -1 || tportno != -1) && ast_format_cap_count(newjointcapability)) {
- /* We are now ready to change the sip session and RTP structures with the offered codecs, since
- they are acceptable */
- unsigned int framing;
- ast_format_cap_remove_by_type(p->jointcaps, AST_MEDIA_TYPE_UNKNOWN);
- ast_format_cap_append_from_cap(p->jointcaps, newjointcapability, AST_MEDIA_TYPE_UNKNOWN); /* Our joint codec profile for this call */
- ast_format_cap_remove_by_type(p->peercaps, AST_MEDIA_TYPE_UNKNOWN);
- ast_format_cap_append_from_cap(p->peercaps, newpeercapability, AST_MEDIA_TYPE_UNKNOWN); /* The other side's capability in latest offer */
- p->jointnoncodeccapability = newnoncodeccapability; /* DTMF capabilities */
-
- tmp_fmt = ast_format_cap_get_format(p->jointcaps, 0);
- framing = ast_format_cap_get_format_framing(p->jointcaps, tmp_fmt);
- /* respond with single most preferred joint codec, limiting the other side's choice */
- if (ast_test_flag(&p->flags[1], SIP_PAGE2_PREFERRED_CODEC)) {
- ast_format_cap_remove_by_type(p->jointcaps, AST_MEDIA_TYPE_UNKNOWN);
- ast_format_cap_append(p->jointcaps, tmp_fmt, framing);
- }
- if (!ast_rtp_codecs_get_framing(&newaudiortp)) {
- /* Peer did not force us to use a specific framing, so use our own */
- ast_rtp_codecs_set_framing(&newaudiortp, framing);
- }
- ao2_ref(tmp_fmt, -1);
- }
-
- /* Setup audio address and port */
- if (p->rtp) {
- if (sa && portno > 0) {
- /* Start ICE negotiation here, only when it is response, and setting that we are conrolling agent,
- as we are offerer */
- set_ice_components(p, p->rtp, remote_rtcp_mux_audio);
- if (req->method == SIP_RESPONSE) {
- start_ice(p->rtp, 1);
- }
- ast_sockaddr_set_port(sa, portno);
- ast_rtp_instance_set_remote_address(p->rtp, sa);
- if (debug) {
- ast_verbose("Peer audio RTP is at port %s\n",
- ast_sockaddr_stringify(sa));
- }
-
- ast_rtp_codecs_payloads_copy(&newaudiortp, ast_rtp_instance_get_codecs(p->rtp), p->rtp);
- /* Ensure RTCP is enabled since it may be inactive
- if we're coming back from a T.38 session */
- configure_rtcp(p, p->rtp, SIP_AUDIO_RTCP_FD, remote_rtcp_mux_audio);
-
- if (ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_AUTO) {
- ast_clear_flag(&p->flags[0], SIP_DTMF);
- if (newnoncodeccapability & AST_RTP_DTMF) {
- /* XXX Would it be reasonable to drop the DSP at this point? XXX */
- ast_set_flag(&p->flags[0], SIP_DTMF_RFC2833);
- /* Since RFC2833 is now negotiated we need to change some properties of the RTP stream */
- ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_DTMF, 1);
- ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_DTMF_COMPENSATE, ast_test_flag(&p->flags[1], SIP_PAGE2_RFC2833_COMPENSATE));
- } else {
- ast_set_flag(&p->flags[0], SIP_DTMF_INBAND);
- }
- }
- } else if (udptlportno > 0) {
- if (debug)
- ast_verbose("Got T.38 Re-invite without audio. Keeping RTP active during T.38 session.\n");
-
- /* Force media to go through us for T.38. */
- memset(&p->redirip, 0, sizeof(p->redirip));
-
- /* Prevent audio RTCP reads */
- if (p->owner) {
- ast_channel_set_fd(p->owner, SIP_AUDIO_RTCP_FD, -1);
- }
- /* Silence RTCP while audio RTP is inactive */
- ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_RTCP, AST_RTP_INSTANCE_RTCP_DISABLED);
- } else {
- ast_rtp_instance_stop(p->rtp);
- if (debug)
- ast_verbose("Peer doesn't provide audio\n");
- }
- }
-
- /* Setup video address and port */
- if (p->vrtp) {
- if (vsa && vportno > 0) {
- set_ice_components(p, p->vrtp, remote_rtcp_mux_video);
- start_ice(p->vrtp, (req->method != SIP_RESPONSE) ? 0 : 1);
- ast_sockaddr_set_port(vsa, vportno);
- ast_rtp_instance_set_remote_address(p->vrtp, vsa);
- if (debug) {
- ast_verbose("Peer video RTP is at port %s\n",
- ast_sockaddr_stringify(vsa));
- }
- ast_rtp_codecs_payloads_copy(&newvideortp, ast_rtp_instance_get_codecs(p->vrtp), p->vrtp);
- configure_rtcp(p, p->vrtp, SIP_VIDEO_RTCP_FD, remote_rtcp_mux_video);
- } else {
- ast_rtp_instance_stop(p->vrtp);
- if (debug)
- ast_verbose("Peer doesn't provide video\n");
- }
- }
-
- /* Setup text address and port */
- if (p->trtp) {
- if (tsa && tportno > 0) {
- start_ice(p->trtp, (req->method != SIP_RESPONSE) ? 0 : 1);
- ast_sockaddr_set_port(tsa, tportno);
- ast_rtp_instance_set_remote_address(p->trtp, tsa);
- if (debug) {
- ast_verbose("Peer T.140 RTP is at port %s\n",
- ast_sockaddr_stringify(tsa));
- }
- if (ast_format_cap_iscompatible_format(p->jointcaps, ast_format_t140_red) != AST_FORMAT_CMP_NOT_EQUAL) {
- p->red = 1;
- ast_rtp_red_init(p->trtp, 300, red_data_pt, 2);
- } else {
- p->red = 0;
- }
- ast_rtp_codecs_payloads_copy(&newtextrtp, ast_rtp_instance_get_codecs(p->trtp), p->trtp);
- } else {
- ast_rtp_instance_stop(p->trtp);
- if (debug)
- ast_verbose("Peer doesn't provide T.140\n");
- }
- }
-
- /* Setup image address and port */
- if (p->udptl) {
- if (isa && udptlportno > 0) {
- if (ast_test_flag(&p->flags[1], SIP_PAGE2_SYMMETRICRTP) && ast_test_flag(&p->flags[1], SIP_PAGE2_UDPTL_DESTINATION)) {
- ast_rtp_instance_get_remote_address(p->rtp, isa);
- if (!ast_sockaddr_isnull(isa) && debug) {
- ast_debug(1, "Peer T.38 UDPTL is set behind NAT and with destination, destination address now %s\n", ast_sockaddr_stringify(isa));
- }
- }
- ast_sockaddr_set_port(isa, udptlportno);
- ast_udptl_set_peer(p->udptl, isa);
- if (debug)
- ast_debug(1, "Peer T.38 UDPTL is at port %s\n", ast_sockaddr_stringify(isa));
-
- /* verify the far max ifp can be calculated. this requires far max datagram to be set. */
- if (!ast_udptl_get_far_max_datagram(p->udptl)) {
- /* setting to zero will force a default if none was provided by the SDP */
- ast_udptl_set_far_max_datagram(p->udptl, 0);
- }
-
- /* Remote party offers T38, we need to update state */
- if ((t38action == SDP_T38_ACCEPT) &&
- (p->t38.state == T38_LOCAL_REINVITE)) {
- change_t38_state(p, T38_ENABLED);
- } else if ((t38action == SDP_T38_INITIATE) &&
- p->owner && p->lastinvite) {
- change_t38_state(p, T38_PEER_REINVITE); /* T38 Offered in re-invite from remote party */
- /* If fax detection is enabled then send us off to the fax extension */
- if (ast_test_flag(&p->flags[1], SIP_PAGE2_FAX_DETECT_T38)) {
- ast_channel_lock(p->owner);
- if (strcmp(ast_channel_exten(p->owner), "fax")) {
- const char *target_context = S_OR(ast_channel_macrocontext(p->owner), ast_channel_context(p->owner));
- ast_channel_unlock(p->owner);
- if (ast_exists_extension(p->owner, target_context, "fax", 1,
- S_COR(ast_channel_caller(p->owner)->id.number.valid, ast_channel_caller(p->owner)->id.number.str, NULL))) {
- ast_verb(2, "Redirecting '%s' to fax extension due to peer T.38 re-INVITE\n", ast_channel_name(p->owner));
- pbx_builtin_setvar_helper(p->owner, "FAXEXTEN", ast_channel_exten(p->owner));
- if (ast_async_goto(p->owner, target_context, "fax", 1)) {
- ast_log(LOG_NOTICE, "Failed to async goto '%s' into fax of '%s'\n", ast_channel_name(p->owner), target_context);
- }
- } else {
- ast_log(LOG_NOTICE, "T.38 re-INVITE detected but no fax extension\n");
- }
- } else {
- ast_channel_unlock(p->owner);
- }
- }
- }
- } else {
- change_t38_state(p, T38_DISABLED);
- ast_udptl_stop(p->udptl);
- if (debug)
- ast_debug(1, "Peer doesn't provide T.38 UDPTL\n");
- }
- }
-
- if ((portno == -1) && (p->t38.state != T38_DISABLED) && (p->t38.state != T38_REJECTED)) {
- ast_debug(3, "Have T.38 but no audio, accepting offer anyway\n");
- res = 0;
- goto process_sdp_cleanup;
- }
-
- /* Ok, we're going with this offer */
- ast_debug(2, "We're settling with these formats: %s\n", ast_format_cap_get_names(p->jointcaps, &codec_buf));
-
- if (!p->owner) { /* There's no open channel owning us so we can return here. For a re-invite or so, we proceed */
- res = 0;
- goto process_sdp_cleanup;
- }
-
- ast_debug(4, "We have an owner, now see if we need to change this call\n");
- if (ast_format_cap_has_type(p->jointcaps, AST_MEDIA_TYPE_AUDIO)) {
- struct ast_format_cap *caps;
- unsigned int framing;
-
- if (debug) {
- struct ast_str *cap_buf = ast_str_alloca(AST_FORMAT_CAP_NAMES_LEN);
- struct ast_str *joint_buf = ast_str_alloca(AST_FORMAT_CAP_NAMES_LEN);
-
- ast_debug(1, "Setting native formats after processing SDP. peer joint formats %s, old nativeformats %s\n",
- ast_format_cap_get_names(p->jointcaps, &joint_buf),
- ast_format_cap_get_names(ast_channel_nativeformats(p->owner), &cap_buf));
- }
-
- caps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
- if (caps) {
- tmp_fmt = ast_format_cap_get_format(p->jointcaps, 0);
- framing = ast_format_cap_get_format_framing(p->jointcaps, tmp_fmt);
- ast_format_cap_append(caps, tmp_fmt, framing);
- ast_format_cap_append_from_cap(caps, vpeercapability, AST_MEDIA_TYPE_VIDEO);
- ast_format_cap_append_from_cap(caps, tpeercapability, AST_MEDIA_TYPE_TEXT);
- ast_channel_nativeformats_set(p->owner, caps);
- ao2_ref(caps, -1);
- ao2_ref(tmp_fmt, -1);
- }
-
- ast_set_read_format(p->owner, ast_channel_readformat(p->owner));
- ast_set_write_format(p->owner, ast_channel_writeformat(p->owner));
- }
-
- if (ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD) && (!ast_sockaddr_isnull(sa) || !ast_sockaddr_isnull(vsa) || !ast_sockaddr_isnull(tsa) || !ast_sockaddr_isnull(isa)) && (!sendonly || sendonly == -1)) {
- if (!ast_test_flag(&p->flags[2], SIP_PAGE3_DISCARD_REMOTE_HOLD_RETRIEVAL)) {
- ast_queue_unhold(p->owner);
- }
- /* Activate a re-invite */
- ast_queue_frame(p->owner, &ast_null_frame);
- change_hold_state(p, req, FALSE, sendonly);
- } else if ((sockaddr_is_null_or_any(sa) && sockaddr_is_null_or_any(vsa) && sockaddr_is_null_or_any(tsa) && sockaddr_is_null_or_any(isa)) || (sendonly && sendonly != -1)) {
- if (!ast_test_flag(&p->flags[2], SIP_PAGE3_DISCARD_REMOTE_HOLD_RETRIEVAL)) {
- ast_queue_hold(p->owner, p->mohsuggest);
- }
- if (sendonly)
- ast_rtp_instance_stop(p->rtp);
- /* RTCP needs to go ahead, even if we're on hold!!! */
- /* Activate a re-invite */
- ast_queue_frame(p->owner, &ast_null_frame);
- change_hold_state(p, req, TRUE, sendonly);
- }
-
-process_sdp_cleanup:
- if (res) {
- offered_media_list_destroy(p);
- }
- ast_rtp_codecs_payloads_destroy(&newtextrtp);
- ast_rtp_codecs_payloads_destroy(&newvideortp);
- ast_rtp_codecs_payloads_destroy(&newaudiortp);
- ao2_cleanup(peercapability);
- ao2_cleanup(vpeercapability);
- ao2_cleanup(tpeercapability);
- ao2_cleanup(newjointcapability);
- ao2_cleanup(newpeercapability);
- return res;
-}
-
-static int process_sdp_o(const char *o, struct sip_pvt *p)
-{
- const char *o_copy_start;
- char *o_copy;
- char *token;
- int offset;
- int64_t sess_version;
- char unique[128];
-
- /* Store the SDP version number of remote UA. This will allow us to
- distinguish between session modifications and session refreshes. If
- the remote UA does not send an incremented SDP version number in a
- subsequent RE-INVITE then that means its not changing media session.
- The RE-INVITE may have been sent to update connected party, remote
- target or to refresh the session (Session-Timers). Asterisk must not
- change media session and increment its own version number in answer
- SDP in this case. */
-
- p->session_modify = TRUE;
-
- if (ast_strlen_zero(o)) {
- ast_log(LOG_WARNING, "SDP syntax error. SDP without an o= line\n");
- return FALSE;
- }
-
- /* o=
- */
-
- o_copy_start = o_copy = ast_strdupa(o);
- token = strsep(&o_copy, " "); /* Skip username */
- if (!o_copy) {
- ast_log(LOG_WARNING, "SDP syntax error in o= line username\n");
- return FALSE;
- }
- token = strsep(&o_copy, " "); /* sess-id */
- if (!o_copy) {
- ast_log(LOG_WARNING, "SDP syntax error in o= line sess-id\n");
- return FALSE;
- }
- token = strsep(&o_copy, " "); /* sess-version */
- if (!o_copy || !sscanf(token, "%30" SCNd64, &sess_version)) {
- ast_log(LOG_WARNING, "SDP syntax error in o= line sess-version\n");
- return FALSE;
- }
-
- /* Copy all after sess-version on top of sess-version into unique.
- * is a numeric string such that the tuple of ,
- * , , , and forms a
- * globally unique identifier for the session.
- * I.e. all except the */
- ast_copy_string(unique, o, sizeof(unique)); /* copy all of o= contents */
- offset = (o_copy - o_copy_start); /* after sess-version */
- if (offset < sizeof(unique)) {
- /* copy all after sess-version on top of sess-version */
- int sess_version_start = token - o_copy_start;
- ast_copy_string(unique + sess_version_start, o + offset, sizeof(unique) - sess_version_start);
- }
-
- /* We need to check the SDP version number the other end sent us;
- * our rules for deciding what to accept are a bit complex.
- *
- * 1) if 'ignoresdpversion' has been set for this dialog, then
- * we will just accept whatever they sent and assume it is
- * a modification of the session, even if it is not
- * 2) otherwise, if this is the first SDP we've seen from them
- * we accept it;
- * note that _them_ may change, in which case the
- * sessionunique_remote will be different
- * 3) otherwise, if the new SDP version number is higher than the
- * old one, we accept it
- * 4) otherwise, if this SDP is in response to us requesting a switch
- * to T.38, we accept the SDP, but also generate a warning message
- * that this peer should have the 'ignoresdpversion' option set,
- * because it is not following the SDP offer/answer RFC; if we did
- * not request a switch to T.38, then we stop parsing the SDP, as it
- * has not changed from the previous version
- */
- if (sip_debug_test_pvt(p)) {
- if (ast_strlen_zero(p->sessionunique_remote)) {
- ast_verbose("Got SDP version %" PRId64 " and unique parts [%s]\n",
- sess_version, unique);
- } else {
- ast_verbose("Comparing SDP version %" PRId64 " -> %" PRId64 " and unique parts [%s] -> [%s]\n",
- p->sessionversion_remote, sess_version, p->sessionunique_remote, unique);
- }
- }
-
- if (ast_test_flag(&p->flags[1], SIP_PAGE2_IGNORESDPVERSION) ||
- sess_version > p->sessionversion_remote ||
- strcmp(unique, S_OR(p->sessionunique_remote, ""))) {
- p->sessionversion_remote = sess_version;
- ast_string_field_set(p, sessionunique_remote, unique);
- } else {
- if (p->t38.state == T38_LOCAL_REINVITE) {
- p->sessionversion_remote = sess_version;
- ast_string_field_set(p, sessionunique_remote, unique);
- ast_log(LOG_WARNING, "Call %s responded to our T.38 reinvite without changing SDP version; 'ignoresdpversion' should be set for this peer.\n", p->callid);
- } else {
- p->session_modify = FALSE;
- ast_debug(2, "Call %s responded to our reinvite without changing SDP version; ignoring SDP.\n", p->callid);
- return FALSE;
- }
- }
-
- return TRUE;
-}
-
-static int process_sdp_c(const char *c, struct ast_sockaddr *addr)
-{
- char proto[4], host[258];
- int af;
-
- /* Check for Media-description-level-address */
- if (sscanf(c, "IN %3s %255s", proto, host) == 2) {
- if (!strcmp("IP4", proto)) {
- af = AF_INET;
- } else if (!strcmp("IP6", proto)) {
- af = AF_INET6;
- } else {
- ast_log(LOG_WARNING, "Unknown protocol '%s'.\n", proto);
- return FALSE;
- }
- if (ast_sockaddr_resolve_first_af(addr, host, 0, af)) {
- ast_log(LOG_WARNING, "Unable to lookup RTP Audio host in c= line, '%s'\n", c);
- return FALSE;
- }
- return TRUE;
- } else {
- ast_log(LOG_WARNING, "Invalid host in c= line, '%s'\n", c);
- return FALSE;
- }
- return FALSE;
-}
-
-static int process_sdp_a_sendonly(const char *a, int *sendonly)
-{
- int found = FALSE;
-
- if (!strcasecmp(a, "sendonly")) {
- if (*sendonly == -1)
- *sendonly = 1;
- found = TRUE;
- } else if (!strcasecmp(a, "inactive")) {
- if (*sendonly == -1)
- *sendonly = 2;
- found = TRUE;
- } else if (!strcasecmp(a, "sendrecv")) {
- if (*sendonly == -1)
- *sendonly = 0;
- found = TRUE;
- }
- return found;
-}
-
-static int process_sdp_a_ice(const char *a, struct sip_pvt *p, struct ast_rtp_instance *instance, int rtcp_mux_offered)
-{
- struct ast_rtp_engine_ice *ice;
- int found = FALSE;
- char ufrag[256], pwd[256], foundation[33], transport[4], address[46], cand_type[6], relay_address[46] = "";
- struct ast_rtp_engine_ice_candidate candidate = { 0, };
- unsigned int port, relay_port = 0;
-
- if (!instance || !(ice = ast_rtp_instance_get_ice(instance))) {
- return found;
- }
-
- if (sscanf(a, "ice-ufrag: %255s", ufrag) == 1) {
- ice->set_authentication(instance, ufrag, NULL);
- found = TRUE;
- } else if (sscanf(a, "ice-pwd: %255s", pwd) == 1) {
- ice->set_authentication(instance, NULL, pwd);
- found = TRUE;
- } else if (sscanf(a, "candidate: %32s %30u %3s %30u %23s %30u typ %5s %*s %23s %*s %30u", foundation, &candidate.id, transport, (unsigned *)&candidate.priority,
- address, &port, cand_type, relay_address, &relay_port) >= 7) {
-
- if (rtcp_mux_offered && ast_test_flag(&p->flags[2], SIP_PAGE3_RTCP_MUX) && candidate.id > 1) {
- /* If we support RTCP-MUX and they offered it, don't consider RTCP candidates */
- return TRUE;
- }
-
- candidate.foundation = foundation;
- candidate.transport = transport;
-
- ast_sockaddr_parse(&candidate.address, address, PARSE_PORT_FORBID);
- ast_sockaddr_set_port(&candidate.address, port);
-
- if (!strcasecmp(cand_type, "host")) {
- candidate.type = AST_RTP_ICE_CANDIDATE_TYPE_HOST;
- } else if (!strcasecmp(cand_type, "srflx")) {
- candidate.type = AST_RTP_ICE_CANDIDATE_TYPE_SRFLX;
- } else if (!strcasecmp(cand_type, "relay")) {
- candidate.type = AST_RTP_ICE_CANDIDATE_TYPE_RELAYED;
- } else {
- return found;
- }
-
- if (!ast_strlen_zero(relay_address)) {
- ast_sockaddr_parse(&candidate.relay_address, relay_address, PARSE_PORT_FORBID);
- }
-
- if (relay_port) {
- ast_sockaddr_set_port(&candidate.relay_address, relay_port);
- }
-
- ice->add_remote_candidate(instance, &candidate);
-
- found = TRUE;
- } else if (!strcasecmp(a, "ice-lite")) {
- ice->ice_lite(instance);
- found = TRUE;
- }
-
- return found;
-}
-
-static int process_sdp_a_rtcp_mux(const char *a, struct sip_pvt *p, int *requested)
-{
- int found = FALSE;
-
- if (!strncasecmp(a, "rtcp-mux", 8)) {
- *requested = TRUE;
- found = TRUE;
- }
-
- return found;
-}
-
-static int process_sdp_a_dtls(const char *a, struct sip_pvt *p, struct ast_rtp_instance *instance)
-{
- struct ast_rtp_engine_dtls *dtls;
- int found = FALSE;
- char value[256], hash[32];
-
- if (!instance || !p->dtls_cfg.enabled || !(dtls = ast_rtp_instance_get_dtls(instance))) {
- return found;
- }
-
- if (sscanf(a, "setup: %255s", value) == 1) {
- found = TRUE;
-
- if (!strcasecmp(value, "active")) {
- dtls->set_setup(instance, AST_RTP_DTLS_SETUP_ACTIVE);
- } else if (!strcasecmp(value, "passive")) {
- dtls->set_setup(instance, AST_RTP_DTLS_SETUP_PASSIVE);
- } else if (!strcasecmp(value, "actpass")) {
- dtls->set_setup(instance, AST_RTP_DTLS_SETUP_ACTPASS);
- } else if (!strcasecmp(value, "holdconn")) {
- dtls->set_setup(instance, AST_RTP_DTLS_SETUP_HOLDCONN);
- } else {
- ast_log(LOG_WARNING, "Unsupported setup attribute value '%s' received on dialog '%s'\n",
- value, p->callid);
- }
- } else if (sscanf(a, "connection: %255s", value) == 1) {
- found = TRUE;
-
- if (!strcasecmp(value, "new")) {
- dtls->reset(instance);
- } else if (!strcasecmp(value, "existing")) {
- /* Since they want to just use what already exists we go on as if nothing happened */
- } else {
- ast_log(LOG_WARNING, "Unsupported connection attribute value '%s' received on dialog '%s'\n",
- value, p->callid);
- }
- } else if (sscanf(a, "fingerprint: %31s %255s", hash, value) == 2) {
- found = TRUE;
-
- if (!strcasecmp(hash, "sha-1")) {
- dtls->set_fingerprint(instance, AST_RTP_DTLS_HASH_SHA1, value);
- } else if (!strcasecmp(hash, "sha-256")) {
- dtls->set_fingerprint(instance, AST_RTP_DTLS_HASH_SHA256, value);
- } else {
- ast_log(LOG_WARNING, "Unsupported fingerprint hash type '%s' received on dialog '%s'\n",
- hash, p->callid);
- }
- }
-
- return found;
-}
-
-static int process_sdp_a_audio(const char *a, struct sip_pvt *p, struct ast_rtp_codecs *newaudiortp, int *last_rtpmap_codec)
-{
- int found = FALSE;
- unsigned int codec;
- char mimeSubtype[128];
- char fmtp_string[256];
- unsigned int sample_rate;
- int debug = sip_debug_test_pvt(p);
-
- if (!strncasecmp(a, "ptime", 5)) {
- char *tmp = strrchr(a, ':');
- long int framing = 0;
- if (tmp) {
- tmp++;
- framing = strtol(tmp, NULL, 10);
- if (framing == LONG_MIN || framing == LONG_MAX) {
- framing = 0;
- ast_debug(1, "Can't read framing from SDP: %s\n", a);
- }
- }
-
- if (framing && p->autoframing) {
- ast_debug(1, "Setting framing to %ld\n", framing);
- ast_format_cap_set_framing(p->caps, framing);
- ast_rtp_codecs_set_framing(newaudiortp, framing);
- }
- found = TRUE;
- } else if (sscanf(a, "rtpmap: %30u %127[^/]/%30u", &codec, mimeSubtype, &sample_rate) == 3) {
- /* We have a rtpmap to handle */
- if (*last_rtpmap_codec < SDP_MAX_RTPMAP_CODECS) {
- if (!(ast_rtp_codecs_payloads_set_rtpmap_type_rate(newaudiortp, NULL, codec, "audio", mimeSubtype,
- ast_test_flag(&p->flags[0], SIP_G726_NONSTANDARD) ? AST_RTP_OPT_G726_NONSTANDARD : 0, sample_rate))) {
- if (debug)
- ast_verbose("Found audio description format %s for ID %u\n", mimeSubtype, codec);
- //found_rtpmap_codecs[last_rtpmap_codec] = codec;
- (*last_rtpmap_codec)++;
- found = TRUE;
- } else {
- ast_rtp_codecs_payloads_unset(newaudiortp, NULL, codec);
- if (debug)
- ast_verbose("Found unknown media description format %s for ID %u\n", mimeSubtype, codec);
- }
- } else {
- if (debug)
- ast_verbose("Discarded description format %s for ID %u\n", mimeSubtype, codec);
- }
- } else if (sscanf(a, "fmtp: %30u %255[^\t\n]", &codec, fmtp_string) == 2) {
- struct ast_format *format;
-
- if ((format = ast_rtp_codecs_get_payload_format(newaudiortp, codec))) {
- unsigned int bit_rate;
- struct ast_format *format_parsed;
-
- format_parsed = ast_format_parse_sdp_fmtp(format, fmtp_string);
- if (format_parsed) {
- ast_rtp_codecs_payload_replace_format(newaudiortp, codec, format_parsed);
- ao2_replace(format, format_parsed);
- ao2_ref(format_parsed, -1);
- found = TRUE;
- } else {
- ast_rtp_codecs_payloads_unset(newaudiortp, NULL, codec);
- }
-
- if (ast_format_cmp(format, ast_format_g719) == AST_FORMAT_CMP_EQUAL) {
- if (sscanf(fmtp_string, "bitrate=%30u", &bit_rate) == 1) {
- if (bit_rate != 64000) {
- ast_log(LOG_WARNING, "Got G.719 offer at %u bps, but only 64000 bps supported; ignoring.\n", bit_rate);
- ast_rtp_codecs_payloads_unset(newaudiortp, NULL, codec);
- } else {
- found = TRUE;
- }
- }
- }
- ao2_ref(format, -1);
- }
- }
-
- return found;
-}
-
-static int process_sdp_a_video(const char *a, struct sip_pvt *p, struct ast_rtp_codecs *newvideortp, int *last_rtpmap_codec)
-{
- int found = FALSE;
- unsigned int codec;
- char mimeSubtype[128];
- unsigned int sample_rate;
- int debug = sip_debug_test_pvt(p);
- char fmtp_string[256];
-
- if (sscanf(a, "rtpmap: %30u %127[^/]/%30u", &codec, mimeSubtype, &sample_rate) == 3) {
- /* We have a rtpmap to handle */
- if (*last_rtpmap_codec < SDP_MAX_RTPMAP_CODECS) {
- /* Note: should really look at the '#chans' params too */
- if (!strncasecmp(mimeSubtype, "H26", 3) || !strncasecmp(mimeSubtype, "MP4", 3)
- || !strncasecmp(mimeSubtype, "VP8", 3)) {
- if (!(ast_rtp_codecs_payloads_set_rtpmap_type_rate(newvideortp, NULL, codec, "video", mimeSubtype, 0, sample_rate))) {
- if (debug)
- ast_verbose("Found video description format %s for ID %u\n", mimeSubtype, codec);
- //found_rtpmap_codecs[last_rtpmap_codec] = codec;
- (*last_rtpmap_codec)++;
- found = TRUE;
- } else {
- ast_rtp_codecs_payloads_unset(newvideortp, NULL, codec);
- if (debug)
- ast_verbose("Found unknown media description format %s for ID %u\n", mimeSubtype, codec);
- }
- }
- } else {
- if (debug)
- ast_verbose("Discarded description format %s for ID %u\n", mimeSubtype, codec);
- }
- } else if (sscanf(a, "fmtp: %30u %255[^\t\n]", &codec, fmtp_string) == 2) {
- struct ast_format *format;
-
- if ((format = ast_rtp_codecs_get_payload_format(newvideortp, codec))) {
- struct ast_format *format_parsed;
-
- format_parsed = ast_format_parse_sdp_fmtp(format, fmtp_string);
-
- if (format_parsed) {
- ast_rtp_codecs_payload_replace_format(newvideortp, codec, format_parsed);
- ao2_replace(format, format_parsed);
- ao2_ref(format_parsed, -1);
- found = TRUE;
- } else {
- ast_rtp_codecs_payloads_unset(newvideortp, NULL, codec);
- }
- ao2_ref(format, -1);
- }
- }
-
- return found;
-}
-
-static int process_sdp_a_text(const char *a, struct sip_pvt *p, struct ast_rtp_codecs *newtextrtp, char *red_fmtp, int *red_num_gen, int *red_data_pt, int *last_rtpmap_codec)
-{
- int found = FALSE;
- unsigned int codec;
- char mimeSubtype[128];
- unsigned int sample_rate;
- char *red_cp;
- int debug = sip_debug_test_pvt(p);
-
- if (sscanf(a, "rtpmap: %30u %127[^/]/%30u", &codec, mimeSubtype, &sample_rate) == 3) {
- /* We have a rtpmap to handle */
- if (*last_rtpmap_codec < SDP_MAX_RTPMAP_CODECS) {
- if (!strncasecmp(mimeSubtype, "T140", 4)) { /* Text */
- if (p->trtp) {
- /* ast_verbose("Adding t140 mimeSubtype to textrtp struct\n"); */
- ast_rtp_codecs_payloads_set_rtpmap_type_rate(newtextrtp, NULL, codec, "text", mimeSubtype, 0, sample_rate);
- found = TRUE;
- }
- } else if (!strncasecmp(mimeSubtype, "RED", 3)) { /* Text with Redudancy */
- if (p->trtp) {
- ast_rtp_codecs_payloads_set_rtpmap_type_rate(newtextrtp, NULL, codec, "text", mimeSubtype, 0, sample_rate);
- sprintf(red_fmtp, "fmtp:%u ", codec);
- if (debug)
- ast_verbose("RED submimetype has payload type: %u\n", codec);
- found = TRUE;
- }
- }
- } else {
- if (debug)
- ast_verbose("Discarded description format %s for ID %u\n", mimeSubtype, codec);
- }
- } else if (!strncmp(a, red_fmtp, strlen(red_fmtp))) {
- char *rest = NULL;
- /* count numbers of generations in fmtp */
- red_cp = &red_fmtp[strlen(red_fmtp)];
- strncpy(red_fmtp, a, 100);
-
- sscanf(red_cp, "%30u", (unsigned *)&red_data_pt[*red_num_gen]);
- red_cp = strtok_r(red_cp, "/", &rest);
- while (red_cp && (*red_num_gen)++ < AST_RED_MAX_GENERATION) {
- sscanf(red_cp, "%30u", (unsigned *)&red_data_pt[*red_num_gen]);
- red_cp = strtok_r(NULL, "/", &rest);
- }
- red_cp = red_fmtp;
- found = TRUE;
- }
-
- return found;
-}
-
-static int process_sdp_a_image(const char *a, struct sip_pvt *p)
-{
- int found = FALSE;
- char s[256];
- unsigned int x;
- char *attrib = ast_strdupa(a);
- char *pos;
-
- if (initialize_udptl(p)) {
- return found;
- }
-
- /* Due to a typo in an IANA registration of one of the T.38 attributes,
- * RFC5347 section 2.5.2 recommends that all T.38 attributes be parsed in
- * a case insensitive manner. Hence, the importance of proof reading (and
- * code reviews).
- */
- for (pos = attrib; *pos; ++pos) {
- *pos = tolower(*pos);
- }
-
- if ((sscanf(attrib, "t38faxmaxbuffer:%30u", &x) == 1)) {
- ast_debug(3, "MaxBufferSize:%u\n", x);
- found = TRUE;
- } else if ((sscanf(attrib, "t38maxbitrate:%30u", &x) == 1) || (sscanf(attrib, "t38faxmaxrate:%30u", &x) == 1)) {
- ast_debug(3, "T38MaxBitRate: %u\n", x);
- switch (x) {
- case 14400:
- p->t38.their_parms.rate = AST_T38_RATE_14400;
- break;
- case 12000:
- p->t38.their_parms.rate = AST_T38_RATE_12000;
- break;
- case 9600:
- p->t38.their_parms.rate = AST_T38_RATE_9600;
- break;
- case 7200:
- p->t38.their_parms.rate = AST_T38_RATE_7200;
- break;
- case 4800:
- p->t38.their_parms.rate = AST_T38_RATE_4800;
- break;
- case 2400:
- p->t38.their_parms.rate = AST_T38_RATE_2400;
- break;
- }
- found = TRUE;
- } else if ((sscanf(attrib, "t38faxversion:%30u", &x) == 1)) {
- ast_debug(3, "FaxVersion: %u\n", x);
- p->t38.their_parms.version = x;
- found = TRUE;
- } else if ((sscanf(attrib, "t38faxmaxdatagram:%30u", &x) == 1) || (sscanf(attrib, "t38maxdatagram:%30u", &x) == 1)) {
- /* override the supplied value if the configuration requests it */
- if (((signed int) p->t38_maxdatagram >= 0) && ((unsigned int) p->t38_maxdatagram > x)) {
- ast_debug(1, "Overriding T38FaxMaxDatagram '%u' with '%d'\n", x, p->t38_maxdatagram);
- x = p->t38_maxdatagram;
- }
- ast_debug(3, "FaxMaxDatagram: %u\n", x);
- ast_udptl_set_far_max_datagram(p->udptl, x);
- found = TRUE;
- } else if ((strncmp(attrib, "t38faxfillbitremoval", sizeof("t38faxfillbitremoval") - 1) == 0)) {
- if (sscanf(attrib, "t38faxfillbitremoval:%30u", &x) == 1) {
- ast_debug(3, "FillBitRemoval: %u\n", x);
- if (x == 1) {
- p->t38.their_parms.fill_bit_removal = TRUE;
- }
- } else {
- ast_debug(3, "FillBitRemoval\n");
- p->t38.their_parms.fill_bit_removal = TRUE;
- }
- found = TRUE;
- } else if ((strncmp(attrib, "t38faxtranscodingmmr", sizeof("t38faxtranscodingmmr") - 1) == 0)) {
- if (sscanf(attrib, "t38faxtranscodingmmr:%30u", &x) == 1) {
- ast_debug(3, "Transcoding MMR: %u\n", x);
- if (x == 1) {
- p->t38.their_parms.transcoding_mmr = TRUE;
- }
- } else {
- ast_debug(3, "Transcoding MMR\n");
- p->t38.their_parms.transcoding_mmr = TRUE;
- }
- found = TRUE;
- } else if ((strncmp(attrib, "t38faxtranscodingjbig", sizeof("t38faxtranscodingjbig") - 1) == 0)) {
- if (sscanf(attrib, "t38faxtranscodingjbig:%30u", &x) == 1) {
- ast_debug(3, "Transcoding JBIG: %u\n", x);
- if (x == 1) {
- p->t38.their_parms.transcoding_jbig = TRUE;
- }
- } else {
- ast_debug(3, "Transcoding JBIG\n");
- p->t38.their_parms.transcoding_jbig = TRUE;
- }
- found = TRUE;
- } else if ((sscanf(attrib, "t38faxratemanagement:%255s", s) == 1)) {
- ast_debug(3, "RateManagement: %s\n", s);
- if (!strcasecmp(s, "localTCF"))
- p->t38.their_parms.rate_management = AST_T38_RATE_MANAGEMENT_LOCAL_TCF;
- else if (!strcasecmp(s, "transferredTCF"))
- p->t38.their_parms.rate_management = AST_T38_RATE_MANAGEMENT_TRANSFERRED_TCF;
- found = TRUE;
- } else if ((sscanf(attrib, "t38faxudpec:%255s", s) == 1)) {
- ast_debug(3, "UDP EC: %s\n", s);
- if (!strcasecmp(s, "t38UDPRedundancy")) {
- ast_udptl_set_error_correction_scheme(p->udptl, UDPTL_ERROR_CORRECTION_REDUNDANCY);
- } else if (!strcasecmp(s, "t38UDPFEC")) {
- ast_udptl_set_error_correction_scheme(p->udptl, UDPTL_ERROR_CORRECTION_FEC);
- } else {
- ast_udptl_set_error_correction_scheme(p->udptl, UDPTL_ERROR_CORRECTION_NONE);
- }
- found = TRUE;
- }
-
- return found;
-}
-
-/*! \brief Add "Supported" header to sip message. Since some options may
- * be disabled in the config, the sip_pvt must be inspected to determine what
- * is supported for this dialog. */
-static int add_supported(struct sip_pvt *pvt, struct sip_request *req)
-{
- char supported_value[SIPBUFSIZE];
- int res;
-
- sprintf(supported_value, "replaces%s%s",
- (st_get_mode(pvt, 0) != SESSION_TIMER_MODE_REFUSE) ? ", timer" : "",
- ast_test_flag(&pvt->flags[0], SIP_USEPATH) ? ", path" : "");
- res = add_header(req, "Supported", supported_value);
-
- return res;
-}
-
-/*! \brief Add header to SIP message */
-static int add_header(struct sip_request *req, const char *var, const char *value)
-{
- if (req->headers == SIP_MAX_HEADERS) {
- ast_log(LOG_WARNING, "Out of SIP header space\n");
- return -1;
- }
-
- if (req->lines) {
- ast_log(LOG_WARNING, "Can't add more headers when lines have been added\n");
- return -1;
- }
-
- if (sip_cfg.compactheaders) {
- var = find_alias(var, var);
- }
-
- ast_str_append(&req->data, 0, "%s: %s\r\n", var, value);
- req->header[req->headers] = ast_str_strlen(req->data);
-
- req->headers++;
-
- return 0;
-}
-
-/*!
- * \pre dialog is assumed to be locked while calling this function
- * \brief Add 'Max-Forwards' header to SIP message
- */
-static int add_max_forwards(struct sip_pvt *dialog, struct sip_request *req)
-{
- char clen[10];
-
- snprintf(clen, sizeof(clen), "%d", dialog->maxforwards);
-
- return add_header(req, "Max-Forwards", clen);
-}
-
-/*! \brief Add 'Content-Length' header and content to SIP message */
-static int finalize_content(struct sip_request *req)
-{
- char clen[10];
-
- if (req->lines) {
- ast_log(LOG_WARNING, "finalize_content() called on a message that has already been finalized\n");
- return -1;
- }
-
- snprintf(clen, sizeof(clen), "%zu", ast_str_strlen(req->content));
- add_header(req, "Content-Length", clen);
-
- if (ast_str_strlen(req->content)) {
- ast_str_append(&req->data, 0, "\r\n%s", ast_str_buffer(req->content));
- }
- req->lines = ast_str_strlen(req->content) ? 1 : 0;
- return 0;
-}
-
-/*! \brief Add content (not header) to SIP message */
-static int add_content(struct sip_request *req, const char *line)
-{
- if (req->lines) {
- ast_log(LOG_WARNING, "Can't add more content when the content has been finalized\n");
- return -1;
- }
-
- ast_str_append(&req->content, 0, "%s", line);
- return 0;
-}
-
-/*! \brief Copy one header field from one request to another */
-static int copy_header(struct sip_request *req, const struct sip_request *orig, const char *field)
-{
- const char *tmp = sip_get_header(orig, field);
-
- if (!ast_strlen_zero(tmp)) /* Add what we're responding to */
- return add_header(req, field, tmp);
- ast_log(LOG_NOTICE, "No field '%s' present to copy\n", field);
- return -1;
-}
-
-/*! \brief Copy all headers from one request to another */
-static int copy_all_header(struct sip_request *req, const struct sip_request *orig, const char *field)
-{
- int start = 0;
- int copied = 0;
- for (;;) {
- const char *tmp = __get_header(orig, field, &start);
-
- if (ast_strlen_zero(tmp))
- break;
- /* Add what we're responding to */
- add_header(req, field, tmp);
- copied++;
- }
- return copied ? 0 : -1;
-}
-
-/*! \brief Copy SIP VIA Headers from the request to the response
-\note If the client indicates that it wishes to know the port we received from,
- it adds ;rport without an argument to the topmost via header. We need to
- add the port number (from our point of view) to that parameter.
-\verbatim
- We always add ;received= to the topmost via header.
-\endverbatim
- Received: RFC 3261, rport RFC 3581 */
-static int copy_via_headers(struct sip_pvt *p, struct sip_request *req, const struct sip_request *orig, const char *field)
-{
- int copied = 0;
- int start = 0;
-
- for (;;) {
- char new[512];
- const char *oh = __get_header(orig, field, &start);
-
- if (ast_strlen_zero(oh))
- break;
-
- if (!copied) { /* Only check for empty rport in topmost via header */
- char leftmost[512], *others, *rport;
-
- /* Only work on leftmost value */
- ast_copy_string(leftmost, oh, sizeof(leftmost));
- others = strchr(leftmost, ',');
- if (others)
- *others++ = '\0';
-
- /* Find ;rport; (empty request) */
- rport = strstr(leftmost, ";rport");
- if (rport && *(rport+6) == '=')
- rport = NULL; /* We already have a parameter to rport */
-
- if (((ast_test_flag(&p->flags[0], SIP_NAT_FORCE_RPORT)) || (rport && ast_test_flag(&p->flags[0], SIP_NAT_RPORT_PRESENT)))) {
- /* We need to add received port - rport */
- char *end;
-
- rport = strstr(leftmost, ";rport");
-
- if (rport) {
- end = strchr(rport + 1, ';');
- if (end)
- memmove(rport, end, strlen(end) + 1);
- else
- *rport = '\0';
- }
-
- /* Add rport to first VIA header if requested */
- snprintf(new, sizeof(new), "%s;received=%s;rport=%d%s%s",
- leftmost, ast_sockaddr_stringify_addr_remote(&p->recv),
- ast_sockaddr_port(&p->recv),
- others ? "," : "", others ? others : "");
- } else {
- /* We should *always* add a received to the topmost via */
- snprintf(new, sizeof(new), "%s;received=%s%s%s",
- leftmost, ast_sockaddr_stringify_addr_remote(&p->recv),
- others ? "," : "", others ? others : "");
- }
- oh = new; /* the header to copy */
- } /* else add the following via headers untouched */
- add_header(req, field, oh);
- copied++;
- }
- if (!copied) {
- ast_log(LOG_NOTICE, "No header field '%s' present to copy\n", field);
- return -1;
- }
- return 0;
-}
-
-/*! \brief Add route header into request per learned route */
-static void add_route(struct sip_request *req, struct sip_route *route, int skip)
-{
- struct ast_str *r;
-
- if (sip_route_empty(route)) {
- return;
- }
-
- if ((r = sip_route_list(route, 0, skip))) {
- if (ast_str_strlen(r)) {
- add_header(req, "Route", ast_str_buffer(r));
- }
- ast_free(r);
- }
-}
-
-/*! \brief Set destination from SIP URI
- *
- * Parse uri to h (host) and port - uri is already just the part inside the <>
- * general form we are expecting is \verbatim sip[s]:username[:password][;parameter]@host[:port][;...] \endverbatim
- * If there's a port given, turn NAPTR/SRV off. NAPTR might indicate SIPS preference even
- * for SIP: uri's
- *
- * If there's a sips: uri scheme, TLS will be required.
- */
-static void set_destination(struct sip_pvt *p, const char *uri)
-{
- char *trans, *maddr, hostname[256];
- const char *h;
- int hn;
- int debug=sip_debug_test_pvt(p);
- int tls_on = FALSE;
-
- if (debug)
- ast_verbose("set_destination: Parsing <%s> for address/port to send to\n", uri);
-
- if ((trans = strcasestr(uri, ";transport="))) {
- trans += strlen(";transport=");
-
- if (!strncasecmp(trans, "ws", 2)) {
- if (debug)
- ast_verbose("set_destination: URI is for WebSocket, we can't set destination\n");
- return;
- }
- }
-
- /* Find and parse hostname */
- h = strchr(uri, '@');
- if (h)
- ++h;
- else {
- h = uri;
- if (!strncasecmp(h, "sip:", 4)) {
- h += 4;
- } else if (!strncasecmp(h, "sips:", 5)) {
- h += 5;
- tls_on = TRUE;
- }
- }
- hn = strcspn(h, ";>") + 1;
- if (hn > sizeof(hostname))
- hn = sizeof(hostname);
- ast_copy_string(hostname, h, hn);
- /* XXX bug here if string has been trimmed to sizeof(hostname) */
- h += hn - 1;
-
- /*! \todo XXX If we have sip_cfg.srvlookup on, then look for NAPTR/SRV,
- * otherwise, just look for A records */
- if (ast_sockaddr_resolve_first_transport(&p->sa, hostname, 0, p->socket.type)) {
- ast_log(LOG_WARNING, "Can't find address for host '%s'\n", hostname);
- return;
- }
-
- /* Got the hostname - but maybe there's a "maddr=" to override address? */
- maddr = strstr(h, "maddr=");
- if (maddr) {
- int port;
-
- maddr += 6;
- hn = strspn(maddr, "abcdefghijklmnopqrstuvwxyzABCDEFGHIJKLMNOPQRSTUVWXYZ"
- "0123456789-.:[]") + 1;
- if (hn > sizeof(hostname))
- hn = sizeof(hostname);
- ast_copy_string(hostname, maddr, hn);
-
- port = ast_sockaddr_port(&p->sa);
-
- /*! \todo XXX If we have sip_cfg.srvlookup on, then look for
- * NAPTR/SRV, otherwise, just look for A records */
- if (ast_sockaddr_resolve_first_transport(&p->sa, hostname, PARSE_PORT_FORBID, p->socket.type)) {
- ast_log(LOG_WARNING, "Can't find address for host '%s'\n", hostname);
- return;
- }
-
- ast_sockaddr_set_port(&p->sa, port);
- }
-
- if (!ast_sockaddr_port(&p->sa)) {
- ast_sockaddr_set_port(&p->sa, tls_on ?
- STANDARD_TLS_PORT : STANDARD_SIP_PORT);
- }
-
- if (debug) {
- ast_verbose("set_destination: set destination to %s\n",
- ast_sockaddr_stringify(&p->sa));
- }
-}
-
-/*! \brief Initialize SIP response, based on SIP request */
-static int init_resp(struct sip_request *resp, const char *msg)
-{
- /* Initialize a response */
- memset(resp, 0, sizeof(*resp));
- resp->method = SIP_RESPONSE;
- if (!(resp->data = ast_str_create(SIP_MIN_PACKET)))
- goto e_return;
- if (!(resp->content = ast_str_create(SIP_MIN_PACKET)))
- goto e_free_data;
- resp->header[0] = 0;
- ast_str_set(&resp->data, 0, "SIP/2.0 %s\r\n", msg);
- resp->headers++;
- return 0;
-
-e_free_data:
- ast_free(resp->data);
- resp->data = NULL;
-e_return:
- return -1;
-}
-
-/*! \brief Initialize SIP request */
-static int init_req(struct sip_request *req, int sipmethod, const char *recip)
-{
- /* Initialize a request */
- memset(req, 0, sizeof(*req));
- if (!(req->data = ast_str_create(SIP_MIN_PACKET)))
- goto e_return;
- if (!(req->content = ast_str_create(SIP_MIN_PACKET)))
- goto e_free_data;
- req->method = sipmethod;
- req->header[0] = 0;
- ast_str_set(&req->data, 0, "%s %s SIP/2.0\r\n", sip_methods[sipmethod].text, recip);
- req->headers++;
- return 0;
-
-e_free_data:
- ast_free(req->data);
- req->data = NULL;
-e_return:
- return -1;
-}
-
-/*! \brief Deinitialize SIP response/request */
-static void deinit_req(struct sip_request *req)
-{
- if (req->data) {
- ast_free(req->data);
- req->data = NULL;
- }
- if (req->content) {
- ast_free(req->content);
- req->content = NULL;
- }
-}
-
-
-/*! \brief Test if this response needs a contact header */
-static inline int resp_needs_contact(const char *msg, enum sipmethod method) {
- /* Requirements for Contact header inclusion in responses generated
- * from the header tables found in the following RFCs. Where the
- * Contact header was marked mandatory (m) or optional (o) this
- * function returns 1.
- *
- * - RFC 3261 (ACK, BYE, CANCEL, INVITE, OPTIONS, REGISTER)
- * - RFC 2976 (INFO)
- * - RFC 3262 (PRACK)
- * - RFC 3265 (SUBSCRIBE, NOTIFY)
- * - RFC 3311 (UPDATE)
- * - RFC 3428 (MESSAGE)
- * - RFC 3515 (REFER)
- * - RFC 3903 (PUBLISH)
- */
-
- switch (method) {
- /* 1xx, 2xx, 3xx, 485 */
- case SIP_INVITE:
- case SIP_UPDATE:
- case SIP_SUBSCRIBE:
- case SIP_NOTIFY:
- if ((msg[0] >= '1' && msg[0] <= '3') || !strncmp(msg, "485", 3))
- return 1;
- break;
-
- /* 2xx, 3xx, 485 */
- case SIP_REGISTER:
- case SIP_OPTIONS:
- if (msg[0] == '2' || msg[0] == '3' || !strncmp(msg, "485", 3))
- return 1;
- break;
-
- /* 3xx, 485 */
- case SIP_BYE:
- case SIP_PRACK:
- case SIP_MESSAGE:
- case SIP_PUBLISH:
- if (msg[0] == '3' || !strncmp(msg, "485", 3))
- return 1;
- break;
-
- /* 2xx, 3xx, 4xx, 5xx, 6xx */
- case SIP_REFER:
- if (msg[0] >= '2' && msg[0] <= '6')
- return 1;
- break;
-
- /* contact will not be included for everything else */
- case SIP_ACK:
- case SIP_CANCEL:
- case SIP_INFO:
- case SIP_PING:
- default:
- return 0;
- }
- return 0;
-}
-
-/*! \brief Prepare SIP response packet */
-static int respprep(struct sip_request *resp, struct sip_pvt *p, const char *msg, const struct sip_request *req)
-{
- char newto[256];
- const char *ot;
-
- init_resp(resp, msg);
- copy_via_headers(p, resp, req, "Via");
- if (msg[0] == '1' || msg[0] == '2')
- copy_all_header(resp, req, "Record-Route");
- copy_header(resp, req, "From");
- ot = sip_get_header(req, "To");
- if (!strcasestr(ot, "tag=") && strncmp(msg, "100", 3)) {
- /* Add the proper tag if we don't have it already. If they have specified
- their tag, use it. Otherwise, use our own tag */
- if (!ast_strlen_zero(p->theirtag) && ast_test_flag(&p->flags[0], SIP_OUTGOING))
- snprintf(newto, sizeof(newto), "%s;tag=%s", ot, p->theirtag);
- else if (p->tag && !ast_test_flag(&p->flags[0], SIP_OUTGOING))
- snprintf(newto, sizeof(newto), "%s;tag=%s", ot, p->tag);
- else
- ast_copy_string(newto, ot, sizeof(newto));
- ot = newto;
- }
- add_header(resp, "To", ot);
- copy_header(resp, req, "Call-ID");
- copy_header(resp, req, "CSeq");
- if (!ast_strlen_zero(global_useragent))
- add_header(resp, "Server", global_useragent);
- add_header(resp, "Allow", ALLOWED_METHODS);
- add_supported(p, resp);
-
- /* If this is an invite, add Session-Timers related headers if the feature is active for this session */
- if (p->method == SIP_INVITE && p->stimer && p->stimer->st_active == TRUE) {
- char se_hdr[256];
- snprintf(se_hdr, sizeof(se_hdr), "%d;refresher=%s", p->stimer->st_interval,
- p->stimer->st_ref == SESSION_TIMER_REFRESHER_US ? "uas" : "uac");
- add_header(resp, "Session-Expires", se_hdr);
- /* RFC 2048, Section 9
- * If the refresher parameter in the Session-Expires header field in the
- * 2xx response has a value of 'uac', the UAS MUST place a Require
- * header field into the response with the value 'timer'.
- * ...
- * If the refresher parameter in
- * the 2xx response has a value of 'uas' and the Supported header field
- * in the request contained the value 'timer', the UAS SHOULD place a
- * Require header field into the response with the value 'timer'
- */
- if (p->stimer->st_ref == SESSION_TIMER_REFRESHER_THEM ||
- (p->stimer->st_ref == SESSION_TIMER_REFRESHER_US &&
- p->stimer->st_active_peer_ua == TRUE)) {
- resp->reqsipoptions |= SIP_OPT_TIMER;
- }
- }
-
- if (msg[0] == '2' && (p->method == SIP_SUBSCRIBE || p->method == SIP_REGISTER || p->method == SIP_PUBLISH)) {
- /* For registration responses, we also need expiry and
- contact info */
- add_expires(resp, p->expiry);
- if (p->expiry) { /* Only add contact if we have an expiry time */
- char contact[SIPBUFSIZE];
- const char *contact_uri = p->method == SIP_SUBSCRIBE ? p->our_contact : p->fullcontact;
- char *brackets = strchr(contact_uri, '<');
- snprintf(contact, sizeof(contact), "%s%s%s;expires=%d", brackets ? "" : "<", contact_uri, brackets ? "" : ">", p->expiry);
- add_header(resp, "Contact", contact); /* Not when we unregister */
- }
- if (p->method == SIP_REGISTER && ast_test_flag(&p->flags[0], SIP_USEPATH)) {
- copy_header(resp, req, "Path");
- }
- } else if (!ast_strlen_zero(p->our_contact) && resp_needs_contact(msg, p->method)) {
- add_header(resp, "Contact", p->our_contact);
- }
-
- if (!ast_strlen_zero(p->url)) {
- add_header(resp, "Access-URL", p->url);
- ast_string_field_set(p, url, NULL);
- }
-
- /* default to routing the response to the address where the request
- * came from. Since we don't have a transport layer, we do this here.
- * The process_via() function will update the port to either the port
- * specified in the via header or the default port later on (per RFC
- * 3261 section 18.2.2).
- */
- p->sa = p->recv;
-
- if (process_via(p, req)) {
- ast_log(LOG_WARNING, "error processing via header, will send response to originating address\n");
- }
-
- return 0;
-}
-
-/*! \brief Initialize a SIP request message (not the initial one in a dialog) */
-static int reqprep(struct sip_request *req, struct sip_pvt *p, int sipmethod, uint32_t seqno, int newbranch)
-{
- struct sip_request *orig = &p->initreq;
- char stripped[80];
- char tmp[80];
- char newto[256];
- const char *c;
- const char *ot, *of;
- int is_strict = FALSE; /*!< Strict routing flag */
- int is_outbound = ast_test_flag(&p->flags[0], SIP_OUTGOING); /* Session direction */
-
- snprintf(p->lastmsg, sizeof(p->lastmsg), "Tx: %s", sip_methods[sipmethod].text);
-
- if (!seqno) {
- p->ocseq++;
- seqno = p->ocseq;
- }
-
- /* A CANCEL must have the same branch as the INVITE that it is canceling. */
- if (sipmethod == SIP_CANCEL) {
- p->branch = p->invite_branch;
- build_via(p);
- } else if (newbranch && (sipmethod == SIP_INVITE)) {
- p->branch ^= ast_random();
- p->invite_branch = p->branch;
- build_via(p);
- } else if (newbranch) {
- p->branch ^= ast_random();
- build_via(p);
- }
-
- /* Check for strict or loose router */
- if (sip_route_is_strict(&p->route)) {
- is_strict = TRUE;
- if (sipdebug)
- ast_debug(1, "Strict routing enforced for session %s\n", p->callid);
- }
-
- if (sipmethod == SIP_CANCEL) {
- c = REQ_OFFSET_TO_STR(&p->initreq, rlpart2); /* Use original URI */
- } else if (sipmethod == SIP_ACK) {
- /* Use URI from Contact: in 200 OK (if INVITE)
- (we only have the contacturi on INVITEs) */
- if (!ast_strlen_zero(p->okcontacturi)) {
- c = is_strict ? sip_route_first_uri(&p->route) : p->okcontacturi;
- } else {
- c = REQ_OFFSET_TO_STR(&p->initreq, rlpart2);
- }
- } else if (!ast_strlen_zero(p->okcontacturi)) {
- /* Use for BYE or REINVITE */
- c = is_strict ? sip_route_first_uri(&p->route) : p->okcontacturi;
- } else if (!ast_strlen_zero(p->uri)) {
- c = p->uri;
- } else {
- char *n;
- /* We have no URI, use To: or From: header as URI (depending on direction) */
- ast_copy_string(stripped, sip_get_header(orig, is_outbound ? "To" : "From"),
- sizeof(stripped));
- n = get_in_brackets(stripped);
- c = remove_uri_parameters(n);
- }
- init_req(req, sipmethod, c);
-
- snprintf(tmp, sizeof(tmp), "%u %s", seqno, sip_methods[sipmethod].text);
-
- add_header(req, "Via", p->via);
- /*
- * Use the learned route set unless this is a CANCEL or an ACK for a non-2xx
- * final response. For a CANCEL or ACK, we have to send to the same destination
- * as the original INVITE.
- * Send UPDATE to the same destination as CANCEL, if call is not in final state.
- */
- if (!sip_route_empty(&p->route) &&
- !(sipmethod == SIP_CANCEL ||
- (sipmethod == SIP_ACK && (p->invitestate == INV_COMPLETED || p->invitestate == INV_CANCELLED)))) {
- if (p->socket.type != AST_TRANSPORT_UDP && p->socket.tcptls_session) {
- /* For TCP/TLS sockets that are connected we won't need
- * to do any hostname/IP lookups */
- } else if (ast_test_flag(&p->flags[0], SIP_NAT_FORCE_RPORT)) {
- /* For NATed traffic, we ignore the contact/route and
- * simply send to the received-from address. No need
- * for lookups. */
- } else if (sipmethod == SIP_UPDATE && (p->invitestate == INV_PROCEEDING || p->invitestate == INV_EARLY_MEDIA)) {
- /* Calling set_destination for an UPDATE in early dialog
- * will result in mangling of the target for a subsequent
- * CANCEL according to ASTERISK-24628 so do not do it.
- */
- } else {
- set_destination(p, sip_route_first_uri(&p->route));
- }
- add_route(req, &p->route, is_strict ? 1 : 0);
- }
- add_max_forwards(p, req);
-
- ot = sip_get_header(orig, "To");
- of = sip_get_header(orig, "From");
-
- /* Add tag *unless* this is a CANCEL, in which case we need to send it exactly
- as our original request, including tag (or presumably lack thereof) */
- if (!strcasestr(ot, "tag=") && sipmethod != SIP_CANCEL) {
- /* Add the proper tag if we don't have it already. If they have specified
- their tag, use it. Otherwise, use our own tag */
- if (is_outbound && !ast_strlen_zero(p->theirtag))
- snprintf(newto, sizeof(newto), "%s;tag=%s", ot, p->theirtag);
- else if (!is_outbound)
- snprintf(newto, sizeof(newto), "%s;tag=%s", ot, p->tag);
- else
- snprintf(newto, sizeof(newto), "%s", ot);
- ot = newto;
- }
-
- if (is_outbound) {
- add_header(req, "From", of);
- add_header(req, "To", ot);
- } else {
- add_header(req, "From", ot);
- add_header(req, "To", of);
- }
- /* Do not add Contact for MESSAGE, BYE and Cancel requests */
- if (sipmethod != SIP_BYE && sipmethod != SIP_CANCEL && sipmethod != SIP_MESSAGE)
- add_header(req, "Contact", p->our_contact);
-
- copy_header(req, orig, "Call-ID");
- add_header(req, "CSeq", tmp);
-
- if (!ast_strlen_zero(global_useragent))
- add_header(req, "User-Agent", global_useragent);
-
- if (!ast_strlen_zero(p->url)) {
- add_header(req, "Access-URL", p->url);
- ast_string_field_set(p, url, NULL);
- }
-
- /* Add Session-Timers related headers if the feature is active for this session.
- An exception to this behavior is the ACK request. Since Asterisk never requires
- session-timers support from a remote end-point (UAS) in an INVITE, it must
- not send 'Require: timer' header in the ACK request.
- */
- if (p->stimer && p->stimer->st_active == TRUE && p->stimer->st_active_peer_ua == TRUE
- && (sipmethod == SIP_INVITE || sipmethod == SIP_UPDATE)) {
- char se_hdr[256];
- snprintf(se_hdr, sizeof(se_hdr), "%d;refresher=%s", p->stimer->st_interval,
- p->stimer->st_ref == SESSION_TIMER_REFRESHER_US ? "uac" : "uas");
- add_header(req, "Session-Expires", se_hdr);
- snprintf(se_hdr, sizeof(se_hdr), "%d", st_get_se(p, FALSE));
- add_header(req, "Min-SE", se_hdr);
- }
-
- return 0;
-}
-
-/*! \brief Base transmit response function */
-static int __transmit_response(struct sip_pvt *p, const char *msg, const struct sip_request *req, enum xmittype reliable)
-{
- struct sip_request resp;
- uint32_t seqno = 0;
-
- if (reliable && (sscanf(sip_get_header(req, "CSeq"), "%30u ", &seqno) != 1)) {
- ast_log(LOG_WARNING, "Unable to determine sequence number from '%s'\n", sip_get_header(req, "CSeq"));
- return -1;
- }
- respprep(&resp, p, msg, req);
-
- if (ast_test_flag(&p->flags[0], SIP_SENDRPID)
- && ast_test_flag(&p->flags[1], SIP_PAGE2_CONNECTLINEUPDATE_PEND)
- && (!strncmp(msg, "180", 3) || !strncmp(msg, "183", 3))) {
- ast_clear_flag(&p->flags[1], SIP_PAGE2_CONNECTLINEUPDATE_PEND);
- add_rpid(&resp, p);
- }
- if (ast_test_flag(&p->flags[0], SIP_OFFER_CC)) {
- add_cc_call_info_to_response(p, &resp);
- }
-
- /* If we are sending a 302 Redirect we can add a diversion header if the redirect information is set */
- if (!strncmp(msg, "302", 3)) {
- add_diversion(&resp, p);
- }
-
- /* If we are cancelling an incoming invite for some reason, add information
- about the reason why we are doing this in clear text */
- if (p->method == SIP_INVITE && msg[0] != '1') {
- char buf[20];
-
- if (ast_test_flag(&p->flags[1], SIP_PAGE2_Q850_REASON)) {
- int hangupcause = 0;
-
- if (p->owner && ast_channel_hangupcause(p->owner)) {
- hangupcause = ast_channel_hangupcause(p->owner);
- } else if (p->hangupcause) {
- hangupcause = p->hangupcause;
- } else {
- int respcode;
- if (sscanf(msg, "%30d ", &respcode))
- hangupcause = hangup_sip2cause(respcode);
- }
-
- if (hangupcause) {
- sprintf(buf, "Q.850;cause=%i", hangupcause & 0x7f);
- add_header(&resp, "Reason", buf);
- }
- }
-
- if (p->owner && ast_channel_hangupcause(p->owner)) {
- add_header(&resp, "X-Asterisk-HangupCause", ast_cause2str(ast_channel_hangupcause(p->owner)));
- snprintf(buf, sizeof(buf), "%d", ast_channel_hangupcause(p->owner));
- add_header(&resp, "X-Asterisk-HangupCauseCode", buf);
- }
- }
- return send_response(p, &resp, reliable, seqno);
-}
-
-static int transmit_response_with_sip_etag(struct sip_pvt *p, const char *msg, const struct sip_request *req, struct sip_esc_entry *esc_entry, int need_new_etag)
-{
- struct sip_request resp;
-
- if (need_new_etag) {
- create_new_sip_etag(esc_entry, 1);
- }
- respprep(&resp, p, msg, req);
- add_header(&resp, "SIP-ETag", esc_entry->entity_tag);
-
- return send_response(p, &resp, 0, 0);
-}
-
-static int temp_pvt_init(void *data)
-{
- struct sip_pvt *p = data;
-
- p->do_history = 0; /* XXX do we need it ? isn't already all 0 ? */
- return ast_string_field_init(p, 512);
-}
-
-static void temp_pvt_cleanup(void *data)
-{
- struct sip_pvt *p = data;
-
- ast_string_field_free_memory(p);
-
- ast_free(data);
-}
-
-/*! \brief Transmit response, no retransmits, using a temporary pvt structure */
-static int transmit_response_using_temp(ast_string_field callid, struct ast_sockaddr *addr, int useglobal_nat, const int intended_method, const struct sip_request *req, const char *msg)
-{
- struct sip_pvt *p = NULL;
-
- if (!(p = ast_threadstorage_get(&ts_temp_pvt, sizeof(*p)))) {
- ast_log(LOG_ERROR, "Failed to get temporary pvt\n");
- return -1;
- }
-
- /* XXX the structure may be dirty from previous usage.
- * Here we should state clearly how we should reinitialize it
- * before using it.
- * E.g. certainly the threadstorage should be left alone,
- * but other thihngs such as flags etc. maybe need cleanup ?
- */
-
- /* Initialize the bare minimum */
- p->method = intended_method;
-
- if (!addr) {
- ast_sockaddr_copy(&p->ourip, &internip);
- } else {
- ast_sockaddr_copy(&p->sa, addr);
- ast_sip_ouraddrfor(&p->sa, &p->ourip, p);
- }
-
- p->branch = ast_random();
- make_our_tag(p);
- p->ocseq = INITIAL_CSEQ;
-
- if (useglobal_nat && addr) {
- ast_copy_flags(&p->flags[0], &global_flags[0], SIP_NAT_FORCE_RPORT);
- ast_copy_flags(&p->flags[2], &global_flags[2], SIP_PAGE3_NAT_AUTO_RPORT);
- ast_sockaddr_copy(&p->recv, addr);
- check_via(p, req);
- }
-
- ast_string_field_set(p, fromdomain, default_fromdomain);
- p->fromdomainport = default_fromdomainport;
- build_via(p);
- ast_string_field_set(p, callid, callid);
-
- copy_socket_data(&p->socket, &req->socket);
-
- /* Use this temporary pvt structure to send the message */
- __transmit_response(p, msg, req, XMIT_UNRELIABLE);
-
- /* Free the string fields, but not the pool space */
- ast_string_field_init(p, 0);
-
- return 0;
-}
-
-/*! \brief Transmit response, no retransmits */
-static int transmit_response(struct sip_pvt *p, const char *msg, const struct sip_request *req)
-{
- return __transmit_response(p, msg, req, XMIT_UNRELIABLE);
-}
-
-/*! \brief Transmit response, no retransmits */
-static int transmit_response_with_unsupported(struct sip_pvt *p, const char *msg, const struct sip_request *req, const char *unsupported)
-{
- struct sip_request resp;
- respprep(&resp, p, msg, req);
- add_date(&resp);
- add_header(&resp, "Unsupported", unsupported);
- return send_response(p, &resp, XMIT_UNRELIABLE, 0);
-}
-
-/*! \brief Transmit 422 response with Min-SE header (Session-Timers) */
-static int transmit_response_with_minse(struct sip_pvt *p, const char *msg, const struct sip_request *req, int minse_int)
-{
- struct sip_request resp;
- char minse_str[20];
-
- respprep(&resp, p, msg, req);
- add_date(&resp);
-
- snprintf(minse_str, sizeof(minse_str), "%d", minse_int);
- add_header(&resp, "Min-SE", minse_str);
- return send_response(p, &resp, XMIT_UNRELIABLE, 0);
-}
-
-
-/*! \brief Transmit response, Make sure you get an ACK
- This is only used for responses to INVITEs, where we need to make sure we get an ACK
-*/
-static int transmit_response_reliable(struct sip_pvt *p, const char *msg, const struct sip_request *req)
-{
- return __transmit_response(p, msg, req, req->ignore ? XMIT_UNRELIABLE : XMIT_CRITICAL);
-}
-
-/*! \brief Add date header to SIP message */
-static void add_date(struct sip_request *req)
-{
- char tmp[256];
- struct tm tm;
- time_t t = time(NULL);
-
- gmtime_r(&t, &tm);
- strftime(tmp, sizeof(tmp), "%a, %d %b %Y %T GMT", &tm);
- add_header(req, "Date", tmp);
-}
-
-/*! \brief Add Expires header to SIP message */
-static void add_expires(struct sip_request *req, int expires)
-{
- char tmp[32];
-
- snprintf(tmp, sizeof(tmp), "%d", expires);
- add_header(req, "Expires", tmp);
-}
-
-/*! \brief Append Retry-After header field when transmitting response */
-static int transmit_response_with_retry_after(struct sip_pvt *p, const char *msg, const struct sip_request *req, const char *seconds)
-{
- struct sip_request resp;
- respprep(&resp, p, msg, req);
- add_header(&resp, "Retry-After", seconds);
- return send_response(p, &resp, XMIT_UNRELIABLE, 0);
-}
-
-/*! \brief Add date before transmitting response */
-static int transmit_response_with_date(struct sip_pvt *p, const char *msg, const struct sip_request *req)
-{
- struct sip_request resp;
- respprep(&resp, p, msg, req);
- add_date(&resp);
- return send_response(p, &resp, XMIT_UNRELIABLE, 0);
-}
-
-/*! \brief Append Accept header, content length before transmitting response */
-static int transmit_response_with_allow(struct sip_pvt *p, const char *msg, const struct sip_request *req, enum xmittype reliable)
-{
- struct sip_request resp;
- respprep(&resp, p, msg, req);
- add_header(&resp, "Accept", "application/sdp");
- return send_response(p, &resp, reliable, 0);
-}
-
-/*! \brief Append Min-Expires header, content length before transmitting response */
-static int transmit_response_with_minexpires(struct sip_pvt *p, const char *msg, const struct sip_request *req, int minexpires)
-{
- struct sip_request resp;
- char tmp[32];
-
- snprintf(tmp, sizeof(tmp), "%d", minexpires);
- respprep(&resp, p, msg, req);
- add_header(&resp, "Min-Expires", tmp);
- return send_response(p, &resp, XMIT_UNRELIABLE, 0);
-}
-
-/*! \brief Respond with authorization request */
-static int transmit_response_with_auth(struct sip_pvt *p, const char *msg, const struct sip_request *req, const char *nonce, enum xmittype reliable, const char *header, int stale)
-{
- struct sip_request resp;
- char tmp[512];
- uint32_t seqno = 0;
-
- if (reliable && (sscanf(sip_get_header(req, "CSeq"), "%30u ", &seqno) != 1)) {
- ast_log(LOG_WARNING, "Unable to determine sequence number from '%s'\n", sip_get_header(req, "CSeq"));
- return -1;
- }
- /* Choose Realm */
- get_realm(p, req);
-
- /* Stale means that they sent us correct authentication, but
- based it on an old challenge (nonce) */
- snprintf(tmp, sizeof(tmp), "Digest algorithm=MD5, realm=\"%s\", nonce=\"%s\"%s", p->realm, nonce, stale ? ", stale=true" : "");
- respprep(&resp, p, msg, req);
- add_header(&resp, header, tmp);
- append_history(p, "AuthChal", "Auth challenge sent for %s - nc %d", p->username, p->noncecount);
- return send_response(p, &resp, reliable, seqno);
-}
-
-/*!
- \brief Extract domain from SIP To/From header
- \retval -1 on error.
- \retval 1 if domain string is empty.
- \retval 0 if domain was properly extracted.
- \note TODO: Such code is all over SIP channel, there is a sense to organize
- this patern in one function
-*/
-static int get_domain(const char *str, char *domain, int len)
-{
- char tmpf[256];
- char *a, *from;
-
- *domain = '\0';
- ast_copy_string(tmpf, str, sizeof(tmpf));
- from = get_in_brackets(tmpf);
- if (!ast_strlen_zero(from)) {
- if (strncasecmp(from, "sip:", 4)) {
- ast_log(LOG_WARNING, "Huh? Not a SIP header (%s)?\n", from);
- return -1;
- }
- from += 4;
- } else
- from = NULL;
-
- if (from) {
- int bracket = 0;
-
- /* Strip any params or options from user */
- if ((a = strchr(from, ';')))
- *a = '\0';
- /* Strip port from domain if present */
- for (a = from; *a != '\0'; ++a) {
- if (*a == ':' && bracket == 0) {
- *a = '\0';
- break;
- } else if (*a == '[') {
- ++bracket;
- } else if (*a == ']') {
- --bracket;
- }
- }
- if ((a = strchr(from, '@'))) {
- *a = '\0';
- ast_copy_string(domain, a + 1, len);
- } else
- ast_copy_string(domain, from, len);
- }
-
- return ast_strlen_zero(domain);
-}
-
-/*!
- \brief Choose realm based on From header and then To header or use globally configured realm.
- Realm from From/To header should be listed among served domains in config file: domain=...
-*/
-static void get_realm(struct sip_pvt *p, const struct sip_request *req)
-{
- char domain[MAXHOSTNAMELEN];
-
- if (!ast_strlen_zero(p->realm))
- return;
-
- if (sip_cfg.domainsasrealm &&
- !AST_LIST_EMPTY(&domain_list))
- {
- /* Check From header first */
- if (!get_domain(sip_get_header(req, "From"), domain, sizeof(domain))) {
- if (check_sip_domain(domain, NULL, 0)) {
- ast_string_field_set(p, realm, domain);
- return;
- }
- }
- /* Check To header */
- if (!get_domain(sip_get_header(req, "To"), domain, sizeof(domain))) {
- if (check_sip_domain(domain, NULL, 0)) {
- ast_string_field_set(p, realm, domain);
- return;
- }
- }
- }
-
- /* Use default realm from config file */
- ast_string_field_set(p, realm, sip_cfg.realm);
-}
-
-/*!
- * \internal
- *
- * \arg msg Only use a string constant for the msg, here, it is shallow copied
- *
- * \note assumes the sip_pvt is locked.
- */
-static int transmit_provisional_response(struct sip_pvt *p, const char *msg, const struct sip_request *req, int with_sdp)
-{
- int res;
-
- if (!(res = with_sdp ? transmit_response_with_sdp(p, msg, req, XMIT_UNRELIABLE, FALSE, FALSE) : transmit_response(p, msg, req))) {
- p->last_provisional = msg;
- update_provisional_keepalive(p, with_sdp);
- }
-
- return res;
-}
-
-/*!
- * \internal
- * \brief Destroy all additional MESSAGE headers.
- *
- * \param pvt SIP private dialog struct.
- */
-static void destroy_msg_headers(struct sip_pvt *pvt)
-{
- struct sip_msg_hdr *doomed;
-
- while ((doomed = AST_LIST_REMOVE_HEAD(&pvt->msg_headers, next))) {
- ast_free(doomed);
- }
-}
-
-/*!
- * \internal
- * \brief Add a MESSAGE header to the dialog.
- *
- * \param pvt SIP private dialog struct.
- * \param hdr_name Name of header for MESSAGE.
- * \param hdr_value Value of header for MESSAGE.
- */
-static void add_msg_header(struct sip_pvt *pvt, const char *hdr_name, const char *hdr_value)
-{
- size_t hdr_len_name;
- size_t hdr_len_value;
- struct sip_msg_hdr *node;
- char *pos;
-
- hdr_len_name = strlen(hdr_name) + 1;
- hdr_len_value = strlen(hdr_value) + 1;
-
- node = ast_calloc(1, sizeof(*node) + hdr_len_name + hdr_len_value);
- if (!node) {
- return;
- }
- pos = node->stuff;
- node->name = pos;
- strcpy(pos, hdr_name);
- pos += hdr_len_name;
- node->value = pos;
- strcpy(pos, hdr_value);
-
- AST_LIST_INSERT_TAIL(&pvt->msg_headers, node, next);
-}
-
-/*! \brief Add text body to SIP message */
-static int add_text(struct sip_request *req, struct sip_pvt *p)
-{
- const char *content_type = NULL;
- struct sip_msg_hdr *node;
-
- /* Add any additional MESSAGE headers. */
- AST_LIST_TRAVERSE(&p->msg_headers, node, next) {
- if (!strcasecmp(node->name, "Content-Type")) {
- /* Save content type */
- content_type = node->value;
- } else {
- add_header(req, node->name, node->value);
- }
- }
- if (ast_strlen_zero(content_type)) {
- /* "Content-Type" not set - use default value */
- content_type = "text/plain;charset=UTF-8";
- }
- add_header(req, "Content-Type", content_type);
-
- /* XXX Convert \n's to \r\n's XXX */
- add_content(req, p->msg_body);
- return 0;
-}
-
-/*! \brief Add DTMF INFO tone to sip message
- Mode = 0 for application/dtmf-relay (Cisco)
- 1 for application/dtmf
-*/
-static int add_digit(struct sip_request *req, char digit, unsigned int duration, int mode)
-{
- char tmp[256];
- int event;
- if (mode) {
- /* Application/dtmf short version used by some implementations */
- if ('0' <= digit && digit <= '9') {
- event = digit - '0';
- } else if (digit == '*') {
- event = 10;
- } else if (digit == '#') {
- event = 11;
- } else if ('A' <= digit && digit <= 'D') {
- event = 12 + digit - 'A';
- } else if ('a' <= digit && digit <= 'd') {
- event = 12 + digit - 'a';
- } else {
- /* Unknown digit */
- event = 0;
- }
- snprintf(tmp, sizeof(tmp), "%d\r\n", event);
- add_header(req, "Content-Type", "application/dtmf");
- add_content(req, tmp);
- } else {
- /* Application/dtmf-relay as documented by Cisco */
- snprintf(tmp, sizeof(tmp), "Signal=%c\r\nDuration=%u\r\n", digit, duration);
- add_header(req, "Content-Type", "application/dtmf-relay");
- add_content(req, tmp);
- }
- return 0;
-}
-
-/*!
- * \pre if p->owner exists, it must be locked
- * \brief Add Remote-Party-ID header to SIP message
- */
-static int add_rpid(struct sip_request *req, struct sip_pvt *p)
-{
- struct ast_str *tmp = ast_str_alloca(256);
- char tmp2[256];
- char lid_name_buf[128];
- char *lid_num;
- char *lid_name;
- int lid_pres;
- const char *fromdomain;
- const char *privacy = NULL;
- const char *screen = NULL;
- struct ast_party_id connected_id;
- const char *anonymous_string = "\"Anonymous\" ";
-
- if (!ast_test_flag(&p->flags[0], SIP_SENDRPID)) {
- return 0;
- }
-
- if (!p->owner) {
- return 0;
- }
- connected_id = ast_channel_connected_effective_id(p->owner);
- lid_num = S_COR(connected_id.number.valid, connected_id.number.str, NULL);
- if (!lid_num) {
- return 0;
- }
- lid_name = S_COR(connected_id.name.valid, connected_id.name.str, NULL);
- if (!lid_name) {
- lid_name = lid_num;
- }
- ast_escape_quoted(lid_name, lid_name_buf, sizeof(lid_name_buf));
- lid_pres = ast_party_id_presentation(&connected_id);
-
- if (((lid_pres & AST_PRES_RESTRICTION) != AST_PRES_ALLOWED) &&
- (ast_test_flag(&p->flags[1], SIP_PAGE2_TRUST_ID_OUTBOUND) == SIP_PAGE2_TRUST_ID_OUTBOUND_NO)) {
- /* If pres is not allowed and we don't trust the peer, we don't apply an RPID header */
- return 0;
- }
-
- fromdomain = p->fromdomain;
- if (!fromdomain ||
- ((ast_test_flag(&p->flags[1], SIP_PAGE2_TRUST_ID_OUTBOUND) == SIP_PAGE2_TRUST_ID_OUTBOUND_YES) &&
- !strcmp("anonymous.invalid", fromdomain))) {
- /* If the fromdomain is NULL or if it was set to anonymous.invalid due to privacy settings and we trust the peer,
- * use the host IP address */
- fromdomain = ast_sockaddr_stringify_host_remote(&p->ourip);
- }
-
- lid_num = ast_uri_encode(lid_num, tmp2, sizeof(tmp2), ast_uri_sip_user);
-
- if (ast_test_flag(&p->flags[0], SIP_SENDRPID_PAI)) {
- if (ast_test_flag(&p->flags[1], SIP_PAGE2_TRUST_ID_OUTBOUND) != SIP_PAGE2_TRUST_ID_OUTBOUND_LEGACY) {
- /* trust_id_outbound = yes - Always give full information even if it's private, but append a privacy header
- * When private data is included */
- ast_str_set(&tmp, -1, "\"%s\" ", lid_name_buf, lid_num, fromdomain);
- if ((lid_pres & AST_PRES_RESTRICTION) != AST_PRES_ALLOWED) {
- add_header(req, "Privacy", "id");
- }
- } else {
- /* trust_id_outbound = legacy - behave in a non RFC-3325 compliant manner and send anonymized data when
- * when handling private data. */
- if ((lid_pres & AST_PRES_RESTRICTION) == AST_PRES_ALLOWED) {
- ast_str_set(&tmp, -1, "\"%s\" ", lid_name_buf, lid_num, fromdomain);
- } else {
- ast_str_set(&tmp, -1, "%s", anonymous_string);
- }
- }
- add_header(req, "P-Asserted-Identity", ast_str_buffer(tmp));
- } else {
- ast_str_set(&tmp, -1, "\"%s\" ;party=%s", lid_name_buf, lid_num, fromdomain, p->outgoing_call ? "calling" : "called");
-
- switch (lid_pres) {
- case AST_PRES_ALLOWED_USER_NUMBER_NOT_SCREENED:
- case AST_PRES_ALLOWED_USER_NUMBER_FAILED_SCREEN:
- privacy = "off";
- screen = "no";
- break;
- case AST_PRES_ALLOWED_USER_NUMBER_PASSED_SCREEN:
- case AST_PRES_ALLOWED_NETWORK_NUMBER:
- privacy = "off";
- screen = "yes";
- break;
- case AST_PRES_PROHIB_USER_NUMBER_NOT_SCREENED:
- case AST_PRES_PROHIB_USER_NUMBER_FAILED_SCREEN:
- privacy = "full";
- screen = "no";
- break;
- case AST_PRES_PROHIB_USER_NUMBER_PASSED_SCREEN:
- case AST_PRES_PROHIB_NETWORK_NUMBER:
- privacy = "full";
- screen = "yes";
- break;
- case AST_PRES_NUMBER_NOT_AVAILABLE:
- break;
- default:
- if ((lid_pres & AST_PRES_RESTRICTION) != AST_PRES_ALLOWED) {
- privacy = "full";
- }
- else
- privacy = "off";
- screen = "no";
- break;
- }
-
- if (!ast_strlen_zero(privacy) && !ast_strlen_zero(screen)) {
- ast_str_append(&tmp, -1, ";privacy=%s;screen=%s", privacy, screen);
- }
-
- add_header(req, "Remote-Party-ID", ast_str_buffer(tmp));
- }
- return 0;
-}
-
-/*! \brief add XML encoded media control with update
- \note XML: The only way to turn 0 bits of information into a few hundred. (markster) */
-static int add_vidupdate(struct sip_request *req)
-{
- const char *xml_is_a_huge_waste_of_space =
- "\r\n"
- " \r\n"
- " \r\n"
- " \r\n"
- " \r\n"
- " \r\n"
- " \r\n"
- " \r\n"
- " \r\n";
- add_header(req, "Content-Type", "application/media_control+xml");
- add_content(req, xml_is_a_huge_waste_of_space);
- return 0;
-}
-
-/*! \brief Add ICE attributes to SDP */
-static void add_ice_to_sdp(struct ast_rtp_instance *instance, struct ast_str **a_buf)
-{
- struct ast_rtp_engine_ice *ice = ast_rtp_instance_get_ice(instance);
- const char *username, *password;
- struct ao2_container *candidates;
- struct ao2_iterator i;
- struct ast_rtp_engine_ice_candidate *candidate;
-
- /* If no ICE support is present we can't very well add the attributes */
- if (!ice || !(candidates = ice->get_local_candidates(instance))) {
- return;
- }
-
- if ((username = ice->get_ufrag(instance))) {
- ast_str_append(a_buf, 0, "a=ice-ufrag:%s\r\n", username);
- }
- if ((password = ice->get_password(instance))) {
- ast_str_append(a_buf, 0, "a=ice-pwd:%s\r\n", password);
- }
-
- i = ao2_iterator_init(candidates, 0);
-
- while ((candidate = ao2_iterator_next(&i))) {
- ast_str_append(a_buf, 0, "a=candidate:%s %u %s %d ", candidate->foundation, candidate->id, candidate->transport, candidate->priority);
- ast_str_append(a_buf, 0, "%s ", ast_sockaddr_stringify_addr_remote(&candidate->address));
-
- ast_str_append(a_buf, 0, "%s typ ", ast_sockaddr_stringify_port(&candidate->address));
-
- if (candidate->type == AST_RTP_ICE_CANDIDATE_TYPE_HOST) {
- ast_str_append(a_buf, 0, "host");
- } else if (candidate->type == AST_RTP_ICE_CANDIDATE_TYPE_SRFLX) {
- ast_str_append(a_buf, 0, "srflx");
- } else if (candidate->type == AST_RTP_ICE_CANDIDATE_TYPE_RELAYED) {
- ast_str_append(a_buf, 0, "relay");
- }
-
- if (!ast_sockaddr_isnull(&candidate->relay_address)) {
- ast_str_append(a_buf, 0, " raddr %s ", ast_sockaddr_stringify_addr_remote(&candidate->relay_address));
- ast_str_append(a_buf, 0, "rport %s", ast_sockaddr_stringify_port(&candidate->relay_address));
- }
-
- ast_str_append(a_buf, 0, "\r\n");
- ao2_ref(candidate, -1);
- }
-
- ao2_iterator_destroy(&i);
-
- ao2_ref(candidates, -1);
-}
-
-/*! \brief Start ICE negotiation on an RTP instance */
-static void start_ice(struct ast_rtp_instance *instance, int offer)
-{
- struct ast_rtp_engine_ice *ice = ast_rtp_instance_get_ice(instance);
-
- if (!ice) {
- return;
- }
-
- /* If we are the offerer then we are the controlling agent, otherwise they are */
- ice->set_role(instance, offer ? AST_RTP_ICE_ROLE_CONTROLLING : AST_RTP_ICE_ROLE_CONTROLLED);
- ice->start(instance);
-}
-
-/*! \brief Add DTLS attributes to SDP */
-static void add_dtls_to_sdp(struct ast_rtp_instance *instance, struct ast_str **a_buf)
-{
- struct ast_rtp_engine_dtls *dtls;
- enum ast_rtp_dtls_hash hash;
- const char *fingerprint;
-
- if (!instance || !(dtls = ast_rtp_instance_get_dtls(instance)) || !dtls->active(instance)) {
- return;
- }
-
- switch (dtls->get_connection(instance)) {
- case AST_RTP_DTLS_CONNECTION_NEW:
- ast_str_append(a_buf, 0, "a=connection:new\r\n");
- break;
- case AST_RTP_DTLS_CONNECTION_EXISTING:
- ast_str_append(a_buf, 0, "a=connection:existing\r\n");
- break;
- default:
- break;
- }
-
- switch (dtls->get_setup(instance)) {
- case AST_RTP_DTLS_SETUP_ACTIVE:
- ast_str_append(a_buf, 0, "a=setup:active\r\n");
- break;
- case AST_RTP_DTLS_SETUP_PASSIVE:
- ast_str_append(a_buf, 0, "a=setup:passive\r\n");
- break;
- case AST_RTP_DTLS_SETUP_ACTPASS:
- ast_str_append(a_buf, 0, "a=setup:actpass\r\n");
- break;
- case AST_RTP_DTLS_SETUP_HOLDCONN:
- ast_str_append(a_buf, 0, "a=setup:holdconn\r\n");
- break;
- default:
- break;
- }
-
- hash = dtls->get_fingerprint_hash(instance);
- fingerprint = dtls->get_fingerprint(instance);
- if (fingerprint && (hash == AST_RTP_DTLS_HASH_SHA1 || hash == AST_RTP_DTLS_HASH_SHA256)) {
- ast_str_append(a_buf, 0, "a=fingerprint:%s %s\r\n", hash == AST_RTP_DTLS_HASH_SHA1 ? "SHA-1" : "SHA-256",
- fingerprint);
- }
-}
-
-/*! \brief Add codec offer to SDP offer/answer body in INVITE or 200 OK */
-static void add_codec_to_sdp(const struct sip_pvt *p,
- struct ast_format *format,
- struct ast_str **m_buf,
- struct ast_str **a_buf,
- int debug,
- int *min_packet_size,
- int *max_packet_size)
-{
- int rtp_code;
- const char *mime;
- unsigned int rate, framing;
-
- if (debug)
- ast_verbose("Adding codec %s to SDP\n", ast_format_get_name(format));
-
- if (((rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(p->rtp), 1, format, 0)) == -1) ||
- !(mime = ast_rtp_lookup_mime_subtype2(1, format, 0, ast_test_flag(&p->flags[0], SIP_G726_NONSTANDARD) ? AST_RTP_OPT_G726_NONSTANDARD : 0)) ||
- !(rate = ast_rtp_lookup_sample_rate2(1, format, 0))) {
- return;
- }
-
- ast_str_append(m_buf, 0, " %d", rtp_code);
- /* Opus mandates 2 channels in rtpmap */
- if (ast_format_cmp(format, ast_format_opus) == AST_FORMAT_CMP_EQUAL) {
- ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%u/2\r\n", rtp_code, mime, rate);
- } else if ((AST_RTP_PT_LAST_STATIC < rtp_code) || !(sip_cfg.compactheaders)) {
- ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%u\r\n", rtp_code, mime, rate);
- }
-
- ast_format_generate_sdp_fmtp(format, rtp_code, a_buf);
-
- framing = ast_format_cap_get_format_framing(p->caps, format);
-
- if (ast_format_cmp(format, ast_format_g723) == AST_FORMAT_CMP_EQUAL) {
- /* Indicate that we don't support VAD (G.723.1 annex A) */
- ast_str_append(a_buf, 0, "a=fmtp:%d annexa=no\r\n", rtp_code);
- } else if (ast_format_cmp(format, ast_format_g719) == AST_FORMAT_CMP_EQUAL) {
- /* Indicate that we only expect 64Kbps */
- ast_str_append(a_buf, 0, "a=fmtp:%d bitrate=64000\r\n", rtp_code);
- }
-
- if (max_packet_size && ast_format_get_maximum_ms(format) &&
- (ast_format_get_maximum_ms(format) < *max_packet_size)) {
- *max_packet_size = ast_format_get_maximum_ms(format);
- }
-
- if (framing && (framing < *min_packet_size)) {
- *min_packet_size = framing;
- }
-
- /* Our first codec packetization processed cannot be zero */
- if ((*min_packet_size) == 0 && framing) {
- *min_packet_size = framing;
- }
-
- if ((*max_packet_size) == 0 && ast_format_get_maximum_ms(format)) {
- *max_packet_size = ast_format_get_maximum_ms(format);
- }
-}
-
-/*! \brief Add video codec offer to SDP offer/answer body in INVITE or 200 OK */
-/* This is different to the audio one now so we can add more caps later */
-static void add_vcodec_to_sdp(const struct sip_pvt *p, struct ast_format *format,
- struct ast_str **m_buf, struct ast_str **a_buf,
- int debug, int *min_packet_size)
-{
- int rtp_code;
- const char *subtype;
- unsigned int rate;
-
- if (!p->vrtp)
- return;
-
- if (debug)
- ast_verbose("Adding video codec %s to SDP\n", ast_format_get_name(format));
-
- if (((rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(p->vrtp), 1, format, 0)) == -1) ||
- !(subtype = ast_rtp_lookup_mime_subtype2(1, format, 0, 0)) ||
- !(rate = ast_rtp_lookup_sample_rate2(1, format, 0))) {
- return;
- }
-
- ast_str_append(m_buf, 0, " %d", rtp_code);
- ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%u\r\n", rtp_code, subtype, rate);
- /* VP8: add RTCP FIR support */
- if (ast_format_cmp(format, ast_format_vp8) == AST_FORMAT_CMP_EQUAL) {
- ast_str_append(a_buf, 0, "a=rtcp-fb:* ccm fir\r\n");
- }
-
- ast_format_generate_sdp_fmtp(format, rtp_code, a_buf);
-}
-
-/*! \brief Add text codec offer to SDP offer/answer body in INVITE or 200 OK */
-static void add_tcodec_to_sdp(const struct sip_pvt *p, struct ast_format *format,
- struct ast_str **m_buf, struct ast_str **a_buf,
- int debug, int *min_packet_size)
-{
- int rtp_code;
-
- if (!p->trtp)
- return;
-
- if (debug)
- ast_verbose("Adding text codec %s to SDP\n", ast_format_get_name(format));
-
- if ((rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(p->trtp), 1, format, 0)) == -1)
- return;
-
- ast_str_append(m_buf, 0, " %d", rtp_code);
- ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%u\r\n", rtp_code,
- ast_rtp_lookup_mime_subtype2(1, format, 0, 0),
- ast_rtp_lookup_sample_rate2(1, format, 0));
- /* Add fmtp code here */
-
- if (ast_format_cmp(format, ast_format_t140_red) == AST_FORMAT_CMP_EQUAL) {
- int t140code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(p->trtp), 1, ast_format_t140, 0);
- ast_str_append(a_buf, 0, "a=fmtp:%d %d/%d/%d\r\n", rtp_code,
- t140code,
- t140code,
- t140code);
-
- }
-}
-
-
-/*! \brief Get Max T.38 Transmission rate from T38 capabilities */
-static unsigned int t38_get_rate(enum ast_control_t38_rate rate)
-{
- switch (rate) {
- case AST_T38_RATE_2400:
- return 2400;
- case AST_T38_RATE_4800:
- return 4800;
- case AST_T38_RATE_7200:
- return 7200;
- case AST_T38_RATE_9600:
- return 9600;
- case AST_T38_RATE_12000:
- return 12000;
- case AST_T38_RATE_14400:
- return 14400;
- default:
- return 0;
- }
-}
-
-/*! \brief Add RFC 2833 DTMF offer to SDP */
-static void add_noncodec_to_sdp(const struct sip_pvt *p, int format,
- struct ast_str **m_buf, struct ast_str **a_buf,
- int debug)
-{
- int rtp_code;
-
- if (debug)
- ast_verbose("Adding non-codec 0x%x (%s) to SDP\n", (unsigned)format, ast_rtp_lookup_mime_subtype2(0, NULL, format, 0));
- if ((rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(p->rtp), 0, NULL, format)) == -1)
- return;
-
- ast_str_append(m_buf, 0, " %d", rtp_code);
- ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%u\r\n", rtp_code,
- ast_rtp_lookup_mime_subtype2(0, NULL, format, 0),
- ast_rtp_lookup_sample_rate2(0, NULL, format));
- if (format == AST_RTP_DTMF) /* Indicate we support DTMF and FLASH... */
- ast_str_append(a_buf, 0, "a=fmtp:%d 0-16\r\n", rtp_code);
-}
-
-/*! \brief Set all IP media addresses for this call
- \note called from add_sdp()
-*/
-static void get_our_media_address(struct sip_pvt *p, int needvideo, int needtext,
- struct ast_sockaddr *addr, struct ast_sockaddr *vaddr,
- struct ast_sockaddr *taddr, struct ast_sockaddr *dest,
- struct ast_sockaddr *vdest, struct ast_sockaddr *tdest)
-{
- int use_externip = 0;
-
- /* First, get our address */
- ast_rtp_instance_get_local_address(p->rtp, addr);
- if (p->vrtp) {
- ast_rtp_instance_get_local_address(p->vrtp, vaddr);
- }
- if (p->trtp) {
- ast_rtp_instance_get_local_address(p->trtp, taddr);
- }
-
- /* If our real IP differs from the local address returned by the RTP engine, use it. */
- /* The premise is that if we are already using that IP to communicate with the client, */
- /* we should be using it for RTP too. */
- use_externip = ast_sockaddr_cmp_addr(&p->ourip, addr);
-
- /* Now, try to figure out where we want them to send data */
- /* Is this a re-invite to move the media out, then use the original offer from caller */
- if (!ast_sockaddr_isnull(&p->redirip)) { /* If we have a redirection IP, use it */
- ast_sockaddr_copy(dest, &p->redirip);
- } else {
- /*
- * Audio Destination IP:
- *
- * 1. Specifically configured media address.
- * 2. Local address as specified by the RTP engine.
- * 3. The local IP as defined by chan_sip.
- *
- * Audio Destination Port:
- *
- * 1. Provided by the RTP engine.
- */
- ast_sockaddr_copy(dest,
- !ast_sockaddr_isnull(&media_address) ? &media_address :
- !ast_sockaddr_is_any(addr) && !use_externip ? addr :
- &p->ourip);
- ast_sockaddr_set_port(dest, ast_sockaddr_port(addr));
- }
-
- if (needvideo) {
- /* Determine video destination */
- if (!ast_sockaddr_isnull(&p->vredirip)) {
- ast_sockaddr_copy(vdest, &p->vredirip);
- } else {
- /*
- * Video Destination IP:
- *
- * 1. Specifically configured media address.
- * 2. Local address as specified by the RTP engine.
- * 3. The local IP as defined by chan_sip.
- *
- * Video Destination Port:
- *
- * 1. Provided by the RTP engine.
- */
- ast_sockaddr_copy(vdest,
- !ast_sockaddr_isnull(&media_address) ? &media_address :
- !ast_sockaddr_is_any(vaddr) && !use_externip ? vaddr :
- &p->ourip);
- ast_sockaddr_set_port(vdest, ast_sockaddr_port(vaddr));
- }
- }
-
- if (needtext) {
- /* Determine text destination */
- if (!ast_sockaddr_isnull(&p->tredirip)) {
- ast_sockaddr_copy(tdest, &p->tredirip);
- } else {
- /*
- * Text Destination IP:
- *
- * 1. Specifically configured media address.
- * 2. Local address as specified by the RTP engine.
- * 3. The local IP as defined by chan_sip.
- *
- * Text Destination Port:
- *
- * 1. Provided by the RTP engine.
- */
- ast_sockaddr_copy(tdest,
- !ast_sockaddr_isnull(&media_address) ? &media_address :
- !ast_sockaddr_is_any(taddr) && !use_externip ? taddr :
- &p->ourip);
- ast_sockaddr_set_port(tdest, ast_sockaddr_port(taddr));
- }
- }
-}
-
-static char *crypto_get_attrib(struct ast_sdp_srtp *srtp, int dtls_enabled, int default_taglen_32)
-{
- struct ast_sdp_srtp *tmp = srtp;
- char *a_crypto;
-
- if (!tmp || dtls_enabled) {
- return NULL;
- }
-
- a_crypto = ast_strdup("");
- if (!a_crypto) {
- return NULL;
- }
-
- do {
- char *copy = a_crypto;
- const char *orig_crypto = ast_sdp_srtp_get_attrib(tmp, dtls_enabled, default_taglen_32);
-
- if (ast_strlen_zero(orig_crypto)) {
- ast_free(copy);
- return NULL;
- }
- if (ast_asprintf(&a_crypto, "%sa=crypto:%s\r\n", copy, orig_crypto) == -1) {
- ast_free(copy);
- return NULL;
- }
-
- ast_free(copy);
- } while ((tmp = AST_LIST_NEXT(tmp, sdp_srtp_list)));
-
- return a_crypto;
-}
-
-/*! \brief Add Session Description Protocol message
-
- If oldsdp is TRUE, then the SDP version number is not incremented. This mechanism
- is used in Session-Timers where RE-INVITEs are used for refreshing SIP sessions
- without modifying the media session in any way.
-*/
-static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int oldsdp, int add_audio, int add_t38)
-{
- struct ast_format_cap *alreadysent = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
- struct ast_format_cap *tmpcap = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
- int res = AST_SUCCESS;
- int doing_directmedia = FALSE;
- struct ast_sockaddr addr = { {0,} };
- struct ast_sockaddr vaddr = { {0,} };
- struct ast_sockaddr taddr = { {0,} };
- struct ast_sockaddr udptladdr = { {0,} };
- struct ast_sockaddr dest = { {0,} };
- struct ast_sockaddr vdest = { {0,} };
- struct ast_sockaddr tdest = { {0,} };
- struct ast_sockaddr udptldest = { {0,} };
-
- /* SDP fields */
- struct offered_media *offer;
- char *version = "v=0\r\n"; /* Protocol version */
- char subject[256]; /* Subject of the session */
- char owner[256]; /* Session owner/creator */
- char connection[256]; /* Connection data */
- char *session_time = "t=0 0\r\n"; /* Time the session is active */
- char bandwidth[256] = ""; /* Max bitrate */
- char *hold = "";
- struct ast_str *m_audio = ast_str_alloca(256); /* Media declaration line for audio */
- struct ast_str *m_video = ast_str_alloca(256); /* Media declaration line for video */
- struct ast_str *m_text = ast_str_alloca(256); /* Media declaration line for text */
- struct ast_str *m_modem = ast_str_alloca(256); /* Media declaration line for modem */
- struct ast_str *a_audio = ast_str_create(256); /* Attributes for audio */
- struct ast_str *a_video = ast_str_create(256); /* Attributes for video */
- struct ast_str *a_text = ast_str_create(256); /* Attributes for text */
- struct ast_str *a_modem = ast_str_alloca(1024); /* Attributes for modem */
- RAII_VAR(char *, a_crypto, NULL, ast_free);
- RAII_VAR(char *, v_a_crypto, NULL, ast_free);
- RAII_VAR(char *, t_a_crypto, NULL, ast_free);
-
- int x;
- struct ast_format *tmp_fmt;
- int needaudio = FALSE;
- int needvideo = FALSE;
- int needtext = FALSE;
- int debug = sip_debug_test_pvt(p);
- int min_audio_packet_size = 0;
- int max_audio_packet_size = 0;
- int min_video_packet_size = 0;
- int min_text_packet_size = 0;
-
- struct ast_str *codec_buf = ast_str_alloca(AST_FORMAT_CAP_NAMES_LEN);
-
- /* Set the SDP session name */
- snprintf(subject, sizeof(subject), "s=%s\r\n", ast_strlen_zero(global_sdpsession) ? "-" : global_sdpsession);
-
- if (!alreadysent || !tmpcap) {
- res = AST_FAILURE;
- goto add_sdp_cleanup;
- }
- if (!p->rtp) {
- ast_log(LOG_WARNING, "No way to add SDP without an RTP structure\n");
- res = AST_FAILURE;
- goto add_sdp_cleanup;
-
- }
- /* XXX We should not change properties in the SIP dialog until
- we have acceptance of the offer if this is a re-invite */
-
- /* Set RTP Session ID and version */
- if (!p->sessionid) {
- p->sessionid = (int)ast_random();
- p->sessionversion = p->sessionid;
- } else {
- if (oldsdp == FALSE)
- p->sessionversion++;
- }
-
- if (add_audio) {
- doing_directmedia = (!ast_sockaddr_isnull(&p->redirip) && (ast_format_cap_count(p->redircaps))) ? TRUE : FALSE;
-
- if (doing_directmedia) {
- ast_format_cap_get_compatible(p->jointcaps, p->redircaps, tmpcap);
- ast_debug(1, "** Our native-bridge filtered capability: %s\n", ast_format_cap_get_names(tmpcap, &codec_buf));
- } else {
- ast_format_cap_append_from_cap(tmpcap, p->jointcaps, AST_MEDIA_TYPE_UNKNOWN);
- }
-
- /* Check if we need audio in this call */
- needaudio = ast_format_cap_has_type(tmpcap, AST_MEDIA_TYPE_AUDIO);
-
- /* Check if we need video in this call */
- if ((ast_format_cap_has_type(tmpcap, AST_MEDIA_TYPE_VIDEO)) && !p->novideo) {
- if (doing_directmedia && !ast_format_cap_has_type(tmpcap, AST_MEDIA_TYPE_VIDEO)) {
- ast_debug(2, "This call needs video offers, but caller probably did not offer it!\n");
- } else if (p->vrtp) {
- needvideo = TRUE;
- ast_debug(2, "This call needs video offers!\n");
- } else {
- ast_debug(2, "This call needs video offers, but there's no video support enabled!\n");
- }
- }
-
- /* Check if we need text in this call */
- if ((ast_format_cap_has_type(p->jointcaps, AST_MEDIA_TYPE_TEXT)) && !p->notext) {
- if (sipdebug_text)
- ast_verbose("We think we can do text\n");
- if (p->trtp) {
- if (sipdebug_text) {
- ast_verbose("And we have a text rtp object\n");
- }
- needtext = TRUE;
- ast_debug(2, "This call needs text offers! \n");
- } else {
- ast_debug(2, "This call needs text offers, but there's no text support enabled ! \n");
- }
- }
-
- /* XXX note, Video and Text are negated - 'true' means 'no' */
- ast_debug(1, "** Our capability: %s Video flag: %s Text flag: %s\n",
- ast_format_cap_get_names(tmpcap, &codec_buf),
- p->novideo ? "True" : "False", p->notext ? "True" : "False");
- ast_debug(1, "** Our prefcodec: %s \n", ast_format_cap_get_names(p->prefcaps, &codec_buf));
- }
-
- get_our_media_address(p, needvideo, needtext, &addr, &vaddr, &taddr, &dest, &vdest, &tdest);
-
- /* We don't use dest here but p->ourip because address in o= field must not change in reINVITE */
- snprintf(owner, sizeof(owner), "o=%s %d %d IN %s %s\r\n",
- ast_strlen_zero(global_sdpowner) ? "-" : global_sdpowner,
- p->sessionid, p->sessionversion,
- (ast_sockaddr_is_ipv6(&p->ourip) && !ast_sockaddr_is_ipv4_mapped(&p->ourip)) ?
- "IP6" : "IP4",
- ast_sockaddr_stringify_addr_remote(&p->ourip));
-
- snprintf(connection, sizeof(connection), "c=IN %s %s\r\n",
- (ast_sockaddr_is_ipv6(&dest) && !ast_sockaddr_is_ipv4_mapped(&dest)) ?
- "IP6" : "IP4",
- ast_sockaddr_stringify_addr_remote(&dest));
-
- if (add_audio) {
- if (ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD) == SIP_PAGE2_CALL_ONHOLD_ONEDIR) {
- hold = "a=recvonly\r\n";
- doing_directmedia = FALSE;
- } else if (ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD) == SIP_PAGE2_CALL_ONHOLD_INACTIVE) {
- hold = "a=inactive\r\n";
- doing_directmedia = FALSE;
- } else {
- hold = "a=sendrecv\r\n";
- }
-
- if (debug) {
- ast_verbose("Audio is at %s\n", ast_sockaddr_stringify_port(&addr));
- }
-
- /* Ok, we need video. Let's add what we need for video and set codecs.
- Video is handled differently than audio since we can not transcode. */
- if (needvideo) {
- v_a_crypto = crypto_get_attrib(p->vsrtp, p->dtls_cfg.enabled,
- ast_test_flag(&p->flags[2], SIP_PAGE3_SRTP_TAG_32));
- ast_str_append(&m_video, 0, "m=video %d %s", ast_sockaddr_port(&vdest),
- ast_sdp_get_rtp_profile(v_a_crypto ? 1 : 0, p->vrtp,
- ast_test_flag(&p->flags[2], SIP_PAGE3_USE_AVPF),
- ast_test_flag(&p->flags[2], SIP_PAGE3_FORCE_AVP)));
-
- /* Build max bitrate string */
- if (p->maxcallbitrate)
- snprintf(bandwidth, sizeof(bandwidth), "b=CT:%d\r\n", p->maxcallbitrate);
- if (debug) {
- ast_verbose("Video is at %s\n", ast_sockaddr_stringify(&vdest));
- }
-
- if (!doing_directmedia) {
- if (ast_test_flag(&p->flags[2], SIP_PAGE3_ICE_SUPPORT)) {
- add_ice_to_sdp(p->vrtp, &a_video);
- }
-
- add_dtls_to_sdp(p->vrtp, &a_video);
- }
- }
-
- /* Ok, we need text. Let's add what we need for text and set codecs.
- Text is handled differently than audio since we can not transcode. */
- if (needtext) {
- if (sipdebug_text)
- ast_verbose("Lets set up the text sdp\n");
- t_a_crypto = crypto_get_attrib(p->tsrtp, p->dtls_cfg.enabled,
- ast_test_flag(&p->flags[2], SIP_PAGE3_SRTP_TAG_32));
- ast_str_append(&m_text, 0, "m=text %d %s", ast_sockaddr_port(&tdest),
- ast_sdp_get_rtp_profile(t_a_crypto ? 1 : 0, p->trtp,
- ast_test_flag(&p->flags[2], SIP_PAGE3_USE_AVPF),
- ast_test_flag(&p->flags[2], SIP_PAGE3_FORCE_AVP)));
- if (debug) { /* XXX should I use tdest below ? */
- ast_verbose("Text is at %s\n", ast_sockaddr_stringify(&taddr));
- }
-
- if (!doing_directmedia) {
- if (ast_test_flag(&p->flags[2], SIP_PAGE3_ICE_SUPPORT)) {
- add_ice_to_sdp(p->trtp, &a_text);
- }
-
- add_dtls_to_sdp(p->trtp, &a_text);
- }
- }
-
- /* Start building generic SDP headers */
-
- /* We break with the "recommendation" and send our IP, in order that our
- peer doesn't have to ast_gethostbyname() us */
-
- a_crypto = crypto_get_attrib(p->srtp, p->dtls_cfg.enabled,
- ast_test_flag(&p->flags[2], SIP_PAGE3_SRTP_TAG_32));
- ast_str_append(&m_audio, 0, "m=audio %d %s", ast_sockaddr_port(&dest),
- ast_sdp_get_rtp_profile(a_crypto ? 1 : 0, p->rtp,
- ast_test_flag(&p->flags[2], SIP_PAGE3_USE_AVPF),
- ast_test_flag(&p->flags[2], SIP_PAGE3_FORCE_AVP)));
-
- /* Now, start adding audio codecs. These are added in this order:
- - First what was requested by the calling channel
- - Then our mutually shared capabilities, determined previous in tmpcap
- */
-
-
- /* Unless otherwise configured, the prefcaps is added before the peer's
- * configured codecs.
- */
- if (!ast_test_flag(&p->flags[2], SIP_PAGE3_IGNORE_PREFCAPS)) {
- for (x = 0; x < ast_format_cap_count(p->prefcaps); x++) {
- tmp_fmt = ast_format_cap_get_format(p->prefcaps, x);
-
- if ((ast_format_get_type(tmp_fmt) != AST_MEDIA_TYPE_AUDIO) ||
- (ast_format_cap_iscompatible_format(tmpcap, tmp_fmt) == AST_FORMAT_CMP_NOT_EQUAL)) {
- ao2_ref(tmp_fmt, -1);
- continue;
- }
-
- add_codec_to_sdp(p, tmp_fmt, &m_audio, &a_audio, debug, &min_audio_packet_size, &max_audio_packet_size);
- ast_format_cap_append(alreadysent, tmp_fmt, 0);
- ao2_ref(tmp_fmt, -1);
- }
- }
-
- /* Now send any other common codecs */
- for (x = 0; x < ast_format_cap_count(tmpcap); x++) {
- tmp_fmt = ast_format_cap_get_format(tmpcap, x);
-
- if (ast_format_cap_iscompatible_format(alreadysent, tmp_fmt) != AST_FORMAT_CMP_NOT_EQUAL) {
- ao2_ref(tmp_fmt, -1);
- continue;
- }
-
- if (ast_format_get_type(tmp_fmt) == AST_MEDIA_TYPE_AUDIO) {
- add_codec_to_sdp(p, tmp_fmt, &m_audio, &a_audio, debug, &min_audio_packet_size, &max_audio_packet_size);
- } else if (needvideo && ast_format_get_type(tmp_fmt) == AST_MEDIA_TYPE_VIDEO) {
- add_vcodec_to_sdp(p, tmp_fmt, &m_video, &a_video, debug, &min_video_packet_size);
- } else if (needtext && ast_format_get_type(tmp_fmt) == AST_MEDIA_TYPE_TEXT) {
- add_tcodec_to_sdp(p, tmp_fmt, &m_text, &a_text, debug, &min_text_packet_size);
- }
-
- ast_format_cap_append(alreadysent, tmp_fmt, 0);
- ao2_ref(tmp_fmt, -1);
- }
-
- /* Now add DTMF RFC2833 telephony-event as a codec */
- for (x = 1LL; x <= AST_RTP_MAX; x <<= 1) {
- if (!(p->jointnoncodeccapability & x))
- continue;
-
- add_noncodec_to_sdp(p, x, &m_audio, &a_audio, debug);
- }
-
- ast_debug(3, "-- Done with adding codecs to SDP\n");
-
- if (!p->owner || ast_channel_timingfd(p->owner) == -1) {
- ast_str_append(&a_audio, 0, "a=silenceSupp:off - - - -\r\n");
- }
-
- if (min_audio_packet_size) {
- ast_str_append(&a_audio, 0, "a=ptime:%d\r\n", min_audio_packet_size);
- }
-
- /* XXX don't think you can have ptime for video */
- if (min_video_packet_size) {
- ast_str_append(&a_video, 0, "a=ptime:%d\r\n", min_video_packet_size);
- }
-
- /* XXX don't think you can have ptime for text */
- if (min_text_packet_size) {
- ast_str_append(&a_text, 0, "a=ptime:%d\r\n", min_text_packet_size);
- }
-
- if (max_audio_packet_size) {
- ast_str_append(&a_audio, 0, "a=maxptime:%d\r\n", max_audio_packet_size);
- }
-
- if (!ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
- ast_debug(1, "Setting framing on incoming call: %u\n", min_audio_packet_size);
- ast_rtp_codecs_set_framing(ast_rtp_instance_get_codecs(p->rtp), min_audio_packet_size);
- }
-
- if (!doing_directmedia) {
- if (ast_test_flag(&p->flags[2], SIP_PAGE3_ICE_SUPPORT)) {
- add_ice_to_sdp(p->rtp, &a_audio);
- /* Start ICE negotiation, and setting that we are controlled agent,
- as this is response to offer */
- if (resp->method == SIP_RESPONSE) {
- start_ice(p->rtp, 0);
- }
- }
-
- add_dtls_to_sdp(p->rtp, &a_audio);
- }
-
- /* If we've got rtcp-mux enabled, just unconditionally offer it in all SDPs */
- if (ast_test_flag(&p->flags[2], SIP_PAGE3_RTCP_MUX)) {
- ast_str_append(&a_audio, 0, "a=rtcp-mux\r\n");
- ast_str_append(&a_video, 0, "a=rtcp-mux\r\n");
- }
- }
-
- if (add_t38) {
- /* Our T.38 end is */
- ast_udptl_get_us(p->udptl, &udptladdr);
-
- /* We don't use directmedia for T.38, so keep the destination the same as our IP address. */
- ast_sockaddr_copy(&udptldest, &p->ourip);
- ast_sockaddr_set_port(&udptldest, ast_sockaddr_port(&udptladdr));
-
- if (debug) {
- ast_debug(1, "T.38 UDPTL is at %s port %d\n", ast_sockaddr_stringify_addr(&p->ourip), ast_sockaddr_port(&udptladdr));
- }
-
- /* We break with the "recommendation" and send our IP, in order that our
- peer doesn't have to ast_gethostbyname() us */
-
- ast_str_append(&m_modem, 0, "m=image %d udptl t38\r\n", ast_sockaddr_port(&udptldest));
-
- if (ast_sockaddr_cmp_addr(&udptldest, &dest)) {
- ast_str_append(&m_modem, 0, "c=IN %s %s\r\n",
- (ast_sockaddr_is_ipv6(&udptldest) && !ast_sockaddr_is_ipv4_mapped(&udptldest)) ?
- "IP6" : "IP4", ast_sockaddr_stringify_addr_remote(&udptldest));
- }
-
- ast_str_append(&a_modem, 0, "a=T38FaxVersion:%u\r\n", p->t38.our_parms.version);
- ast_str_append(&a_modem, 0, "a=T38MaxBitRate:%u\r\n", t38_get_rate(p->t38.our_parms.rate));
- if (p->t38.our_parms.fill_bit_removal) {
- ast_str_append(&a_modem, 0, "a=T38FaxFillBitRemoval\r\n");
- }
- if (p->t38.our_parms.transcoding_mmr) {
- ast_str_append(&a_modem, 0, "a=T38FaxTranscodingMMR\r\n");
- }
- if (p->t38.our_parms.transcoding_jbig) {
- ast_str_append(&a_modem, 0, "a=T38FaxTranscodingJBIG\r\n");
- }
- switch (p->t38.our_parms.rate_management) {
- case AST_T38_RATE_MANAGEMENT_TRANSFERRED_TCF:
- ast_str_append(&a_modem, 0, "a=T38FaxRateManagement:transferredTCF\r\n");
- break;
- case AST_T38_RATE_MANAGEMENT_LOCAL_TCF:
- ast_str_append(&a_modem, 0, "a=T38FaxRateManagement:localTCF\r\n");
- break;
- }
- ast_str_append(&a_modem, 0, "a=T38FaxMaxDatagram:%u\r\n", ast_udptl_get_local_max_datagram(p->udptl));
- switch (ast_udptl_get_error_correction_scheme(p->udptl)) {
- case UDPTL_ERROR_CORRECTION_NONE:
- break;
- case UDPTL_ERROR_CORRECTION_FEC:
- ast_str_append(&a_modem, 0, "a=T38FaxUdpEC:t38UDPFEC\r\n");
- break;
- case UDPTL_ERROR_CORRECTION_REDUNDANCY:
- ast_str_append(&a_modem, 0, "a=T38FaxUdpEC:t38UDPRedundancy\r\n");
- break;
- }
- }
-
- if (needaudio)
- ast_str_append(&m_audio, 0, "\r\n");
- if (needvideo)
- ast_str_append(&m_video, 0, "\r\n");
- if (needtext)
- ast_str_append(&m_text, 0, "\r\n");
-
- add_header(resp, "Content-Type", "application/sdp");
- add_content(resp, version);
- add_content(resp, owner);
- add_content(resp, subject);
- add_content(resp, connection);
- /* only if video response is appropriate */
- if (needvideo) {
- add_content(resp, bandwidth);
- }
- add_content(resp, session_time);
- /* if this is a response to an invite, order our offers properly */
- if (!AST_LIST_EMPTY(&p->offered_media)) {
- AST_LIST_TRAVERSE(&p->offered_media, offer, next) {
- switch (offer->type) {
- case SDP_AUDIO:
- if (needaudio) {
- add_content(resp, ast_str_buffer(m_audio));
- if (a_crypto) {
- add_content(resp, a_crypto);
- }
- add_content(resp, ast_str_buffer(a_audio));
- add_content(resp, hold);
- } else {
- add_content(resp, offer->decline_m_line);
- }
- break;
- case SDP_VIDEO:
- if (needvideo) { /* only if video response is appropriate */
- add_content(resp, ast_str_buffer(m_video));
- add_content(resp, ast_str_buffer(a_video));
- add_content(resp, hold); /* Repeat hold for the video stream */
- if (v_a_crypto) {
- add_content(resp, v_a_crypto);
- }
- } else {
- add_content(resp, offer->decline_m_line);
- }
- break;
- case SDP_TEXT:
- if (needtext) { /* only if text response is appropriate */
- add_content(resp, ast_str_buffer(m_text));
- add_content(resp, ast_str_buffer(a_text));
- add_content(resp, hold); /* Repeat hold for the text stream */
- if (t_a_crypto) {
- add_content(resp, t_a_crypto);
- }
- } else {
- add_content(resp, offer->decline_m_line);
- }
- break;
- case SDP_IMAGE:
- if (add_t38) {
- add_content(resp, ast_str_buffer(m_modem));
- add_content(resp, ast_str_buffer(a_modem));
- } else {
- add_content(resp, offer->decline_m_line);
- }
- break;
- case SDP_UNKNOWN:
- add_content(resp, offer->decline_m_line);
- break;
- }
- }
- } else {
- /* generate new SDP from scratch, no offers */
- if (needaudio) {
- add_content(resp, ast_str_buffer(m_audio));
- if (a_crypto) {
- add_content(resp, a_crypto);
- }
- add_content(resp, ast_str_buffer(a_audio));
- add_content(resp, hold);
- }
- if (needvideo) { /* only if video response is appropriate */
- add_content(resp, ast_str_buffer(m_video));
- add_content(resp, ast_str_buffer(a_video));
- add_content(resp, hold); /* Repeat hold for the video stream */
- if (v_a_crypto) {
- add_content(resp, v_a_crypto);
- }
- }
- if (needtext) { /* only if text response is appropriate */
- add_content(resp, ast_str_buffer(m_text));
- add_content(resp, ast_str_buffer(a_text));
- add_content(resp, hold); /* Repeat hold for the text stream */
- if (t_a_crypto) {
- add_content(resp, t_a_crypto);
- }
- }
- if (add_t38) {
- add_content(resp, ast_str_buffer(m_modem));
- add_content(resp, ast_str_buffer(a_modem));
- }
- }
-
- /* Update lastrtprx when we send our SDP */
- p->lastrtprx = p->lastrtptx = time(NULL); /* XXX why both ? */
-
- /*
- * We unlink this dialog and link again into the
- * dialogs_rtpcheck container so its not in there twice.
- */
- ao2_lock(dialogs_rtpcheck);
- ao2_t_unlink(dialogs_rtpcheck, p, "unlink pvt into dialogs_rtpcheck container");
- ao2_t_link(dialogs_rtpcheck, p, "link pvt into dialogs_rtpcheck container");
- ao2_unlock(dialogs_rtpcheck);
-
- ast_debug(3, "Done building SDP. Settling with this capability: %s\n",
- ast_format_cap_get_names(tmpcap, &codec_buf));
-
-add_sdp_cleanup:
- ast_free(a_text);
- ast_free(a_video);
- ast_free(a_audio);
- ao2_cleanup(alreadysent);
- ao2_cleanup(tmpcap);
-
- return res;
-}
-
-/*! \brief Used for 200 OK and 183 early media */
-static int transmit_response_with_t38_sdp(struct sip_pvt *p, char *msg, struct sip_request *req, int retrans)
-{
- struct sip_request resp;
- uint32_t seqno;
-
- if (sscanf(sip_get_header(req, "CSeq"), "%30u ", &seqno) != 1) {
- ast_log(LOG_WARNING, "Unable to get seqno from '%s'\n", sip_get_header(req, "CSeq"));
- return -1;
- }
- respprep(&resp, p, msg, req);
- if (p->udptl) {
- add_sdp(&resp, p, 0, 0, 1);
- } else
- ast_log(LOG_ERROR, "Can't add SDP to response, since we have no UDPTL session allocated. Call-ID %s\n", p->callid);
- if (retrans && !p->pendinginvite)
- p->pendinginvite = seqno; /* Buggy clients sends ACK on RINGING too */
- return send_response(p, &resp, retrans, seqno);
-}
-
-/*! \brief copy SIP request (mostly used to save request for responses) */
-static void copy_request(struct sip_request *dst, const struct sip_request *src)
-{
- /* XXX this function can encounter memory allocation errors, perhaps it
- * should return a value */
-
- struct ast_str *duplicate = dst->data;
- struct ast_str *duplicate_content = dst->content;
-
- /* copy the entire request then restore the original data and content
- * members from the dst request */
- *dst = *src;
- dst->data = duplicate;
- dst->content = duplicate_content;
-
- /* copy the data into the dst request */
- if (!dst->data && !(dst->data = ast_str_create(ast_str_strlen(src->data) + 1))) {
- return;
- }
- ast_str_copy_string(&dst->data, src->data);
-
- /* copy the content into the dst request (if it exists) */
- if (src->content) {
- if (!dst->content && !(dst->content = ast_str_create(ast_str_strlen(src->content) + 1))) {
- return;
- }
- ast_str_copy_string(&dst->content, src->content);
- }
-}
-
-static void add_cc_call_info_to_response(struct sip_pvt *p, struct sip_request *resp)
-{
- char uri[SIPBUFSIZE];
- struct ast_str *header = ast_str_alloca(SIPBUFSIZE);
- struct ast_cc_agent *agent = find_sip_cc_agent_by_original_callid(p);
- struct sip_cc_agent_pvt *agent_pvt;
-
- if (!agent) {
- /* Um, what? How could the SIP_OFFER_CC flag be set but there not be an
- * agent? Oh well, we'll just warn and return without adding the header.
- */
- ast_log(LOG_WARNING, "Can't find SIP CC agent for call '%s' even though OFFER_CC flag was set?\n", p->callid);
- return;
- }
-
- agent_pvt = agent->private_data;
-
- if (!ast_strlen_zero(agent_pvt->subscribe_uri)) {
- ast_copy_string(uri, agent_pvt->subscribe_uri, sizeof(uri));
- } else {
- generate_uri(p, uri, sizeof(uri));
- ast_copy_string(agent_pvt->subscribe_uri, uri, sizeof(agent_pvt->subscribe_uri));
- }
- /* XXX Hardcode "NR" as the m reason for now. This should perhaps be changed
- * to be more accurate. This parameter has no bearing on the actual operation
- * of the feature; it's just there for informational purposes.
- */
- ast_str_set(&header, 0, "<%s>;purpose=call-completion;m=%s", uri, "NR");
- add_header(resp, "Call-Info", ast_str_buffer(header));
- ao2_ref(agent, -1);
-}
-
-/*! \brief Used for 200 OK and 183 early media
- \retval XMIT_ERROR for network errors.
-*/
-static int transmit_response_with_sdp(struct sip_pvt *p, const char *msg, const struct sip_request *req, enum xmittype reliable, int oldsdp, int rpid)
-{
- struct sip_request resp;
- uint32_t seqno;
- if (sscanf(sip_get_header(req, "CSeq"), "%30u ", &seqno) != 1) {
- ast_log(LOG_WARNING, "Unable to get seqno from '%s'\n", sip_get_header(req, "CSeq"));
- return -1;
- }
- respprep(&resp, p, msg, req);
- if (rpid == TRUE) {
- add_rpid(&resp, p);
- }
- if (ast_test_flag(&p->flags[0], SIP_OFFER_CC)) {
- add_cc_call_info_to_response(p, &resp);
- }
- if (p->rtp) {
- ast_rtp_instance_activate(p->rtp);
- try_suggested_sip_codec(p);
- if (p->t38.state == T38_ENABLED) {
- add_sdp(&resp, p, oldsdp, TRUE, TRUE);
- } else {
- add_sdp(&resp, p, oldsdp, TRUE, FALSE);
- }
- } else
- ast_log(LOG_ERROR, "Can't add SDP to response, since we have no RTP session allocated. Call-ID %s\n", p->callid);
- if (reliable && !p->pendinginvite)
- p->pendinginvite = seqno; /* Buggy clients sends ACK on RINGING too */
- add_required_respheader(&resp);
- return send_response(p, &resp, reliable, seqno);
-}
-
-/*! \brief Parse first line of incoming SIP request */
-static int determine_firstline_parts(struct sip_request *req)
-{
- char *e = ast_skip_blanks(ast_str_buffer(req->data)); /* there shouldn't be any */
- char *local_rlpart1;
-
- if (!*e)
- return -1;
- req->rlpart1 = e - ast_str_buffer(req->data); /* method or protocol */
- local_rlpart1 = e;
- e = ast_skip_nonblanks(e);
- if (*e)
- *e++ = '\0';
- /* Get URI or status code */
- e = ast_skip_blanks(e);
- if ( !*e )
- return -1;
- ast_trim_blanks(e);
-
- if (!strcasecmp(local_rlpart1, "SIP/2.0") ) { /* We have a response */
- if (strlen(e) < 3) /* status code is 3 digits */
- return -1;
- req->rlpart2 = e - ast_str_buffer(req->data);
- } else { /* We have a request */
- if ( *e == '<' ) { /* XXX the spec says it must not be in <> ! */
- ast_debug(3, "Oops. Bogus uri in <> %s\n", e);
- e++;
- if (!*e)
- return -1;
- }
- req->rlpart2 = e - ast_str_buffer(req->data); /* URI */
- e = ast_skip_nonblanks(e);
- if (*e)
- *e++ = '\0';
- e = ast_skip_blanks(e);
- if (strcasecmp(e, "SIP/2.0") ) {
- ast_debug(3, "Skipping packet - Bad request protocol %s\n", e);
- return -1;
- }
- }
- return 1;
-}
-
-/*! \brief Transmit reinvite with SDP
-\note A re-invite is basically a new INVITE with the same CALL-ID and TAG as the
- INVITE that opened the SIP dialogue
- We reinvite so that the audio stream (RTP) go directly between
- the SIP UAs. SIP Signalling stays with * in the path.
-
- If t38version is TRUE, we send T38 SDP for re-invite from audio/video to
- T38 UDPTL transmission on the channel
-
- If oldsdp is TRUE then the SDP version number is not incremented. This
- is needed for Session-Timers so we can send a re-invite to refresh the
- SIP session without modifying the media session.
-*/
-static int transmit_reinvite_with_sdp(struct sip_pvt *p, int t38version, int oldsdp)
-{
- struct sip_request req;
-
- if (t38version) {
- /* Force media to go through us for T.38. */
- memset(&p->redirip, 0, sizeof(p->redirip));
- }
- if (p->rtp) {
- if (t38version) {
- /* Silence RTCP while audio RTP is inactive */
- ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_RTCP, AST_RTP_INSTANCE_RTCP_DISABLED);
- if (p->owner) {
- /* Prevent audio RTCP reads */
- ast_channel_set_fd(p->owner, SIP_AUDIO_RTCP_FD, -1);
- }
- } else if (ast_sockaddr_isnull(&p->redirip)) {
- /* Enable RTCP since it will be inactive if we're coming back
- * with this reinvite */
- ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_RTCP, AST_RTP_INSTANCE_RTCP_STANDARD);
- if (p->owner) {
- /* Enable audio RTCP reads */
- ast_channel_set_fd(p->owner, SIP_AUDIO_RTCP_FD, ast_rtp_instance_fd(p->rtp, 1));
- }
- }
- }
-
- reqprep(&req, p, ast_test_flag(&p->flags[0], SIP_REINVITE_UPDATE) ? SIP_UPDATE : SIP_INVITE, 0, 1);
-
- add_header(&req, "Allow", ALLOWED_METHODS);
- add_supported(p, &req);
- if (sipdebug) {
- if (oldsdp == TRUE)
- add_header(&req, "X-asterisk-Info", "SIP re-invite (Session-Timers)");
- else
- add_header(&req, "X-asterisk-Info", "SIP re-invite (External RTP bridge)");
- }
-
- if (ast_test_flag(&p->flags[0], SIP_SENDRPID))
- add_rpid(&req, p);
-
- if (p->do_history) {
- append_history(p, "ReInv", "Re-invite sent");
- }
-
- offered_media_list_destroy(p);
-
- try_suggested_sip_codec(p);
- if (t38version) {
- add_sdp(&req, p, oldsdp, FALSE, TRUE);
- } else {
- add_sdp(&req, p, oldsdp, TRUE, FALSE);
- }
-
- /* Use this as the basis */
- initialize_initreq(p, &req);
- p->lastinvite = p->ocseq;
- ast_set_flag(&p->flags[0], SIP_OUTGOING); /* Change direction of this dialog */
- p->ongoing_reinvite = 1;
- return send_request(p, &req, XMIT_CRITICAL, p->ocseq);
-}
-
-/*! \brief Remove URI parameters at end of URI, not in username part though */
-static char *remove_uri_parameters(char *uri)
-{
- char *atsign;
- atsign = strchr(uri, '@'); /* First, locate the at sign */
- if (!atsign) {
- atsign = uri; /* Ok hostname only, let's stick with the rest */
- }
- atsign = strchr(atsign, ';'); /* Locate semi colon */
- if (atsign)
- *atsign = '\0'; /* Kill at the semi colon */
- return uri;
-}
-
-/*! \brief Check Contact: URI of SIP message */
-static void extract_uri(struct sip_pvt *p, struct sip_request *req)
-{
- char stripped[SIPBUFSIZE];
- char *c;
-
- ast_copy_string(stripped, sip_get_header(req, "Contact"), sizeof(stripped));
- c = get_in_brackets(stripped);
- /* Cut the URI at the at sign after the @, not in the username part */
- c = remove_uri_parameters(c);
- if (!ast_strlen_zero(c)) {
- ast_string_field_set(p, uri, c);
- }
-}
-
-/*!
- * \brief Determine if, as a UAS, we need to use a SIPS Contact.
- *
- * This uses the rules defined in RFC 3261 section 12.1.1 to
- * determine if a SIPS URI should be used as the Contact header
- * when responding to incoming SIP requests.
- *
- * \param req The incoming SIP request
- * \retval 0 SIPS is not required
- * \retval 1 SIPS is required
- */
-static int uas_sips_contact(struct sip_request *req)
-{
- const char *record_route = sip_get_header(req, "Record-Route");
-
- if (!strncmp(REQ_OFFSET_TO_STR(req, rlpart2), "sips:", 5)) {
- return 1;
- }
-
- if (record_route) {
- char *record_route_uri = get_in_brackets(ast_strdupa(record_route));
-
- if (!strncmp(record_route_uri, "sips:", 5)) {
- return 1;
- }
- } else {
- const char *contact = sip_get_header(req, "Contact");
- char *contact_uri = get_in_brackets(ast_strdupa(contact));
-
- if (!strncmp(contact_uri, "sips:", 5)) {
- return 1;
- }
- }
-
- return 0;
-}
-
-/*!
- * \brief Determine if, as a UAC, we need to use a SIPS Contact.
- *
- * This uses the rules defined in RFC 3621 section 8.1.1.8 to
- * determine if a SIPS URI should be used as the Contact header
- * on our outgoing request.
- *
- * \param req The outgoing SIP request
- * \retval 0 SIPS is not required
- * \retval 1 SIPS is required
- */
-static int uac_sips_contact(struct sip_request *req)
-{
- const char *route = sip_get_header(req, "Route");
-
- if (!strncmp(REQ_OFFSET_TO_STR(req, rlpart2), "sips:", 5)) {
- return 1;
- }
-
- if (route) {
- char *route_uri = get_in_brackets(ast_strdupa(route));
-
- if (!strncmp(route_uri, "sips:", 5)) {
- return 1;
- }
- }
-
- return 0;
-}
-
-/*!
- * \brief Build contact header
- *
- * This is the Contact header that we send out in SIP requests and responses
- * involving this sip_pvt.
- *
- * The incoming parameter is used to tell if we are building the request parameter
- * is an incoming SIP request that we are building the Contact header in response to,
- * or if the req parameter is an outbound SIP request that we will later be adding
- * the Contact header to.
- *
- * \param p The sip_pvt where the built Contact will be saved.
- * \param req The request that triggered the creation of a Contact header.
- * \param incoming Indicates if the Contact header is being created for a response to an incoming request
- */
-static void build_contact(struct sip_pvt *p, struct sip_request *req, int incoming)
-{
- char tmp[SIPBUFSIZE];
- char *user = ast_uri_encode(p->exten, tmp, sizeof(tmp), ast_uri_sip_user);
- int use_sips;
- char *transport = ast_strdupa(sip_get_transport(p->socket.type));
-
- if (incoming) {
- use_sips = uas_sips_contact(req);
- } else {
- use_sips = uac_sips_contact(req);
- }
-
- if (p->socket.type == AST_TRANSPORT_UDP) {
- ast_string_field_build(p, our_contact, "<%s:%s%s%s>", use_sips ? "sips" : "sip",
- user, ast_strlen_zero(user) ? "" : "@",
- ast_sockaddr_stringify_remote(&p->ourip));
- } else {
- ast_string_field_build(p, our_contact, "<%s:%s%s%s;transport=%s>",
- use_sips ? "sips" : "sip", user, ast_strlen_zero(user) ? "" : "@",
- ast_sockaddr_stringify_remote(&p->ourip), ast_str_to_lower(transport));
- }
-}
-
-/*! \brief Initiate new SIP request to peer/user */
-static void initreqprep(struct sip_request *req, struct sip_pvt *p, int sipmethod, const char * const explicit_uri)
-{
-#define SIPHEADER 256
- struct ast_str *invite = ast_str_create(SIPHEADER);
- struct ast_str *from = ast_str_create(SIPHEADER);
- struct ast_str *to = ast_str_create(SIPHEADER);
- char tmp_n[SIPBUFSIZE/2]; /* build a local copy of 'n' if needed */
- char tmp_l[SIPBUFSIZE/2]; /* build a local copy of 'l' if needed */
- const char *l = NULL; /* XXX what is this, exactly ? */
- const char *n = NULL; /* XXX what is this, exactly ? */
- const char *d = NULL; /* domain in from header */
- const char *urioptions = "";
- int ourport;
- int cid_has_name = 1;
- int cid_has_num = 1;
- struct ast_party_id connected_id;
- int ret;
-
- if (ast_test_flag(&p->flags[0], SIP_USEREQPHONE)) {
- const char *s = p->username; /* being a string field, cannot be NULL */
-
- /* Test p->username against allowed characters in AST_DIGIT_ANY
- If it matches the allowed characters list, then sipuser = ";user=phone"
- If not, then sipuser = ""
- */
- /* + is allowed in first position in a tel: uri */
- if (*s == '+')
- s++;
- for (; *s; s++) {
- if (!strchr(AST_DIGIT_ANYNUM, *s) )
- break;
- }
- /* If we have only digits, add ;user=phone to the uri */
- if (!*s)
- urioptions = ";user=phone";
- }
-
-
- snprintf(p->lastmsg, sizeof(p->lastmsg), "Init: %s", sip_methods[sipmethod].text);
-
- if (ast_strlen_zero(p->fromdomain)) {
- d = ast_sockaddr_stringify_host_remote(&p->ourip);
- }
- if (p->owner) {
- connected_id = ast_channel_connected_effective_id(p->owner);
-
- if ((ast_party_id_presentation(&connected_id) & AST_PRES_RESTRICTION) == AST_PRES_ALLOWED) {
- if (connected_id.number.valid) {
- l = connected_id.number.str;
- }
- if (connected_id.name.valid) {
- n = connected_id.name.str;
- }
- } else {
- /* Even if we are using RPID, we shouldn't leak information in the From if the user wants
- * their callerid restricted */
- l = "anonymous";
- n = CALLERID_UNKNOWN;
- d = FROMDOMAIN_INVALID;
- }
- }
-
- /* Hey, it's a NOTIFY! See if they've configured a mwi_from.
- * XXX Right now, this logic works because the only place that mwi_from
- * is set on the sip_pvt is in sip_send_mwi_to_peer. If things changed, then
- * we might end up putting the mwi_from setting into other types of NOTIFY
- * messages as well.
- */
- if (sipmethod == SIP_NOTIFY && !ast_strlen_zero(p->mwi_from)) {
- l = p->mwi_from;
- }
-
- if (ast_strlen_zero(l)) {
- cid_has_num = 0;
- l = default_callerid;
- }
- if (ast_strlen_zero(n)) {
- cid_has_name = 0;
- n = l;
- }
-
- /* Allow user to be overridden */
- if (!ast_strlen_zero(p->fromuser))
- l = p->fromuser;
- else /* Save for any further attempts */
- ast_string_field_set(p, fromuser, l);
-
- /* Allow user to be overridden */
- if (!ast_strlen_zero(p->fromname))
- n = p->fromname;
- else /* Save for any further attempts */
- ast_string_field_set(p, fromname, n);
-
- /* Allow domain to be overridden */
- if (!ast_strlen_zero(p->fromdomain))
- d = p->fromdomain;
- else /* Save for any further attempts */
- ast_string_field_set(p, fromdomain, d);
-
- ast_copy_string(tmp_l, l, sizeof(tmp_l));
- if (sip_cfg.pedanticsipchecking) {
- ast_uri_encode(l, tmp_l, sizeof(tmp_l), ast_uri_sip_user);
- }
-
- ourport = (p->fromdomainport && (p->fromdomainport != STANDARD_SIP_PORT)) ? p->fromdomainport : ast_sockaddr_port(&p->ourip);
-
- if (!sip_standard_port(p->socket.type, ourport)) {
- ret = ast_str_set(&from, 0, ";tag=%s", tmp_l, d, ourport, p->tag);
- } else {
- ret = ast_str_set(&from, 0, ";tag=%s", tmp_l, d, p->tag);
- }
- if (ret == AST_DYNSTR_BUILD_FAILED) {
- /* We don't have an escape path from here... */
- ast_log(LOG_ERROR, "The From header was truncated in call '%s'. This call setup will fail.\n", p->callid);
- /* Make sure that the field contains something non-broken.
- See https://issues.asterisk.org/jira/browse/ASTERISK-26069
- */
- ast_str_set(&from, 3, "<>");
-
- }
-
- /* If a caller id name was specified, prefix a display name, if there is enough room. */
- if (cid_has_name || !cid_has_num) {
- size_t written = ast_str_strlen(from);
- size_t name_len;
- if (sip_cfg.pedanticsipchecking) {
- ast_escape_quoted(n, tmp_n, sizeof(tmp_n));
- n = tmp_n;
- }
- name_len = strlen(n);
- ret = ast_str_make_space(&from, name_len + written + 4);
-
- if (ret == 0) {
- /* needed again, as ast_str_make_space coud've changed the pointer */
- char *from_buf = ast_str_buffer(from);
-
- memmove(from_buf + name_len + 3, from_buf, written + 1);
- from_buf[0] = '"';
- memcpy(from_buf + 1, n, name_len);
- from_buf[name_len + 1] = '"';
- from_buf[name_len + 2] = ' ';
- }
- }
-
- if (!ast_strlen_zero(explicit_uri)) {
- ast_str_set(&invite, 0, "%s", explicit_uri);
- } else {
- /* If we're calling a registered SIP peer, use the fullcontact to dial to the peer */
- if (!ast_strlen_zero(p->fullcontact)) {
- /* If we have full contact, trust it */
- ast_str_append(&invite, 0, "%s", p->fullcontact);
- } else {
- /* Otherwise, use the username while waiting for registration */
- ast_str_append(&invite, 0, "sip:");
- if (!ast_strlen_zero(p->username)) {
- n = p->username;
- if (sip_cfg.pedanticsipchecking) {
- ast_uri_encode(n, tmp_n, sizeof(tmp_n), ast_uri_sip_user);
- n = tmp_n;
- }
- ast_str_append(&invite, 0, "%s@", n);
- }
- ast_str_append(&invite, 0, "%s", p->tohost);
- if (p->portinuri) {
- ast_str_append(&invite, 0, ":%d", ast_sockaddr_port(&p->sa));
- }
- ast_str_append(&invite, 0, "%s", urioptions);
- }
- }
-
- /* If custom URI options have been provided, append them */
- if (p->options && !ast_strlen_zero(p->options->uri_options))
- ast_str_append(&invite, 0, ";%s", p->options->uri_options);
-
- /* This is the request URI, which is the next hop of the call
- which may or may not be the destination of the call
- */
- ast_string_field_set(p, uri, ast_str_buffer(invite));
-
- if (!ast_strlen_zero(p->todnid)) {
- /*! \todo Need to add back the VXML URL here at some point, possibly use build_string for all this junk */
- if (!strchr(p->todnid, '@')) {
- /* We have no domain in the dnid */
- ret = ast_str_set(&to, 0, "%s%s", p->todnid, p->tohost, ast_strlen_zero(p->theirtag) ? "" : ";tag=", p->theirtag);
- } else {
- ret = ast_str_set(&to, 0, "%s%s", p->todnid, ast_strlen_zero(p->theirtag) ? "" : ";tag=", p->theirtag);
- }
- } else {
- if (sipmethod == SIP_NOTIFY && !ast_strlen_zero(p->theirtag)) {
- /* If this is a NOTIFY, use the From: tag in the subscribe (RFC 3265) */
- ret = ast_str_set(&to, 0, "<%s%s>;tag=%s", (strncasecmp(p->uri, "sip:", 4) ? "sip:" : ""), p->uri, p->theirtag);
- } else if (p->options && p->options->vxml_url) {
- /* If there is a VXML URL append it to the SIP URL */
- ret = ast_str_set(&to, 0, "<%s>;%s", p->uri, p->options->vxml_url);
- } else {
- ret = ast_str_set(&to, 0, "<%s>", p->uri);
- }
- }
- if (ret == AST_DYNSTR_BUILD_FAILED) {
- /* We don't have an escape path from here... */
- ast_log(LOG_ERROR, "The To header was truncated in call '%s'. This call setup will fail.\n", p->callid);
- /* Make sure that the field contains something non-broken.
- See https://issues.asterisk.org/jira/browse/ASTERISK-26069
- */
- ast_str_set(&to, 3, "<>");
- }
-
- init_req(req, sipmethod, p->uri);
- /* now tmp_n is available so reuse it to build the CSeq */
- snprintf(tmp_n, sizeof(tmp_n), "%u %s", ++p->ocseq, sip_methods[sipmethod].text);
-
- add_header(req, "Via", p->via);
- add_max_forwards(p, req);
- /* This will be a no-op most of the time. However, under certain circumstances,
- * NOTIFY messages will use this function for preparing the request and should
- * have Route headers present.
- */
- add_route(req, &p->route, 0);
-
- add_header(req, "From", ast_str_buffer(from));
- add_header(req, "To", ast_str_buffer(to));
- ast_string_field_set(p, exten, l);
- build_contact(p, req, 0);
- add_header(req, "Contact", p->our_contact);
- add_header(req, "Call-ID", p->callid);
- add_header(req, "CSeq", tmp_n);
- if (!ast_strlen_zero(global_useragent)) {
- add_header(req, "User-Agent", global_useragent);
- }
-
- ast_free(from);
- ast_free(to);
- ast_free(invite);
-}
-
-/*! \brief Add "Diversion" header to outgoing message
- *
- * We need to add a Diversion header if the owner channel of
- * this dialog has redirecting information associated with it.
- *
- * \param req The request/response to which we will add the header
- * \param pvt The sip_pvt which represents the call-leg
- */
-static void add_diversion(struct sip_request *req, struct sip_pvt *pvt)
-{
- struct ast_party_id diverting_from;
- const char *reason;
- const char *quote_str;
- char header_text[256];
- char encoded_number[SIPBUFSIZE/2];
-
- /* We skip this entirely if the configuration doesn't allow diversion headers */
- if (!sip_cfg.send_diversion) {
- return;
- }
-
- if (!pvt->owner) {
- return;
- }
-
- diverting_from = ast_channel_redirecting_effective_from(pvt->owner);
- if (!diverting_from.number.valid
- || ast_strlen_zero(diverting_from.number.str)) {
- return;
- }
-
- if (sip_cfg.pedanticsipchecking) {
- ast_uri_encode(diverting_from.number.str, encoded_number, sizeof(encoded_number), ast_uri_sip_user);
- } else {
- ast_copy_string(encoded_number, diverting_from.number.str, sizeof(encoded_number));
- }
-
- reason = sip_reason_code_to_str(&ast_channel_redirecting(pvt->owner)->reason);
-
- /* Reason is either already quoted or it is a token to not need quotes added. */
- quote_str = *reason == '\"' || sip_is_token(reason) ? "" : "\"";
-
- /* We at least have a number to place in the Diversion header, which is enough */
- if (!diverting_from.name.valid
- || ast_strlen_zero(diverting_from.name.str)) {
- snprintf(header_text, sizeof(header_text), ";reason=%s%s%s",
- encoded_number,
- ast_sockaddr_stringify_host_remote(&pvt->ourip),
- quote_str, reason, quote_str);
- } else {
- char escaped_name[SIPBUFSIZE/2];
- if (sip_cfg.pedanticsipchecking) {
- ast_escape_quoted(diverting_from.name.str, escaped_name, sizeof(escaped_name));
- } else {
- ast_copy_string(escaped_name, diverting_from.name.str, sizeof(escaped_name));
- }
- snprintf(header_text, sizeof(header_text), "\"%s\" ;reason=%s%s%s",
- escaped_name,
- encoded_number,
- ast_sockaddr_stringify_host_remote(&pvt->ourip),
- quote_str, reason, quote_str);
- }
-
- add_header(req, "Diversion", header_text);
-}
-
-static int transmit_publish(struct sip_epa_entry *epa_entry, enum sip_publish_type publish_type, const char * const explicit_uri)
-{
- struct sip_pvt *pvt;
- int expires;
-
- epa_entry->publish_type = publish_type;
-
- if (!(pvt = sip_alloc(NULL, NULL, 0, SIP_PUBLISH, NULL, 0))) {
- return -1;
- }
-
- sip_pvt_lock(pvt);
-
- if (create_addr(pvt, epa_entry->destination, NULL, TRUE)) {
- sip_pvt_unlock(pvt);
- dialog_unlink_all(pvt);
- dialog_unref(pvt, "create_addr failed in transmit_publish. Unref dialog");
- return -1;
- }
- ast_sip_ouraddrfor(&pvt->sa, &pvt->ourip, pvt);
- ast_set_flag(&pvt->flags[0], SIP_OUTGOING);
- expires = (publish_type == SIP_PUBLISH_REMOVE) ? 0 : DEFAULT_PUBLISH_EXPIRES;
- pvt->expiry = expires;
-
- /* Bump refcount for sip_pvt's reference */
- ao2_ref(epa_entry, +1);
- pvt->epa_entry = epa_entry;
-
- transmit_invite(pvt, SIP_PUBLISH, FALSE, 2, explicit_uri);
- sip_pvt_unlock(pvt);
- sip_scheddestroy(pvt, DEFAULT_TRANS_TIMEOUT);
- dialog_unref(pvt, "Done with the sip_pvt allocated for transmitting PUBLISH");
- return 0;
-}
-
-/*!
- * \brief Build REFER/INVITE/OPTIONS/SUBSCRIBE message and transmit it
- * \param p sip_pvt structure
- * \param sipmethod
- * \param sdp unknown
- * \param init 0 = Prepare request within dialog, 1= prepare request, new branch,
- * 2= prepare new request and new dialog. do_proxy_auth calls this with init!=2
- * \param explicit_uri
-*/
-static int transmit_invite(struct sip_pvt *p, int sipmethod, int sdp, int init, const char * const explicit_uri)
-{
- struct sip_request req;
- struct ast_variable *var;
-
- if (init) {/* Bump branch even on initial requests */
- p->branch ^= ast_random();
- p->invite_branch = p->branch;
- build_via(p);
- }
- if (init > 1) {
- initreqprep(&req, p, sipmethod, explicit_uri);
- } else {
- /* If init=1, we should not generate a new branch. If it's 0, we need a new branch. */
- reqprep(&req, p, sipmethod, 0, init ? 0 : 1);
- }
-
- if (p->options && p->options->auth) {
- add_header(&req, p->options->authheader, p->options->auth);
- }
- add_date(&req);
- if (sipmethod == SIP_REFER && p->refer) { /* Call transfer */
- if (!ast_strlen_zero(p->refer->refer_to)) {
- add_header(&req, "Refer-To", p->refer->refer_to);
- }
- if (!ast_strlen_zero(p->refer->referred_by)) {
- add_header(&req, "Referred-By", p->refer->referred_by);
- }
- } else if (sipmethod == SIP_SUBSCRIBE) {
- if (p->subscribed == MWI_NOTIFICATION) {
- add_header(&req, "Event", "message-summary");
- add_header(&req, "Accept", "application/simple-message-summary");
- } else if (p->subscribed == CALL_COMPLETION) {
- add_header(&req, "Event", "call-completion");
- add_header(&req, "Accept", "application/call-completion");
- }
- add_expires(&req, p->expiry);
- }
-
- /* This new INVITE is part of an attended transfer. Make sure that the
- other end knows and replace the current call with this new call */
- if (p->options && !ast_strlen_zero(p->options->replaces)) {
- add_header(&req, "Replaces", p->options->replaces);
- add_header(&req, "Require", "replaces");
- }
-
- /* Add Session-Timers related headers if not already there */
- if (ast_strlen_zero(sip_get_header(&req, "Session-Expires")) &&
- (sipmethod == SIP_INVITE || sipmethod == SIP_UPDATE) &&
- (st_get_mode(p, 0) == SESSION_TIMER_MODE_ORIGINATE
- || (st_get_mode(p, 0) == SESSION_TIMER_MODE_ACCEPT
- && st_get_se(p, FALSE) != DEFAULT_MIN_SE))) {
- char i2astr[10];
-
- if (!p->stimer->st_interval) {
- p->stimer->st_interval = st_get_se(p, TRUE);
- }
-
- p->stimer->st_active = TRUE;
- if (st_get_mode(p, 0) == SESSION_TIMER_MODE_ORIGINATE) {
- snprintf(i2astr, sizeof(i2astr), "%d", p->stimer->st_interval);
- add_header(&req, "Session-Expires", i2astr);
- }
-
- snprintf(i2astr, sizeof(i2astr), "%d", st_get_se(p, FALSE));
- add_header(&req, "Min-SE", i2astr);
- }
-
- add_header(&req, "Allow", ALLOWED_METHODS);
- add_supported(p, &req);
-
- if (p->owner && ((p->options && p->options->addsipheaders)
- || (p->refer && global_refer_addheaders))) {
- struct ast_channel *chan = p->owner; /* The owner channel */
- struct varshead *headp;
-
- ast_channel_lock(chan);
-
- headp = ast_channel_varshead(chan);
-
- if (!headp) {
- ast_log(LOG_WARNING, "No Headp for the channel...ooops!\n");
- } else {
- const struct ast_var_t *current;
- AST_LIST_TRAVERSE(headp, current, entries) {
- /* SIPADDHEADER: Add SIP header to outgoing call */
- if (!strncmp(ast_var_name(current), "SIPADDHEADER", strlen("SIPADDHEADER"))) {
- char *content, *end;
- const char *header = ast_var_value(current);
- char *headdup = ast_strdupa(header);
-
- /* Strip of the starting " (if it's there) */
- if (*headdup == '"') {
- headdup++;
- }
- if ((content = strchr(headdup, ':'))) {
- *content++ = '\0';
- content = ast_skip_blanks(content); /* Skip white space */
- /* Strip the ending " (if it's there) */
- end = content + strlen(content) -1;
- if (*end == '"') {
- *end = '\0';
- }
-
- add_header(&req, headdup, content);
- if (sipdebug) {
- ast_debug(1, "Adding SIP Header \"%s\" with content :%s: \n", headdup, content);
- }
- }
- }
- }
- }
-
- ast_channel_unlock(chan);
- }
- if ((sipmethod == SIP_INVITE || sipmethod == SIP_UPDATE) && ast_test_flag(&p->flags[0], SIP_SENDRPID))
- add_rpid(&req, p);
- if (sipmethod == SIP_INVITE) {
- add_diversion(&req, p);
- }
- if (sdp) {
- offered_media_list_destroy(p);
- if (p->udptl && p->t38.state == T38_LOCAL_REINVITE) {
- ast_debug(1, "T38 is in state %u on channel %s\n", p->t38.state, p->owner ? ast_channel_name(p->owner) : "");
- add_sdp(&req, p, FALSE, FALSE, TRUE);
- } else if (p->rtp) {
- try_suggested_sip_codec(p);
- add_sdp(&req, p, FALSE, TRUE, FALSE);
- }
- } else if (sipmethod == SIP_NOTIFY && p->notify) {
- for (var = p->notify->headers; var; var = var->next) {
- add_header(&req, var->name, var->value);
- }
- if (ast_str_strlen(p->notify->content)) {
- add_content(&req, ast_str_buffer(p->notify->content));
- }
- } else if (sipmethod == SIP_PUBLISH) {
- switch (p->epa_entry->static_data->event) {
- case CALL_COMPLETION:
- add_header(&req, "Event", "call-completion");
- add_expires(&req, p->expiry);
- if (p->epa_entry->publish_type != SIP_PUBLISH_INITIAL) {
- add_header(&req, "SIP-If-Match", p->epa_entry->entity_tag);
- }
-
- if (!ast_strlen_zero(p->epa_entry->body)) {
- add_header(&req, "Content-Type", "application/pidf+xml");
- add_content(&req, p->epa_entry->body);
- }
- default:
- break;
- }
- }
-
- if (!p->initreq.headers || init > 2) {
- initialize_initreq(p, &req);
- }
- if (sipmethod == SIP_INVITE || sipmethod == SIP_SUBSCRIBE) {
- p->lastinvite = p->ocseq;
- }
- return send_request(p, &req, init ? XMIT_CRITICAL : XMIT_RELIABLE, p->ocseq);
-}
-
-/*!
- * \brief Send a subscription or resubscription for MWI
- *
- * \note Run by the sched thread.
- */
-static int sip_subscribe_mwi_do(const void *data)
-{
- struct sip_subscription_mwi *mwi = (struct sip_subscription_mwi *) data;
-
- mwi->resub = -1;
- __sip_subscribe_mwi_do(mwi);
- ao2_t_ref(mwi, -1, "Scheduled mwi resub complete");
-
- return 0;
-}
-
-/* Run by the sched thread. */
-static int __shutdown_mwi_subscription(const void *data)
-{
- struct sip_subscription_mwi *mwi = (void *) data;
-
- AST_SCHED_DEL_UNREF(sched, mwi->resub,
- ao2_t_ref(mwi, -1, "Stop scheduled mwi resub"));
-
- if (mwi->dnsmgr) {
- ast_dnsmgr_release(mwi->dnsmgr);
- mwi->dnsmgr = NULL;
- ao2_t_ref(mwi, -1, "dnsmgr release");
- }
-
- ao2_t_ref(mwi, -1, "Shutdown MWI subscription action");
- return 0;
-}
-
-static void shutdown_mwi_subscription(struct sip_subscription_mwi *mwi)
-{
- ao2_t_ref(mwi, +1, "Shutdown MWI subscription action");
- if (ast_sched_add(sched, 0, __shutdown_mwi_subscription, mwi) < 0) {
- /* Uh Oh. Expect bad behavior. */
- ao2_t_ref(mwi, -1, "Failed to schedule shutdown MWI subscription action");
- }
-}
-
-struct mwi_subscription_data {
- struct sip_subscription_mwi *mwi;
- int ms;
-};
-
-/* Run by the sched thread. */
-static int __start_mwi_subscription(const void *data)
-{
- struct mwi_subscription_data *sched_data = (void *) data;
- struct sip_subscription_mwi *mwi = sched_data->mwi;
- int ms = sched_data->ms;
-
- ast_free(sched_data);
-
- AST_SCHED_DEL_UNREF(sched, mwi->resub,
- ao2_t_ref(mwi, -1, "Stop scheduled mwi resub"));
-
- ao2_t_ref(mwi, +1, "Schedule mwi resub");
- mwi->resub = ast_sched_add(sched, ms, sip_subscribe_mwi_do, mwi);
- if (mwi->resub < 0) {
- /* Uh Oh. Expect bad behavior. */
- ao2_t_ref(mwi, -1, "Failed to schedule mwi resub");
- }
-
- ao2_t_ref(mwi, -1, "Start MWI subscription action");
- return 0;
-}
-
-static void start_mwi_subscription(struct sip_subscription_mwi *mwi, int ms)
-{
- struct mwi_subscription_data *sched_data;
-
- sched_data = ast_malloc(sizeof(*sched_data));
- if (!sched_data) {
- /* Uh Oh. Expect bad behavior. */
- return;
- }
- sched_data->mwi = mwi;
- sched_data->ms = ms;
- ao2_t_ref(mwi, +1, "Start MWI subscription action");
- if (ast_sched_add(sched, 0, __start_mwi_subscription, sched_data) < 0) {
- /* Uh Oh. Expect bad behavior. */
- ao2_t_ref(mwi, -1, "Failed to schedule start MWI subscription action");
- ast_free(sched_data);
- }
-}
-
-static void on_dns_update_registry(struct ast_sockaddr *old, struct ast_sockaddr *new, void *data)
-{
- struct sip_registry *reg = data;
- const char *old_str;
-
- /* This shouldn't happen, but just in case */
- if (ast_sockaddr_isnull(new)) {
- ast_debug(1, "Empty sockaddr change...ignoring!\n");
- return;
- }
-
- if (!ast_sockaddr_port(new)) {
- ast_sockaddr_set_port(new, reg->portno);
- }
-
- old_str = ast_strdupa(ast_sockaddr_stringify(old));
-
- ast_debug(1, "Changing registry %s from %s to %s\n", S_OR(reg->peername, reg->hostname), old_str, ast_sockaddr_stringify(new));
- ast_sockaddr_copy(®->us, new);
-}
-
-static void on_dns_update_peer(struct ast_sockaddr *old, struct ast_sockaddr *new, void *data)
-{
- struct sip_peer *peer = data;
- const char *old_str;
-
- /* This shouldn't happen, but just in case */
- if (ast_sockaddr_isnull(new)) {
- ast_debug(1, "Empty sockaddr change...ignoring!\n");
- return;
- }
-
- if (!ast_sockaddr_isnull(&peer->addr)) {
- ao2_unlink(peers_by_ip, peer);
- }
-
- if (!ast_sockaddr_port(new)) {
- ast_sockaddr_set_port(new, default_sip_port(peer->socket.type));
- }
-
- old_str = ast_strdupa(ast_sockaddr_stringify(old));
- ast_debug(1, "Changing peer %s address from %s to %s\n", peer->name, old_str, ast_sockaddr_stringify(new));
-
- ao2_lock(peer);
- ast_sockaddr_copy(&peer->addr, new);
- ao2_unlock(peer);
-
- ao2_link(peers_by_ip, peer);
-}
-
-static void on_dns_update_mwi(struct ast_sockaddr *old, struct ast_sockaddr *new, void *data)
-{
- struct sip_subscription_mwi *mwi = data;
- const char *old_str;
-
- /* This shouldn't happen, but just in case */
- if (ast_sockaddr_isnull(new)) {
- ast_debug(1, "Empty sockaddr change...ignoring!\n");
- return;
- }
-
- old_str = ast_strdupa(ast_sockaddr_stringify(old));
- ast_debug(1, "Changing mwi %s from %s to %s\n", mwi->hostname, old_str, ast_sockaddr_stringify(new));
- ast_sockaddr_copy(&mwi->us, new);
-}
-
-/*! \brief Actually setup an MWI subscription or resubscribe */
-static int __sip_subscribe_mwi_do(struct sip_subscription_mwi *mwi)
-{
- /* If we have no DNS manager let's do a lookup */
- if (!mwi->dnsmgr) {
- char transport[MAXHOSTNAMELEN];
- snprintf(transport, sizeof(transport), "_%s._%s", get_srv_service(mwi->transport), get_srv_protocol(mwi->transport));
-
- mwi->us.ss.ss_family = get_address_family_filter(mwi->transport); /* Filter address family */
- ao2_t_ref(mwi, +1, "dnsmgr reference to mwi");
- ast_dnsmgr_lookup_cb(mwi->hostname, &mwi->us, &mwi->dnsmgr, sip_cfg.srvlookup ? transport : NULL, on_dns_update_mwi, mwi);
- if (!mwi->dnsmgr) {
- ao2_t_ref(mwi, -1, "dnsmgr disabled, remove reference");
- }
- }
-
- /* If we already have a subscription up simply send a resubscription */
- if (mwi->call) {
- transmit_invite(mwi->call, SIP_SUBSCRIBE, 0, 0, NULL);
- return 0;
- }
-
- /* Create a dialog that we will use for the subscription */
- if (!(mwi->call = sip_alloc(NULL, NULL, 0, SIP_SUBSCRIBE, NULL, 0))) {
- return -1;
- }
-
- ref_proxy(mwi->call, obproxy_get(mwi->call, NULL));
-
- if (!ast_sockaddr_port(&mwi->us) && mwi->portno) {
- ast_sockaddr_set_port(&mwi->us, mwi->portno);
- }
-
- /* Setup the destination of our subscription */
- if (create_addr(mwi->call, mwi->hostname, &mwi->us, 0)) {
- dialog_unlink_all(mwi->call);
- mwi->call = dialog_unref(mwi->call, "unref dialog after unlink_all");
- return 0;
- }
-
- mwi->call->expiry = mwi_expiry;
-
- if (!mwi->dnsmgr && mwi->portno) {
- ast_sockaddr_set_port(&mwi->call->sa, mwi->portno);
- ast_sockaddr_set_port(&mwi->call->recv, mwi->portno);
- } else {
- mwi->portno = ast_sockaddr_port(&mwi->call->sa);
- }
-
- /* Set various other information */
- if (!ast_strlen_zero(mwi->authuser)) {
- ast_string_field_set(mwi->call, peername, mwi->authuser);
- ast_string_field_set(mwi->call, authname, mwi->authuser);
- ast_string_field_set(mwi->call, fromuser, mwi->authuser);
- } else {
- ast_string_field_set(mwi->call, peername, mwi->username);
- ast_string_field_set(mwi->call, authname, mwi->username);
- ast_string_field_set(mwi->call, fromuser, mwi->username);
- }
- ast_string_field_set(mwi->call, username, mwi->username);
- if (!ast_strlen_zero(mwi->secret)) {
- ast_string_field_set(mwi->call, peersecret, mwi->secret);
- }
- set_socket_transport(&mwi->call->socket, mwi->transport);
- ast_sip_ouraddrfor(&mwi->call->sa, &mwi->call->ourip, mwi->call);
- build_via(mwi->call);
-
- /* Change the dialog callid. */
- change_callid_pvt(mwi->call, NULL);
-
- ast_set_flag(&mwi->call->flags[0], SIP_OUTGOING);
-
- /* Associate the call with us */
- mwi->call->mwi = ao2_t_bump(mwi, "Reference mwi from it's call");
-
- mwi->call->subscribed = MWI_NOTIFICATION;
-
- /* Actually send the packet */
- transmit_invite(mwi->call, SIP_SUBSCRIBE, 0, 2, NULL);
-
- return 0;
-}
-
-/*!
- * \internal
- * \brief Find the channel that is causing the RINGING update, ref'd
- */
-static struct ast_channel *find_ringing_channel(struct ao2_container *device_state_info, struct sip_pvt *p)
-{
- struct ao2_iterator citer;
- struct ast_device_state_info *device_state;
- struct ast_channel *c = NULL;
- struct timeval tv = {0,};
-
- /* iterate ringing devices and get the oldest of all causing channels */
- citer = ao2_iterator_init(device_state_info, 0);
- for (; (device_state = ao2_iterator_next(&citer)); ao2_ref(device_state, -1)) {
- if (!device_state->causing_channel || (device_state->device_state != AST_DEVICE_RINGING &&
- device_state->device_state != AST_DEVICE_RINGINUSE)) {
- continue;
- }
- ast_channel_lock(device_state->causing_channel);
- if (ast_tvzero(tv) || ast_tvcmp(ast_channel_creationtime(device_state->causing_channel), tv) < 0) {
- c = device_state->causing_channel;
- tv = ast_channel_creationtime(c);
- }
- ast_channel_unlock(device_state->causing_channel);
- }
- ao2_iterator_destroy(&citer);
- return c ? ast_channel_ref(c) : NULL;
-}
-
-/* XXX Candidate for moving into its own file */
-static int allow_notify_user_presence(struct sip_pvt *p)
-{
- return (strstr(p->useragent, "Digium")) ? 1 : 0;
-}
-
-/*! \brief Builds XML portion of NOTIFY messages for presence or dialog updates */
-static void state_notify_build_xml(struct state_notify_data *data, int full, const char *exten, const char *context, struct ast_str **tmp, struct sip_pvt *p, int subscribed, const char *mfrom, const char *mto)
-{
- enum state { NOTIFY_OPEN, NOTIFY_INUSE, NOTIFY_CLOSED } local_state = NOTIFY_OPEN;
- const char *statestring = "terminated";
- const char *pidfstate = "--";
- const char *pidfnote ="Ready";
- char hint[AST_MAX_EXTENSION];
-
- switch (data->state) {
- case (AST_EXTENSION_RINGING | AST_EXTENSION_INUSE):
- statestring = (sip_cfg.notifyringing == NOTIFYRINGING_ENABLED) ? "early" : "confirmed";
- local_state = NOTIFY_INUSE;
- pidfstate = "busy";
- pidfnote = "Ringing";
- break;
- case AST_EXTENSION_RINGING:
- statestring = "early";
- local_state = NOTIFY_INUSE;
- pidfstate = "busy";
- pidfnote = "Ringing";
- break;
- case AST_EXTENSION_INUSE:
- statestring = "confirmed";
- local_state = NOTIFY_INUSE;
- pidfstate = "busy";
- pidfnote = "On the phone";
- break;
- case AST_EXTENSION_BUSY:
- statestring = "confirmed";
- local_state = NOTIFY_CLOSED;
- pidfstate = "busy";
- pidfnote = "On the phone";
- break;
- case AST_EXTENSION_UNAVAILABLE:
- statestring = "terminated";
- local_state = NOTIFY_CLOSED;
- pidfstate = "away";
- pidfnote = "Unavailable";
- break;
- case AST_EXTENSION_ONHOLD:
- statestring = "confirmed";
- local_state = NOTIFY_CLOSED;
- pidfstate = "busy";
- pidfnote = "On hold";
- break;
- case AST_EXTENSION_NOT_INUSE:
- default:
- /* Default setting */
- break;
- }
-
- /* Check which device/devices we are watching and if they are registered */
- if (ast_get_hint(hint, sizeof(hint), NULL, 0, NULL, context, exten)) {
- char *hint2;
- char *individual_hint = NULL;
- int hint_count = 0, unavailable_count = 0;
-
- /* strip off any possible PRESENCE providers from hint */
- if ((hint2 = strrchr(hint, ','))) {
- *hint2 = '\0';
- }
- hint2 = hint;
-
- while ((individual_hint = strsep(&hint2, "&"))) {
- hint_count++;
-
- if (ast_device_state(individual_hint) == AST_DEVICE_UNAVAILABLE)
- unavailable_count++;
- }
-
- /* If none of the hinted devices are registered, we will
- * override notification and show no availability.
- */
- if (hint_count > 0 && hint_count == unavailable_count) {
- local_state = NOTIFY_CLOSED;
- pidfstate = "away";
- pidfnote = "Not online";
- }
- }
-
- switch (subscribed) {
- case XPIDF_XML:
- case CPIM_PIDF_XML:
- ast_str_append(tmp, 0,
- "\n"
- "\n"
- "\n");
- ast_str_append(tmp, 0, "\n", exten);
- ast_str_append(tmp, 0, "\n", mto);
- ast_str_append(tmp, 0, " \n \n \n");
- break;
- case PIDF_XML: /* Eyebeam supports this format */
- ast_str_append(tmp, 0,
- "\n"
- "\n", mfrom);
- ast_str_append(tmp, 0, "\n");
- if (pidfstate[0] != '-') {
- ast_str_append(tmp, 0, " %s \n", pidfnote); /* Note */
- ast_str_append(tmp, 0, "\n", exten); /* Tuple start */
- ast_str_append(tmp, 0, "%s \n", mto);
- if (pidfstate[0] == 'b') /* Busy? Still open ... */
- ast_str_append(tmp, 0, "open %s \n");
- ast_str_append(tmp, 0, "\n");
- ast_str_append(tmp, 0, "\n");
- ast_str_append(tmp, 0, "%s \n",
- ast_presence_state2str(data->presence_state),
- S_OR(data->presence_subtype, ""),
- S_OR(data->presence_message, ""));
- ast_str_append(tmp, 0, " \n");
- ast_test_suite_event_notify("DIGIUM_PRESENCE_SENT",
- "PresenceState: %s\r\n"
- "Subtype: %s\r\n"
- "Message: %s",
- ast_presence_state2str(data->presence_state),
- S_OR(data->presence_subtype, ""),
- S_OR(data->presence_message, ""));
- }
- ast_str_append(tmp, 0, " \n \n");
- break;
- case DIALOG_INFO_XML: /* SNOM subscribes in this format */
- ast_str_append(tmp, 0, "\n");
- ast_str_append(tmp, 0, "\n", p->dialogver, full ? "full" : "partial", mto);
- if (data->state > 0 && (data->state & AST_EXTENSION_RINGING) && sip_cfg.notifyringing) {
- /* Twice the extension length should be enough for XML encoding */
- char local_display[AST_MAX_EXTENSION * 2];
- char remote_display[AST_MAX_EXTENSION * 2];
- char *local_target = ast_strdupa(mto);
- /* It may seem odd to base the remote_target on the To header here,
- * but testing by reporters on issue ASTERISK-16735 found that basing
- * on the From header would cause ringing state hints to not work
- * properly on certain SNOM devices. If you are using notifycid properly
- * (i.e. in the same extension and context as the dialed call) then this
- * should not be an issue since the data will be overwritten shortly
- * with channel caller ID
- */
- char *remote_target = ast_strdupa(mto);
-
- ast_xml_escape(exten, local_display, sizeof(local_display));
- ast_xml_escape(exten, remote_display, sizeof(remote_display));
-
- /* There are some limitations to how this works. The primary one is that the
- callee must be dialing the same extension that is being monitored. Simply dialing
- the hint'd device is not sufficient. */
- if (sip_cfg.notifycid) {
- struct ast_channel *callee;
-
- callee = find_ringing_channel(data->device_state_info, p);
- if (callee) {
- static char *anonymous = "anonymous";
- static char *invalid = "anonymous.invalid";
- char *cid_num;
- char *connected_num;
- int need;
- int cid_num_restricted, connected_num_restricted;
-
- ast_channel_lock(callee);
-
- cid_num_restricted = (ast_channel_caller(callee)->id.number.presentation &
- AST_PRES_RESTRICTION) == AST_PRES_RESTRICTED;
- cid_num = S_COR(ast_channel_caller(callee)->id.number.valid,
- S_COR(cid_num_restricted, anonymous,
- ast_channel_caller(callee)->id.number.str), "");
-
- need = strlen(cid_num) + (cid_num_restricted ? strlen(invalid) :
- strlen(p->fromdomain)) + sizeof("sip:@");
- local_target = ast_alloca(need);
-
- snprintf(local_target, need, "sip:%s@%s", cid_num,
- cid_num_restricted ? invalid : p->fromdomain);
-
- ast_xml_escape(S_COR(ast_channel_caller(callee)->id.name.valid,
- S_COR((ast_channel_caller(callee)->id.name.presentation &
- AST_PRES_RESTRICTION) == AST_PRES_RESTRICTED, anonymous,
- ast_channel_caller(callee)->id.name.str), ""),
- local_display, sizeof(local_display));
-
- connected_num_restricted = (ast_channel_connected(callee)->id.number.presentation &
- AST_PRES_RESTRICTION) == AST_PRES_RESTRICTED;
- connected_num = S_COR(ast_channel_connected(callee)->id.number.valid,
- S_COR(connected_num_restricted, anonymous,
- ast_channel_connected(callee)->id.number.str), "");
-
- need = strlen(connected_num) + (connected_num_restricted ? strlen(invalid) :
- strlen(p->fromdomain)) + sizeof("sip:@");
- remote_target = ast_alloca(need);
-
- snprintf(remote_target, need, "sip:%s@%s", connected_num,
- connected_num_restricted ? invalid : p->fromdomain);
-
- ast_xml_escape(S_COR(ast_channel_connected(callee)->id.name.valid,
- S_COR((ast_channel_connected(callee)->id.name.presentation &
- AST_PRES_RESTRICTION) == AST_PRES_RESTRICTED, anonymous,
- ast_channel_connected(callee)->id.name.str), ""),
- remote_display, sizeof(remote_display));
-
- ast_channel_unlock(callee);
- callee = ast_channel_unref(callee);
- }
-
- /* We create a fake call-id which the phone will send back in an INVITE
- Replaces header which we can grab and do some magic with. */
- if (sip_cfg.pedanticsipchecking) {
- ast_str_append(tmp, 0, "\n",
- exten, p->callid, p->theirtag, p->tag);
- } else {
- ast_str_append(tmp, 0, "\n",
- exten, p->callid);
- }
- ast_str_append(tmp, 0,
- "\n"
- /* See the limitations of this above. Luckily the phone seems to still be
- happy when these values are not correct. */
- "%s \n"
- "\n"
- " \n"
- "\n"
- "%s \n"
- "\n"
- " \n",
- remote_display, remote_target, remote_target, local_display, local_target, local_target);
- } else {
- ast_str_append(tmp, 0, "\n", exten);
- }
-
- } else {
- ast_str_append(tmp, 0, "\n", exten);
- }
- ast_str_append(tmp, 0, "%s \n", statestring);
- if (data->state == AST_EXTENSION_ONHOLD) {
- ast_str_append(tmp, 0, "\n\n"
- " \n \n", mto);
- }
- ast_str_append(tmp, 0, " \n \n");
- break;
- case NONE:
- default:
- break;
- }
-}
-
-static int transmit_cc_notify(struct ast_cc_agent *agent, struct sip_pvt *subscription, enum sip_cc_notify_state state)
-{
- struct sip_request req;
- struct sip_cc_agent_pvt *agent_pvt = agent->private_data;
- char uri[SIPBUFSIZE + sizeof("cc-URI: \r\n") - 1];
- char state_str[64];
- char subscription_state_hdr[64];
-
- if (state < CC_QUEUED || state > CC_READY) {
- ast_log(LOG_WARNING, "Invalid state provided for transmit_cc_notify (%u)\n", state);
- return -1;
- }
-
- reqprep(&req, subscription, SIP_NOTIFY, 0, TRUE);
- snprintf(state_str, sizeof(state_str), "%s\r\n", sip_cc_notify_state_map[state].state_string);
- add_header(&req, "Event", "call-completion");
- add_header(&req, "Content-Type", "application/call-completion");
- snprintf(subscription_state_hdr, sizeof(subscription_state_hdr), "active;expires=%d", subscription->expiry);
- add_header(&req, "Subscription-State", subscription_state_hdr);
- if (state == CC_READY) {
- generate_uri(subscription, agent_pvt->notify_uri, sizeof(agent_pvt->notify_uri));
- snprintf(uri, sizeof(uri), "cc-URI: %s\r\n", agent_pvt->notify_uri);
- }
- add_content(&req, state_str);
- if (state == CC_READY) {
- add_content(&req, uri);
- }
- return send_request(subscription, &req, XMIT_RELIABLE, subscription->ocseq);
-}
-
-/*! \brief Used in the SUBSCRIBE notification subsystem (RFC3265) */
-static int transmit_state_notify(struct sip_pvt *p, struct state_notify_data *data, int full, int timeout)
-{
- struct ast_str *tmp = ast_str_alloca(4000);
- char from[256], to[256];
- char *c, *mfrom, *mto;
- struct sip_request req;
- const struct cfsubscription_types *subscriptiontype;
-
- /* If the subscription has not yet been accepted do not send a NOTIFY */
- if (!ast_test_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED)) {
- return 0;
- }
-
- memset(from, 0, sizeof(from));
- memset(to, 0, sizeof(to));
-
- subscriptiontype = find_subscription_type(p->subscribed);
-
- ast_copy_string(from, sip_get_header(&p->initreq, "From"), sizeof(from));
- c = get_in_brackets(from);
- if (strncasecmp(c, "sip:", 4) && strncasecmp(c, "sips:", 5)) {
- ast_log(LOG_WARNING, "Huh? Not a SIP header (%s)?\n", c);
- return -1;
- }
-
- mfrom = remove_uri_parameters(c);
-
- ast_copy_string(to, sip_get_header(&p->initreq, "To"), sizeof(to));
- c = get_in_brackets(to);
- if (strncasecmp(c, "sip:", 4) && strncasecmp(c, "sips:", 5)) {
- ast_log(LOG_WARNING, "Huh? Not a SIP header (%s)?\n", c);
- return -1;
- }
- mto = remove_uri_parameters(c);
-
- reqprep(&req, p, SIP_NOTIFY, 0, 1);
-
- switch(data->state) {
- case AST_EXTENSION_DEACTIVATED:
- if (timeout)
- add_header(&req, "Subscription-State", "terminated;reason=timeout");
- else {
- add_header(&req, "Subscription-State", "terminated;reason=probation");
- add_header(&req, "Retry-After", "60");
- }
- break;
- case AST_EXTENSION_REMOVED:
- add_header(&req, "Subscription-State", "terminated;reason=noresource");
- break;
- default:
- if (p->expiry)
- add_header(&req, "Subscription-State", "active");
- else /* Expired */
- add_header(&req, "Subscription-State", "terminated;reason=timeout");
- }
-
- switch (p->subscribed) {
- case XPIDF_XML:
- case CPIM_PIDF_XML:
- add_header(&req, "Event", subscriptiontype->event);
- state_notify_build_xml(data, full, p->exten, p->context, &tmp, p, p->subscribed, mfrom, mto);
- add_header(&req, "Content-Type", subscriptiontype->mediatype);
- p->dialogver++;
- break;
- case PIDF_XML: /* Eyebeam supports this format */
- add_header(&req, "Event", subscriptiontype->event);
- state_notify_build_xml(data, full, p->exten, p->context, &tmp, p, p->subscribed, mfrom, mto);
- add_header(&req, "Content-Type", subscriptiontype->mediatype);
- p->dialogver++;
- break;
- case DIALOG_INFO_XML: /* SNOM subscribes in this format */
- add_header(&req, "Event", subscriptiontype->event);
- state_notify_build_xml(data, full, p->exten, p->context, &tmp, p, p->subscribed, mfrom, mto);
- add_header(&req, "Content-Type", subscriptiontype->mediatype);
- p->dialogver++;
- break;
- case NONE:
- default:
- break;
- }
-
- add_content(&req, ast_str_buffer(tmp));
-
- p->pendinginvite = p->ocseq; /* Remember that we have a pending NOTIFY in order not to confuse the NOTIFY subsystem */
-
- /* Send as XMIT_CRITICAL as we may never receive a 200 OK Response which clears p->pendinginvite.
- *
- * extensionstate_update() uses p->pendinginvite for queuing control.
- * Updates stall if pendinginvite <> 0.
- *
- * The most appropriate solution is to remove the subscription when the NOTIFY transaction fails.
- * The client will re-subscribe after restarting or maxexpiry timeout.
- */
-
- /* RFC6665 4.2.2. Sending State Information to Subscribers
- * If the NOTIFY request fails due to expiration of SIP Timer F (transaction timeout),
- * the notifier SHOULD remove the subscription.
- */
- return send_request(p, &req, XMIT_CRITICAL, p->ocseq);
-}
-
-/*! \brief Notify user of messages waiting in voicemail (RFC3842)
-\note - Notification only works for registered peers with mailbox= definitions
- in sip.conf
- - We use the SIP Event package message-summary
- MIME type defaults to "application/simple-message-summary";
- */
-static int transmit_notify_with_mwi(struct sip_pvt *p, int newmsgs, int oldmsgs, const char *vmexten)
-{
- struct sip_request req;
- struct ast_str *out = ast_str_alloca(500);
- int ourport = (p->fromdomainport && (p->fromdomainport != STANDARD_SIP_PORT)) ? p->fromdomainport : ast_sockaddr_port(&p->ourip);
- const char *domain;
- const char *exten = S_OR(vmexten, default_vmexten);
-
- initreqprep(&req, p, SIP_NOTIFY, NULL);
- add_header(&req, "Event", "message-summary");
- add_header(&req, "Content-Type", default_notifymime);
- ast_str_append(&out, 0, "Messages-Waiting: %s\r\n", newmsgs ? "yes" : "no");
-
- /* domain initialization occurs here because initreqprep changes ast_sockaddr_stringify string. */
- domain = S_OR(p->fromdomain, ast_sockaddr_stringify_host_remote(&p->ourip));
-
- if (!sip_standard_port(p->socket.type, ourport)) {
- if (p->socket.type == AST_TRANSPORT_UDP) {
- ast_str_append(&out, 0, "Message-Account: sip:%s@%s:%d\r\n", exten, domain, ourport);
- } else {
- ast_str_append(&out, 0, "Message-Account: sip:%s@%s:%d;transport=%s\r\n", exten, domain, ourport, sip_get_transport(p->socket.type));
- }
- } else {
- if (p->socket.type == AST_TRANSPORT_UDP) {
- ast_str_append(&out, 0, "Message-Account: sip:%s@%s\r\n", exten, domain);
- } else {
- ast_str_append(&out, 0, "Message-Account: sip:%s@%s;transport=%s\r\n", exten, domain, sip_get_transport(p->socket.type));
- }
- }
- /* Cisco has a bug in the SIP stack where it can't accept the
- (0/0) notification. This can temporarily be disabled in
- sip.conf with the "buggymwi" option */
- ast_str_append(&out, 0, "Voice-Message: %d/%d%s\r\n",
- newmsgs, oldmsgs, (ast_test_flag(&p->flags[1], SIP_PAGE2_BUGGY_MWI) ? "" : " (0/0)"));
-
- if (p->subscribed) {
- if (p->expiry) {
- add_header(&req, "Subscription-State", "active");
- } else { /* Expired */
- add_header(&req, "Subscription-State", "terminated;reason=timeout");
- }
- }
-
- add_content(&req, ast_str_buffer(out));
-
- if (!p->initreq.headers) {
- initialize_initreq(p, &req);
- }
- return send_request(p, &req, XMIT_RELIABLE, p->ocseq);
-}
-
-/*! \brief Notify a transferring party of the status of transfer (RFC3515) */
-static int transmit_notify_with_sipfrag(struct sip_pvt *p, int cseq, char *message, int terminate)
-{
- struct sip_request req;
- char tmp[SIPBUFSIZE/2];
-
- reqprep(&req, p, SIP_NOTIFY, 0, 1);
- snprintf(tmp, sizeof(tmp), "refer;id=%d", cseq);
- add_header(&req, "Event", tmp);
- add_header(&req, "Subscription-state", terminate ? "terminated;reason=noresource" : "active");
- add_header(&req, "Content-Type", "message/sipfrag;version=2.0");
- add_header(&req, "Allow", ALLOWED_METHODS);
- add_supported(p, &req);
-
- snprintf(tmp, sizeof(tmp), "SIP/2.0 %s\r\n", message);
- add_content(&req, tmp);
-
- if (!p->initreq.headers) {
- initialize_initreq(p, &req);
- }
-
- return send_request(p, &req, XMIT_RELIABLE, p->ocseq);
-}
-
-static int manager_sipnotify(struct mansession *s, const struct message *m)
-{
- const char *channame = astman_get_header(m, "Channel");
- struct ast_variable *vars = astman_get_variables_order(m, ORDER_NATURAL);
- const char *callid = astman_get_header(m, "Call-ID");
- struct sip_pvt *p;
- struct ast_variable *header, *var;
-
- if (ast_strlen_zero(channame)) {
- astman_send_error(s, m, "SIPNotify requires a channel name");
- ast_variables_destroy(vars);
- return 0;
- }
-
- if (!strncasecmp(channame, "sip/", 4)) {
- channame += 4;
- }
-
- /* check if Call-ID header is set */
- if (!ast_strlen_zero(callid)) {
- struct sip_pvt tmp_dialog = {
- .callid = callid,
- };
-
- p = ao2_find(dialogs, &tmp_dialog, OBJ_SEARCH_OBJECT);
- if (!p) {
- astman_send_error(s, m, "Call-ID not found");
- ast_variables_destroy(vars);
- return 0;
- }
-
- if (!(p->notify)) {
- sip_notify_alloc(p);
- } else {
- ast_variables_destroy(p->notify->headers);
- }
- } else {
- if (!(p = sip_alloc(NULL, NULL, 0, SIP_NOTIFY, NULL, 0))) {
- astman_send_error(s, m, "Unable to build sip pvt data for notify (memory/socket error)");
- ast_variables_destroy(vars);
- return 0;
- }
-
- if (create_addr(p, channame, NULL, 1)) {
- /* Maybe they're not registered, etc. */
- dialog_unlink_all(p);
- dialog_unref(p, "unref dialog inside for loop" );
- /* sip_destroy(p); */
- astman_send_error(s, m, "Could not create address");
- ast_variables_destroy(vars);
- return 0;
- }
-
- /* Notify is outgoing call */
- ast_set_flag(&p->flags[0], SIP_OUTGOING);
- sip_notify_alloc(p);
-
- }
-
- p->notify->headers = header = ast_variable_new("Subscription-State", "terminated", "");
-
- for (var = vars; var; var = var->next) {
- if (!strcasecmp(var->name, "Content")) {
- if (ast_str_strlen(p->notify->content))
- ast_str_append(&p->notify->content, 0, "\r\n");
- ast_str_append(&p->notify->content, 0, "%s", var->value);
- } else if (!strcasecmp(var->name, "Content-Length")) {
- ast_log(LOG_WARNING, "it is not necessary to specify Content-Length, ignoring\n");
- } else {
- header->next = ast_variable_new(var->name, var->value, "");
- header = header->next;
- }
- }
-
- if (ast_strlen_zero(callid)) {
- /* Now that we have the peer's address, set our ip and change callid */
- ast_sip_ouraddrfor(&p->sa, &p->ourip, p);
- build_via(p);
-
- change_callid_pvt(p, NULL);
-
- sip_scheddestroy(p, SIP_TRANS_TIMEOUT);
- transmit_invite(p, SIP_NOTIFY, 0, 2, NULL);
- } else {
- sip_scheddestroy(p, SIP_TRANS_TIMEOUT);
- transmit_invite(p, SIP_NOTIFY, 0, 1, NULL);
- }
- dialog_unref(p, "bump down the count of p since we're done with it.");
-
- astman_send_ack(s, m, "Notify Sent");
- ast_variables_destroy(vars);
- return 0;
-}
-
-/*! \brief Send a provisional response indicating that a call was redirected
- */
-static void update_redirecting(struct sip_pvt *p, const void *data, size_t datalen)
-{
- struct sip_request resp;
-
- if (ast_channel_state(p->owner) == AST_STATE_UP || ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
- return;
- }
-
- respprep(&resp, p, "181 Call is being forwarded", &p->initreq);
- add_diversion(&resp, p);
- send_response(p, &resp, XMIT_UNRELIABLE, 0);
-}
-
-/*! \brief Notify peer that the connected line has changed */
-static void update_connectedline(struct sip_pvt *p, const void *data, size_t datalen)
-{
- struct ast_party_id connected_id = ast_channel_connected_effective_id(p->owner);
-
- if (!ast_test_flag(&p->flags[0], SIP_SENDRPID)) {
- return;
- }
- if (!connected_id.number.valid
- || ast_strlen_zero(connected_id.number.str)) {
- return;
- }
-
- append_history(p, "ConnectedLine", "%s party is now %s <%s>",
- ast_test_flag(&p->flags[0], SIP_OUTGOING) ? "Calling" : "Called",
- S_COR(connected_id.name.valid, connected_id.name.str, ""),
- S_COR(connected_id.number.valid, connected_id.number.str, ""));
-
- if (ast_channel_state(p->owner) == AST_STATE_UP || ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
- struct sip_request req;
-
- if (!p->pendinginvite && (p->invitestate == INV_CONFIRMED || p->invitestate == INV_TERMINATED)) {
- reqprep(&req, p, ast_test_flag(&p->flags[0], SIP_REINVITE_UPDATE) ? SIP_UPDATE : SIP_INVITE, 0, 1);
-
- add_header(&req, "Allow", ALLOWED_METHODS);
- add_supported(p, &req);
- add_rpid(&req, p);
- add_sdp(&req, p, FALSE, TRUE, FALSE);
-
- initialize_initreq(p, &req);
- p->lastinvite = p->ocseq;
- ast_set_flag(&p->flags[0], SIP_OUTGOING);
- send_request(p, &req, XMIT_CRITICAL, p->ocseq);
- } else if ((is_method_allowed(&p->allowed_methods, SIP_UPDATE)) && (!ast_strlen_zero(p->okcontacturi))) {
- reqprep(&req, p, SIP_UPDATE, 0, 1);
- add_rpid(&req, p);
- add_header(&req, "X-Asterisk-rpid-update", "Yes");
- send_request(p, &req, XMIT_CRITICAL, p->ocseq);
- } else {
- /* We cannot send the update yet, so we have to wait until we can */
- ast_set_flag(&p->flags[0], SIP_NEEDREINVITE);
- }
- } else {
- ast_set_flag(&p->flags[1], SIP_PAGE2_CONNECTLINEUPDATE_PEND);
- if (ast_test_flag(&p->flags[1], SIP_PAGE2_RPID_IMMEDIATE)) {
- struct sip_request resp;
-
- if ((ast_channel_state(p->owner) == AST_STATE_RING) && !ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT)) {
- ast_clear_flag(&p->flags[1], SIP_PAGE2_CONNECTLINEUPDATE_PEND);
- respprep(&resp, p, "180 Ringing", &p->initreq);
- add_rpid(&resp, p);
- send_response(p, &resp, XMIT_UNRELIABLE, 0);
- ast_set_flag(&p->flags[0], SIP_RINGING);
- } else if (ast_channel_state(p->owner) == AST_STATE_RINGING) {
- ast_clear_flag(&p->flags[1], SIP_PAGE2_CONNECTLINEUPDATE_PEND);
- respprep(&resp, p, "183 Session Progress", &p->initreq);
- add_rpid(&resp, p);
- send_response(p, &resp, XMIT_UNRELIABLE, 0);
- ast_set_flag(&p->flags[0], SIP_PROGRESS_SENT);
- } else {
- ast_debug(1, "Unable able to send update to '%s' in state '%s'\n", ast_channel_name(p->owner), ast_state2str(ast_channel_state(p->owner)));
- }
- }
- }
-}
-
-static const struct _map_x_s regstatestrings[] = {
- { REG_STATE_FAILED, "Failed" },
- { REG_STATE_UNREGISTERED, "Unregistered"},
- { REG_STATE_REGSENT, "Request Sent"},
- { REG_STATE_AUTHSENT, "Auth. Sent"},
- { REG_STATE_REGISTERED, "Registered"},
- { REG_STATE_REJECTED, "Rejected"},
- { REG_STATE_TIMEOUT, "Registered"},/* Hidden state. We are renewing registration. */
- { REG_STATE_NOAUTH, "No Authentication"},
- { -1, NULL } /* terminator */
-};
-
-/*! \brief Convert registration state status to string */
-static const char *regstate2str(enum sipregistrystate regstate)
-{
- return map_x_s(regstatestrings, regstate, "Unknown");
-}
-
-static void sip_publish_registry(const char *username, const char *domain, const char *status)
-{
- ast_system_publish_registry("SIP", username, domain, status, NULL);
-}
-
-/*!
- * \brief Update registration with SIP Proxy.
- *
- * \details
- * Called from the scheduler when the previous registration expires,
- * so we don't have to cancel the pending event.
- * We assume the reference so the sip_registry is valid, since it
- * is stored in the scheduled event anyways.
- *
- * \note Run by the sched thread.
- */
-static int sip_reregister(const void *data)
-{
- /* if we are here, we know that we need to reregister. */
- struct sip_registry *r = (struct sip_registry *) data;
-
- if (r->call && r->call->do_history) {
- append_history(r->call, "RegistryRenew", "Account: %s@%s", r->username, r->hostname);
- }
- /* Since registry's are only added/removed by the monitor thread, this
- may be overkill to reference/dereference at all here */
- if (sipdebug) {
- ast_log(LOG_NOTICE, " -- Re-registration for %s@%s\n", r->username, r->hostname);
- }
-
- r->expire = -1;
- r->expiry = r->configured_expiry;
- switch (r->regstate) {
- case REG_STATE_UNREGISTERED:
- case REG_STATE_REGSENT:
- case REG_STATE_AUTHSENT:
- break;
- case REG_STATE_REJECTED:
- case REG_STATE_NOAUTH:
- case REG_STATE_FAILED:
- /* Restarting registration as unregistered */
- r->regstate = REG_STATE_UNREGISTERED;
- break;
- case REG_STATE_TIMEOUT:
- case REG_STATE_REGISTERED:
- /* Registration needs to be renewed. */
- r->regstate = REG_STATE_TIMEOUT;
- break;
- }
- __sip_do_register(r);
- ao2_t_ref(r, -1, "Scheduled reregister timeout complete");
- return 0;
-}
-
-/*! \brief Register with SIP proxy
- \return see \ref __sip_xmit
-*/
-static int __sip_do_register(struct sip_registry *r)
-{
- int res;
-
- res = transmit_register(r, SIP_REGISTER, NULL, NULL);
- return res;
-}
-
-struct reregister_data {
- struct sip_registry *reg;
- int ms;
-};
-
-/* Run by the sched thread. */
-static int __start_reregister_timeout(const void *data)
-{
- struct reregister_data *sched_data = (void *) data;
- struct sip_registry *reg = sched_data->reg;
- int ms = sched_data->ms;
-
- ast_free(sched_data);
-
- AST_SCHED_DEL_UNREF(sched, reg->expire,
- ao2_t_ref(reg, -1, "Stop scheduled reregister timeout"));
-
- ao2_t_ref(reg, +1, "Schedule reregister timeout");
- reg->expire = ast_sched_add(sched, ms, sip_reregister, reg);
- if (reg->expire < 0) {
- /* Uh Oh. Expect bad behavior. */
- ao2_t_ref(reg, -1, "Failed to schedule reregister timeout");
- }
-
- ao2_t_ref(reg, -1, "Start reregister timeout action");
- return 0;
-}
-
-static void start_reregister_timeout(struct sip_registry *reg, int ms)
-{
- struct reregister_data *sched_data;
-
- sched_data = ast_malloc(sizeof(*sched_data));
- if (!sched_data) {
- /* Uh Oh. Expect bad behavior. */
- return;
- }
- sched_data->reg = reg;
- sched_data->ms = ms;
- ao2_t_ref(reg, +1, "Start reregister timeout action");
- if (ast_sched_add(sched, 0, __start_reregister_timeout, sched_data) < 0) {
- /* Uh Oh. Expect bad behavior. */
- ao2_t_ref(reg, -1, "Failed to schedule start reregister timeout action");
- ast_free(sched_data);
- }
-}
-
-/*!
- * \brief Registration request timeout, register again
- *
- * \details
- * Registered as a timeout handler during transmit_register(),
- * to retransmit the packet if a reply does not come back.
- *
- * \note This is called by the scheduler so the event is not pending anymore when
- * we are called.
- *
- * \note Run by the sched thread.
- */
-static int sip_reg_timeout(const void *data)
-{
- struct sip_registry *r = (struct sip_registry *)data; /* the ref count should have been bumped when the sched item was added */
- struct sip_pvt *p;
-
- switch (r->regstate) {
- case REG_STATE_UNREGISTERED:
- case REG_STATE_REGSENT:
- case REG_STATE_AUTHSENT:
- case REG_STATE_TIMEOUT:
- break;
- default:
- /*
- * Registration completed because we got a request response
- * and we couldn't stop the scheduled entry in time.
- */
- r->timeout = -1;
- ao2_t_ref(r, -1, "Scheduled register timeout completed early");
- return 0;
- }
-
- if (r->dnsmgr) {
- /* If the registration has timed out, maybe the IP changed. Force a refresh. */
- ast_dnsmgr_refresh(r->dnsmgr);
- }
-
- /* If the initial tranmission failed, we may not have an existing dialog,
- * so it is possible that r->call == NULL.
- * Otherwise destroy it, as we have a timeout so we don't want it.
- */
- if (r->call) {
- /* Unlink us, destroy old call. Locking is not relevant here because all this happens
- in the single SIP manager thread. */
- p = r->call;
- sip_pvt_lock(p);
- pvt_set_needdestroy(p, "registration timeout");
- /* Pretend to ACK anything just in case */
- __sip_pretend_ack(p);
- sip_pvt_unlock(p);
-
- /* decouple the two objects */
- /* p->registry == r, so r has 2 refs, and the unref won't take the object away */
- ao2_t_replace(p->registry, NULL, "p->registry unreffed");
- r->call = dialog_unref(r->call, "unrefing r->call");
- }
- /* If we have a limit, stop registration and give up */
- r->timeout = -1;
- if (global_regattempts_max && r->regattempts >= global_regattempts_max) {
- /* Ok, enough is enough. Don't try any more */
- /* We could add an external notification here...
- steal it from app_voicemail :-) */
- ast_log(LOG_NOTICE, " -- Last Registration Attempt #%d failed, Giving up forever trying to register '%s@%s'\n", r->regattempts, r->username, r->hostname);
- r->regstate = REG_STATE_FAILED;
- } else {
- r->regstate = REG_STATE_UNREGISTERED;
- transmit_register(r, SIP_REGISTER, NULL, NULL);
- ast_log(LOG_NOTICE, " -- Registration for '%s@%s' timed out, trying again (Attempt #%d)\n", r->username, r->hostname, r->regattempts);
- }
- sip_publish_registry(r->username, r->hostname, regstate2str(r->regstate));
- ao2_t_ref(r, -1, "Scheduled register timeout complete");
- return 0;
-}
-
-/* Run by the sched thread. */
-static int __stop_register_timeout(const void *data)
-{
- struct sip_registry *reg = (struct sip_registry *) data;
-
- AST_SCHED_DEL_UNREF(sched, reg->timeout,
- ao2_t_ref(reg, -1, "Stop scheduled register timeout"));
- ao2_t_ref(reg, -1, "Stop register timeout action");
- return 0;
-}
-
-static void stop_register_timeout(struct sip_registry *reg)
-{
- ao2_t_ref(reg, +1, "Stop register timeout action");
- if (ast_sched_add(sched, 0, __stop_register_timeout, reg) < 0) {
- /* Uh Oh. Expect bad behavior. */
- ao2_t_ref(reg, -1, "Failed to schedule stop register timeout action");
- }
-}
-
-/* Run by the sched thread. */
-static int __start_register_timeout(const void *data)
-{
- struct sip_registry *reg = (struct sip_registry *) data;
-
- AST_SCHED_DEL_UNREF(sched, reg->timeout,
- ao2_t_ref(reg, -1, "Stop scheduled register timeout"));
-
- ao2_t_ref(reg, +1, "Schedule register timeout");
- reg->timeout = ast_sched_add(sched, global_reg_timeout * 1000, sip_reg_timeout, reg);
- if (reg->timeout < 0) {
- /* Uh Oh. Expect bad behavior. */
- ao2_t_ref(reg, -1, "Failed to schedule register timeout");
- }
- ast_debug(1, "Scheduled a registration timeout for %s id #%d \n",
- reg->hostname, reg->timeout);
-
- ao2_t_ref(reg, -1, "Start register timeout action");
- return 0;
-}
-
-static void start_register_timeout(struct sip_registry *reg)
-{
- ao2_t_ref(reg, +1, "Start register timeout action");
- if (ast_sched_add(sched, 0, __start_register_timeout, reg) < 0) {
- /* Uh Oh. Expect bad behavior. */
- ao2_t_ref(reg, -1, "Failed to schedule start register timeout action");
- }
-}
-
-static const char *sip_sanitized_host(const char *host)
-{
- struct ast_sockaddr addr;
-
- /* peer/sip_pvt->tohost and sip_registry->hostname should never have a port
- * in them, so we use PARSE_PORT_FORBID here. If this lookup fails, we return
- * the original host which is most likely a host name and not an IP. */
- memset(&addr, 0, sizeof(addr));
- if (!ast_sockaddr_parse(&addr, host, PARSE_PORT_FORBID)) {
- return host;
- }
- return ast_sockaddr_stringify_host_remote(&addr);
-}
-
-/*! \brief Transmit register to SIP proxy or UA
- * auth = NULL on the initial registration (from sip_reregister())
- */
-static int transmit_register(struct sip_registry *r, int sipmethod, const char *auth, const char *authheader)
-{
- struct sip_request req;
- char from[256];
- char to[256];
- char tmp[80];
- char addr[80];
- struct sip_pvt *p;
- struct sip_peer *peer = NULL;
- int res;
- int portno = 0;
-
- /* exit if we are already in process with this registrar ?*/
- if (r == NULL || ((auth == NULL) && (r->regstate == REG_STATE_REGSENT || r->regstate == REG_STATE_AUTHSENT))) {
- if (r) {
- ast_log(LOG_NOTICE, "Strange, trying to register %s@%s when registration already pending\n", r->username, r->hostname);
- }
- return 0;
- }
-
- if (r->dnsmgr == NULL) {
- char transport[MAXHOSTNAMELEN];
- peer = sip_find_peer(r->hostname, NULL, TRUE, FINDPEERS, FALSE, 0);
- snprintf(transport, sizeof(transport), "_%s._%s",get_srv_service(r->transport), get_srv_protocol(r->transport)); /* have to use static sip_get_transport function */
- r->us.ss.ss_family = get_address_family_filter(r->transport); /* Filter address family */
-
- /* No point in doing a DNS lookup of the register hostname if we're just going to
- * end up using an outbound proxy. obproxy_get is safe to call with either of r->call
- * or peer NULL. Since we're only concerned with its existence, we're not going to
- * bother getting a ref to the proxy*/
- if (!obproxy_get(r->call, peer)) {
- ao2_t_ref(r, +1, "add reg ref for dnsmgr");
- ast_dnsmgr_lookup_cb(peer ? peer->tohost : r->hostname, &r->us, &r->dnsmgr, sip_cfg.srvlookup ? transport : NULL, on_dns_update_registry, r);
- if (!r->dnsmgr) {
- /*dnsmgr refresh disabled, no reference added! */
- ao2_t_ref(r, -1, "remove reg ref, dnsmgr disabled");
- }
- }
- if (peer) {
- peer = sip_unref_peer(peer, "removing peer ref for dnsmgr_lookup");
- }
- }
-
- if (r->call) { /* We have a registration */
- if (!auth) {
- ast_log(LOG_WARNING, "Already have a REGISTER going on to %s@%s?? \n", r->username, r->hostname);
- return 0;
- } else {
- p = dialog_ref(r->call, "getting a copy of the r->call dialog in transmit_register");
- ast_string_field_set(p, theirtag, NULL); /* forget their old tag, so we don't match tags when getting response */
- }
- } else {
- /* Build callid for registration if we haven't registered before */
- if (!r->callid_valid) {
- build_callid_registry(r, &internip, default_fromdomain);
- build_localtag_registry(r);
- r->callid_valid = TRUE;
- }
- /* Allocate SIP dialog for registration */
- if (!(p = sip_alloc( r->callid, NULL, 0, SIP_REGISTER, NULL, 0))) {
- ast_log(LOG_WARNING, "Unable to allocate registration transaction (memory or socket error)\n");
- return 0;
- }
-
- /* reset tag to consistent value from registry */
- ast_string_field_set(p, tag, r->localtag);
-
- if (p->do_history) {
- append_history(p, "RegistryInit", "Account: %s@%s", r->username, r->hostname);
- }
-
- p->socket.type = r->transport;
-
- /* Use port number specified if no SRV record was found */
- if (!ast_sockaddr_isnull(&r->us)) {
- if (!ast_sockaddr_port(&r->us) && r->portno) {
- ast_sockaddr_set_port(&r->us, r->portno);
- }
-
- /* It is possible that DNS was unavailable at the time the peer was created.
- * Here, if we've updated the address in the registry via manually calling
- * ast_dnsmgr_lookup_cb() above, then we call the same function that dnsmgr would
- * call if it was updating a peer's address */
- if ((peer = sip_find_peer(S_OR(r->peername, r->hostname), NULL, TRUE, FINDPEERS, FALSE, 0))) {
- if (ast_sockaddr_cmp(&peer->addr, &r->us)) {
- on_dns_update_peer(&peer->addr, &r->us, peer);
- }
- peer = sip_unref_peer(peer, "unref after sip_find_peer");
- }
- }
-
- /* Find address to hostname */
- if (create_addr(p, S_OR(r->peername, r->hostname), &r->us, 0)) {
- /* we have what we hope is a temporary network error,
- * probably DNS. We need to reschedule a registration try */
- dialog_unlink_all(p);
- p = dialog_unref(p, "unref dialog after unlink_all");
- ast_log(LOG_WARNING, "Probably a DNS error for registration to %s@%s, trying REGISTER again (after %d seconds)\n",
- r->username, r->hostname, global_reg_timeout);
- start_register_timeout(r);
- r->regattempts++;
- return 0;
- }
-
- /* Copy back Call-ID in case create_addr changed it */
- ast_string_field_set(r, callid, p->callid);
-
- if (!r->dnsmgr && r->portno) {
- ast_sockaddr_set_port(&p->sa, r->portno);
- ast_sockaddr_set_port(&p->recv, r->portno);
- }
- if (!ast_strlen_zero(p->fromdomain)) {
- portno = (p->fromdomainport) ? p->fromdomainport : STANDARD_SIP_PORT;
- } else if (!ast_strlen_zero(r->regdomain)) {
- portno = (r->regdomainport) ? r->regdomainport : STANDARD_SIP_PORT;
- } else {
- portno = ast_sockaddr_port(&p->sa);
- }
-
- ast_set_flag(&p->flags[0], SIP_OUTGOING); /* Registration is outgoing call */
- r->call = dialog_ref(p, "copying dialog into registry r->call"); /* Save pointer to SIP dialog */
- p->registry = ao2_t_bump(r, "transmit_register: addref to p->registry in transmit_register"); /* Add pointer to registry in packet */
- if (!ast_strlen_zero(r->secret)) { /* Secret (password) */
- ast_string_field_set(p, peersecret, r->secret);
- }
- if (!ast_strlen_zero(r->md5secret))
- ast_string_field_set(p, peermd5secret, r->md5secret);
- /* User name in this realm
- - if authuser is set, use that, otherwise use username */
- if (!ast_strlen_zero(r->authuser)) {
- ast_string_field_set(p, peername, r->authuser);
- ast_string_field_set(p, authname, r->authuser);
- } else if (!ast_strlen_zero(r->username)) {
- ast_string_field_set(p, peername, r->username);
- ast_string_field_set(p, authname, r->username);
- ast_string_field_set(p, fromuser, r->username);
- }
- if (!ast_strlen_zero(r->username)) {
- ast_string_field_set(p, username, r->username);
- }
- /* Save extension in packet */
- if (!ast_strlen_zero(r->callback)) {
- ast_string_field_set(p, exten, r->callback);
- }
-
- /* Set transport so the correct contact is built */
- set_socket_transport(&p->socket, r->transport);
-
- /*
- check which address we should use in our contact header
- based on whether the remote host is on the external or
- internal network so we can register through nat
- */
- ast_sip_ouraddrfor(&p->sa, &p->ourip, p);
- }
-
- /* set up a timeout */
- if (auth == NULL) {
- start_register_timeout(r);
- }
-
- snprintf(from, sizeof(from), ";tag=%s", r->username, S_OR(r->regdomain, sip_sanitized_host(p->tohost)), p->tag);
- if (!ast_strlen_zero(p->theirtag)) {
- snprintf(to, sizeof(to), ";tag=%s", r->username, S_OR(r->regdomain, sip_sanitized_host(p->tohost)), p->theirtag);
- } else {
- snprintf(to, sizeof(to), "", r->username, S_OR(r->regdomain, sip_sanitized_host(p->tohost)));
- }
-
- /* Fromdomain is what we are registering to, regardless of actual
- host name from SRV */
- if (portno && portno != STANDARD_SIP_PORT) {
- snprintf(addr, sizeof(addr), "sip:%s:%d", S_OR(p->fromdomain,S_OR(r->regdomain, sip_sanitized_host(r->hostname))), portno);
- } else {
- snprintf(addr, sizeof(addr), "sip:%s", S_OR(p->fromdomain,S_OR(r->regdomain, sip_sanitized_host(r->hostname))));
- }
-
- ast_string_field_set(p, uri, addr);
-
- p->branch ^= ast_random();
-
- init_req(&req, sipmethod, addr);
-
- /* Add to CSEQ */
- snprintf(tmp, sizeof(tmp), "%u %s", ++r->ocseq, sip_methods[sipmethod].text);
- p->ocseq = r->ocseq;
-
- build_via(p);
- add_header(&req, "Via", p->via);
- add_max_forwards(p, &req);
- add_header(&req, "From", from);
- add_header(&req, "To", to);
- add_header(&req, "Call-ID", p->callid);
- add_header(&req, "CSeq", tmp);
- add_supported(p, &req);
- if (!ast_strlen_zero(global_useragent))
- add_header(&req, "User-Agent", global_useragent);
-
- if (auth) { /* Add auth header */
- add_header(&req, authheader, auth);
- } else if (!ast_strlen_zero(r->nonce)) {
- char digest[1024];
-
- /* We have auth data to reuse, build a digest header.
- * Note, this is not always useful because some parties do not
- * like nonces to be reused (for good reasons!) so they will
- * challenge us anyways.
- */
- if (sipdebug) {
- ast_debug(1, " >>> Re-using Auth data for %s@%s\n", r->username, r->hostname);
- }
- ast_string_field_set(p, realm, r->realm);
- ast_string_field_set(p, nonce, r->nonce);
- ast_string_field_set(p, domain, r->authdomain);
- ast_string_field_set(p, opaque, r->opaque);
- ast_string_field_set(p, qop, r->qop);
- p->noncecount = ++r->noncecount;
-
- memset(digest, 0, sizeof(digest));
- if(!build_reply_digest(p, sipmethod, digest, sizeof(digest))) {
- add_header(&req, "Authorization", digest);
- } else {
- ast_log(LOG_NOTICE, "No authorization available for authentication of registration to %s@%s\n", r->username, r->hostname);
- }
- }
-
- add_expires(&req, r->expiry);
- build_contact(p, &req, 0);
- add_header(&req, "Contact", p->our_contact);
-
- initialize_initreq(p, &req);
- if (sip_debug_test_pvt(p)) {
- ast_verbose("REGISTER %d headers, %d lines\n", p->initreq.headers, p->initreq.lines);
- }
- r->regstate = auth ? REG_STATE_AUTHSENT : REG_STATE_REGSENT;
- r->regattempts++; /* Another attempt */
- ast_debug(4, "REGISTER attempt %d to %s@%s\n", r->regattempts, r->username, r->hostname);
- res = send_request(p, &req, XMIT_CRITICAL, p->ocseq);
- dialog_unref(p, "p is finished here at the end of transmit_register");
- return res;
-}
-
-/*!
- * \brief Transmit with SIP MESSAGE method
- * \note The p->msg_headers and p->msg_body are already setup.
- */
-static int transmit_message(struct sip_pvt *p, int init, int auth)
-{
- struct sip_request req;
-
- if (init) {
- initreqprep(&req, p, SIP_MESSAGE, NULL);
- initialize_initreq(p, &req);
- } else {
- reqprep(&req, p, SIP_MESSAGE, 0, 1);
- }
- if (auth) {
- return transmit_request_with_auth(p, SIP_MESSAGE, p->ocseq, XMIT_RELIABLE, 0);
- } else {
- add_text(&req, p);
- return send_request(p, &req, XMIT_RELIABLE, p->ocseq);
- }
-}
-
-/*! \brief Allocate SIP refer structure */
-static int sip_refer_alloc(struct sip_pvt *p)
-{
- sip_refer_destroy(p);
- p->refer = ast_calloc_with_stringfields(1, struct sip_refer, 512);
- return p->refer ? 1 : 0;
-}
-
-/*! \brief Destroy SIP refer structure */
-static void sip_refer_destroy(struct sip_pvt *p)
-{
- if (p->refer) {
- ast_string_field_free_memory(p->refer);
- ast_free(p->refer);
- p->refer = NULL;
- }
-}
-
-/*! \brief Allocate SIP refer structure */
-static int sip_notify_alloc(struct sip_pvt *p)
-{
- p->notify = ast_calloc(1, sizeof(struct sip_notify));
- if (p->notify) {
- p->notify->content = ast_str_create(128);
- }
- return p->notify ? 1 : 0;
-}
-
-/*! \brief Transmit SIP REFER message (initiated by the transfer() dialplan application
- \note this is currently broken as we have no way of telling the dialplan
- engine whether a transfer succeeds or fails.
- \todo Fix the transfer() dialplan function so that a transfer may fail
-*/
-static int transmit_refer(struct sip_pvt *p, const char *dest)
-{
- char from[256];
- const char *of;
- char *c;
- char referto[256];
- int use_tls=FALSE;
-
- if (sipdebug) {
- ast_debug(1, "SIP transfer of %s to %s\n", p->callid, dest);
- }
-
- /* Are we transfering an inbound or outbound call ? */
- if (ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
- of = sip_get_header(&p->initreq, "To");
- } else {
- of = sip_get_header(&p->initreq, "From");
- }
-
- ast_copy_string(from, of, sizeof(from));
- of = get_in_brackets(from);
- ast_string_field_set(p, from, of);
- if (!strncasecmp(of, "sip:", 4)) {
- of += 4;
- } else if (!strncasecmp(of, "sips:", 5)) {
- of += 5;
- use_tls = TRUE;
- } else {
- ast_log(LOG_NOTICE, "From address missing 'sip(s):', assuming sip:\n");
- }
- /* Get just the username part */
- if (strchr(dest, '@')) {
- c = NULL;
- } else if ((c = strchr(of, '@'))) {
- *c++ = '\0';
- }
- if (c) {
- snprintf(referto, sizeof(referto), "", use_tls ? "s" : "", dest, c);
- } else {
- snprintf(referto, sizeof(referto), "", use_tls ? "s" : "", dest);
- }
-
- /* save in case we get 407 challenge */
- sip_refer_alloc(p);
- ast_string_field_set(p->refer, refer_to, referto);
- ast_string_field_set(p->refer, referred_by, p->our_contact);
- p->refer->status = REFER_SENT; /* Set refer status */
-
- return transmit_invite(p, SIP_REFER, FALSE, 0, NULL);
- /* We should propably wait for a NOTIFY here until we ack the transfer */
- /* Maybe fork a new thread and wait for a STATUS of REFER_200OK on the refer status before returning to app_transfer */
-
- /*! \todo In theory, we should hang around and wait for a reply, before
- returning to the dial plan here. Don't know really how that would
- affect the transfer() app or the pbx, but, well, to make this
- useful we should have a STATUS code on transfer().
- */
-}
-
-/*! \brief Send SIP INFO advice of charge message */
-static int transmit_info_with_aoc(struct sip_pvt *p, struct ast_aoc_decoded *decoded)
-{
- struct sip_request req;
- struct ast_str *str = ast_str_alloca(512);
- const struct ast_aoc_unit_entry *unit_entry = ast_aoc_get_unit_info(decoded, 0);
- enum ast_aoc_charge_type charging = ast_aoc_get_charge_type(decoded);
-
- reqprep(&req, p, SIP_INFO, 0, 1);
-
- if (ast_aoc_get_msg_type(decoded) == AST_AOC_D) {
- ast_str_append(&str, 0, "type=active;");
- } else if (ast_aoc_get_msg_type(decoded) == AST_AOC_E) {
- ast_str_append(&str, 0, "type=terminated;");
- } else {
- /* unsupported message type */
- return -1;
- }
-
- switch (charging) {
- case AST_AOC_CHARGE_FREE:
- ast_str_append(&str, 0, "free-of-charge;");
- break;
- case AST_AOC_CHARGE_CURRENCY:
- ast_str_append(&str, 0, "charging;");
- ast_str_append(&str, 0, "charging-info=currency;");
- ast_str_append(&str, 0, "amount=%u;", ast_aoc_get_currency_amount(decoded));
- ast_str_append(&str, 0, "multiplier=%s;", ast_aoc_get_currency_multiplier_decimal(decoded));
- if (!ast_strlen_zero(ast_aoc_get_currency_name(decoded))) {
- ast_str_append(&str, 0, "currency=%s;", ast_aoc_get_currency_name(decoded));
- }
- break;
- case AST_AOC_CHARGE_UNIT:
- ast_str_append(&str, 0, "charging;");
- ast_str_append(&str, 0, "charging-info=pulse;");
- if (unit_entry) {
- ast_str_append(&str, 0, "recorded-units=%u;", unit_entry->amount);
- }
- break;
- default:
- ast_str_append(&str, 0, "not-available;");
- };
-
- add_header(&req, "AOC", ast_str_buffer(str));
-
- return send_request(p, &req, XMIT_RELIABLE, p->ocseq);
-}
-
-/*! \brief Send SIP INFO dtmf message, see Cisco documentation on cisco.com */
-static int transmit_info_with_digit(struct sip_pvt *p, const char digit, unsigned int duration)
-{
- struct sip_request req;
-
- reqprep(&req, p, SIP_INFO, 0, 1);
- add_digit(&req, digit, duration, (ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_SHORTINFO));
- return send_request(p, &req, XMIT_RELIABLE, p->ocseq);
-}
-
-/*! \brief Send SIP INFO with video update request */
-static int transmit_info_with_vidupdate(struct sip_pvt *p)
-{
- struct sip_request req;
-
- reqprep(&req, p, SIP_INFO, 0, 1);
- add_vidupdate(&req);
- return send_request(p, &req, XMIT_RELIABLE, p->ocseq);
-}
-
-/*! \brief Transmit generic SIP request
- returns XMIT_ERROR if transmit failed with a critical error (don't retry)
-*/
-static int transmit_request(struct sip_pvt *p, int sipmethod, uint32_t seqno, enum xmittype reliable, int newbranch)
-{
- struct sip_request resp;
-
- reqprep(&resp, p, sipmethod, seqno, newbranch);
- if (sipmethod == SIP_CANCEL && p->answered_elsewhere) {
- add_header(&resp, "Reason", "SIP;cause=200;text=\"Call completed elsewhere\"");
- }
-
- if (sipmethod == SIP_ACK) {
- p->invitestate = INV_CONFIRMED;
- }
-
- return send_request(p, &resp, reliable, seqno ? seqno : p->ocseq);
-}
-
-/*! \brief return the request and response header for a 401 or 407 code */
-void sip_auth_headers(enum sip_auth_type code, char **header, char **respheader)
-{
- if (code == WWW_AUTH) { /* 401 */
- *header = "WWW-Authenticate";
- *respheader = "Authorization";
- } else if (code == PROXY_AUTH) { /* 407 */
- *header = "Proxy-Authenticate";
- *respheader = "Proxy-Authorization";
- } else {
- ast_verbose("-- wrong response code %u\n", code);
- *header = *respheader = "Invalid";
- }
-}
-
-/*! \brief Transmit SIP request, auth added */
-static int transmit_request_with_auth(struct sip_pvt *p, int sipmethod, uint32_t seqno, enum xmittype reliable, int newbranch)
-{
- struct sip_request resp;
-
- reqprep(&resp, p, sipmethod, seqno, newbranch);
- if (!ast_strlen_zero(p->realm)) {
- char digest[1024];
-
- memset(digest, 0, sizeof(digest));
- if(!build_reply_digest(p, sipmethod, digest, sizeof(digest))) {
- char *dummy, *response;
- enum sip_auth_type code = p->options ? p->options->auth_type : PROXY_AUTH; /* XXX force 407 if unknown */
- sip_auth_headers(code, &dummy, &response);
- add_header(&resp, response, digest);
- } else {
- ast_log(LOG_WARNING, "No authentication available for call %s\n", p->callid);
- }
- }
-
- switch (sipmethod) {
- case SIP_BYE:
- {
- char buf[20];
-
- /*
- * We are hanging up. If we know a cause for that, send it in
- * clear text to make debugging easier.
- */
- if (ast_test_flag(&p->flags[1], SIP_PAGE2_Q850_REASON) && p->hangupcause) {
- snprintf(buf, sizeof(buf), "Q.850;cause=%d", p->hangupcause & 0x7f);
- add_header(&resp, "Reason", buf);
- }
-
- add_header(&resp, "X-Asterisk-HangupCause", ast_cause2str(p->hangupcause));
- snprintf(buf, sizeof(buf), "%d", p->hangupcause);
- add_header(&resp, "X-Asterisk-HangupCauseCode", buf);
- break;
- }
- case SIP_MESSAGE:
- add_text(&resp, p);
- break;
- default:
- break;
- }
-
- return send_request(p, &resp, reliable, seqno ? seqno : p->ocseq);
-}
-
-/*! \brief Remove registration data from realtime database or AST/DB when registration expires */
-static void destroy_association(struct sip_peer *peer)
-{
- int realtimeregs = ast_check_realtime("sipregs");
- char *tablename = (realtimeregs) ? "sipregs" : "sippeers";
-
- if (!sip_cfg.ignore_regexpire) {
- if (peer->rt_fromcontact && sip_cfg.peer_rtupdate) {
- ast_update_realtime(tablename, "name", peer->name, "fullcontact", "", "ipaddr", "", "port", "0", "regseconds", "0", "regserver", "", "useragent", "", "lastms", "0", SENTINEL);
- } else {
- ast_db_del("SIP/Registry", peer->name);
- ast_db_del("SIP/RegistryPath", peer->name);
- ast_db_del("SIP/PeerMethods", peer->name);
- }
- }
-}
-
-static void set_socket_transport(struct sip_socket *socket, int transport)
-{
- /* if the transport type changes, clear all socket data */
- if (socket->type != transport) {
- socket->fd = -1;
- socket->type = transport;
- if (socket->tcptls_session) {
- ao2_ref(socket->tcptls_session, -1);
- socket->tcptls_session = NULL;
- } else if (socket->ws_session) {
- ast_websocket_unref(socket->ws_session);
- socket->ws_session = NULL;
- }
- }
-}
-
-/*! \brief Expire registration of SIP peer */
-static int expire_register(const void *data)
-{
- struct sip_peer *peer = (struct sip_peer *)data;
-
- if (!peer) { /* Hmmm. We have no peer. Weird. */
- return 0;
- }
-
- peer->expire = -1;
- peer->portinuri = 0;
-
- destroy_association(peer); /* remove registration data from storage */
- set_socket_transport(&peer->socket, peer->default_outbound_transport);
-
- if (peer->socket.tcptls_session) {
- ao2_ref(peer->socket.tcptls_session, -1);
- peer->socket.tcptls_session = NULL;
- } else if (peer->socket.ws_session) {
- ast_websocket_unref(peer->socket.ws_session);
- peer->socket.ws_session = NULL;
- }
-
- if (peer->endpoint) {
- RAII_VAR(struct ast_json *, blob, NULL, ast_json_unref);
- ast_endpoint_set_state(peer->endpoint, AST_ENDPOINT_OFFLINE);
- blob = ast_json_pack("{s: s, s: s}",
- "peer_status", "Unregistered",
- "cause", "Expired");
- ast_endpoint_blob_publish(peer->endpoint, ast_endpoint_state_type(), blob);
- }
- register_peer_exten(peer, FALSE); /* Remove regexten */
- ast_devstate_changed(AST_DEVICE_UNKNOWN, AST_DEVSTATE_CACHABLE, "SIP/%s", peer->name);
-
- /* Do we need to release this peer from memory?
- Only for realtime peers and autocreated peers
- */
- if (peer->is_realtime) {
- ast_debug(3, "-REALTIME- peer expired registration. Name: %s. Realtime peer objects now %d\n", peer->name, rpeerobjs);
- }
-
- if (peer->selfdestruct ||
- ast_test_flag(&peer->flags[1], SIP_PAGE2_RTAUTOCLEAR)) {
- ao2_t_unlink(peers, peer, "ao2_unlink of peer from peers table");
- }
- if (!ast_sockaddr_isnull(&peer->addr)) {
- /* We still need to unlink the peer from the peers_by_ip table,
- * otherwise we end up with multiple copies hanging around each
- * time a registration expires and the peer re-registers. */
- ao2_t_unlink(peers_by_ip, peer, "ao2_unlink of peer from peers_by_ip table");
- }
-
- /* Only clear the addr after we check for destruction. The addr must remain
- * in order to unlink from the peers_by_ip container correctly */
- memset(&peer->addr, 0, sizeof(peer->addr));
-
- sip_unref_peer(peer, "removing peer ref for expire_register");
-
- return 0;
-}
-
-/*! \brief Poke peer (send qualify to check if peer is alive and well) */
-static int sip_poke_peer_s(const void *data)
-{
- struct sip_peer *peer = (struct sip_peer *)data;
- struct sip_peer *foundpeer;
-
- peer->pokeexpire = -1;
-
- foundpeer = ao2_find(peers, peer, OBJ_POINTER);
- if (!foundpeer) {
- sip_unref_peer(peer, "removing poke peer ref");
- return 0;
- } else if (foundpeer->name != peer->name) {
- sip_unref_peer(foundpeer, "removing above peer ref");
- sip_unref_peer(peer, "removing poke peer ref");
- return 0;
- }
-
- sip_unref_peer(foundpeer, "removing above peer ref");
- sip_poke_peer(peer, 0);
- sip_unref_peer(peer, "removing poke peer ref");
-
- return 0;
-}
-
-static int sip_poke_peer_now(const void *data)
-{
- struct sip_peer *peer = (struct sip_peer *) data;
-
- peer->pokeexpire = -1;
- sip_poke_peer(peer, 0);
- sip_unref_peer(peer, "removing poke peer ref");
-
- return 0;
-}
-
-/*! \brief Get registration details from Asterisk DB */
-static void reg_source_db(struct sip_peer *peer)
-{
- char data[256];
- char path[SIPBUFSIZE * 2];
- struct ast_sockaddr sa;
- int expire;
- char full_addr[128];
- AST_DECLARE_APP_ARGS(args,
- AST_APP_ARG(addr);
- AST_APP_ARG(port);
- AST_APP_ARG(expiry_str);
- AST_APP_ARG(username);
- AST_APP_ARG(contact);
- );
-
- /* If read-only RT backend, then refresh from local DB cache */
- if (peer->rt_fromcontact && sip_cfg.peer_rtupdate) {
- return;
- }
- if (ast_db_get("SIP/Registry", peer->name, data, sizeof(data))) {
- return;
- }
-
- AST_NONSTANDARD_RAW_ARGS(args, data, ':');
-
- snprintf(full_addr, sizeof(full_addr), "%s:%s", args.addr, args.port);
-
- if (!ast_sockaddr_parse(&sa, full_addr, 0)) {
- return;
- }
-
- if (args.expiry_str) {
- expire = atoi(args.expiry_str);
- } else {
- return;
- }
-
- if (args.username) {
- ast_string_field_set(peer, username, args.username);
- }
- if (args.contact) {
- ast_string_field_set(peer, fullcontact, args.contact);
- }
-
- ast_debug(2, "SIP Seeding peer from astdb: '%s' at %s@%s for %d\n",
- peer->name, peer->username, ast_sockaddr_stringify_host(&sa), expire);
-
- ast_sockaddr_copy(&peer->addr, &sa);
- if (peer->maxms) {
- /* Don't poke peer immediately, just schedule it within qualifyfreq */
- AST_SCHED_REPLACE_UNREF(peer->pokeexpire, sched,
- ast_random() % ((peer->qualifyfreq) ? peer->qualifyfreq : global_qualifyfreq) + 1,
- sip_poke_peer_s, peer,
- sip_unref_peer(_data, "removing poke peer ref"),
- sip_unref_peer(peer, "removing poke peer ref"),
- sip_ref_peer(peer, "adding poke peer ref"));
- }
- AST_SCHED_REPLACE_UNREF(peer->expire, sched, (expire + 10) * 1000, expire_register, peer,
- sip_unref_peer(_data, "remove registration ref"),
- sip_unref_peer(peer, "remove registration ref"),
- sip_ref_peer(peer, "add registration ref"));
- register_peer_exten(peer, TRUE);
- if (!ast_db_get("SIP/RegistryPath", peer->name, path, sizeof(path))) {
- build_path(NULL, peer, NULL, path);
- }
-}
-
-/*! \brief Save contact header for 200 OK on INVITE */
-static int parse_ok_contact(struct sip_pvt *pvt, struct sip_request *req)
-{
- char contact[SIPBUFSIZE];
- char *c;
-
- /* Look for brackets */
- ast_copy_string(contact, sip_get_header(req, "Contact"), sizeof(contact));
- c = get_in_brackets(contact);
-
- /* Save full contact to call pvt for later bye or re-invite */
- ast_string_field_set(pvt, fullcontact, c);
-
- /* Save URI for later ACKs, BYE or RE-invites */
- ast_string_field_set(pvt, okcontacturi, c);
-
- /* We should return false for URI:s we can't handle,
- like tel:, mailto:,ldap: etc */
- return TRUE;
-}
-
-/*!
- * \brief Parses SIP reason header according to RFC3326 and sets channel's hangupcause if configured so
- * and header present
- *
- * \note This is used in BYE and CANCEL request and SIP response, but according to RFC3326 it could
- * appear in any request, but makes not a lot of sense in others than BYE or CANCEL.
- * Currently only implemented for Q.850 status codes.
- * \retval 0 success
- * \retval -1 on failure or if not configured
- */
-static int use_reason_header(struct sip_pvt *pvt, struct sip_request *req)
-{
- int ret, cause;
- const char *rp, *rh;
-
- if (!pvt->owner) {
- return -1;
- }
-
- if (!ast_test_flag(&pvt->flags[1], SIP_PAGE2_Q850_REASON) ||
- !(rh = sip_get_header(req, "Reason"))) {
- return -1;
- }
-
- rh = ast_skip_blanks(rh);
- if (strncasecmp(rh, "Q.850", 5)) {
- return -1;
- }
-
- ret = -1;
- cause = ast_channel_hangupcause(pvt->owner);
- rp = strstr(rh, "cause=");
- if (rp && sscanf(rp + 6, "%3d", &cause) == 1) {
- ret = 0;
- ast_channel_hangupcause_set(pvt->owner, cause & 0x7f);
- if (req->debug) {
- ast_verbose("Using Reason header for cause code: %d\n",
- ast_channel_hangupcause(pvt->owner));
- }
- }
- return ret;
-}
-
-/*! \brief parse uri in a way that allows semicolon stripping if legacy mode is enabled
- *
- * \note This calls parse_uri which has the unexpected property that passing more
- * arguments results in more splitting. Most common is to leave out the pass
- * argument, causing user to contain user:pass if available.
- */
-static int parse_uri_legacy_check(char *uri, const char *scheme, char **user, char **pass, char **hostport, char **transport)
-{
- int ret = parse_uri(uri, scheme, user, pass, hostport, transport);
- if (sip_cfg.legacy_useroption_parsing) { /* if legacy mode is active, strip semis from the user field */
- char *p;
- if ((p = strchr(uri, (int)';'))) {
- *p = '\0';
- }
- }
- return ret;
-}
-
-static int __set_address_from_contact(const char *fullcontact, struct ast_sockaddr *addr, int tcp)
-{
- char *hostport, *transport;
- char contact_buf[256];
- char *contact;
-
- /* Work on a copy */
- ast_copy_string(contact_buf, fullcontact, sizeof(contact_buf));
- contact = contact_buf;
-
- /*
- * We have only the part in here so we just need to parse a SIP URI.
- *
- * Note: The outbound proxy could be using UDP between the proxy and Asterisk.
- * We still need to be able to send to the remote agent through the proxy.
- */
-
- if (parse_uri_legacy_check(contact, "sip:,sips:", &contact, NULL, &hostport,
- &transport)) {
- ast_log(LOG_WARNING, "Invalid contact uri %s (missing sip: or sips:), attempting to use anyway\n", fullcontact);
- }
-
- /* XXX This could block for a long time XXX */
- /* We should only do this if it's a name, not an IP */
- /* \todo - if there's no PORT number in contact - we are required to check NAPTR/SRV records
- to find transport, port address and hostname. If there's a port number, we have to
- assume that the hostport part is a host name and only look for an A/AAAA record in DNS.
- */
-
- /* If we took in an invalid URI, hostport may not have been initialized */
- /* ast_sockaddr_resolve requires an initialized hostport string. */
- if (ast_strlen_zero(hostport)) {
- ast_log(LOG_WARNING, "Invalid URI: parse_uri failed to acquire hostport\n");
- return -1;
- }
-
- if (ast_sockaddr_resolve_first_transport(addr, hostport, 0, get_transport_str2enum(transport))) {
- ast_log(LOG_WARNING, "Invalid host name in Contact: (can't "
- "resolve in DNS) : '%s'\n", hostport);
- return -1;
- }
-
- /* set port */
- if (!ast_sockaddr_port(addr)) {
- ast_sockaddr_set_port(addr,
- (get_transport_str2enum(transport) ==
- AST_TRANSPORT_TLS ||
- !strncasecmp(fullcontact, "sips", 4)) ?
- STANDARD_TLS_PORT : STANDARD_SIP_PORT);
- }
-
- return 0;
-}
-
-/*! \brief Change the other partys IP address based on given contact */
-static int set_address_from_contact(struct sip_pvt *pvt)
-{
- if (ast_test_flag(&pvt->flags[0], SIP_NAT_FORCE_RPORT)) {
- /* NAT: Don't trust the contact field. Just use what they came to us
- with. */
- /*! \todo We need to save the TRANSPORT here too */
- pvt->sa = pvt->recv;
- return 0;
- }
-
- return __set_address_from_contact(pvt->fullcontact, &pvt->sa, pvt->socket.type == AST_TRANSPORT_TLS ? 1 : 0);
-}
-
-/*! \brief Parse contact header and save registration (peer registration) */
-static enum parse_register_result parse_register_contact(struct sip_pvt *pvt, struct sip_peer *peer, struct sip_request *req)
-{
- char contact[SIPBUFSIZE];
- char data[SIPBUFSIZE];
- const char *expires = sip_get_header(req, "Expires");
- int expire = atoi(expires);
- char *curi = NULL, *hostport = NULL, *transport = NULL;
- int transport_type;
- const char *useragent;
- struct ast_sockaddr oldsin, testsa;
- char *firstcuri = NULL;
- int start = 0;
- int wildcard_found = 0;
- int single_binding_found = 0;
-
- ast_copy_string(contact, __get_header(req, "Contact", &start), sizeof(contact));
-
- if (ast_strlen_zero(expires)) { /* No expires header, try look in Contact: */
- char *s = strcasestr(contact, ";expires=");
- if (s) {
- expires = strsep(&s, ";"); /* trim ; and beyond */
- if (sscanf(expires + 9, "%30d", &expire) != 1) {
- expire = default_expiry;
- }
- } else {
- /* Nothing has been specified */
- expire = default_expiry;
- }
- }
-
- if (expire > max_expiry) {
- expire = max_expiry;
- }
- if (expire < min_expiry && expire != 0) {
- expire = min_expiry;
- }
- pvt->expiry = expire;
-
- copy_socket_data(&pvt->socket, &req->socket);
-
- do {
- /* Look for brackets */
- curi = contact;
- if (strchr(contact, '<') == NULL) /* No <, check for ; and strip it */
- strsep(&curi, ";"); /* This is Header options, not URI options */
- curi = get_in_brackets(contact);
- if (!firstcuri) {
- firstcuri = ast_strdupa(curi);
- }
-
- if (!strcasecmp(curi, "*")) {
- wildcard_found = 1;
- } else {
- single_binding_found = 1;
- }
-
- if (wildcard_found && (ast_strlen_zero(expires) || expire != 0 || single_binding_found)) {
- /* Contact header parameter "*" detected, so punt if: Expires header is missing,
- * Expires value is not zero, or another Contact header is present. */
- return PARSE_REGISTER_FAILED;
- }
-
- ast_copy_string(contact, __get_header(req, "Contact", &start), sizeof(contact));
- } while (!ast_strlen_zero(contact));
- curi = firstcuri;
-
- /* if they did not specify Contact: or Expires:, they are querying
- what we currently have stored as their contact address, so return
- it
- */
- if (ast_strlen_zero(curi) && ast_strlen_zero(expires)) {
- /* If we have an active registration, tell them when the registration is going to expire */
- if (peer->expire > -1 && !ast_strlen_zero(peer->fullcontact)) {
- pvt->expiry = ast_sched_when(sched, peer->expire);
- }
- return PARSE_REGISTER_QUERY;
- } else if (!strcasecmp(curi, "*") || !expire) { /* Unregister this peer */
- /* This means remove all registrations and return OK */
- AST_SCHED_DEL_UNREF(sched, peer->expire,
- sip_unref_peer(peer, "remove register expire ref"));
- ast_verb(3, "Unregistered SIP '%s'\n", peer->name);
- expire_register(sip_ref_peer(peer,"add ref for explicit expire_register"));
- return PARSE_REGISTER_UPDATE;
- }
-
- /* Store whatever we got as a contact from the client */
- ast_string_field_set(peer, fullcontact, curi);
-
- /* For the 200 OK, we should use the received contact */
- ast_string_field_build(pvt, our_contact, "<%s>", curi);
-
- /* Make sure it's a SIP URL */
- if (ast_strlen_zero(curi) || parse_uri_legacy_check(curi, "sip:,sips:", &curi, NULL, &hostport, &transport)) {
- ast_log(LOG_NOTICE, "Not a valid SIP contact (missing sip:/sips:) trying to use anyway\n");
- }
-
- /* handle the transport type specified in Contact header. */
- if (!(transport_type = get_transport_str2enum(transport))) {
- transport_type = pvt->socket.type;
- }
-
- /* if the peer's socket type is different than the Registration
- * transport type, change it. If it got this far, it is a
- * supported type, but check just in case */
- if ((peer->socket.type != transport_type) && (peer->transports & transport_type)) {
- set_socket_transport(&peer->socket, transport_type);
- }
-
- oldsin = peer->addr;
-
- /* If we were already linked into the peers_by_ip container unlink ourselves so nobody can find us */
- if (!ast_sockaddr_isnull(&peer->addr) && (!peer->is_realtime || ast_test_flag(&global_flags[1], SIP_PAGE2_RTCACHEFRIENDS))) {
- ao2_t_unlink(peers_by_ip, peer, "ao2_unlink of peer from peers_by_ip table");
- }
-
- if ((transport_type != AST_TRANSPORT_WS) && (transport_type != AST_TRANSPORT_WSS) &&
- (!ast_test_flag(&peer->flags[0], SIP_NAT_FORCE_RPORT) && !ast_test_flag(&pvt->flags[0], SIP_NAT_RPORT_PRESENT))) {
- /* use the data provided in the Contact header for call routing */
- ast_debug(1, "Store REGISTER's Contact header for call routing.\n");
- /* XXX This could block for a long time XXX */
- /*! \todo Check NAPTR/SRV if we have not got a port in the URI */
- if (ast_sockaddr_resolve_first_transport(&testsa, hostport, 0, peer->socket.type)) {
- ast_log(LOG_WARNING, "Invalid hostport '%s'\n", hostport);
- ast_string_field_set(peer, fullcontact, "");
- ast_string_field_set(pvt, our_contact, "");
- return PARSE_REGISTER_FAILED;
- }
-
- /* If we have a port number in the given URI, make sure we do remember to not check for NAPTR/SRV records.
- The hostport part is actually a host. */
- peer->portinuri = ast_sockaddr_port(&testsa) ? TRUE : FALSE;
-
- if (!ast_sockaddr_port(&testsa)) {
- ast_sockaddr_set_port(&testsa, default_sip_port(transport_type));
- }
-
- ast_sockaddr_copy(&peer->addr, &testsa);
- } else {
- /* Don't trust the contact field. Just use what they came to us
- with */
- ast_debug(1, "Store REGISTER's src-IP:port for call routing.\n");
- peer->addr = pvt->recv;
- }
-
- /* Check that they're allowed to register at this IP */
- if (ast_apply_acl(sip_cfg.contact_acl, &peer->addr, "SIP contact ACL: ") != AST_SENSE_ALLOW ||
- ast_apply_acl(peer->contactacl, &peer->addr, "SIP contact ACL: ") != AST_SENSE_ALLOW) {
- ast_log(LOG_WARNING, "Domain '%s' disallowed by contact ACL (violating IP %s)\n", hostport,
- ast_sockaddr_stringify_addr(&peer->addr));
- ast_string_field_set(peer, fullcontact, "");
- ast_string_field_set(pvt, our_contact, "");
- return PARSE_REGISTER_DENIED;
- }
-
- /* if the Contact header information copied into peer->addr matches the
- * received address, and the transport types are the same, then copy socket
- * data into the peer struct */
- if ((peer->socket.type == pvt->socket.type) &&
- !ast_sockaddr_cmp(&peer->addr, &pvt->recv)) {
- copy_socket_data(&peer->socket, &pvt->socket);
- }
-
- /* Now that our address has been updated put ourselves back into the container for lookups */
- if (!peer->is_realtime || ast_test_flag(&peer->flags[1], SIP_PAGE2_RTCACHEFRIENDS)) {
- ao2_t_link(peers_by_ip, peer, "ao2_link into peers_by_ip table");
- }
-
- /* Save SIP options profile */
- peer->sipoptions = pvt->sipoptions;
-
- if (!ast_strlen_zero(curi) && ast_strlen_zero(peer->username)) {
- ast_string_field_set(peer, username, curi);
- }
-
- AST_SCHED_DEL_UNREF(sched, peer->expire,
- sip_unref_peer(peer, "remove register expire ref"));
-
- if (peer->is_realtime && !ast_test_flag(&peer->flags[1], SIP_PAGE2_RTCACHEFRIENDS)) {
- peer->expire = -1;
- } else {
- peer->expire = ast_sched_add(sched, (expire + 10) * 1000, expire_register,
- sip_ref_peer(peer, "add registration ref"));
- if (peer->expire == -1) {
- sip_unref_peer(peer, "remote registration ref");
- }
- }
- if (!build_path(pvt, peer, req, NULL)) {
- /* Tell the dialog to use the Path header in the response */
- ast_set2_flag(&pvt->flags[0], 1, SIP_USEPATH);
- }
- snprintf(data, sizeof(data), "%s:%d:%s:%s", ast_sockaddr_stringify(&peer->addr),
- expire, peer->username, peer->fullcontact);
- /* We might not immediately be able to reconnect via TCP, but try caching it anyhow */
- if (!peer->rt_fromcontact || !sip_cfg.peer_rtupdate) {
- if (!sip_route_empty(&peer->path)) {
- struct ast_str *r = sip_route_list(&peer->path, 0, 0);
- if (r) {
- ast_db_put("SIP/RegistryPath", peer->name, ast_str_buffer(r));
- ast_free(r);
- }
- }
- ast_db_put("SIP/Registry", peer->name, data);
- }
-
- if (peer->endpoint) {
- RAII_VAR(struct ast_json *, blob, NULL, ast_json_unref);
- ast_endpoint_set_state(peer->endpoint, AST_ENDPOINT_ONLINE);
- blob = ast_json_pack("{s: s, s: s}",
- "peer_status", "Registered",
- "address", ast_sockaddr_stringify(&peer->addr));
- ast_endpoint_blob_publish(peer->endpoint, ast_endpoint_state_type(), blob);
- }
-
- /* Is this a new IP address for us? */
- if (ast_sockaddr_cmp(&peer->addr, &oldsin)) {
- ast_verb(3, "Registered SIP '%s' at %s\n", peer->name,
- ast_sockaddr_stringify(&peer->addr));
- }
- sip_pvt_unlock(pvt);
- sip_poke_peer(peer, 0);
- sip_pvt_lock(pvt);
- register_peer_exten(peer, 1);
-
- /* Save User agent */
- useragent = sip_get_header(req, "User-Agent");
- if (strcasecmp(useragent, peer->useragent)) {
- ast_string_field_set(peer, useragent, useragent);
- ast_verb(4, "Saved useragent \"%s\" for peer %s\n", peer->useragent, peer->name);
- }
- return PARSE_REGISTER_UPDATE;
-}
-
-/*! \brief Build route list from Record-Route header
- *
- * \param p
- * \param req
- * \param backwards
- * \param resp the SIP response code or 0 for a request
- *
- */
-static void build_route(struct sip_pvt *p, struct sip_request *req, int backwards, int resp)
-{
- int start = 0;
- const char *header;
-
- /* Once a persistent route is set, don't fool with it */
- if (!sip_route_empty(&p->route) && p->route_persistent) {
- ast_debug(1, "build_route: Retaining previous route: <%s>\n", sip_route_first_uri(&p->route));
- return;
- }
-
- sip_route_clear(&p->route);
-
- /* We only want to create the route set the first time this is called except
- it is called from a provisional response.*/
- if ((resp < 100) || (resp > 199)) {
- p->route_persistent = 1;
- }
-
- /* Build a tailq, then assign it to p->route when done.
- * If backwards, we add entries from the head so they end up
- * in reverse order. However, we do need to maintain a correct
- * tail pointer because the contact is always at the end.
- */
- /* 1st we pass through all the hops in any Record-Route headers */
- for (;;) {
- header = __get_header(req, "Record-Route", &start);
- if (*header == '\0') {
- break;
- }
- sip_route_process_header(&p->route, header, backwards);
- }
-
- /* Only append the contact if we are dealing with a strict router or have no route */
- if (sip_route_empty(&p->route) || sip_route_is_strict(&p->route)) {
- /* 2nd append the Contact: if there is one */
- /* Can be multiple Contact headers, comma separated values - we just take the first */
- int len;
- header = sip_get_header(req, "Contact");
- if (strchr(header, '<')) {
- get_in_brackets_const(header, &header, &len);
- } else {
- len = strlen(header);
- }
- if (header && len) {
- sip_route_add(&p->route, header, len, 0);
- }
- }
-
- /* For debugging dump what we ended up with */
- if (sip_debug_test_pvt(p)) {
- sip_route_dump(&p->route);
- }
-}
-
-/*! \brief Build route list from Path header
- * RFC 3327 requires that the Path header contains SIP URIs with lr paramter.
- * Thus, we do not care about strict routing SIP routers
- */
-static int build_path(struct sip_pvt *p, struct sip_peer *peer, struct sip_request *req, const char *pathbuf)
-{
- sip_route_clear(&peer->path);
-
- if (!ast_test_flag(&peer->flags[0], SIP_USEPATH)) {
- ast_debug(2, "build_path: do not use Path headers\n");
- return -1;
- }
- ast_debug(2, "build_path: try to build pre-loaded route-set by parsing Path headers\n");
-
- if (req) {
- int start = 0;
- const char *header;
- for (;;) {
- header = __get_header(req, "Path", &start);
- if (*header == '\0') {
- break;
- }
- sip_route_process_header(&peer->path, header, 0);
- }
- } else if (pathbuf) {
- sip_route_process_header(&peer->path, pathbuf, 0);
- }
-
- /* Caches result for any dialog->route copied from peer->path */
- sip_route_is_strict(&peer->path);
-
- /* For debugging dump what we ended up with */
- if (p && sip_debug_test_pvt(p)) {
- sip_route_dump(&peer->path);
- }
- return 0;
-}
-
-/*! \brief builds the sip_pvt's nonce field which is used for the authentication
- * challenge. When forceupdate is not set, the nonce is only updated if
- * the current one is stale. In this case, a stalenonce is one which
- * has already received a response, if a nonce has not received a response
- * it is not always necessary or beneficial to create a new one. */
-
-static void build_nonce(struct sip_pvt *p, int forceupdate)
-{
- if (p->stalenonce || forceupdate || ast_strlen_zero(p->nonce)) {
- ast_string_field_build(p, nonce, "%08lx", (unsigned long)ast_random()); /* Create nonce for challenge */
- p->stalenonce = 0;
- }
-}
-
-/*! \brief Takes the digest response and parses it */
-void sip_digest_parser(char *c, struct digestkeys *keys)
-{
- struct digestkeys *i = i;
-
- while(c && *(c = ast_skip_blanks(c)) ) { /* lookup for keys */
- for (i = keys; i->key != NULL; i++) {
- const char *separator = ","; /* default */
-
- if (strncasecmp(c, i->key, strlen(i->key)) != 0) {
- continue;
- }
- /* Found. Skip keyword, take text in quotes or up to the separator. */
- c += strlen(i->key);
- if (*c == '"') { /* in quotes. Skip first and look for last */
- c++;
- separator = "\"";
- }
- i->s = c;
- strsep(&c, separator);
- break;
- }
- if (i->key == NULL) { /* not found, jump after space or comma */
- strsep(&c, " ,");
- }
- }
-}
-
-/*! \brief Check user authorization from peer definition
- Some actions, like REGISTER and INVITEs from peers require
- authentication (if peer have secret set)
- \return 0 on success, non-zero on error
-*/
-static enum check_auth_result check_auth(struct sip_pvt *p, struct sip_request *req, const char *username,
- const char *secret, const char *md5secret, int sipmethod,
- const char *uri, enum xmittype reliable)
-{
- const char *response;
- char *reqheader, *respheader;
- const char *authtoken;
- char a1_hash[256];
- char resp_hash[256]="";
- char *c;
- int is_bogus_peer = 0;
- int wrongnonce = FALSE;
- int good_response;
- const char *usednonce = p->nonce;
- struct ast_str *buf;
- int res;
-
- /* table of recognised keywords, and their value in the digest */
- struct digestkeys keys[] = {
- [K_RESP] = { "response=", "" },
- [K_URI] = { "uri=", "" },
- [K_USER] = { "username=", "" },
- [K_NONCE] = { "nonce=", "" },
- [K_LAST] = { NULL, NULL}
- };
-
- /* Always OK if no secret */
- if (ast_strlen_zero(secret) && ast_strlen_zero(md5secret)) {
- return AUTH_SUCCESSFUL;
- }
-
- /* Always auth with WWW-auth since we're NOT a proxy */
- /* Using proxy-auth in a B2BUA may block proxy authorization in the same transaction */
- response = "401 Unauthorized";
-
- /*
- * Note the apparent swap of arguments below, compared to other
- * usages of sip_auth_headers().
- */
- sip_auth_headers(WWW_AUTH, &respheader, &reqheader);
-
- authtoken = sip_get_header(req, reqheader);
- if (req->ignore && !ast_strlen_zero(p->nonce) && ast_strlen_zero(authtoken)) {
- /* This is a retransmitted invite/register/etc, don't reconstruct authentication
- information */
- if (!reliable) {
- /* Resend message if this was NOT a reliable delivery. Otherwise the
- retransmission should get it */
- transmit_response_with_auth(p, response, req, p->nonce, reliable, respheader, 0);
- /* Schedule auto destroy in 32 seconds (according to RFC 3261) */
- sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
- }
- return AUTH_CHALLENGE_SENT;
- } else if (ast_strlen_zero(p->nonce) || ast_strlen_zero(authtoken)) {
- /* We have no auth, so issue challenge and request authentication */
- build_nonce(p, 1); /* Create nonce for challenge */
- transmit_response_with_auth(p, response, req, p->nonce, reliable, respheader, 0);
- /* Schedule auto destroy in 32 seconds */
- sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
- return AUTH_CHALLENGE_SENT;
- }
-
- /* --- We have auth, so check it */
-
- /* Whoever came up with the authentication section of SIP can suck my %&#$&* for not putting
- an example in the spec of just what it is you're doing a hash on. */
-
- if (!(buf = ast_str_thread_get(&check_auth_buf, CHECK_AUTH_BUF_INITLEN))) {
- return AUTH_SECRET_FAILED; /*! XXX \todo need a better return code here */
- }
-
- /* Make a copy of the response and parse it */
- res = ast_str_set(&buf, 0, "%s", authtoken);
-
- if (res == AST_DYNSTR_BUILD_FAILED) {
- return AUTH_SECRET_FAILED; /*! XXX \todo need a better return code here */
- }
-
- c = ast_str_buffer(buf);
-
- sip_digest_parser(c, keys);
-
- /* We cannot rely on the bogus_peer having a bad md5 value. Someone could
- * use it to construct valid auth. */
- if (md5secret && strcmp(md5secret, BOGUS_PEER_MD5SECRET) == 0) {
- is_bogus_peer = 1;
- }
-
- /* Verify that digest username matches the username we auth as */
- if (strcmp(username, keys[K_USER].s) && !is_bogus_peer) {
- ast_log(LOG_WARNING, "username mismatch, have <%s>, digest has <%s>\n",
- username, keys[K_USER].s);
- /* Oops, we're trying something here */
- return AUTH_USERNAME_MISMATCH;
- }
-
- /* Verify nonce from request matches our nonce, and the nonce has not already been responded to.
- * If this check fails, send 401 with new nonce */
- if (strcasecmp(p->nonce, keys[K_NONCE].s) || p->stalenonce) { /* XXX it was 'n'casecmp ? */
- wrongnonce = TRUE;
- usednonce = keys[K_NONCE].s;
- } else {
- p->stalenonce = 1; /* now, since the nonce has a response, mark it as stale so it can't be sent or responded to again */
- }
-
- if (!ast_strlen_zero(md5secret)) {
- ast_copy_string(a1_hash, md5secret, sizeof(a1_hash));
- } else {
- char a1[256];
-
- snprintf(a1, sizeof(a1), "%s:%s:%s", username, p->realm, secret);
- ast_md5_hash(a1_hash, a1);
- }
-
- /* compute the expected response to compare with what we received */
- {
- char a2[256];
- char a2_hash[256];
- char resp[256];
-
- snprintf(a2, sizeof(a2), "%s:%s", sip_methods[sipmethod].text,
- S_OR(keys[K_URI].s, uri));
- ast_md5_hash(a2_hash, a2);
- snprintf(resp, sizeof(resp), "%s:%s:%s", a1_hash, usednonce, a2_hash);
- ast_md5_hash(resp_hash, resp);
- }
-
- good_response = keys[K_RESP].s &&
- !strncasecmp(keys[K_RESP].s, resp_hash, strlen(resp_hash)) &&
- !is_bogus_peer; /* lastly, check that the peer isn't the fake peer */
- if (wrongnonce) {
- if (good_response) {
- if (sipdebug)
- ast_log(LOG_NOTICE, "Correct auth, but based on stale nonce received from '%s'\n", sip_get_header(req, "From"));
- /* We got working auth token, based on stale nonce . */
- build_nonce(p, 0);
- transmit_response_with_auth(p, response, req, p->nonce, reliable, respheader, TRUE);
- } else {
- /* Everything was wrong, so give the device one more try with a new challenge */
- if (!req->ignore) {
- if (sipdebug) {
- ast_log(LOG_NOTICE, "Bad authentication received from '%s'\n", sip_get_header(req, "To"));
- }
- build_nonce(p, 1);
- } else {
- if (sipdebug) {
- ast_log(LOG_NOTICE, "Duplicate authentication received from '%s'\n", sip_get_header(req, "To"));
- }
- }
- transmit_response_with_auth(p, response, req, p->nonce, reliable, respheader, FALSE);
- }
-
- /* Schedule auto destroy in 32 seconds */
- sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
- return AUTH_CHALLENGE_SENT;
- }
- if (good_response) {
- append_history(p, "AuthOK", "Auth challenge successful for %s", username);
- return AUTH_SUCCESSFUL;
- }
-
- /* Ok, we have a bad username/secret pair */
- /* Tell the UAS not to re-send this authentication data, because
- it will continue to fail
- */
-
- return AUTH_SECRET_FAILED;
-}
-
-/*! \brief Change onhold state of a peer using a pvt structure */
-static void sip_peer_hold(struct sip_pvt *p, int hold)
-{
- if (!p->relatedpeer) {
- return;
- }
-
- /* If they put someone on hold, increment the value... otherwise decrement it */
- ast_atomic_fetchadd_int(&p->relatedpeer->onhold, (hold ? +1 : -1));
-
- /* Request device state update */
- ast_devstate_changed(AST_DEVICE_UNKNOWN, (ast_test_flag(ast_channel_flags(p->owner), AST_FLAG_DISABLE_DEVSTATE_CACHE) ? AST_DEVSTATE_NOT_CACHABLE : AST_DEVSTATE_CACHABLE),
- "SIP/%s", p->relatedpeer->name);
-
- return;
-}
-
-/*! \brief Receive MWI events that we have subscribed to */
-static void mwi_event_cb(void *userdata, struct stasis_subscription *sub, struct stasis_message *msg)
-{
- struct sip_peer *peer = userdata;
-
- /*
- * peer can't be NULL here but the peer can be in the process of being
- * destroyed. If it is, we don't want to send any messages. In most cases,
- * the peer is actually gone and there's no sense sending NOTIFYs that will
- * never be answered.
- */
- if (stasis_subscription_final_message(sub, msg) || peer_in_destruction(peer)) {
- return;
- }
-
- if (ast_mwi_state_type() == stasis_message_type(msg)) {
- sip_send_mwi_to_peer(peer, 0);
- }
-}
-
-static void network_change_stasis_subscribe(void)
-{
- if (!network_change_sub) {
- network_change_sub = stasis_subscribe(ast_system_topic(),
- network_change_stasis_cb, NULL);
- stasis_subscription_accept_message_type(network_change_sub, ast_network_change_type());
- stasis_subscription_set_filter(network_change_sub, STASIS_SUBSCRIPTION_FILTER_SELECTIVE);
- }
-}
-
-static void network_change_stasis_unsubscribe(void)
-{
- network_change_sub = stasis_unsubscribe_and_join(network_change_sub);
-}
-
-static void acl_change_stasis_subscribe(void)
-{
- if (!acl_change_sub) {
- acl_change_sub = stasis_subscribe(ast_security_topic(),
- acl_change_stasis_cb, NULL);
- stasis_subscription_accept_message_type(acl_change_sub, ast_named_acl_change_type());
- stasis_subscription_set_filter(acl_change_sub, STASIS_SUBSCRIPTION_FILTER_SELECTIVE);
- }
-
-}
-
-static void acl_change_event_stasis_unsubscribe(void)
-{
- acl_change_sub = stasis_unsubscribe_and_join(acl_change_sub);
-}
-
-/* Run by the sched thread. */
-static int network_change_sched_cb(const void *data)
-{
- network_change_sched_id = -1;
- sip_send_all_registers();
- sip_send_all_mwi_subscriptions();
- return 0;
-}
-
-static void network_change_stasis_cb(void *data, struct stasis_subscription *sub, struct stasis_message *message)
-{
- /* This callback is only concerned with network change messages from the system topic. */
- if (stasis_message_type(message) != ast_network_change_type()) {
- return;
- }
-
- ast_verb(1, "SIP, got a network change message, renewing all SIP registrations.\n");
- if (network_change_sched_id == -1) {
- network_change_sched_id = ast_sched_add(sched, 1000, network_change_sched_cb, NULL);
- }
-}
-
-static void cb_extensionstate_destroy(int id, void *data)
-{
- struct sip_pvt *p = data;
-
- dialog_unref(p, "the extensionstate containing this dialog ptr was destroyed");
-}
-
-/*! \brief Callback for the devicestate notification (SUBSCRIBE) support subsystem
-\note If you add an "hint" priority to the extension in the dial plan,
- you will get notifications on device state changes */
-static int extensionstate_update(const char *context, const char *exten, struct state_notify_data *data, struct sip_pvt *p, int force)
-{
- sip_pvt_lock(p);
-
- switch (data->state) {
- case AST_EXTENSION_DEACTIVATED: /* Retry after a while */
- case AST_EXTENSION_REMOVED: /* Extension is gone */
- sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT); /* Delete subscription in 32 secs */
- ast_verb(2, "Extension state: Watcher for hint %s %s. Notify User %s\n", exten, data->state == AST_EXTENSION_DEACTIVATED ? "deactivated" : "removed", p->username);
- p->subscribed = NONE;
- append_history(p, "Subscribestatus", "%s", data->state == AST_EXTENSION_REMOVED ? "HintRemoved" : "Deactivated");
- break;
- default: /* Tell user */
- if (force) {
- /* we must skip the next two checks for a queued state change or resubscribe */
- } else if ((p->laststate == data->state && (~data->state & AST_EXTENSION_RINGING)) &&
- (p->last_presence_state == data->presence_state &&
- !strcmp(p->last_presence_subtype, data->presence_subtype) &&
- !strcmp(p->last_presence_message, data->presence_message))) {
- /* don't notify unchanged state or unchanged early-state causing parties again */
- sip_pvt_unlock(p);
- return 0;
- } else if (data->state & AST_EXTENSION_RINGING) {
- /* check if another channel than last time is ringing now to be notified */
- struct ast_channel *ringing = find_ringing_channel(data->device_state_info, p);
- if (ringing) {
- if (!ast_tvcmp(ast_channel_creationtime(ringing), p->last_ringing_channel_time)) {
- /* we assume here that no two channels have the exact same creation time */
- ao2_ref(ringing, -1);
- sip_pvt_unlock(p);
- return 0;
- } else {
- p->last_ringing_channel_time = ast_channel_creationtime(ringing);
- ao2_ref(ringing, -1);
- }
- }
- /* If no ringing channel was found, it doesn't necessarily indicate anything bad.
- * Likely, a device state change occurred for a custom device state, which does not
- * correspond to any channel. In such a case, just go ahead and pass the notification
- * along.
- */
- }
- /* ref before unref because the new could be the same as the old one. Don't risk destruction! */
- if (data->device_state_info) {
- ao2_ref(data->device_state_info, 1);
- }
- ao2_cleanup(p->last_device_state_info);
- p->last_device_state_info = data->device_state_info;
- p->laststate = data->state;
- p->last_presence_state = data->presence_state;
- ast_string_field_set(p, last_presence_subtype, S_OR(data->presence_subtype, ""));
- ast_string_field_set(p, last_presence_message, S_OR(data->presence_message, ""));
- break;
- }
- if (p->subscribed != NONE) { /* Only send state NOTIFY if we know the format */
- if (!p->pendinginvite) {
- transmit_state_notify(p, data, 1, FALSE);
- if (p->last_device_state_info) {
- /* We don't need the saved ref anymore, don't keep channels ref'd. */
- ao2_ref(p->last_device_state_info, -1);
- p->last_device_state_info = NULL;
- }
- } else {
- /* We already have a NOTIFY sent that is not answered. Queue the state up.
- if many state changes happen meanwhile, we will only send a notification of the last one */
- ast_set_flag(&p->flags[1], SIP_PAGE2_STATECHANGEQUEUE);
- }
- }
-
- if (!force) {
- ast_verb(2, "Extension Changed %s[%s] new state %s for Notify User %s %s\n", exten, context, ast_extension_state2str(data->state), p->username,
- ast_test_flag(&p->flags[1], SIP_PAGE2_STATECHANGEQUEUE) ? "(queued)" : "");
- }
-
- sip_pvt_unlock(p);
-
- return 0;
-}
-
-/*! \brief Callback for the devicestate notification (SUBSCRIBE) support subsystem
-\note If you add an "hint" priority to the extension in the dial plan,
- you will get notifications on device state changes */
-static int cb_extensionstate(const char *context, const char *exten, struct ast_state_cb_info *info, void *data)
-{
- struct sip_pvt *p = data;
- struct state_notify_data notify_data = {
- .state = info->exten_state,
- .device_state_info = info->device_state_info,
- .presence_state = info->presence_state,
- .presence_subtype = info->presence_subtype,
- .presence_message = info->presence_message,
- };
-
- if ((info->reason == AST_HINT_UPDATE_PRESENCE) && !(allow_notify_user_presence(p))) {
- /* ignore a presence triggered update if we know the useragent doesn't care */
- return 0;
- }
-
- return extensionstate_update(context, exten, ¬ify_data, p, FALSE);
-}
-
-/*! \brief Send a fake 401 Unauthorized response when the administrator
- wants to hide the names of local devices from fishers
- */
-static void transmit_fake_auth_response(struct sip_pvt *p, struct sip_request *req, enum xmittype reliable)
-{
- /* We have to emulate EXACTLY what we'd get with a good peer
- * and a bad password, or else we leak information. */
- const char *response = "401 Unauthorized";
- const char *reqheader = "Authorization";
- const char *respheader = "WWW-Authenticate";
- const char *authtoken;
- struct ast_str *buf;
- char *c;
-
- /* table of recognised keywords, and their value in the digest */
- enum keys { K_NONCE, K_LAST };
- struct x {
- const char *key;
- const char *s;
- } *i, keys[] = {
- [K_NONCE] = { "nonce=", "" },
- [K_LAST] = { NULL, NULL}
- };
-
- authtoken = sip_get_header(req, reqheader);
- if (req->ignore && !ast_strlen_zero(p->nonce) && ast_strlen_zero(authtoken)) {
- /* This is a retransmitted invite/register/etc, don't reconstruct authentication
- * information */
- transmit_response_with_auth(p, response, req, p->nonce, reliable, respheader, 0);
- /* Schedule auto destroy in 32 seconds (according to RFC 3261) */
- sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
- return;
- } else if (ast_strlen_zero(p->nonce) || ast_strlen_zero(authtoken)) {
- /* We have no auth, so issue challenge and request authentication */
- build_nonce(p, 1);
- transmit_response_with_auth(p, response, req, p->nonce, reliable, respheader, 0);
- /* Schedule auto destroy in 32 seconds */
- sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
- return;
- }
-
- if (!(buf = ast_str_thread_get(&check_auth_buf, CHECK_AUTH_BUF_INITLEN))) {
- __transmit_response(p, "403 Forbidden", &p->initreq, reliable);
- return;
- }
-
- /* Make a copy of the response and parse it */
- if (ast_str_set(&buf, 0, "%s", authtoken) == AST_DYNSTR_BUILD_FAILED) {
- __transmit_response(p, "403 Forbidden", &p->initreq, reliable);
- return;
- }
-
- c = ast_str_buffer(buf);
-
- while (c && *(c = ast_skip_blanks(c))) { /* lookup for keys */
- for (i = keys; i->key != NULL; i++) {
- const char *separator = ","; /* default */
-
- if (strncasecmp(c, i->key, strlen(i->key)) != 0) {
- continue;
- }
- /* Found. Skip keyword, take text in quotes or up to the separator. */
- c += strlen(i->key);
- if (*c == '"') { /* in quotes. Skip first and look for last */
- c++;
- separator = "\"";
- }
- i->s = c;
- strsep(&c, separator);
- break;
- }
- if (i->key == NULL) { /* not found, jump after space or comma */
- strsep(&c, " ,");
- }
- }
-
- /* Verify nonce from request matches our nonce. If not, send 401 with new nonce */
- if (strcasecmp(p->nonce, keys[K_NONCE].s)) {
- if (!req->ignore) {
- build_nonce(p, 1);
- }
- transmit_response_with_auth(p, response, req, p->nonce, reliable, respheader, FALSE);
-
- /* Schedule auto destroy in 32 seconds */
- sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
- } else {
- __transmit_response(p, "403 Forbidden", &p->initreq, reliable);
- }
-}
-
-/*!
- * Terminate the uri at the first ';' or space.
- * Technically we should ignore escaped space per RFC3261 (19.1.1 etc)
- * but don't do it for the time being. Remember the uri format is:
- * (User-parameters was added after RFC 3261)
- *\verbatim
- *
- * sip:user:password;user-parameters@host:port;uri-parameters?headers
- * sips:user:password;user-parameters@host:port;uri-parameters?headers
- *
- *\endverbatim
- * \todo As this function does not support user-parameters, it's considered broken
- * and needs fixing.
- */
-static char *terminate_uri(char *uri)
-{
- char *t = uri;
- while (*t && *t > ' ' && *t != ';') {
- t++;
- }
- *t = '\0';
- return uri;
-}
-
-/*! \brief Terminate a host:port at the ':'
- * \param hostport The address of the hostport string
- *
- * \note In the case of a bracket-enclosed IPv6 address, the hostport variable
- * will contain the non-bracketed host as a result of calling this function.
- */
-static void extract_host_from_hostport(char **hostport)
-{
- char *dont_care;
- ast_sockaddr_split_hostport(*hostport, hostport, &dont_care, PARSE_PORT_IGNORE);
-}
-
-/*!
- * \internal
- * \brief Helper function to update a peer's lastmsgssent value
- */
-static void update_peer_lastmsgssent(struct sip_peer *peer, int value, int locked)
-{
- if (!locked) {
- ao2_lock(peer);
- }
- peer->lastmsgssent = value;
- if (!locked) {
- ao2_unlock(peer);
- }
-}
-
-
-/*!
- * \brief Verify registration of user
- *
- * \details
- * - Registration is done in several steps, first a REGISTER without auth
- * to get a challenge (nonce) then a second one with auth
- * - Registration requests are only matched with peers that are marked as "dynamic"
- */
-static enum check_auth_result register_verify(struct sip_pvt *p, struct ast_sockaddr *addr,
- struct sip_request *req, const char *uri)
-{
- enum check_auth_result res = AUTH_NOT_FOUND;
- struct sip_peer *peer;
- char tmp[256];
- char *c, *name, *unused_password, *domain;
- char *uri2 = ast_strdupa(uri);
- int send_mwi = 0;
-
- terminate_uri(uri2);
-
- ast_copy_string(tmp, sip_get_header(req, "To"), sizeof(tmp));
-
- c = get_in_brackets(tmp);
- c = remove_uri_parameters(c);
-
- if (parse_uri_legacy_check(c, "sip:,sips:", &name, &unused_password, &domain, NULL)) {
- ast_log(LOG_NOTICE, "Invalid to address: '%s' from %s (missing sip:) trying to use anyway...\n", c, ast_sockaddr_stringify_addr(addr));
- return -1;
- }
-
- SIP_PEDANTIC_DECODE(name);
- SIP_PEDANTIC_DECODE(domain);
-
- extract_host_from_hostport(&domain);
-
- if (ast_strlen_zero(domain)) {
- /* , never good */
- transmit_response(p, "404 Not found", &p->initreq);
- return AUTH_UNKNOWN_DOMAIN;
- }
-
- if (ast_strlen_zero(name)) {
- /* , unsure whether valid for
- * registration. RFC 3261, 10.2 states:
- * "The To header field and the Request-URI field typically
- * differ, as the former contains a user name."
- * But, Asterisk has always treated the domain-only uri as a
- * username: we allow admins to create accounts described by
- * domain name. */
- name = domain;
- }
-
- /* This here differs from 1.4 and 1.6: the domain matching ACLs were
- * skipped if it was a domain-only URI (used as username). Here we treat
- * as and won't forget to test the
- * domain ACLs against host. */
- if (!AST_LIST_EMPTY(&domain_list)) {
- if (!check_sip_domain(domain, NULL, 0)) {
- if (sip_cfg.alwaysauthreject) {
- transmit_fake_auth_response(p, &p->initreq, XMIT_UNRELIABLE);
- } else {
- transmit_response(p, "404 Not found (unknown domain)", &p->initreq);
- }
- return AUTH_UNKNOWN_DOMAIN;
- }
- }
-
- ast_string_field_set(p, exten, name);
- build_contact(p, req, 1);
- if (req->ignore) {
- /* Expires is a special case, where we only want to load the peer if this isn't a deregistration attempt */
- const char *expires = sip_get_header(req, "Expires");
- int expire = atoi(expires);
-
- if (ast_strlen_zero(expires)) { /* No expires header; look in Contact */
- if ((expires = strcasestr(sip_get_header(req, "Contact"), ";expires="))) {
- expire = atoi(expires + 9);
- }
- }
- if (!ast_strlen_zero(expires) && expire == 0) {
- transmit_response_with_date(p, "200 OK", req);
- return 0;
- }
- }
- peer = sip_find_peer(name, NULL, TRUE, FINDPEERS, FALSE, 0);
-
- /* If we don't want username disclosure, use the bogus_peer when a user
- * is not found. */
- if (!peer && sip_cfg.alwaysauthreject && sip_cfg.autocreatepeer == AUTOPEERS_DISABLED) {
- peer = ao2_t_global_obj_ref(g_bogus_peer, "register_verify: Get the bogus peer.");
- }
-
- if (!(peer && ast_apply_acl(peer->acl, addr, "SIP Peer ACL: "))) {
- /* Peer fails ACL check */
- if (peer) {
- sip_unref_peer(peer, "register_verify: sip_unref_peer: from sip_find_peer operation");
- peer = NULL;
- res = AUTH_ACL_FAILED;
- } else {
- res = AUTH_NOT_FOUND;
- }
- }
-
- if (peer) {
- ao2_lock(peer);
- if (!peer->host_dynamic) {
- ast_log(LOG_ERROR, "Peer '%s' is trying to register, but not configured as host=dynamic\n", peer->name);
- res = AUTH_PEER_NOT_DYNAMIC;
- } else {
-
- set_peer_nat(p, peer);
-
- ast_copy_flags(&p->flags[0], &peer->flags[0], SIP_NAT_FORCE_RPORT);
-
- if (!(res = check_auth(p, req, peer->name, peer->secret, peer->md5secret, SIP_REGISTER, uri2, XMIT_UNRELIABLE))) {
- sip_cancel_destroy(p);
-
- if (check_request_transport(peer, req)) {
- ast_set_flag(&p->flags[0], SIP_PENDINGBYE);
- transmit_response_with_date(p, "403 Forbidden", req);
- res = AUTH_BAD_TRANSPORT;
- } else {
-
- /* We have a successful registration attempt with proper authentication,
- now, update the peer */
- switch (parse_register_contact(p, peer, req)) {
- case PARSE_REGISTER_DENIED:
- ast_log(LOG_WARNING, "Registration denied because of contact ACL\n");
- transmit_response_with_date(p, "603 Denied", req);
- res = 0;
- break;
- case PARSE_REGISTER_FAILED:
- ast_log(LOG_WARNING, "Failed to parse contact info\n");
- transmit_response_with_date(p, "400 Bad Request", req);
- res = 0;
- break;
- case PARSE_REGISTER_QUERY:
- ast_string_field_set(p, fullcontact, peer->fullcontact);
- transmit_response_with_date(p, "200 OK", req);
- res = 0;
- send_mwi = 1;
- break;
- case PARSE_REGISTER_UPDATE:
- ast_string_field_set(p, fullcontact, peer->fullcontact);
- /* If expiry is 0, peer has been unregistered already */
- if (p->expiry != 0) {
- update_peer(peer, p->expiry);
- }
- /* Say OK and ask subsystem to retransmit msg counter */
- transmit_response_with_date(p, "200 OK", req);
- send_mwi = 1;
- res = 0;
- break;
- }
- }
-
- }
- }
- ao2_unlock(peer);
- }
- if (!peer && sip_cfg.autocreatepeer != AUTOPEERS_DISABLED) {
- /* Create peer if we have autocreate mode enabled */
- peer = temp_peer(name);
- if (peer && !(peer->endpoint = ast_endpoint_create("SIP", name))) {
- ao2_t_ref(peer, -1, "failed to allocate Stasis endpoint, drop peer");
- peer = NULL;
- }
- if (peer) {
- ao2_t_link(peers, peer, "link peer into peer table");
- if (!ast_sockaddr_isnull(&peer->addr)) {
- ao2_t_link(peers_by_ip, peer, "link peer into peers-by-ip table");
- }
- ao2_lock(peer);
- sip_cancel_destroy(p);
- switch (parse_register_contact(p, peer, req)) {
- case PARSE_REGISTER_DENIED:
- ast_log(LOG_WARNING, "Registration denied because of contact ACL\n");
- transmit_response_with_date(p, "403 Forbidden", req);
- res = 0;
- break;
- case PARSE_REGISTER_FAILED:
- ast_log(LOG_WARNING, "Failed to parse contact info\n");
- transmit_response_with_date(p, "400 Bad Request", req);
- res = 0;
- break;
- case PARSE_REGISTER_QUERY:
- ast_string_field_set(p, fullcontact, peer->fullcontact);
- transmit_response_with_date(p, "200 OK", req);
- send_mwi = 1;
- res = 0;
- break;
- case PARSE_REGISTER_UPDATE:
- ast_string_field_set(p, fullcontact, peer->fullcontact);
- /* Say OK and ask subsystem to retransmit msg counter */
- transmit_response_with_date(p, "200 OK", req);
- if (peer->endpoint) {
- RAII_VAR(struct ast_json *, blob, NULL, ast_json_unref);
- ast_endpoint_set_state(peer->endpoint, AST_ENDPOINT_ONLINE);
- blob = ast_json_pack("{s: s, s: s}",
- "peer_status", "Registered",
- "address", ast_sockaddr_stringify(addr));
- ast_endpoint_blob_publish(peer->endpoint, ast_endpoint_state_type(), blob);
- }
- send_mwi = 1;
- res = 0;
- break;
- }
- ao2_unlock(peer);
- }
- }
- if (!res) {
- if (send_mwi) {
- sip_pvt_unlock(p);
- sip_send_mwi_to_peer(peer, 0);
- sip_pvt_lock(p);
- } else {
- update_peer_lastmsgssent(peer, -1, 0);
- }
- ast_devstate_changed(AST_DEVICE_UNKNOWN, AST_DEVSTATE_CACHABLE, "SIP/%s", peer->name);
- }
- if (res < 0) {
- RAII_VAR(struct ast_json *, blob, NULL, ast_json_unref);
-
- switch (res) {
- case AUTH_SECRET_FAILED:
- /* Wrong password in authentication. Go away, don't try again until you fixed it */
- transmit_response(p, "403 Forbidden", &p->initreq);
- if (global_authfailureevents) {
- const char *peer_addr = ast_strdupa(ast_sockaddr_stringify_addr(addr));
- const char *peer_port = ast_strdupa(ast_sockaddr_stringify_port(addr));
-
- blob = ast_json_pack("{s: s, s: s, s: s, s: s}",
- "peer_status", "Rejected",
- "cause", "AUTH_SECRET_FAILED",
- "address", peer_addr,
- "port", peer_port);
- }
- break;
- case AUTH_USERNAME_MISMATCH:
- /* Username and digest username does not match.
- Asterisk uses the From: username for authentication. We need the
- devices to use the same authentication user name until we support
- proper authentication by digest auth name */
- case AUTH_NOT_FOUND:
- case AUTH_PEER_NOT_DYNAMIC:
- case AUTH_ACL_FAILED:
- if (sip_cfg.alwaysauthreject) {
- transmit_fake_auth_response(p, &p->initreq, XMIT_UNRELIABLE);
- if (global_authfailureevents) {
- const char *peer_addr = ast_strdupa(ast_sockaddr_stringify_addr(addr));
- const char *peer_port = ast_strdupa(ast_sockaddr_stringify_port(addr));
-
- blob = ast_json_pack("{s: s, s: s, s: s, s: s}",
- "peer_status", "Rejected",
- "cause", res == AUTH_PEER_NOT_DYNAMIC ? "AUTH_PEER_NOT_DYNAMIC" : "URI_NOT_FOUND",
- "address", peer_addr,
- "port", peer_port);
- }
- } else {
- /* URI not found */
- if (res == AUTH_PEER_NOT_DYNAMIC) {
- transmit_response(p, "403 Forbidden", &p->initreq);
- if (global_authfailureevents) {
- const char *peer_addr = ast_strdupa(ast_sockaddr_stringify_addr(addr));
- const char *peer_port = ast_strdupa(ast_sockaddr_stringify_port(addr));
-
- blob = ast_json_pack("{s: s, s: s, s: s, s: s}",
- "peer_status", "Rejected",
- "cause", "AUTH_PEER_NOT_DYNAMIC",
- "address", peer_addr,
- "port", peer_port);
- }
- } else {
- transmit_response(p, "404 Not found", &p->initreq);
- if (global_authfailureevents) {
- const char *peer_addr = ast_strdupa(ast_sockaddr_stringify_addr(addr));
- const char *peer_port = ast_strdupa(ast_sockaddr_stringify_port(addr));
-
- blob = ast_json_pack("{s: s, s: s, s: s, s: s}",
- "peer_status", "Rejected",
- "cause", (res == AUTH_USERNAME_MISMATCH) ? "AUTH_USERNAME_MISMATCH" : "URI_NOT_FOUND",
- "address", peer_addr,
- "port", peer_port);
- }
- }
- }
- break;
- case AUTH_BAD_TRANSPORT:
- default:
- break;
- }
-
- if (peer && peer->endpoint) {
- ast_endpoint_blob_publish(peer->endpoint, ast_endpoint_state_type(), blob);
- }
- }
- if (peer) {
- sip_unref_peer(peer, "register_verify: sip_unref_peer: tossing stack peer pointer at end of func");
- }
-
- return res;
-}
-
-/*! \brief Translate referring cause */
-static void sip_set_redirstr(struct sip_pvt *p, char *reason) {
-
- if (!strcmp(reason, "unknown")) {
- ast_string_field_set(p, redircause, "UNKNOWN");
- } else if (!strcmp(reason, "user-busy")) {
- ast_string_field_set(p, redircause, "BUSY");
- } else if (!strcmp(reason, "no-answer")) {
- ast_string_field_set(p, redircause, "NOANSWER");
- } else if (!strcmp(reason, "unavailable")) {
- ast_string_field_set(p, redircause, "UNREACHABLE");
- } else if (!strcmp(reason, "unconditional")) {
- ast_string_field_set(p, redircause, "UNCONDITIONAL");
- } else if (!strcmp(reason, "time-of-day")) {
- ast_string_field_set(p, redircause, "UNKNOWN");
- } else if (!strcmp(reason, "do-not-disturb")) {
- ast_string_field_set(p, redircause, "UNKNOWN");
- } else if (!strcmp(reason, "deflection")) {
- ast_string_field_set(p, redircause, "UNKNOWN");
- } else if (!strcmp(reason, "follow-me")) {
- ast_string_field_set(p, redircause, "UNKNOWN");
- } else if (!strcmp(reason, "out-of-service")) {
- ast_string_field_set(p, redircause, "UNREACHABLE");
- } else if (!strcmp(reason, "away")) {
- ast_string_field_set(p, redircause, "UNREACHABLE");
- } else {
- ast_string_field_set(p, redircause, "UNKNOWN");
- }
-}
-
-/*! \brief Parse the parts of the P-Asserted-Identity header
- * on an incoming packet. Returns 1 if a valid header is found
- * and it is different from the current caller id.
- */
-static int get_pai(struct sip_pvt *p, struct sip_request *req)
-{
- char pai[256];
- char privacy[64];
- char *cid_num = NULL;
- char *cid_name = NULL;
- char emptyname[1] = "";
- int callingpres = AST_PRES_ALLOWED_USER_NUMBER_NOT_SCREENED;
- char *uri = NULL;
- int is_anonymous = 0, do_update = 1, no_name = 0;
-
- ast_copy_string(pai, sip_get_header(req, "P-Asserted-Identity"), sizeof(pai));
-
- if (ast_strlen_zero(pai)) {
- return 0;
- }
-
- /* use the reqresp_parser function get_name_and_number*/
- if (get_name_and_number(pai, &cid_name, &cid_num)) {
- return 0;
- }
-
- if (global_shrinkcallerid && ast_is_shrinkable_phonenumber(cid_num)) {
- ast_shrink_phone_number(cid_num);
- }
-
- uri = get_in_brackets(pai);
- if (!strncasecmp(uri, "sip:anonymous@anonymous.invalid", 31)) {
- callingpres = AST_PRES_PROHIB_USER_NUMBER_NOT_SCREENED;
- /*XXX Assume no change in cid_num. Perhaps it should be
- * blanked?
- */
- ast_free(cid_num);
- is_anonymous = 1;
- cid_num = (char *)p->cid_num;
- }
-
- ast_copy_string(privacy, sip_get_header(req, "Privacy"), sizeof(privacy));
- if (!ast_strlen_zero(privacy) && strcasecmp(privacy, "none")) {
- callingpres = AST_PRES_PROHIB_USER_NUMBER_NOT_SCREENED;
- }
- if (!cid_name) {
- no_name = 1;
- cid_name = (char *)emptyname;
- }
- /* Only return true if the supplied caller id is different */
- if (!strcasecmp(p->cid_num, cid_num) && !strcasecmp(p->cid_name, cid_name) && p->callingpres == callingpres) {
- do_update = 0;
- } else {
-
- ast_string_field_set(p, cid_num, cid_num);
- ast_string_field_set(p, cid_name, cid_name);
- p->callingpres = callingpres;
-
- if (p->owner) {
- ast_set_callerid(p->owner, cid_num, cid_name, NULL);
- ast_channel_caller(p->owner)->id.name.presentation = callingpres;
- ast_channel_caller(p->owner)->id.number.presentation = callingpres;
- }
- }
-
- /* get_name_and_number allocates memory for cid_num and cid_name so we have to free it */
- if (!is_anonymous) {
- ast_free(cid_num);
- }
- if (!no_name) {
- ast_free(cid_name);
- }
-
- return do_update;
-}
-
-/*! \brief Get name, number and presentation from remote party id header,
- * returns true if a valid header was found and it was different from the
- * current caller id.
- */
-static int get_rpid(struct sip_pvt *p, struct sip_request *oreq)
-{
- char tmp[256];
- struct sip_request *req;
- char *cid_num = "";
- char *cid_name = "";
- int callingpres = AST_PRES_ALLOWED_USER_NUMBER_NOT_SCREENED;
- char *privacy = "";
- char *screen = "";
- char *start, *end;
-
- if (!ast_test_flag(&p->flags[0], SIP_TRUSTRPID))
- return 0;
- req = oreq;
- if (!req)
- req = &p->initreq;
- ast_copy_string(tmp, sip_get_header(req, "Remote-Party-ID"), sizeof(tmp));
- if (ast_strlen_zero(tmp)) {
- return get_pai(p, req);
- }
-
- /*
- * RPID is not:
- * rpid = (name-addr / addr-spec) *(SEMI rpi-token)
- * But it is:
- * rpid = [display-name] LAQUOT addr-spec RAQUOT *(SEMI rpi-token)
- * Ergo, calling parse_name_andor_addr() on it wouldn't be
- * correct because that would allow addr-spec style too.
- */
- start = tmp;
- /* Quoted (note that we're not dealing with escapes properly) */
- if (*start == '"') {
- *start++ = '\0';
- end = strchr(start, '"');
- if (!end)
- return 0;
- *end++ = '\0';
- cid_name = start;
- start = ast_skip_blanks(end);
- /* Unquoted */
- } else {
- cid_name = start;
- start = end = strchr(start, '<');
- if (!start) {
- return 0;
- }
- /* trim blanks if there are any. the mandatory NUL is done below */
- while (--end >= cid_name && *end < 33) {
- *end = '\0';
- }
- }
-
- if (*start != '<')
- return 0;
- *start++ = '\0';
- end = strchr(start, '@');
- if (!end)
- return 0;
- *end++ = '\0';
- if (strncasecmp(start, "sip:", 4))
- return 0;
- cid_num = start + 4;
- if (global_shrinkcallerid && ast_is_shrinkable_phonenumber(cid_num))
- ast_shrink_phone_number(cid_num);
- start = end;
-
- end = strchr(start, '>');
- if (!end)
- return 0;
- *end++ = '\0';
- if (*end) {
- start = end;
- if (*start != ';')
- return 0;
- *start++ = '\0';
- while (!ast_strlen_zero(start)) {
- end = strchr(start, ';');
- if (end)
- *end++ = '\0';
- if (!strncasecmp(start, "privacy=", 8))
- privacy = start + 8;
- else if (!strncasecmp(start, "screen=", 7))
- screen = start + 7;
- start = end;
- }
-
- if (!strcasecmp(privacy, "full")) {
- if (!strcasecmp(screen, "yes"))
- callingpres = AST_PRES_PROHIB_USER_NUMBER_PASSED_SCREEN;
- else if (!strcasecmp(screen, "no"))
- callingpres = AST_PRES_PROHIB_USER_NUMBER_NOT_SCREENED;
- } else {
- if (!strcasecmp(screen, "yes"))
- callingpres = AST_PRES_ALLOWED_USER_NUMBER_PASSED_SCREEN;
- else if (!strcasecmp(screen, "no"))
- callingpres = AST_PRES_ALLOWED_USER_NUMBER_NOT_SCREENED;
- }
- }
-
- /* Only return true if the supplied caller id is different */
- if (!strcasecmp(p->cid_num, cid_num) && !strcasecmp(p->cid_name, cid_name) && p->callingpres == callingpres)
- return 0;
-
- ast_string_field_set(p, cid_num, cid_num);
- ast_string_field_set(p, cid_name, cid_name);
- p->callingpres = callingpres;
-
- if (p->owner) {
- ast_set_callerid(p->owner, cid_num, cid_name, NULL);
- ast_channel_caller(p->owner)->id.name.presentation = callingpres;
- ast_channel_caller(p->owner)->id.number.presentation = callingpres;
- }
-
- return 1;
-}
-
-/*! \brief Get referring dnis
- *
- * \param p dialog information
- * \param oreq The request to retrieve RDNIS from
- * \param[out] name The name of the party redirecting the call.
- * \param[out] number The number of the party redirecting the call.
- * \param[out] reason_code The numerical code corresponding to the reason for the redirection.
- * \param[out] reason_str A string describing the reason for redirection. Will never be zero-length
- *
- * \retval -1 Could not retrieve RDNIS information
- * \retval 0 RDNIS successfully retrieved
- */
-static int get_rdnis(struct sip_pvt *p, struct sip_request *oreq, char **name, char **number, int *reason_code, char **reason_str)
-{
- char tmp[256], *exten, *rexten, *rdomain, *rname = NULL;
- char *params, *reason_param = NULL;
- struct sip_request *req;
-
- ast_assert(reason_code != NULL);
- ast_assert(reason_str != NULL);
-
- req = oreq ? oreq : &p->initreq;
-
- ast_copy_string(tmp, sip_get_header(req, "Diversion"), sizeof(tmp));
- if (ast_strlen_zero(tmp))
- return -1;
-
- if ((params = strchr(tmp, '>'))) {
- params = strchr(params, ';');
- }
-
- exten = get_in_brackets(tmp);
- if (!strncasecmp(exten, "sip:", 4)) {
- exten += 4;
- } else if (!strncasecmp(exten, "sips:", 5)) {
- exten += 5;
- } else {
- ast_log(LOG_WARNING, "Huh? Not an RDNIS SIP header (%s)?\n", exten);
- return -1;
- }
-
- /* Get diversion-reason param if present */
- if (params) {
- *params = '\0'; /* Cut off parameters */
- params++;
- while (*params == ';' || *params == ' ')
- params++;
- /* Check if we have a reason parameter */
- if ((reason_param = strcasestr(params, "reason="))) {
- char *end;
- reason_param+=7;
- if ((end = strchr(reason_param, ';'))) {
- *end = '\0';
- }
- }
- }
-
- rdomain = exten;
- rexten = strsep(&rdomain, "@"); /* trim anything after @ */
- if (p->owner)
- pbx_builtin_setvar_helper(p->owner, "__SIPRDNISDOMAIN", rdomain);
-
- if (sip_debug_test_pvt(p)) {
- ast_verbose("RDNIS for this call is %s (reason %s)\n", exten, S_OR(reason_param, ""));
- }
- /*ast_string_field_set(p, rdnis, rexten);*/
-
- if (*tmp == '\"') {
- char *end_quote;
- rname = tmp + 1;
- end_quote = strchr(rname, '\"');
- if (end_quote) {
- *end_quote = '\0';
- }
- }
-
- if (number) {
- *number = ast_strdup(rexten);
- }
-
- if (name && rname) {
- *name = ast_strdup(rname);
- }
-
- if (!ast_strlen_zero(reason_param)) {
- *reason_str = ast_strdup(reason_param);
-
- /* Remove any enclosing double-quotes */
- if (*reason_param == '"') {
- reason_param = ast_strip_quoted(reason_param, "\"", "\"");
- }
-
- *reason_code = ast_redirecting_reason_parse(reason_param);
- if (*reason_code < 0) {
- *reason_code = AST_REDIRECTING_REASON_UNKNOWN;
- } else {
- ast_free(*reason_str);
- *reason_str = ast_strdup("");
- }
-
- if (!ast_strlen_zero(reason_param)) {
- sip_set_redirstr(p, reason_param);
- if (p->owner) {
- pbx_builtin_setvar_helper(p->owner, "__PRIREDIRECTREASON", p->redircause);
- pbx_builtin_setvar_helper(p->owner, "__SIPREDIRECTREASON", reason_param);
- }
- }
- }
-
- return 0;
-}
-
-/*!
- * \brief Find out who the call is for.
- *
- * \details
- * We use the request uri as a destination.
- * This code assumes authentication has been done, so that the
- * device (peer/user) context is already set.
- *
- * \return 0 on success (found a matching extension), non-zero on failure
- *
- * \note If the incoming uri is a SIPS: uri, we are required to carry this across
- * the dialplan, so that the outbound call also is a sips: call or encrypted
- * IAX2 call. If that's not available, the call should FAIL.
- */
-static enum sip_get_dest_result get_destination(struct sip_pvt *p, struct sip_request *oreq, int *cc_recall_core_id)
-{
- char tmp[256] = "", *uri, *unused_password, *domain;
- RAII_VAR(char *, tmpf, NULL, ast_free);
- char *from = NULL;
- struct sip_request *req;
- char *decoded_uri;
- RAII_VAR(struct ast_features_pickup_config *, pickup_cfg, ast_get_chan_features_pickup_config(p->owner), ao2_cleanup);
- const char *pickupexten;
-
- if (!pickup_cfg) {
- ast_log(LOG_ERROR, "Unable to retrieve pickup configuration options. Unable to detect call pickup extension\n");
- pickupexten = "";
- } else {
- /* Don't need to duplicate since channel is locked for the duration of this function */
- pickupexten = pickup_cfg->pickupexten;
- }
-
- req = oreq;
- if (!req) {
- req = &p->initreq;
- }
-
- /* Find the request URI */
- if (req->rlpart2) {
- ast_copy_string(tmp, REQ_OFFSET_TO_STR(req, rlpart2), sizeof(tmp));
- }
-
- uri = ast_strdupa(get_in_brackets(tmp));
-
- if (parse_uri_legacy_check(uri, "sip:,sips:,tel:", &uri, &unused_password, &domain, NULL)) {
- ast_log(LOG_WARNING, "Not a SIP header (%s)?\n", uri);
- return SIP_GET_DEST_INVALID_URI;
- }
-
- SIP_PEDANTIC_DECODE(domain);
- SIP_PEDANTIC_DECODE(uri);
-
- extract_host_from_hostport(&domain);
-
- if (strncasecmp(get_in_brackets(tmp), "tel:", 4)) {
- ast_string_field_set(p, domain, domain);
- } else {
- ast_string_field_set(p, tel_phone_context, domain);
- }
-
- if (ast_strlen_zero(uri)) {
- /*
- * Either there really was no extension found or the request
- * URI had encoded nulls that made the string "empty". Use "s"
- * as the extension.
- */
- uri = "s";
- }
-
- /* Now find the From: caller ID and name */
- /* XXX Why is this done in get_destination? Isn't it already done?
- Needs to be checked
- */
- tmpf = ast_strdup(sip_get_header(req, "From"));
- if (!ast_strlen_zero(tmpf)) {
- from = get_in_brackets(tmpf);
- if (parse_uri_legacy_check(from, "sip:,sips:,tel:", &from, NULL, &domain, NULL)) {
- ast_log(LOG_WARNING, "Not a SIP header (%s)?\n", from);
- return SIP_GET_DEST_INVALID_URI;
- }
-
- SIP_PEDANTIC_DECODE(from);
- SIP_PEDANTIC_DECODE(domain);
-
- extract_host_from_hostport(&domain);
-
- ast_string_field_set(p, fromdomain, domain);
- }
-
- if (!AST_LIST_EMPTY(&domain_list)) {
- char domain_context[AST_MAX_EXTENSION];
-
- domain_context[0] = '\0';
- if (!check_sip_domain(p->domain, domain_context, sizeof(domain_context))) {
- if (!sip_cfg.allow_external_domains && (req->method == SIP_INVITE || req->method == SIP_REFER)) {
- ast_debug(1, "Got SIP %s to non-local domain '%s'; refusing request.\n", sip_methods[req->method].text, p->domain);
- return SIP_GET_DEST_REFUSED;
- }
- }
- /* If we don't have a peer (i.e. we're a guest call),
- * overwrite the original context */
- if (!ast_test_flag(&p->flags[1], SIP_PAGE2_HAVEPEERCONTEXT) && !ast_strlen_zero(domain_context)) {
- ast_string_field_set(p, context, domain_context);
- }
- }
-
- /* If the request coming in is a subscription and subscribecontext has been specified use it */
- if (req->method == SIP_SUBSCRIBE && !ast_strlen_zero(p->subscribecontext)) {
- ast_string_field_set(p, context, p->subscribecontext);
- }
-
- if (sip_debug_test_pvt(p)) {
- ast_verbose("Looking for %s in %s (domain %s)\n", uri, p->context, p->domain);
- }
-
- /* Since extensions.conf can have unescaped characters, try matching a
- * decoded uri in addition to the non-decoded uri. */
- decoded_uri = ast_strdupa(uri);
- ast_uri_decode(decoded_uri, ast_uri_sip_user);
-
- /* If this is a subscription we actually just need to see if a hint exists for the extension */
- if (req->method == SIP_SUBSCRIBE) {
- int which = 0;
-
- if (ast_get_hint(NULL, 0, NULL, 0, NULL, p->context, uri)
- || (ast_get_hint(NULL, 0, NULL, 0, NULL, p->context, decoded_uri)
- && (which = 1))) {
- if (!oreq) {
- ast_string_field_set(p, exten, which ? decoded_uri : uri);
- }
- return SIP_GET_DEST_EXTEN_FOUND;
- } else {
- return SIP_GET_DEST_EXTEN_NOT_FOUND;
- }
- } else {
- struct ast_cc_agent *agent;
- /* Check the dialplan for the username part of the request URI,
- the domain will be stored in the SIPDOMAIN variable
- Return 0 if we have a matching extension */
- if (ast_exists_extension(NULL, p->context, uri, 1, S_OR(p->cid_num, from))) {
- if (!oreq) {
- ast_string_field_set(p, exten, uri);
- }
- return SIP_GET_DEST_EXTEN_FOUND;
- }
- if (ast_exists_extension(NULL, p->context, decoded_uri, 1, S_OR(p->cid_num, from))
- || !strcmp(decoded_uri, pickupexten)) {
- if (!oreq) {
- ast_string_field_set(p, exten, decoded_uri);
- }
- return SIP_GET_DEST_EXTEN_FOUND;
- }
- if ((agent = find_sip_cc_agent_by_notify_uri(tmp))) {
- struct sip_cc_agent_pvt *agent_pvt = agent->private_data;
- /* This is a CC recall. We can set p's extension to the exten from
- * the original INVITE
- */
- ast_string_field_set(p, exten, agent_pvt->original_exten);
- /* And we need to let the CC core know that the caller is attempting
- * his recall
- */
- ast_cc_agent_recalling(agent->core_id, "SIP caller %s is attempting recall",
- agent->device_name);
- if (cc_recall_core_id) {
- *cc_recall_core_id = agent->core_id;
- }
- ao2_ref(agent, -1);
- return SIP_GET_DEST_EXTEN_FOUND;
- }
- }
-
- if (ast_test_flag(&global_flags[1], SIP_PAGE2_ALLOWOVERLAP)
- && (ast_canmatch_extension(NULL, p->context, uri, 1, S_OR(p->cid_num, from))
- || ast_canmatch_extension(NULL, p->context, decoded_uri, 1, S_OR(p->cid_num, from))
- || !strncmp(decoded_uri, pickupexten, strlen(decoded_uri)))) {
- /* Overlap dialing is enabled and we need more digits to match an extension. */
- return SIP_GET_DEST_EXTEN_MATCHMORE;
- }
-
- return SIP_GET_DEST_EXTEN_NOT_FOUND;
-}
-
-/*! \brief Find a companion dialog based on Replaces information
- *
- * This information may come from a Refer-To header in a REFER or from
- * a Replaces header in an INVITE.
- *
- * This function will find the appropriate sip_pvt and increment the refcount
- * of both the sip_pvt and its owner channel. These two references are returned
- * in the out parameters
- *
- * \param callid Callid to search for
- * \param totag to-tag parameter from Replaces
- * \param fromtag from-tag parameter from Replaces
- * \param[out] out_pvt The found sip_pvt.
- * \param[out] out_chan The found sip_pvt's owner channel.
- * \retval 0 Success
- * \retval non-zero Failure
- */
-static int get_sip_pvt_from_replaces(const char *callid, const char *totag,
- const char *fromtag, struct sip_pvt **out_pvt, struct ast_channel **out_chan)
-{
- RAII_VAR(struct sip_pvt *, sip_pvt_ptr, NULL, ao2_cleanup);
- struct sip_pvt tmp_dialog = {
- .callid = callid,
- };
-
- if (totag) {
- ast_debug(4, "Looking for callid %s (fromtag %s totag %s)\n", callid, fromtag ? fromtag : "", totag ? totag : "");
- }
-
- /* Search dialogs and find the match */
-
- sip_pvt_ptr = ao2_t_find(dialogs, &tmp_dialog, OBJ_POINTER, "ao2_find of dialog in dialogs table");
- if (sip_pvt_ptr) {
- /* Go ahead and lock it (and its owner) before returning */
- SCOPED_LOCK(lock, sip_pvt_ptr, sip_pvt_lock, sip_pvt_unlock);
- if (sip_cfg.pedanticsipchecking) {
- unsigned char frommismatch = 0, tomismatch = 0;
-
- if (ast_strlen_zero(fromtag)) {
- ast_debug(4, "Matched %s call for callid=%s - no from tag specified, pedantic check fails\n",
- sip_pvt_ptr->outgoing_call == TRUE ? "OUTGOING": "INCOMING", sip_pvt_ptr->callid);
- return -1;
- }
-
- if (ast_strlen_zero(totag)) {
- ast_debug(4, "Matched %s call for callid=%s - no to tag specified, pedantic check fails\n",
- sip_pvt_ptr->outgoing_call == TRUE ? "OUTGOING": "INCOMING", sip_pvt_ptr->callid);
- return -1;
- }
- /* RFC 3891
- * > 3. User Agent Server Behavior: Receiving a Replaces Header
- * > The Replaces header contains information used to match an existing
- * > SIP dialog (call-id, to-tag, and from-tag). Upon receiving an INVITE
- * > with a Replaces header, the User Agent (UA) attempts to match this
- * > information with a confirmed or early dialog. The User Agent Server
- * > (UAS) matches the to-tag and from-tag parameters as if they were tags
- * > present in an incoming request. In other words, the to-tag parameter
- * > is compared to the local tag, and the from-tag parameter is compared
- * > to the remote tag.
- *
- * Thus, the totag is always compared to the local tag, regardless if
- * this our call is an incoming or outgoing call.
- */
- frommismatch = !!strcmp(fromtag, sip_pvt_ptr->theirtag);
- tomismatch = !!strcmp(totag, sip_pvt_ptr->tag);
-
- /* Don't check from if the dialog is not established, due to multi forking the from
- * can change when the call is not answered yet.
- */
- if ((frommismatch && ast_test_flag(&sip_pvt_ptr->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED)) || tomismatch) {
- if (frommismatch) {
- ast_debug(4, "Matched %s call for callid=%s - pedantic from tag check fails; their tag is %s our tag is %s\n",
- sip_pvt_ptr->outgoing_call == TRUE ? "OUTGOING": "INCOMING", sip_pvt_ptr->callid,
- fromtag, sip_pvt_ptr->theirtag);
- }
- if (tomismatch) {
- ast_debug(4, "Matched %s call for callid=%s - pedantic to tag check fails; their tag is %s our tag is %s\n",
- sip_pvt_ptr->outgoing_call == TRUE ? "OUTGOING": "INCOMING", sip_pvt_ptr->callid,
- totag, sip_pvt_ptr->tag);
- }
- return -1;
- }
- }
-
- if (totag)
- ast_debug(4, "Matched %s call - their tag is %s Our tag is %s\n",
- sip_pvt_ptr->outgoing_call == TRUE ? "OUTGOING": "INCOMING",
- sip_pvt_ptr->theirtag, sip_pvt_ptr->tag);
-
- *out_pvt = sip_pvt_ptr;
- if (out_chan) {
- *out_chan = sip_pvt_ptr->owner ? ast_channel_ref(sip_pvt_ptr->owner) : NULL;
- }
- }
-
- if (!sip_pvt_ptr) {
- /* return error if sip_pvt was not found */
- return -1;
- }
-
- /* If we're here sip_pvt_ptr has been copied to *out_pvt, prevent RAII_VAR cleanup */
- sip_pvt_ptr = NULL;
-
- return 0;
-}
-
-static void extract_transferrer_headers(const char *prefix, struct ast_channel *peer, const struct sip_request *req)
-{
- struct ast_str *pbxvar = ast_str_alloca(120);
- int i;
-
- /* The '*' alone matches all headers. */
- if (strcmp(prefix, "*") == 0) {
- prefix = "";
- }
-
- for (i = 0; i < req->headers; i++) {
- const char *header = REQ_OFFSET_TO_STR(req, header[i]);
- if (ast_begins_with(header, prefix)) {
- int hdrlen = strcspn(header, " \t:");
- const char *val = ast_skip_blanks(header + hdrlen);
- if (hdrlen > 0 && *val == ':') {
- ast_str_set(&pbxvar, -1, "~HASH~TRANSFER_DATA~%.*s~", hdrlen, header);
- pbx_builtin_setvar_helper(peer, ast_str_buffer(pbxvar), ast_skip_blanks(val + 1));
- }
- }
- }
-}
-
-/*! \brief Call transfer support (the REFER method)
- * Extracts Refer headers into pvt dialog structure
- *
- * \note If we get a SIPS uri in the refer-to header, we're required to set up a secure signalling path
- * to that extension. As a minimum, this needs to be added to a channel variable, if not a channel
- * flag.
- */
-static int get_refer_info(struct sip_pvt *transferer, struct sip_request *outgoing_req)
-{
- const char *p_referred_by = NULL;
- char *h_refer_to = NULL;
- char *h_referred_by = NULL;
- char *refer_to;
- const char *p_refer_to;
- char *referred_by_uri = NULL;
- char *ptr;
- struct sip_request *req = NULL;
- const char *transfer_context = NULL;
- struct sip_refer *refer;
-
- req = outgoing_req;
- refer = transferer->refer;
-
- if (!req) {
- req = &transferer->initreq;
- }
-
- p_refer_to = sip_get_header(req, "Refer-To");
- if (ast_strlen_zero(p_refer_to)) {
- ast_log(LOG_WARNING, "Refer-To Header missing. Skipping transfer.\n");
- return -2; /* Syntax error */
- }
- h_refer_to = ast_strdupa(p_refer_to);
- refer_to = get_in_brackets(h_refer_to);
- if (!strncasecmp(refer_to, "sip:", 4)) {
- refer_to += 4; /* Skip sip: */
- } else if (!strncasecmp(refer_to, "sips:", 5)) {
- refer_to += 5;
- } else {
- ast_log(LOG_WARNING, "Can't transfer to non-sip: URI. (Refer-to: %s)?\n", refer_to);
- return -3;
- }
-
- /* Get referred by header if it exists */
- p_referred_by = sip_get_header(req, "Referred-By");
-
- /* Give useful transfer information to the dialplan */
- if (transferer->owner) {
- RAII_VAR(struct ast_channel *, peer, NULL, ast_channel_cleanup);
- RAII_VAR(struct ast_channel *, owner_relock, NULL, ast_channel_cleanup);
- RAII_VAR(struct ast_channel *, owner_ref, NULL, ast_channel_cleanup);
-
- /* Grab a reference to transferer->owner to prevent it from going away */
- owner_ref = ast_channel_ref(transferer->owner);
-
- /* Established locking order here is bridge, channel, pvt
- * and the bridge will be locked during ast_channel_bridge_peer */
- ast_channel_unlock(owner_ref);
- sip_pvt_unlock(transferer);
-
- peer = ast_channel_bridge_peer(owner_ref);
- if (peer) {
- const char *get_xfrdata;
-
- pbx_builtin_setvar_helper(peer, "SIPREFERRINGCONTEXT",
- S_OR(transferer->context, NULL));
- pbx_builtin_setvar_helper(peer, "__SIPREFERREDBYHDR",
- S_OR(p_referred_by, NULL));
-
- ast_channel_lock(peer);
- get_xfrdata = pbx_builtin_getvar_helper(peer, "GET_TRANSFERRER_DATA");
- if (!ast_strlen_zero(get_xfrdata)) {
- extract_transferrer_headers(get_xfrdata, peer, req);
- }
- ast_channel_unlock(peer);
- }
-
- owner_relock = sip_pvt_lock_full(transferer);
- if (!owner_relock) {
- ast_debug(3, "Unable to reacquire owner channel lock, channel is gone\n");
- return -5;
- }
- }
-
- if (!ast_strlen_zero(p_referred_by)) {
- h_referred_by = ast_strdupa(p_referred_by);
-
- referred_by_uri = get_in_brackets(h_referred_by);
-
- if (!strncasecmp(referred_by_uri, "sip:", 4)) {
- referred_by_uri += 4; /* Skip sip: */
- } else if (!strncasecmp(referred_by_uri, "sips:", 5)) {
- referred_by_uri += 5; /* Skip sips: */
- } else {
- ast_log(LOG_WARNING, "Huh? Not a sip: header (Referred-by: %s). Skipping.\n", referred_by_uri);
- referred_by_uri = NULL;
- }
- }
-
- /* Check for arguments in the refer_to header */
- if ((ptr = strcasestr(refer_to, "replaces="))) {
- char *to = NULL, *from = NULL, *callid;
-
- /* This is an attended transfer */
- refer->attendedtransfer = 1;
-
- callid = ast_strdupa(ptr + 9);
- ast_uri_decode(callid, ast_uri_sip_user);
- if ((ptr = strchr(callid, ';'))) { /* Find options */
- *ptr++ = '\0';
- }
- ast_string_field_set(refer, replaces_callid, callid);
-
- if (ptr) {
- /* Find the different tags before we destroy the string */
- to = strcasestr(ptr, "to-tag=");
- from = strcasestr(ptr, "from-tag=");
- }
-
- /* Grab the to header */
- if (to) {
- ptr = to + 7;
- if ((to = strchr(ptr, '&'))) {
- *to = '\0';
- }
- if ((to = strchr(ptr, ';'))) {
- *to = '\0';
- }
- ast_string_field_set(refer, replaces_callid_totag, ptr);
- }
-
- if (from) {
- ptr = from + 9;
- if ((from = strchr(ptr, '&'))) {
- *from = '\0';
- }
- if ((from = strchr(ptr, ';'))) {
- *from = '\0';
- }
- ast_string_field_set(refer, replaces_callid_fromtag, ptr);
- }
-
- if (!strcmp(refer->replaces_callid, transferer->callid) &&
- (!sip_cfg.pedanticsipchecking ||
- (!strcmp(refer->replaces_callid_fromtag, transferer->theirtag) &&
- !strcmp(refer->replaces_callid_totag, transferer->tag)))) {
- ast_log(LOG_WARNING, "Got an attempt to replace own Call-ID on %s\n", transferer->callid);
- return -4;
- }
-
- if (!sip_cfg.pedanticsipchecking) {
- ast_debug(2, "Attended transfer: Will use Replace-Call-ID : %s (No check of from/to tags)\n", refer->replaces_callid);
- } else {
- ast_debug(2, "Attended transfer: Will use Replace-Call-ID : %s F-tag: %s T-tag: %s\n", refer->replaces_callid, refer->replaces_callid_fromtag ? refer->replaces_callid_fromtag : "", refer->replaces_callid_totag ? refer->replaces_callid_totag : "");
- }
- }
-
- if ((ptr = strchr(refer_to, '@'))) { /* Separate domain */
- char *urioption = NULL, *domain;
- int bracket = 0;
- *ptr++ = '\0';
-
- if ((urioption = strchr(ptr, ';'))) { /* Separate urioptions */
- *urioption++ = '\0';
- }
-
- domain = ptr;
-
- /* Remove :port */
- for (; *ptr != '\0'; ++ptr) {
- if (*ptr == ':' && bracket == 0) {
- *ptr = '\0';
- break;
- } else if (*ptr == '[') {
- ++bracket;
- } else if (*ptr == ']') {
- --bracket;
- }
- }
-
- SIP_PEDANTIC_DECODE(domain);
- SIP_PEDANTIC_DECODE(urioption);
-
- /* Save the domain for the dial plan */
- ast_string_field_set(refer, refer_to_domain, domain);
- if (urioption) {
- ast_string_field_set(refer, refer_to_urioption, urioption);
- }
- }
-
- if ((ptr = strchr(refer_to, ';'))) /* Remove options */
- *ptr = '\0';
-
- SIP_PEDANTIC_DECODE(refer_to);
- ast_string_field_set(refer, refer_to, refer_to);
-
- if (referred_by_uri) {
- if ((ptr = strchr(referred_by_uri, ';'))) /* Remove options */
- *ptr = '\0';
- SIP_PEDANTIC_DECODE(referred_by_uri);
- ast_string_field_build(refer, referred_by, "", referred_by_uri);
- } else {
- ast_string_field_set(refer, referred_by, "");
- }
-
- /* Determine transfer context */
- if (transferer->owner) {
- /* By default, use the context in the channel sending the REFER */
- transfer_context = pbx_builtin_getvar_helper(transferer->owner, "TRANSFER_CONTEXT");
- if (ast_strlen_zero(transfer_context)) {
- transfer_context = ast_channel_macrocontext(transferer->owner);
- }
- }
- if (ast_strlen_zero(transfer_context)) {
- transfer_context = S_OR(transferer->context, sip_cfg.default_context);
- }
-
- ast_string_field_set(refer, refer_to_context, transfer_context);
-
- /* Either an existing extension or the parking extension */
- if (refer->attendedtransfer || ast_exists_extension(NULL, transfer_context, refer_to, 1, NULL)) {
- if (sip_debug_test_pvt(transferer)) {
- ast_verbose("SIP transfer to extension %s@%s by %s\n", refer_to, transfer_context, S_OR(referred_by_uri, "Unknown"));
- }
- /* We are ready to transfer to the extension */
- return 0;
- }
- if (sip_debug_test_pvt(transferer))
- ast_verbose("Failed SIP Transfer to non-existing extension %s in context %s\n n", refer_to, transfer_context);
-
- /* Failure, we can't find this extension */
- return -1;
-}
-
-
-/*! \brief Call transfer support (old way, deprecated by the IETF)
- * \note does not account for SIPS: uri requirements, nor check transport
- */
-static int get_also_info(struct sip_pvt *p, struct sip_request *oreq)
-{
- char tmp[256] = "", *c, *a;
- struct sip_request *req = oreq ? oreq : &p->initreq;
- struct sip_refer *refer = NULL;
- const char *transfer_context = NULL;
-
- if (!sip_refer_alloc(p)) {
- return -1;
- }
-
- refer = p->refer;
-
- ast_copy_string(tmp, sip_get_header(req, "Also"), sizeof(tmp));
- c = get_in_brackets(tmp);
-
- if (parse_uri_legacy_check(c, "sip:,sips:", &c, NULL, &a, NULL)) {
- ast_log(LOG_WARNING, "Huh? Not a SIP header in Also: transfer (%s)?\n", c);
- return -1;
- }
-
- SIP_PEDANTIC_DECODE(c);
- SIP_PEDANTIC_DECODE(a);
-
- if (!ast_strlen_zero(a)) {
- ast_string_field_set(refer, refer_to_domain, a);
- }
-
- if (sip_debug_test_pvt(p))
- ast_verbose("Looking for %s in %s\n", c, p->context);
-
- /* Determine transfer context */
- if (p->owner) {
- /* By default, use the context in the channel sending the REFER */
- transfer_context = pbx_builtin_getvar_helper(p->owner, "TRANSFER_CONTEXT");
- if (ast_strlen_zero(transfer_context)) {
- transfer_context = ast_channel_macrocontext(p->owner);
- }
- }
- if (ast_strlen_zero(transfer_context)) {
- transfer_context = S_OR(p->context, sip_cfg.default_context);
- }
-
- if (ast_exists_extension(NULL, transfer_context, c, 1, NULL)) {
- /* This is a blind transfer */
- ast_debug(1, "SIP Bye-also transfer to Extension %s@%s \n", c, transfer_context);
- ast_string_field_set(refer, refer_to, c);
- ast_string_field_set(refer, referred_by, "");
- ast_string_field_set(refer, refer_contact, "");
- /* Set new context */
- ast_string_field_set(p, context, transfer_context);
- return 0;
- } else if (ast_canmatch_extension(NULL, p->context, c, 1, NULL)) {
- return 1;
- }
-
- return -1;
-}
-
-/*! \brief Set the peers nat flags if they are using auto_* settings */
-static void set_peer_nat(const struct sip_pvt *p, struct sip_peer *peer)
-{
-
- if (!p || !peer) {
- return;
- }
-
- if (ast_test_flag(&peer->flags[2], SIP_PAGE3_NAT_AUTO_RPORT)) {
- if (p->natdetected) {
- ast_set_flag(&peer->flags[0], SIP_NAT_FORCE_RPORT);
- } else {
- ast_clear_flag(&peer->flags[0], SIP_NAT_FORCE_RPORT);
- }
- }
-
- if (ast_test_flag(&peer->flags[2], SIP_PAGE3_NAT_AUTO_COMEDIA)) {
- if (p->natdetected) {
- ast_set_flag(&peer->flags[1], SIP_PAGE2_SYMMETRICRTP);
- } else {
- ast_clear_flag(&peer->flags[1], SIP_PAGE2_SYMMETRICRTP);
- }
- }
-}
-
-/*! \brief Check and see if the requesting UA is likely to be behind a NAT.
- *
- * If the requesting NAT is behind NAT, set the * natdetected flag so that
- * later, peers with nat=auto_* can use the value. Also, set the flags so
- * that Asterisk responds identically whether or not a peer exists so as
- * not to leak peer name information.
- */
-static void check_for_nat(const struct ast_sockaddr *addr, struct sip_pvt *p)
-{
-
- if (!addr || !p) {
- return;
- }
-
- if (ast_sockaddr_cmp_addr(addr, &p->recv)) {
- char *tmp_str = ast_strdupa(ast_sockaddr_stringify_addr(addr));
- ast_debug(3, "NAT detected for %s / %s\n", tmp_str, ast_sockaddr_stringify_addr(&p->recv));
- p->natdetected = 1;
- if (ast_test_flag(&p->flags[2], SIP_PAGE3_NAT_AUTO_RPORT)) {
- ast_set_flag(&p->flags[0], SIP_NAT_FORCE_RPORT);
- }
- if (ast_test_flag(&p->flags[2], SIP_PAGE3_NAT_AUTO_COMEDIA)) {
- ast_set_flag(&p->flags[1], SIP_PAGE2_SYMMETRICRTP);
- }
- } else {
- p->natdetected = 0;
- if (ast_test_flag(&p->flags[2], SIP_PAGE3_NAT_AUTO_RPORT)) {
- ast_clear_flag(&p->flags[0], SIP_NAT_FORCE_RPORT);
- }
- if (ast_test_flag(&p->flags[2], SIP_PAGE3_NAT_AUTO_COMEDIA)) {
- ast_clear_flag(&p->flags[1], SIP_PAGE2_SYMMETRICRTP);
- }
- }
-
-}
-
-/*! \brief check Via: header for hostname, port and rport request/answer */
-static void check_via(struct sip_pvt *p, const struct sip_request *req)
-{
- char via[512];
- char *c, *maddr;
- struct ast_sockaddr tmp = { { 0, } };
- uint16_t port;
-
- ast_copy_string(via, sip_get_header(req, "Via"), sizeof(via));
-
- /* If this is via WebSocket we don't use the Via header contents at all */
- if (!strncasecmp(via, "SIP/2.0/WS", 10)) {
- return;
- }
-
- /* Work on the leftmost value of the topmost Via header */
- c = strchr(via, ',');
- if (c)
- *c = '\0';
-
- /* Check for rport */
- c = strstr(via, ";rport");
- if (c && (c[6] != '=')) { /* rport query, not answer */
- ast_set_flag(&p->flags[1], SIP_PAGE2_RPORT_PRESENT);
- ast_set_flag(&p->flags[0], SIP_NAT_RPORT_PRESENT);
- }
-
- /* Check for maddr */
- maddr = strstr(via, "maddr=");
- if (maddr) {
- maddr += 6;
- c = maddr + strspn(maddr, "abcdefghijklmnopqrstuvwxyz"
- "ABCDEFGHIJKLMNOPQRSTUVWXYZ0123456789-.:[]");
- *c = '\0';
- }
-
- c = strchr(via, ';');
- if (c)
- *c = '\0';
-
- c = strchr(via, ' ');
- if (c) {
- *c = '\0';
- c = ast_strip(c+1);
- if (strcasecmp(via, "SIP/2.0/UDP") && strcasecmp(via, "SIP/2.0/TCP") && strcasecmp(via, "SIP/2.0/TLS")) {
- ast_log(LOG_WARNING, "Don't know how to respond via '%s'\n", via);
- return;
- }
-
- if (maddr && ast_sockaddr_resolve_first(&p->sa, maddr, 0)) {
- p->sa = p->recv;
- }
-
- if (ast_sockaddr_resolve_first(&tmp, c, 0)) {
- ast_log(LOG_WARNING, "Could not resolve socket address for '%s'\n", c);
- port = STANDARD_SIP_PORT;
- } else if (!(port = ast_sockaddr_port(&tmp))) {
- port = STANDARD_SIP_PORT;
- ast_sockaddr_set_port(&tmp, port);
- }
-
- ast_sockaddr_set_port(&p->sa, port);
-
- check_for_nat(&tmp, p);
-
- if (sip_debug_test_pvt(p)) {
- ast_verbose("Sending to %s (%s)\n",
- ast_sockaddr_stringify(sip_real_dst(p)),
- sip_nat_mode(p));
- }
- }
-}
-
-/*! \brief Validate device authentication */
-static enum check_auth_result check_peer_ok(struct sip_pvt *p, char *of,
- struct sip_request *req, int sipmethod, struct ast_sockaddr *addr,
- struct sip_peer **authpeer,
- enum xmittype reliable, char *calleridname, char *uri2)
-{
- enum check_auth_result res;
- int debug = sip_debug_test_addr(addr);
- struct sip_peer *peer;
- struct sip_peer *bogus_peer;
-
- if (sipmethod == SIP_SUBSCRIBE) {
- /* For subscribes, match on device name only; for other methods,
- * match on IP address-port of the incoming request.
- */
- peer = sip_find_peer(of, NULL, TRUE, FINDALLDEVICES, FALSE, 0);
- } else {
- /* First find devices based on username (avoid all type=peer's) */
- peer = sip_find_peer(of, NULL, TRUE, FINDUSERS, FALSE, 0);
-
- /* Then find devices based on IP */
- if (!peer) {
- char *uri_tmp, *callback = NULL, *dummy;
- uri_tmp = ast_strdupa(uri2);
- parse_uri(uri_tmp, "sip:,sips:,tel:", &callback, &dummy, &dummy, &dummy);
- if (!ast_strlen_zero(callback) && (peer = sip_find_peer_by_ip_and_exten(&p->recv, callback, p->socket.type))) {
- ; /* found, fall through */
- } else {
- peer = sip_find_peer(NULL, &p->recv, TRUE, FINDPEERS, FALSE, p->socket.type);
- }
- }
- }
-
- if (!peer) {
- if (debug) {
- ast_verbose("No matching peer for '%s' from '%s'\n",
- of, ast_sockaddr_stringify(&p->recv));
- }
-
- /* If you don't mind, we can return 404s for devices that do
- * not exist: username disclosure. If we allow guests, there
- * is no way around that. */
- if (sip_cfg.allowguest || !sip_cfg.alwaysauthreject) {
- return AUTH_DONT_KNOW;
- }
-
- /* If you do mind, we use a peer that will never authenticate.
- * This ensures that we follow the same code path as regular
- * auth: less chance for username disclosure. */
- peer = ao2_t_global_obj_ref(g_bogus_peer, "check_peer_ok: Get the bogus peer.");
- if (!peer) {
- return AUTH_DONT_KNOW;
- }
- bogus_peer = peer;
- } else {
- bogus_peer = NULL;
- }
-
- if (!ast_apply_acl(peer->acl, addr, "SIP Peer ACL: ")) {
- ast_debug(2, "Found peer '%s' for '%s', but fails host access\n", peer->name, of);
- sip_unref_peer(peer, "sip_unref_peer: check_peer_ok: from sip_find_peer call, early return of AUTH_ACL_FAILED");
- return AUTH_ACL_FAILED;
- }
- if (debug && peer != bogus_peer) {
- ast_verbose("Found peer '%s' for '%s' from %s\n",
- peer->name, of, ast_sockaddr_stringify(&p->recv));
- }
-
- /* Set Frame packetization */
- if (p->rtp) {
- ast_rtp_codecs_set_framing(ast_rtp_instance_get_codecs(p->rtp), ast_format_cap_get_framing(peer->caps));
- p->autoframing = peer->autoframing;
- }
-
- /* Take the peer */
- ast_copy_flags(&p->flags[0], &peer->flags[0], SIP_FLAGS_TO_COPY);
- ast_copy_flags(&p->flags[1], &peer->flags[1], SIP_PAGE2_FLAGS_TO_COPY);
- ast_copy_flags(&p->flags[2], &peer->flags[2], SIP_PAGE3_FLAGS_TO_COPY);
-
- if (ast_test_flag(&p->flags[1], SIP_PAGE2_T38SUPPORT) && p->udptl) {
- p->t38_maxdatagram = peer->t38_maxdatagram;
- set_t38_capabilities(p);
- }
-
- ast_rtp_dtls_cfg_copy(&peer->dtls_cfg, &p->dtls_cfg);
-
- /* Copy SIP extensions profile to peer */
- /* XXX is this correct before a successful auth ? */
- if (p->sipoptions)
- peer->sipoptions = p->sipoptions;
-
- do_setnat(p);
-
- ast_string_field_set(p, peersecret, peer->secret);
- ast_string_field_set(p, peermd5secret, peer->md5secret);
- ast_string_field_set(p, subscribecontext, peer->subscribecontext);
- ast_string_field_set(p, mohinterpret, peer->mohinterpret);
- ast_string_field_set(p, mohsuggest, peer->mohsuggest);
- if (!ast_strlen_zero(peer->parkinglot)) {
- ast_string_field_set(p, parkinglot, peer->parkinglot);
- }
- ast_string_field_set(p, engine, peer->engine);
- p->disallowed_methods = peer->disallowed_methods;
- set_pvt_allowed_methods(p, req);
- ast_cc_copy_config_params(p->cc_params, peer->cc_params);
- if (peer->callingpres) /* Peer calling pres setting will override RPID */
- p->callingpres = peer->callingpres;
- if (peer->maxms && peer->lastms)
- p->timer_t1 = peer->lastms < global_t1min ? global_t1min : peer->lastms;
- else
- p->timer_t1 = peer->timer_t1;
-
- /* Set timer B to control transaction timeouts */
- if (peer->timer_b)
- p->timer_b = peer->timer_b;
- else
- p->timer_b = 64 * p->timer_t1;
-
- p->allowtransfer = peer->allowtransfer;
-
- if (ast_test_flag(&peer->flags[0], SIP_INSECURE_INVITE)) {
- /* Pretend there is no required authentication */
- ast_string_field_set(p, peersecret, NULL);
- ast_string_field_set(p, peermd5secret, NULL);
- }
- if (!(res = check_auth(p, req, peer->name, p->peersecret, p->peermd5secret, sipmethod, uri2, reliable))) {
-
- /* build_peer, called through sip_find_peer, is not able to check the
- * sip_pvt->natdetected flag in order to determine if the peer is behind
- * NAT or not when SIP_PAGE3_NAT_AUTO_RPORT or SIP_PAGE3_NAT_AUTO_COMEDIA
- * are set on the peer. So we check for that here and set the peer's
- * address accordingly. The address should ONLY be set once we are sure
- * authentication was a success. If, for example, an INVITE was sent that
- * matched the peer name but failed the authentication check, the address
- * would be updated, which is bad.
- */
- set_peer_nat(p, peer);
- if (p->natdetected && ast_test_flag(&peer->flags[2], SIP_PAGE3_NAT_AUTO_RPORT)) {
- ast_sockaddr_copy(&peer->addr, &p->recv);
- }
-
- /* If we have a call limit, set flag */
- if (peer->call_limit)
- ast_set_flag(&p->flags[0], SIP_CALL_LIMIT);
- ast_string_field_set(p, peername, peer->name);
- ast_string_field_set(p, authname, peer->name);
-
- ast_rtp_dtls_cfg_copy(&peer->dtls_cfg, &p->dtls_cfg);
-
- if (sipmethod == SIP_INVITE) {
- /* destroy old channel vars and copy in new ones. */
- ast_variables_destroy(p->chanvars);
- p->chanvars = copy_vars(peer->chanvars);
- }
-
- if (authpeer) {
- ao2_t_ref(peer, 1, "copy pointer into (*authpeer)");
- (*authpeer) = peer; /* Add a ref to the object here, to keep it in memory a bit longer if it is realtime */
- }
-
- if (!ast_strlen_zero(peer->username)) {
- ast_string_field_set(p, username, peer->username);
- /* Use the default username for authentication on outbound calls */
- /* XXX this takes the name from the caller... can we override ? */
- ast_string_field_set(p, authname, peer->username);
- }
- if (!get_rpid(p, req)) {
- if (!ast_strlen_zero(peer->cid_num)) {
- char *tmp = ast_strdupa(peer->cid_num);
- if (global_shrinkcallerid && ast_is_shrinkable_phonenumber(tmp))
- ast_shrink_phone_number(tmp);
- ast_string_field_set(p, cid_num, tmp);
- }
- if (!ast_strlen_zero(peer->cid_name))
- ast_string_field_set(p, cid_name, peer->cid_name);
- if (peer->callingpres)
- p->callingpres = peer->callingpres;
- }
- if (!ast_strlen_zero(peer->cid_tag)) {
- ast_string_field_set(p, cid_tag, peer->cid_tag);
- }
- ast_string_field_set(p, fullcontact, peer->fullcontact);
- if (!ast_strlen_zero(peer->context)) {
- ast_string_field_set(p, context, peer->context);
- }
- if (!ast_strlen_zero(peer->messagecontext)) {
- ast_string_field_set(p, messagecontext, peer->messagecontext);
- }
- if (!ast_strlen_zero(peer->mwi_from)) {
- ast_string_field_set(p, mwi_from, peer->mwi_from);
- }
- ast_string_field_set(p, peersecret, peer->secret);
- ast_string_field_set(p, peermd5secret, peer->md5secret);
- ast_string_field_set(p, language, peer->language);
- ast_string_field_set(p, accountcode, peer->accountcode);
- p->amaflags = peer->amaflags;
- p->callgroup = peer->callgroup;
- p->pickupgroup = peer->pickupgroup;
- ast_unref_namedgroups(p->named_callgroups);
- p->named_callgroups = ast_ref_namedgroups(peer->named_callgroups);
- ast_unref_namedgroups(p->named_pickupgroups);
- p->named_pickupgroups = ast_ref_namedgroups(peer->named_pickupgroups);
- ast_format_cap_remove_by_type(p->caps, AST_MEDIA_TYPE_UNKNOWN);
- ast_format_cap_append_from_cap(p->caps, peer->caps, AST_MEDIA_TYPE_UNKNOWN);
- ast_format_cap_remove_by_type(p->jointcaps, AST_MEDIA_TYPE_UNKNOWN);
- ast_format_cap_append_from_cap(p->jointcaps, peer->caps, AST_MEDIA_TYPE_UNKNOWN);
- ast_copy_string(p->zone, peer->zone, sizeof(p->zone));
- if (peer->maxforwards > 0) {
- p->maxforwards = peer->maxforwards;
- }
- if (ast_format_cap_count(p->peercaps)) {
- struct ast_format_cap *joint;
-
- joint = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
- if (joint) {
- ast_format_cap_get_compatible(p->jointcaps, p->peercaps, joint);
- ao2_ref(p->jointcaps, -1);
- p->jointcaps = joint;
- }
- }
- p->maxcallbitrate = peer->maxcallbitrate;
- if ((ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833) ||
- (ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_AUTO))
- p->noncodeccapability |= AST_RTP_DTMF;
- else
- p->noncodeccapability &= ~AST_RTP_DTMF;
- p->jointnoncodeccapability = p->noncodeccapability;
- p->rtptimeout = peer->rtptimeout;
- p->rtpholdtimeout = peer->rtpholdtimeout;
- p->rtpkeepalive = peer->rtpkeepalive;
- if (!dialog_initialize_rtp(p)) {
- if (p->rtp) {
- ast_rtp_codecs_set_framing(ast_rtp_instance_get_codecs(p->rtp), ast_format_cap_get_framing(peer->caps));
- p->autoframing = peer->autoframing;
- }
- } else {
- res = AUTH_RTP_FAILED;
- }
- }
- sip_unref_peer(peer, "check_peer_ok: sip_unref_peer: tossing temp ptr to peer from sip_find_peer");
-
- return res;
-}
-
-
-/*! \brief Check if matching user or peer is defined
- Match user on From: user name and peer on IP/port
- This is used on first invite (not re-invites) and subscribe requests
- \return 0 on success, non-zero on failure
-*/
-static enum check_auth_result check_user_full(struct sip_pvt *p, struct sip_request *req,
- int sipmethod, const char *uri, enum xmittype reliable,
- struct ast_sockaddr *addr, struct sip_peer **authpeer)
-{
- char *of, *name, *unused_password, *domain;
- RAII_VAR(char *, ofbuf, NULL, ast_free); /* beware, everyone starts pointing to this */
- RAII_VAR(char *, namebuf, NULL, ast_free);
- enum check_auth_result res = AUTH_DONT_KNOW;
- char calleridname[256];
- char *uri2 = ast_strdupa(uri);
-
- terminate_uri(uri2); /* trim extra stuff */
-
- ofbuf = ast_strdup(sip_get_header(req, "From"));
- /* XXX here tries to map the username for invite things */
-
- /* strip the display-name portion off the beginning of the FROM header. */
- if (!(of = (char *) get_calleridname(ofbuf, calleridname, sizeof(calleridname)))) {
- ast_log(LOG_ERROR, "FROM header can not be parsed\n");
- return res;
- }
-
- if (calleridname[0]) {
- ast_string_field_set(p, cid_name, calleridname);
- }
-
- if (ast_strlen_zero(p->exten)) {
- char *t = uri2;
- if (!strncasecmp(t, "sip:", 4)) {
- t += 4;
- } else if (!strncasecmp(t, "sips:", 5)) {
- t += 5;
- } else if (!strncasecmp(t, "tel:", 4)) { /* TEL URI INVITE */
- t += 4;
- }
- ast_string_field_set(p, exten, t);
- t = strchr(p->exten, '@');
- if (t)
- *t = '\0';
-
- if (ast_strlen_zero(p->our_contact)) {
- build_contact(p, req, 1);
- }
- }
-
- of = get_in_brackets(of);
-
- /* save the URI part of the From header */
- ast_string_field_set(p, from, of);
-
- if (parse_uri_legacy_check(of, "sip:,sips:,tel:", &name, &unused_password, &domain, NULL)) {
- ast_log(LOG_NOTICE, "From address missing 'sip:', using it anyway\n");
- }
-
- SIP_PEDANTIC_DECODE(name);
- SIP_PEDANTIC_DECODE(domain);
-
- extract_host_from_hostport(&domain);
-
- if (ast_strlen_zero(domain)) {
- /* , never good */
- ast_log(LOG_ERROR, "Empty domain name in FROM header\n");
- return res;
- }
-
- if (ast_strlen_zero(name)) {
- /* . Asterisk 1.4 and 1.6 have always
- * treated that as a username, so we continue the tradition:
- * uri is now . */
- name = domain;
- } else {
- /* Non-empty name, try to get caller id from it */
- char *tmp = ast_strdupa(name);
- /* We need to be able to handle from-headers looking like
-
- */
- tmp = strsep(&tmp, ";");
- if (global_shrinkcallerid && ast_is_shrinkable_phonenumber(tmp)) {
- ast_shrink_phone_number(tmp);
- }
- ast_string_field_set(p, cid_num, tmp);
- }
-
- if (global_match_auth_username) {
- /*
- * XXX This is experimental code to grab the search key from the
- * Auth header's username instead of the 'From' name, if available.
- * Do not enable this block unless you understand the side effects (if any!)
- * Note, the search for "username" should be done in a more robust way.
- * Note2, at the moment we check both fields, though maybe we should
- * pick one or another depending on the request ? XXX
- */
- const char *hdr = sip_get_header(req, "Authorization");
- if (ast_strlen_zero(hdr)) {
- hdr = sip_get_header(req, "Proxy-Authorization");
- }
-
- if (!ast_strlen_zero(hdr) && (hdr = strstr(hdr, "username=\""))) {
- namebuf = name = ast_strdup(hdr + strlen("username=\""));
- name = strsep(&name, "\"");
- }
- }
-
- res = check_peer_ok(p, name, req, sipmethod, addr,
- authpeer, reliable, calleridname, uri2);
- if (res != AUTH_DONT_KNOW) {
- return res;
- }
-
- /* Finally, apply the guest policy */
- if (sip_cfg.allowguest) {
- /* Ignore check_return warning from Coverity for get_rpid below. */
- get_rpid(p, req);
- p->rtptimeout = global_rtptimeout;
- p->rtpholdtimeout = global_rtpholdtimeout;
- p->rtpkeepalive = global_rtpkeepalive;
- if (!dialog_initialize_rtp(p)) {
- res = AUTH_SUCCESSFUL;
- } else {
- res = AUTH_RTP_FAILED;
- }
- } else {
- res = AUTH_SECRET_FAILED; /* we don't want any guests, authentication will fail */
- }
-
- if (ast_test_flag(&p->flags[1], SIP_PAGE2_RPORT_PRESENT)) {
- ast_set_flag(&p->flags[0], SIP_NAT_RPORT_PRESENT);
- }
-
- return res;
-}
-
-/*! \brief Find user
- If we get a match, this will add a reference pointer to the user object, that needs to be unreferenced
-*/
-static int check_user(struct sip_pvt *p, struct sip_request *req, int sipmethod, const char *uri, enum xmittype reliable, struct ast_sockaddr *addr)
-{
- return check_user_full(p, req, sipmethod, uri, reliable, addr, NULL);
-}
-
-static void send_check_user_failure_response(struct sip_pvt *p, struct sip_request *req, int res, enum xmittype reliable)
-{
- const char *response;
-
- switch (res) {
- case AUTH_SECRET_FAILED:
- case AUTH_USERNAME_MISMATCH:
- case AUTH_NOT_FOUND:
- case AUTH_UNKNOWN_DOMAIN:
- case AUTH_PEER_NOT_DYNAMIC:
- case AUTH_BAD_TRANSPORT:
- case AUTH_ACL_FAILED:
- ast_log(LOG_NOTICE, "Failed to authenticate device %s for %s, code = %d\n",
- sip_get_header(req, "From"), sip_methods[p->method].text, res);
- response = "403 Forbidden";
- break;
- case AUTH_SESSION_LIMIT:
- /* Unexpected here, actually. As it's handled elsewhere. */
- ast_log(LOG_NOTICE, "Call limit reached for device %s for %s, code = %d\n",
- sip_get_header(req, "From"), sip_methods[p->method].text, res);
- response = "480 Temporarily Unavailable";
- break;
- case AUTH_RTP_FAILED:
- /* We don't want to send a 403 in the RTP_FAILED case.
- * The cause could be any one of:
- * - out of memory or rtp ports
- * - dtls/srtp requested but not loaded/invalid
- * Neither of them warrant a 403. A 503 makes more
- * sense, as this node is broken/overloaded. */
- ast_log(LOG_NOTICE, "RTP init failure for device %s for %s, code = %d\n",
- sip_get_header(req, "From"), sip_methods[p->method].text, res);
- response = "503 Service Unavailable";
- break;
- case AUTH_SUCCESSFUL:
- case AUTH_CHALLENGE_SENT:
- /* These should have been handled elsewhere. */
- default:
- ast_log(LOG_NOTICE, "Unexpected error for device %s for %s, code = %d\n",
- sip_get_header(req, "From"), sip_methods[p->method].text, res);
- response = "503 Service Unavailable";
- }
-
- if (reliable == XMIT_RELIABLE) {
- transmit_response_reliable(p, response, req);
- } else if (reliable == XMIT_UNRELIABLE) {
- transmit_response(p, response, req);
- }
-}
-
-static int set_message_vars_from_req(struct ast_msg *msg, struct sip_request *req)
-{
- size_t x;
- char name_buf[1024];
- char val_buf[1024];
- const char *name;
- char *c;
- int res = 0;
-
- for (x = 0; x < req->headers; x++) {
- const char *header = REQ_OFFSET_TO_STR(req, header[x]);
-
- if ((c = strchr(header, ':'))) {
- ast_copy_string(name_buf, header, MIN((c - header + 1), sizeof(name_buf)));
- ast_copy_string(val_buf, ast_skip_blanks(c + 1), sizeof(val_buf));
- ast_trim_blanks(name_buf);
-
- /* Convert header name to full name alias. */
- name = find_full_alias(name_buf, name_buf);
-
- res = ast_msg_set_var(msg, name, val_buf);
- if (res) {
- break;
- }
- }
- }
- return res;
-}
-
-/*! \brief Receive SIP MESSAGE method messages
-\note We only handle messages within current calls currently
- Reference: RFC 3428 */
-static void receive_message(struct sip_pvt *p, struct sip_request *req, struct ast_sockaddr *addr, const char *e)
-{
- char *buf;
- size_t len;
- struct ast_frame f;
- const char *content_type = sip_get_header(req, "Content-Type");
- struct ast_msg *msg;
- int res;
- char *from;
- char *to;
- char from_name[50];
- char stripped[SIPBUFSIZE];
- enum sip_get_dest_result dest_result;
-
- if (strncmp(content_type, "text/plain", strlen("text/plain"))) { /* No text/plain attachment */
- transmit_response(p, "415 Unsupported Media Type", req); /* Good enough, or? */
- if (!p->owner) {
- sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
- }
- return;
- }
-
- if (!(buf = get_content(req))) {
- ast_log(LOG_WARNING, "Unable to retrieve text from %s\n", p->callid);
- transmit_response(p, "500 Internal Server Error", req);
- if (!p->owner) {
- sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
- }
- return;
- }
-
- /* Strip trailing line feeds from message body. (get_content may add
- * a trailing linefeed and we don't need any at the end) */
- len = strlen(buf);
- while (len > 0) {
- if (buf[--len] != '\n') {
- ++len;
- break;
- }
- }
- buf[len] = '\0';
-
- if (p->owner) {
- if (sip_debug_test_pvt(p)) {
- ast_verbose("SIP Text message received: '%s'\n", buf);
- }
- memset(&f, 0, sizeof(f));
- f.frametype = AST_FRAME_TEXT;
- f.subclass.integer = 0;
- f.offset = 0;
- f.data.ptr = buf;
- f.datalen = strlen(buf) + 1;
- ast_queue_frame(p->owner, &f);
- transmit_response(p, "202 Accepted", req); /* We respond 202 accepted, since we relay the message */
- return;
- }
-
- /*
- * At this point MESSAGE is outside of a call.
- *
- * NOTE: p->owner is NULL so no additional check is needed after
- * this point.
- */
-
- if (!sip_cfg.accept_outofcall_message) {
- /* Message outside of a call, we do not support that */
- ast_debug(1, "MESSAGE outside of a call administratively disabled.\n");
- transmit_response(p, "405 Method Not Allowed", req);
- sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
- return;
- }
-
- copy_request(&p->initreq, req);
-
- if (sip_cfg.auth_message_requests) {
- int res;
-
- set_pvt_allowed_methods(p, req);
- res = check_user(p, req, SIP_MESSAGE, e, XMIT_UNRELIABLE, addr);
- if (res == AUTH_CHALLENGE_SENT) {
- sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
- return;
- }
- if (res < 0) { /* Something failed in authentication */
- send_check_user_failure_response(p, req, res, XMIT_UNRELIABLE);
- sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
- return;
- }
- /* Auth was successful. Proceed. */
- } else {
- struct sip_peer *peer;
-
- /*
- * MESSAGE outside of a call, not authenticating it.
- * Check to see if we match a peer anyway so that we can direct
- * it to the right context.
- */
-
- peer = sip_find_peer(NULL, &p->recv, TRUE, FINDPEERS, 0, p->socket.type);
- if (peer) {
- /* Only if no auth is required. */
- if (ast_strlen_zero(peer->secret) && ast_strlen_zero(peer->md5secret)) {
- ast_string_field_set(p, context, peer->context);
- }
- if (!ast_strlen_zero(peer->messagecontext)) {
- ast_string_field_set(p, messagecontext, peer->messagecontext);
- }
- ast_string_field_set(p, peername, peer->name);
- peer = sip_unref_peer(peer, "from sip_find_peer() in receive_message");
- }
- }
-
- /* Override the context with the message context _BEFORE_
- * getting the destination. This way we can guarantee the correct
- * extension is used in the message context when it is present. */
- if (!ast_strlen_zero(p->messagecontext)) {
- ast_string_field_set(p, context, p->messagecontext);
- } else if (!ast_strlen_zero(sip_cfg.messagecontext)) {
- ast_string_field_set(p, context, sip_cfg.messagecontext);
- }
-
- dest_result = get_destination(p, NULL, NULL);
- switch (dest_result) {
- case SIP_GET_DEST_REFUSED:
- /* Okay to send 403 since this is after auth processing */
- transmit_response(p, "403 Forbidden", req);
- sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
- return;
- case SIP_GET_DEST_INVALID_URI:
- transmit_response(p, "416 Unsupported URI Scheme", req);
- sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
- return;
- default:
- /* We may have something other than dialplan who wants
- * the message, so defer further error handling for now */
- break;
- }
-
- if (!(msg = ast_msg_alloc())) {
- transmit_response(p, "500 Internal Server Error", req);
- sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
- return;
- }
-
- to = ast_strdupa(REQ_OFFSET_TO_STR(req, rlpart2));
- from = ast_strdupa(sip_get_header(req, "From"));
-
- res = ast_msg_set_to(msg, "%s", to);
-
- /* Build "display" for from string. */
- from = (char *) get_calleridname(from, from_name, sizeof(from_name));
- from = get_in_brackets(from);
- if (from_name[0]) {
- char from_buf[128];
-
- ast_escape_quoted(from_name, from_buf, sizeof(from_buf));
- res |= ast_msg_set_from(msg, "\"%s\" <%s>", from_buf, from);
- } else {
- res |= ast_msg_set_from(msg, "<%s>", from);
- }
-
- res |= ast_msg_set_body(msg, "%s", buf);
- res |= ast_msg_set_context(msg, "%s", p->context);
-
- res |= ast_msg_set_var(msg, "SIP_RECVADDR", ast_sockaddr_stringify(&p->recv));
- res |= ast_msg_set_tech(msg, "%s", "SIP");
- if (!ast_strlen_zero(p->peername)) {
- res |= ast_msg_set_endpoint(msg, "%s", p->peername);
- res |= ast_msg_set_var(msg, "SIP_PEERNAME", p->peername);
- }
-
- ast_copy_string(stripped, sip_get_header(req, "Contact"), sizeof(stripped));
- res |= ast_msg_set_var(msg, "SIP_FULLCONTACT", get_in_brackets(stripped));
-
- res |= ast_msg_set_exten(msg, "%s", p->exten);
- res |= set_message_vars_from_req(msg, req);
-
- if (res) {
- ast_msg_destroy(msg);
- transmit_response(p, "500 Internal Server Error", req);
- sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
- return;
- }
-
- if (ast_msg_has_destination(msg)) {
- ast_msg_queue(msg);
- transmit_response(p, "202 Accepted", req);
- sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
- return;
- }
-
- /* Find a specific error cause to send */
- switch (dest_result) {
- case SIP_GET_DEST_EXTEN_NOT_FOUND:
- case SIP_GET_DEST_EXTEN_MATCHMORE:
- transmit_response(p, "404 Not Found", req);
- break;
- case SIP_GET_DEST_EXTEN_FOUND:
- default:
- /* We should have sent the message already! */
- ast_assert(0);
- transmit_response(p, "500 Internal Server Error", req);
- break;
- }
- sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
- ast_msg_destroy(msg);
-}
-
-/*! \brief CLI Command to show calls within limits set by call_limit */
-static char *sip_show_inuse(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
-{
-#define FORMAT "%-25.25s %-15.15s %-15.15s \n"
-#define FORMAT2 "%-25.25s %-15.15s %-15.15s \n"
- char ilimits[40];
- char iused[40];
- int showall = FALSE;
- struct ao2_iterator i;
- struct sip_peer *peer;
-
- switch (cmd) {
- case CLI_INIT:
- e->command = "sip show inuse [all]";
- e->usage =
- "Usage: sip show inuse [all]\n"
- " List all SIP devices usage counters and limits.\n"
- " Add option \"all\" to show all devices, not only those with a limit.\n";
- return NULL;
- case CLI_GENERATE:
- return NULL;
- }
-
- if (a->argc < 3)
- return CLI_SHOWUSAGE;
-
- if (a->argc == 4 && !strcmp(a->argv[3], "all"))
- showall = TRUE;
-
- ast_cli(a->fd, FORMAT, "* Peer name", "In use", "Limit");
-
- i = ao2_iterator_init(peers, 0);
- while ((peer = ao2_t_iterator_next(&i, "iterate thru peer table"))) {
- ao2_lock(peer);
- if (peer->call_limit)
- snprintf(ilimits, sizeof(ilimits), "%d", peer->call_limit);
- else
- ast_copy_string(ilimits, "N/A", sizeof(ilimits));
- snprintf(iused, sizeof(iused), "%d/%d/%d", peer->inuse, peer->ringing, peer->onhold);
- if (showall || peer->call_limit)
- ast_cli(a->fd, FORMAT2, peer->name, iused, ilimits);
- ao2_unlock(peer);
- sip_unref_peer(peer, "toss iterator pointer");
- }
- ao2_iterator_destroy(&i);
-
- return CLI_SUCCESS;
-#undef FORMAT
-#undef FORMAT2
-}
-
-
-/*! \brief Convert transfer mode to text string */
-static char *transfermode2str(enum transfermodes mode)
-{
- if (mode == TRANSFER_OPENFORALL)
- return "open";
- else if (mode == TRANSFER_CLOSED)
- return "closed";
- return "strict";
-}
-
-/*! \brief Report Peer status in character string
- * \retval 0 if peer is unreachable.
- * \retval 1 if peer is online.
- * \retval -1 if unmonitored.
- */
-
-
-/* Session-Timer Modes */
-static const struct _map_x_s stmodes[] = {
- { SESSION_TIMER_MODE_ACCEPT, "Accept"},
- { SESSION_TIMER_MODE_ORIGINATE, "Originate"},
- { SESSION_TIMER_MODE_REFUSE, "Refuse"},
- { -1, NULL},
-};
-
-static const char *stmode2str(enum st_mode m)
-{
- return map_x_s(stmodes, m, "Unknown");
-}
-
-static enum st_mode str2stmode(const char *s)
-{
- return map_s_x(stmodes, s, -1);
-}
-
-/* Session-Timer Refreshers */
-static const struct _map_x_s strefresher_params[] = {
- { SESSION_TIMER_REFRESHER_PARAM_UNKNOWN, "unknown" },
- { SESSION_TIMER_REFRESHER_PARAM_UAC, "uac" },
- { SESSION_TIMER_REFRESHER_PARAM_UAS, "uas" },
- { -1, NULL },
-};
-
-static const struct _map_x_s strefreshers[] = {
- { SESSION_TIMER_REFRESHER_AUTO, "auto" },
- { SESSION_TIMER_REFRESHER_US, "us" },
- { SESSION_TIMER_REFRESHER_THEM, "them" },
- { -1, NULL },
-};
-
-static const char *strefresherparam2str(enum st_refresher_param r)
-{
- return map_x_s(strefresher_params, r, "Unknown");
-}
-
-static enum st_refresher_param str2strefresherparam(const char *s)
-{
- return map_s_x(strefresher_params, s, -1);
-}
-
-/* Autocreatepeer modes */
-static struct _map_x_s autopeermodes[] = {
- { AUTOPEERS_DISABLED, "Off"},
- { AUTOPEERS_VOLATILE, "Volatile"},
- { AUTOPEERS_PERSIST, "Persisted"},
- { -1, NULL},
-};
-
-static const char *strefresher2str(enum st_refresher r)
-{
- return map_x_s(strefreshers, r, "Unknown");
-}
-
-static const char *autocreatepeer2str(enum autocreatepeer_mode r)
-{
- return map_x_s(autopeermodes, r, "Unknown");
-}
-
-static int peer_status(struct sip_peer *peer, char *status, int statuslen)
-{
- int res = 0;
- if (peer->maxms) {
- if (peer->lastms < 0) {
- ast_copy_string(status, "UNREACHABLE", statuslen);
- } else if (peer->lastms > peer->maxms) {
- snprintf(status, statuslen, "LAGGED (%d ms)", peer->lastms);
- res = 1;
- } else if (peer->lastms) {
- snprintf(status, statuslen, "OK (%d ms)", peer->lastms);
- res = 1;
- } else {
- ast_copy_string(status, "UNKNOWN", statuslen);
- }
- } else {
- ast_copy_string(status, "Unmonitored", statuslen);
- /* Checking if port is 0 */
- res = -1;
- }
- return res;
-}
-
-/*! \brief Show active TCP connections */
-static char *sip_show_tcp(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
-{
- struct sip_threadinfo *th;
- struct ao2_iterator i;
-
-#define FORMAT2 "%-47.47s %9.9s %6.6s\n"
-#define FORMAT "%-47.47s %-9.9s %-6.6s\n"
-
- switch (cmd) {
- case CLI_INIT:
- e->command = "sip show tcp";
- e->usage =
- "Usage: sip show tcp\n"
- " Lists all active TCP/TLS sessions.\n";
- return NULL;
- case CLI_GENERATE:
- return NULL;
- }
-
- if (a->argc != 3)
- return CLI_SHOWUSAGE;
-
- ast_cli(a->fd, FORMAT2, "Address", "Transport", "Type");
-
- i = ao2_iterator_init(threadt, 0);
- while ((th = ao2_t_iterator_next(&i, "iterate through tcp threads for 'sip show tcp'"))) {
- ast_cli(a->fd, FORMAT,
- ast_sockaddr_stringify(&th->tcptls_session->remote_address),
- sip_get_transport(th->type),
- (th->tcptls_session->client ? "Client" : "Server"));
- ao2_t_ref(th, -1, "decrement ref from iterator");
- }
- ao2_iterator_destroy(&i);
-
- return CLI_SUCCESS;
-#undef FORMAT
-#undef FORMAT2
-}
-
-/*! \brief CLI Command 'SIP Show Users' */
-static char *sip_show_users(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
-{
- regex_t regexbuf;
- int havepattern = FALSE;
- struct ao2_iterator user_iter;
- struct sip_peer *user;
-
-#define FORMAT "%-25.25s %-15.15s %-15.15s %-15.15s %-5.5s%-10.10s\n"
-
- switch (cmd) {
- case CLI_INIT:
- e->command = "sip show users [like]";
- e->usage =
- "Usage: sip show users [like ]\n"
- " Lists all known SIP users.\n"
- " Optional regular expression pattern is used to filter the user list.\n";
- return NULL;
- case CLI_GENERATE:
- return NULL;
- }
-
- switch (a->argc) {
- case 5:
- if (!strcasecmp(a->argv[3], "like")) {
- if (regcomp(®exbuf, a->argv[4], REG_EXTENDED | REG_NOSUB))
- return CLI_SHOWUSAGE;
- havepattern = TRUE;
- } else
- return CLI_SHOWUSAGE;
- case 3:
- break;
- default:
- return CLI_SHOWUSAGE;
- }
-
- ast_cli(a->fd, FORMAT, "Username", "Secret", "Accountcode", "Def.Context", "ACL", "Forcerport");
-
- user_iter = ao2_iterator_init(peers, 0);
- while ((user = ao2_t_iterator_next(&user_iter, "iterate thru peers table"))) {
- ao2_lock(user);
- if (!(user->type & SIP_TYPE_USER)) {
- ao2_unlock(user);
- sip_unref_peer(user, "sip show users");
- continue;
- }
-
- if (havepattern && regexec(®exbuf, user->name, 0, NULL, 0)) {
- ao2_unlock(user);
- sip_unref_peer(user, "sip show users");
- continue;
- }
-
- ast_cli(a->fd, FORMAT, user->name,
- user->secret,
- user->accountcode,
- user->context,
- AST_CLI_YESNO(ast_acl_list_is_empty(user->acl) == 0),
- AST_CLI_YESNO(ast_test_flag(&user->flags[0], SIP_NAT_FORCE_RPORT)));
- ao2_unlock(user);
- sip_unref_peer(user, "sip show users");
- }
- ao2_iterator_destroy(&user_iter);
-
- if (havepattern)
- regfree(®exbuf);
-
- return CLI_SUCCESS;
-#undef FORMAT
-}
-
-/*! \brief Show SIP registrations in the manager API */
-static int manager_show_registry(struct mansession *s, const struct message *m)
-{
- const char *id = astman_get_header(m, "ActionID");
- char idtext[256] = "";
- int total = 0;
- struct ao2_iterator iter;
- struct sip_registry *iterator;
-
- if (!ast_strlen_zero(id))
- snprintf(idtext, sizeof(idtext), "ActionID: %s\r\n", id);
-
- astman_send_listack(s, m, "Registrations will follow", "start");
-
- iter = ao2_iterator_init(registry_list, 0);
- while ((iterator = ao2_t_iterator_next(&iter, "manager_show_registry iter"))) {
- ao2_lock(iterator);
-
- astman_append(s,
- "Event: RegistryEntry\r\n"
- "%s"
- "Host: %s\r\n"
- "Port: %d\r\n"
- "Username: %s\r\n"
- "Domain: %s\r\n"
- "DomainPort: %d\r\n"
- "Refresh: %d\r\n"
- "State: %s\r\n"
- "RegistrationTime: %ld\r\n"
- "\r\n",
- idtext,
- iterator->hostname,
- iterator->portno ? iterator->portno : STANDARD_SIP_PORT,
- iterator->username,
- S_OR(iterator->regdomain,iterator->hostname),
- iterator->regdomainport ? iterator->regdomainport : STANDARD_SIP_PORT,
- iterator->refresh,
- regstate2str(iterator->regstate),
- (long) iterator->regtime.tv_sec);
-
- ao2_unlock(iterator);
- ao2_t_ref(iterator, -1, "manager_show_registry iter");
- total++;
- }
- ao2_iterator_destroy(&iter);
-
- astman_send_list_complete_start(s, m, "RegistrationsComplete", total);
- astman_send_list_complete_end(s);
-
- return 0;
-}
-
-/*! \brief Show SIP peers in the manager API */
-/* Inspired from chan_iax2 */
-static int manager_sip_show_peers(struct mansession *s, const struct message *m)
-{
- const char *id = astman_get_header(m, "ActionID");
- const char *a[] = {"sip", "show", "peers"};
- char idtext[256] = "";
- int total = 0;
-
- if (!ast_strlen_zero(id))
- snprintf(idtext, sizeof(idtext), "ActionID: %s\r\n", id);
-
- astman_send_listack(s, m, "Peer status list will follow", "start");
-
- /* List the peers in separate manager events */
- _sip_show_peers(-1, &total, s, m, 3, a);
-
- /* Send final confirmation */
- astman_send_list_complete_start(s, m, "PeerlistComplete", total);
- astman_send_list_complete_end(s);
- return 0;
-}
-
-/*! \brief CLI Show Peers command */
-static char *sip_show_peers(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
-{
- switch (cmd) {
- case CLI_INIT:
- e->command = "sip show peers [like]";
- e->usage =
- "Usage: sip show peers [like ]\n"
- " Lists all known SIP peers.\n"
- " Optional regular expression pattern is used to filter the peer list.\n";
- return NULL;
- case CLI_GENERATE:
- return NULL;
- }
-
- return _sip_show_peers(a->fd, NULL, NULL, NULL, a->argc, (const char **) a->argv);
-}
-
-int peercomparefunc(const void *a, const void *b);
-
-int peercomparefunc(const void *a, const void *b)
-{
- struct sip_peer **ap = (struct sip_peer **)a;
- struct sip_peer **bp = (struct sip_peer **)b;
- return strcmp((*ap)->name, (*bp)->name);
-}
-
-/* the last argument is left-aligned, so we don't need a size anyways */
-#define PEERS_FORMAT2 "%-25.25s %-39.39s %-3.3s %-10.10s %-10.10s %-3.3s %-8s %-11s %-32.32s %s\n"
-
-/*! \brief Used in the sip_show_peers functions to pass parameters */
-struct show_peers_context {
- regex_t regexbuf;
- int havepattern;
- char idtext[256];
- int realtimepeers;
- int peers_mon_online;
- int peers_mon_offline;
- int peers_unmon_offline;
- int peers_unmon_online;
-};
-
-/*! \brief Execute sip show peers command */
-static char *_sip_show_peers(int fd, int *total, struct mansession *s, const struct message *m, int argc, const char *argv[])
-{
- struct show_peers_context cont = {
- .havepattern = FALSE,
- .idtext = "",
-
- .peers_mon_online = 0,
- .peers_mon_offline = 0,
- .peers_unmon_online = 0,
- .peers_unmon_offline = 0,
- };
-
- struct sip_peer *peer;
- struct ao2_iterator* it_peers;
-
- int total_peers = 0;
- const char *id;
- struct sip_peer **peerarray;
- int k;
-
- cont.realtimepeers = ast_check_realtime("sippeers");
-
- if (s) { /* Manager - get ActionID */
- id = astman_get_header(m, "ActionID");
- if (!ast_strlen_zero(id)) {
- snprintf(cont.idtext, sizeof(cont.idtext), "ActionID: %s\r\n", id);
- }
- }
-
- switch (argc) {
- case 5:
- if (!strcasecmp(argv[3], "like")) {
- if (regcomp(&cont.regexbuf, argv[4], REG_EXTENDED | REG_NOSUB)) {
- return CLI_SHOWUSAGE;
- }
- cont.havepattern = TRUE;
- } else {
- return CLI_SHOWUSAGE;
- }
- case 3:
- break;
- default:
- return CLI_SHOWUSAGE;
- }
-
- if (!s) {
- /* Normal list */
- ast_cli(fd, PEERS_FORMAT2, "Name/username", "Host", "Dyn", "Forcerport", "Comedia", "ACL", "Port", "Status", "Description", (cont.realtimepeers ? "Realtime" : ""));
- }
-
- ao2_lock(peers);
- if (!(it_peers = ao2_callback(peers, OBJ_MULTIPLE, NULL, NULL))) {
- ast_log(AST_LOG_ERROR, "Unable to create iterator for peers container for sip show peers\n");
- ao2_unlock(peers);
- return CLI_FAILURE;
- }
- if (!(peerarray = ast_calloc(sizeof(struct sip_peer *), ao2_container_count(peers)))) {
- ast_log(AST_LOG_ERROR, "Unable to allocate peer array for sip show peers\n");
- ao2_iterator_destroy(it_peers);
- ao2_unlock(peers);
- return CLI_FAILURE;
- }
- ao2_unlock(peers);
-
- while ((peer = ao2_t_iterator_next(it_peers, "iterate thru peers table"))) {
- ao2_lock(peer);
-
- if (!(peer->type & SIP_TYPE_PEER)) {
- ao2_unlock(peer);
- sip_unref_peer(peer, "unref peer because it's actually a user");
- continue;
- }
-
- if (cont.havepattern && regexec(&cont.regexbuf, peer->name, 0, NULL, 0)) {
- ao2_unlock(peer);
- sip_unref_peer(peer, "toss iterator peer ptr before continue");
- continue;
- }
-
- peerarray[total_peers++] = peer;
- ao2_unlock(peer);
- }
- ao2_iterator_destroy(it_peers);
-
- qsort(peerarray, total_peers, sizeof(struct sip_peer *), peercomparefunc);
-
- for(k = 0; k < total_peers; k++) {
- peerarray[k] = _sip_show_peers_one(fd, s, &cont, peerarray[k]);
- }
-
- if (!s) {
- ast_cli(fd, "%d sip peers [Monitored: %d online, %d offline Unmonitored: %d online, %d offline]\n",
- total_peers, cont.peers_mon_online, cont.peers_mon_offline, cont.peers_unmon_online, cont.peers_unmon_offline);
- }
-
- if (cont.havepattern) {
- regfree(&cont.regexbuf);
- }
-
- if (total) {
- *total = total_peers;
- }
-
- ast_free(peerarray);
-
- return CLI_SUCCESS;
-}
-
-/*! \brief Emit informations for one peer during sip show peers command */
-static struct sip_peer *_sip_show_peers_one(int fd, struct mansession *s, struct show_peers_context *cont, struct sip_peer *peer)
-{
- /* _sip_show_peers_one() is separated from _sip_show_peers() to properly free the ast_strdupa
- * (this is executed in a loop in _sip_show_peers() )
- */
-
- char name[256];
- char status[20] = "";
- char pstatus;
-
- /*
- * tmp_port and tmp_host store copies of ast_sockaddr_stringify strings since the
- * string pointers for that function aren't valid between subsequent calls to
- * ast_sockaddr_stringify functions
- */
- char *tmp_port;
- char *tmp_host;
-
- tmp_port = ast_sockaddr_isnull(&peer->addr) ?
- "0" : ast_strdupa(ast_sockaddr_stringify_port(&peer->addr));
-
- tmp_host = ast_sockaddr_isnull(&peer->addr) ?
- "(Unspecified)" : ast_strdupa(ast_sockaddr_stringify_addr(&peer->addr));
-
- ao2_lock(peer);
- if (cont->havepattern && regexec(&cont->regexbuf, peer->name, 0, NULL, 0)) {
- ao2_unlock(peer);
- return sip_unref_peer(peer, "toss iterator peer ptr no match");
- }
-
- if (!ast_strlen_zero(peer->username) && !s) {
- snprintf(name, sizeof(name), "%s/%s", peer->name, peer->username);
- } else {
- ast_copy_string(name, peer->name, sizeof(name));
- }
-
- pstatus = peer_status(peer, status, sizeof(status));
- if (pstatus == 1) {
- cont->peers_mon_online++;
- } else if (pstatus == 0) {
- cont->peers_mon_offline++;
- } else {
- if (ast_sockaddr_isnull(&peer->addr) ||
- !ast_sockaddr_port(&peer->addr)) {
- cont->peers_unmon_offline++;
- } else {
- cont->peers_unmon_online++;
- }
- }
-
- if (!s) { /* Normal CLI list */
- ast_cli(fd, PEERS_FORMAT2, name,
- tmp_host,
- peer->host_dynamic ? " D " : " ", /* Dynamic or not? */
- force_rport_string(peer->flags),
- comedia_string(peer->flags),
- (!ast_acl_list_is_empty(peer->acl)) ? " A " : " ", /* permit/deny */
- tmp_port, status,
- peer->description ? peer->description : "",
- cont->realtimepeers ? (peer->is_realtime ? "Cached RT" : "") : "");
- } else { /* Manager format */
- /* The names here need to be the same as other channels */
- astman_append(s,
- "Event: PeerEntry\r\n%s"
- "Channeltype: SIP\r\n"
- "ObjectName: %s\r\n"
- "ChanObjectType: peer\r\n" /* "peer" or "user" */
- "IPaddress: %s\r\n"
- "IPport: %s\r\n"
- "Dynamic: %s\r\n"
- "AutoForcerport: %s\r\n"
- "Forcerport: %s\r\n"
- "AutoComedia: %s\r\n"
- "Comedia: %s\r\n"
- "VideoSupport: %s\r\n"
- "TextSupport: %s\r\n"
- "ACL: %s\r\n"
- "Status: %s\r\n"
- "RealtimeDevice: %s\r\n"
- "Description: %s\r\n"
- "Accountcode: %s\r\n"
- "\r\n",
- cont->idtext,
- peer->name,
- ast_sockaddr_isnull(&peer->addr) ? "-none-" : tmp_host,
- ast_sockaddr_isnull(&peer->addr) ? "0" : tmp_port,
- peer->host_dynamic ? "yes" : "no", /* Dynamic or not? */
- ast_test_flag(&peer->flags[2], SIP_PAGE3_NAT_AUTO_RPORT) ? "yes" : "no",
- ast_test_flag(&peer->flags[0], SIP_NAT_FORCE_RPORT) ? "yes" : "no",
- ast_test_flag(&peer->flags[2], SIP_PAGE3_NAT_AUTO_COMEDIA) ? "yes" : "no",
- ast_test_flag(&peer->flags[1], SIP_PAGE2_SYMMETRICRTP) ? "yes" : "no",
- ast_test_flag(&peer->flags[1], SIP_PAGE2_VIDEOSUPPORT) ? "yes" : "no", /* VIDEOSUPPORT=yes? */
- ast_test_flag(&peer->flags[1], SIP_PAGE2_TEXTSUPPORT) ? "yes" : "no", /* TEXTSUPPORT=yes? */
- ast_acl_list_is_empty(peer->acl) ? "no" : "yes", /* permit/deny/acl */
- status,
- cont->realtimepeers ? (peer->is_realtime ? "yes" : "no") : "no",
- peer->description,
- peer->accountcode);
- }
- ao2_unlock(peer);
-
- return sip_unref_peer(peer, "toss iterator peer ptr");
-}
-#undef PEERS_FORMAT2
-
-static int peer_dump_func(void *userobj, void *arg, int flags)
-{
- struct sip_peer *peer = userobj;
- int refc = ao2_t_ref(userobj, 0, "");
- struct ast_cli_args *a = (struct ast_cli_args *) arg;
-
- ast_cli(a->fd, "name: %s\ntype: peer\nobjflags: %d\nrefcount: %d\n\n",
- peer->name, 0, refc);
- return 0;
-}
-
-static int dialog_dump_func(void *userobj, void *arg, int flags)
-{
- struct sip_pvt *pvt = userobj;
- int refc = ao2_t_ref(userobj, 0, "");
- struct ast_cli_args *a = (struct ast_cli_args *) arg;
-
- ast_cli(a->fd, "name: %s\ntype: dialog\nobjflags: %d\nrefcount: %d\n\n",
- pvt->callid, 0, refc);
- return 0;
-}
-
-
-/*! \brief List all allocated SIP Objects (realtime or static) */
-static char *sip_show_objects(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
-{
- struct sip_registry *reg;
- struct ao2_iterator iter;
-
- switch (cmd) {
- case CLI_INIT:
- e->command = "sip show objects";
- e->usage =
- "Usage: sip show objects\n"
- " Lists status of known SIP objects\n";
- return NULL;
- case CLI_GENERATE:
- return NULL;
- }
-
- if (a->argc != 3)
- return CLI_SHOWUSAGE;
- ast_cli(a->fd, "-= Peer objects: %d static, %d realtime, %d autocreate =-\n\n", speerobjs, rpeerobjs, apeerobjs);
- ao2_t_callback(peers, OBJ_NODATA, peer_dump_func, a, "initiate ao2_callback to dump peers");
- ast_cli(a->fd, "-= Peer objects by IP =-\n\n");
- ao2_t_callback(peers_by_ip, OBJ_NODATA, peer_dump_func, a, "initiate ao2_callback to dump peers_by_ip");
-
- iter = ao2_iterator_init(registry_list, 0);
- ast_cli(a->fd, "-= Registry objects: %d =-\n\n", ao2_container_count(registry_list));
- while ((reg = ao2_t_iterator_next(&iter, "sip_show_objects iter"))) {
- ao2_lock(reg);
- ast_cli(a->fd, "name: %s\n", reg->configvalue);
- ao2_unlock(reg);
- ao2_t_ref(reg, -1, "sip_show_objects iter");
- }
- ao2_iterator_destroy(&iter);
-
- ast_cli(a->fd, "-= Dialog objects:\n\n");
- ao2_t_callback(dialogs, OBJ_NODATA, dialog_dump_func, a, "initiate ao2_callback to dump dialogs");
- return CLI_SUCCESS;
-}
-/*! \brief Print call group and pickup group */
-static void print_group(int fd, ast_group_t group, int crlf)
-{
- char buf[256];
- ast_cli(fd, crlf ? "%s\r\n" : "%s\n", ast_print_group(buf, sizeof(buf), group) );
-}
-
-/*! \brief Print named call groups and pickup groups */
-static void print_named_groups(int fd, struct ast_namedgroups *group, int crlf)
-{
- struct ast_str *buf = ast_str_create(1024);
- if (buf) {
- ast_cli(fd, crlf ? "%s\r\n" : "%s\n", ast_print_namedgroups(&buf, group) );
- ast_free(buf);
- }
-}
-
-/*! \brief mapping between dtmf flags and strings */
-static const struct _map_x_s dtmfstr[] = {
- { SIP_DTMF_RFC2833, "rfc2833" },
- { SIP_DTMF_INFO, "info" },
- { SIP_DTMF_SHORTINFO, "shortinfo" },
- { SIP_DTMF_INBAND, "inband" },
- { SIP_DTMF_AUTO, "auto" },
- { -1, NULL }, /* terminator */
-};
-
-/*! \brief Convert DTMF mode to printable string */
-static const char *dtmfmode2str(int mode)
-{
- return map_x_s(dtmfstr, mode, "");
-}
-
-/*! \brief maps a string to dtmfmode, returns -1 on error */
-static int str2dtmfmode(const char *str)
-{
- return map_s_x(dtmfstr, str, -1);
-}
-
-static const struct _map_x_s insecurestr[] = {
- { SIP_INSECURE_PORT, "port" },
- { SIP_INSECURE_INVITE, "invite" },
- { SIP_INSECURE_PORT | SIP_INSECURE_INVITE, "port,invite" },
- { 0, "no" },
- { -1, NULL }, /* terminator */
-};
-
-/*! \brief Convert Insecure setting to printable string */
-static const char *insecure2str(int mode)
-{
- return map_x_s(insecurestr, mode, "");
-}
-
-static const struct _map_x_s allowoverlapstr[] = {
- { SIP_PAGE2_ALLOWOVERLAP_YES, "Yes" },
- { SIP_PAGE2_ALLOWOVERLAP_DTMF, "DTMF" },
- { SIP_PAGE2_ALLOWOVERLAP_NO, "No" },
- { -1, NULL }, /* terminator */
-};
-
-/*! \brief Convert AllowOverlap setting to printable string */
-static const char *allowoverlap2str(int mode)
-{
- return map_x_s(allowoverlapstr, mode, "");
-}
-
-static const struct _map_x_s trust_id_outboundstr[] = {
- { SIP_PAGE2_TRUST_ID_OUTBOUND_LEGACY, "Legacy" },
- { SIP_PAGE2_TRUST_ID_OUTBOUND_NO, "No" },
- { SIP_PAGE2_TRUST_ID_OUTBOUND_YES, "Yes" },
- { -1, NULL }, /* terminator */
-};
-
-static const char *trust_id_outbound2str(int mode)
-{
- return map_x_s(trust_id_outboundstr, mode, "");
-}
-
-/*! \brief Destroy disused contexts between reloads
- Only used in reload_config so the code for regcontext doesn't get ugly
-*/
-static void cleanup_stale_contexts(char *new, char *old)
-{
- char *oldcontext, *newcontext, *stalecontext, *stringp, newlist[AST_MAX_CONTEXT];
-
- while ((oldcontext = strsep(&old, "&"))) {
- stalecontext = NULL;
- ast_copy_string(newlist, new, sizeof(newlist));
- stringp = newlist;
- while ((newcontext = strsep(&stringp, "&"))) {
- if (!strcmp(newcontext, oldcontext)) {
- /* This is not the context you're looking for */
- stalecontext = NULL;
- break;
- } else if (strcmp(newcontext, oldcontext)) {
- stalecontext = oldcontext;
- }
-
- }
- ast_context_destroy_by_name(stalecontext, "SIP");
- }
-}
-
-/*!
- * \brief Check RTP Timeout on dialogs
- *
- * \details This is used with ao2_callback to check rtptimeout
- * rtponholdtimeout and send rtpkeepalive packets.
- *
- * \return CMP_MATCH for items to be unlinked from dialogs_rtpcheck.
- */
-static int dialog_checkrtp_cb(void *dialogobj, void *arg, int flags)
-{
- struct sip_pvt *dialog = dialogobj;
- time_t *t = arg;
- int match_status;
-
- if (sip_pvt_trylock(dialog)) {
- return 0;
- }
-
- if (dialog->rtp || dialog->vrtp) {
- match_status = check_rtp_timeout(dialog, *t);
- } else {
- /* Dialog has no active RTP or VRTP. unlink it from dialogs_rtpcheck. */
- match_status = CMP_MATCH;
- }
- sip_pvt_unlock(dialog);
-
- return match_status;
-}
-
-/*!
- * \brief Match dialogs that need to be destroyed
- *
- * \details This is used with ao2_callback to unlink/delete all dialogs that
- * are marked needdestroy.
- *
- * \todo Re-work this to improve efficiency. Currently, this function is called
- * on _every_ dialog after processing _every_ incoming SIP/UDP packet, or
- * potentially even more often when the scheduler has entries to run.
- */
-static int dialog_needdestroy(void *dialogobj, void *arg, int flags)
-{
- struct sip_pvt *dialog = dialogobj;
-
- if (sip_pvt_trylock(dialog)) {
- /* Don't block the monitor thread. This function is called often enough
- * that we can wait for the next time around. */
- return 0;
- }
-
- /* If we have sessions that needs to be destroyed, do it now */
- /* Check if we have outstanding requests not responsed to or an active call
- - if that's the case, wait with destruction */
- if (dialog->needdestroy && !dialog->packets && !dialog->owner) {
- /* We absolutely cannot destroy the rtp struct while a bridge is active or we WILL crash */
- if (dialog->rtp && ast_rtp_instance_get_bridged(dialog->rtp)) {
- ast_debug(2, "Bridge still active. Delaying destruction of SIP dialog '%s' Method: %s\n", dialog->callid, sip_methods[dialog->method].text);
- sip_pvt_unlock(dialog);
- return 0;
- }
-
- if (dialog->vrtp && ast_rtp_instance_get_bridged(dialog->vrtp)) {
- ast_debug(2, "Bridge still active. Delaying destroy of SIP dialog '%s' Method: %s\n", dialog->callid, sip_methods[dialog->method].text);
- sip_pvt_unlock(dialog);
- return 0;
- }
-
- sip_pvt_unlock(dialog);
- /* no, the unlink should handle this: dialog_unref(dialog, "needdestroy: one more refcount decrement to allow dialog to be destroyed"); */
- /* the CMP_MATCH will unlink this dialog from the dialog hash table */
- dialog_unlink_all(dialog);
- return 0; /* the unlink_all should unlink this from the table, so.... no need to return a match */
- }
-
- sip_pvt_unlock(dialog);
-
- return 0;
-}
-
-/*! \brief Remove temporary realtime objects from memory (CLI) */
-/*! \todo XXXX Propably needs an overhaul after removal of the devices */
-static char *sip_prune_realtime(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
-{
- struct sip_peer *peer, *pi;
- int prunepeer = FALSE;
- int multi = FALSE;
- const char *name = NULL;
- regex_t regexbuf;
- int havepattern = 0;
- struct ao2_iterator i;
- static const char * const choices[] = { "all", "like", NULL };
- char *cmplt;
-
- if (cmd == CLI_INIT) {
- e->command = "sip prune realtime [peer|all]";
- e->usage =
- "Usage: sip prune realtime [peer [|all|like ]|all]\n"
- " Prunes object(s) from the cache.\n"
- " Optional regular expression pattern is used to filter the objects.\n";
- return NULL;
- } else if (cmd == CLI_GENERATE) {
- if (a->pos == 4 && !strcasecmp(a->argv[3], "peer")) {
- cmplt = ast_cli_complete(a->word, choices, a->n);
- if (!cmplt)
- cmplt = complete_sip_peer(a->word, a->n - sizeof(choices), SIP_PAGE2_RTCACHEFRIENDS);
- return cmplt;
- }
- if (a->pos == 5 && !strcasecmp(a->argv[4], "like"))
- return complete_sip_peer(a->word, a->n, SIP_PAGE2_RTCACHEFRIENDS);
- return NULL;
- }
- switch (a->argc) {
- case 4:
- name = a->argv[3];
- /* we accept a name in position 3, but keywords are not good. */
- if (!strcasecmp(name, "peer") || !strcasecmp(name, "like"))
- return CLI_SHOWUSAGE;
- prunepeer = TRUE;
- if (!strcasecmp(name, "all")) {
- multi = TRUE;
- name = NULL;
- }
- /* else a single name, already set */
- break;
- case 5:
- /* sip prune realtime {peer|like} name */
- name = a->argv[4];
- if (!strcasecmp(a->argv[3], "peer"))
- prunepeer = TRUE;
- else if (!strcasecmp(a->argv[3], "like")) {
- prunepeer = TRUE;
- multi = TRUE;
- } else
- return CLI_SHOWUSAGE;
- if (!strcasecmp(name, "like"))
- return CLI_SHOWUSAGE;
- if (!multi && !strcasecmp(name, "all")) {
- multi = TRUE;
- name = NULL;
- }
- break;
- case 6:
- name = a->argv[5];
- multi = TRUE;
- /* sip prune realtime {peer} like name */
- if (strcasecmp(a->argv[4], "like"))
- return CLI_SHOWUSAGE;
- if (!strcasecmp(a->argv[3], "peer")) {
- prunepeer = TRUE;
- } else
- return CLI_SHOWUSAGE;
- break;
- default:
- return CLI_SHOWUSAGE;
- }
-
- if (multi && name) {
- if (regcomp(®exbuf, name, REG_EXTENDED | REG_NOSUB)) {
- return CLI_SHOWUSAGE;
- }
- havepattern = 1;
- }
-
- if (multi) {
- if (prunepeer) {
- int pruned = 0;
-
- i = ao2_iterator_init(peers, 0);
- while ((pi = ao2_t_iterator_next(&i, "iterate thru peers table"))) {
- ao2_lock(pi);
- if (name && regexec(®exbuf, pi->name, 0, NULL, 0)) {
- ao2_unlock(pi);
- sip_unref_peer(pi, "toss iterator peer ptr before continue");
- continue;
- };
- if (ast_test_flag(&pi->flags[1], SIP_PAGE2_RTCACHEFRIENDS)) {
- pi->the_mark = 1;
- pruned++;
- }
- ao2_unlock(pi);
- sip_unref_peer(pi, "toss iterator peer ptr");
- }
- ao2_iterator_destroy(&i);
- if (pruned) {
- unlink_marked_peers_from_tables();
- ast_cli(a->fd, "%d peers pruned.\n", pruned);
- } else
- ast_cli(a->fd, "No peers found to prune.\n");
- }
- } else {
- if (prunepeer) {
- struct sip_peer tmp;
- ast_copy_string(tmp.name, name, sizeof(tmp.name));
- if ((peer = ao2_t_find(peers, &tmp, OBJ_POINTER | OBJ_UNLINK, "finding to unlink from peers"))) {
- if (!ast_sockaddr_isnull(&peer->addr)) {
- ao2_t_unlink(peers_by_ip, peer, "unlinking peer from peers_by_ip also");
- }
- if (!ast_test_flag(&peer->flags[1], SIP_PAGE2_RTCACHEFRIENDS)) {
- ast_cli(a->fd, "Peer '%s' is not a Realtime peer, cannot be pruned.\n", name);
- /* put it back! */
- ao2_t_link(peers, peer, "link peer into peer table");
- if (!ast_sockaddr_isnull(&peer->addr)) {
- ao2_t_link(peers_by_ip, peer, "link peer into peers_by_ip table");
- }
- } else
- ast_cli(a->fd, "Peer '%s' pruned.\n", name);
- sip_unref_peer(peer, "sip_prune_realtime: sip_unref_peer: tossing temp peer ptr");
- } else
- ast_cli(a->fd, "Peer '%s' not found.\n", name);
- }
- }
-
- if (havepattern) {
- regfree(®exbuf);
- }
-
- return CLI_SUCCESS;
-}
-
-/*! \brief Print domain mode to cli */
-static const char *domain_mode_to_text(const enum domain_mode mode)
-{
- switch (mode) {
- case SIP_DOMAIN_AUTO:
- return "[Automatic]";
- case SIP_DOMAIN_CONFIG:
- return "[Configured]";
- }
-
- return "";
-}
-
-/*! \brief CLI command to list local domains */
-static char *sip_show_domains(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
-{
- struct domain *d;
-#define FORMAT "%-40.40s %-20.20s %-16.16s\n"
-
- switch (cmd) {
- case CLI_INIT:
- e->command = "sip show domains";
- e->usage =
- "Usage: sip show domains\n"
- " Lists all configured SIP local domains.\n"
- " Asterisk only responds to SIP messages to local domains.\n";
- return NULL;
- case CLI_GENERATE:
- return NULL;
- }
-
- if (AST_LIST_EMPTY(&domain_list)) {
- ast_cli(a->fd, "SIP Domain support not enabled.\n\n");
- return CLI_SUCCESS;
- } else {
- ast_cli(a->fd, FORMAT, "Our local SIP domains:", "Context", "Set by");
- AST_LIST_LOCK(&domain_list);
- AST_LIST_TRAVERSE(&domain_list, d, list)
- ast_cli(a->fd, FORMAT, d->domain, S_OR(d->context, "(default)"),
- domain_mode_to_text(d->mode));
- AST_LIST_UNLOCK(&domain_list);
- ast_cli(a->fd, "\n");
- return CLI_SUCCESS;
- }
-}
-#undef FORMAT
-
-/*! \brief Show SIP peers in the manager API */
-static int manager_sip_show_peer(struct mansession *s, const struct message *m)
-{
- const char *a[4];
- const char *peer;
-
- peer = astman_get_header(m, "Peer");
- if (ast_strlen_zero(peer)) {
- astman_send_error(s, m, "Peer: missing.");
- return 0;
- }
- a[0] = "sip";
- a[1] = "show";
- a[2] = "peer";
- a[3] = peer;
-
- _sip_show_peer(1, -1, s, m, 4, a);
- astman_append(s, "\r\n" );
- return 0;
-}
-
-/*! \brief Show one peer in detail */
-static char *sip_show_peer(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
-{
- switch (cmd) {
- case CLI_INIT:
- e->command = "sip show peer";
- e->usage =
- "Usage: sip show peer [load]\n"
- " Shows all details on one SIP peer and the current status.\n"
- " Option \"load\" forces lookup of peer in realtime storage.\n";
- return NULL;
- case CLI_GENERATE:
- if (a->pos == 4) {
- static const char * const completions[] = { "load", NULL };
- return ast_cli_complete(a->word, completions, a->n);
- } else {
- return complete_sip_show_peer(a->line, a->word, a->pos, a->n);
- }
- }
- return _sip_show_peer(0, a->fd, NULL, NULL, a->argc, (const char **) a->argv);
-}
-
-static void send_manager_peer_status(struct mansession *s, struct sip_peer *peer, const char *idText)
-{
- char time[128] = "";
- char status[128] = "";
- if (peer->maxms) {
- if (peer->lastms < 0) {
- snprintf(status, sizeof(status), "PeerStatus: Unreachable\r\n");
- } else if (peer->lastms > peer->maxms) {
- snprintf(status, sizeof(status), "PeerStatus: Lagged\r\n");
- snprintf(time, sizeof(time), "Time: %d\r\n", peer->lastms);
- } else if (peer->lastms) {
- snprintf(status, sizeof(status), "PeerStatus: Reachable\r\n");
- snprintf(time, sizeof(time), "Time: %d\r\n", peer->lastms);
- } else {
- snprintf(status, sizeof(status), "PeerStatus: Unknown\r\n");
- }
- } else {
- snprintf(status, sizeof(status), "PeerStatus: Unmonitored\r\n");
- }
-
- astman_append(s,
- "Event: PeerStatus\r\n"
- "Privilege: System\r\n"
- "ChannelType: SIP\r\n"
- "Peer: SIP/%s\r\n"
- "%s"
- "%s"
- "%s"
- "\r\n",
- peer->name, status, time, idText);
-}
-
-/*! \brief Show SIP peers in the manager API */
-static int manager_sip_peer_status(struct mansession *s, const struct message *m)
-{
- const char *id = astman_get_header(m,"ActionID");
- const char *peer_name = astman_get_header(m,"Peer");
- char idText[256];
- struct sip_peer *peer = NULL;
- int num_peers = 0;
-
- idText[0] = '\0';
- if (!ast_strlen_zero(id)) {
- snprintf(idText, sizeof(idText), "ActionID: %s\r\n", id);
- }
-
- if (!ast_strlen_zero(peer_name)) {
- /* strip SIP/ from the begining of the peer name */
- if (strlen(peer_name) >= 4 && !strncasecmp("SIP/", peer_name, 4)) {
- peer_name += 4;
- }
-
- peer = sip_find_peer(peer_name, NULL, TRUE, FINDPEERS, FALSE, 0);
- if (!peer) {
- astman_send_error(s, m, "No such peer");
- return 0;
- }
- }
-
- astman_send_listack(s, m, "Peer status will follow", "start");
-
- if (!peer) {
- struct ao2_iterator i = ao2_iterator_init(peers, 0);
-
- while ((peer = ao2_t_iterator_next(&i, "iterate thru peers table for SIPpeerstatus"))) {
- ao2_lock(peer);
- send_manager_peer_status(s, peer, idText);
- ao2_unlock(peer);
- sip_unref_peer(peer, "unref peer for SIPpeerstatus");
- ++num_peers;
- }
- ao2_iterator_destroy(&i);
- } else {
- ao2_lock(peer);
- send_manager_peer_status(s, peer, idText);
- ao2_unlock(peer);
- sip_unref_peer(peer, "unref peer for SIPpeerstatus");
- ++num_peers;
- }
-
- astman_send_list_complete_start(s, m, "SIPpeerstatusComplete", num_peers);
- astman_send_list_complete_end(s);
-
- return 0;
-}
-
-static void publish_qualify_peer_done(const char *id, const char *peer)
-{
- RAII_VAR(struct ast_json *, body, NULL, ast_json_unref);
-
- if (ast_strlen_zero(id)) {
- body = ast_json_pack("{s: s}", "Peer", peer);
- } else {
- body = ast_json_pack("{s: s, s: s}", "Peer", peer, "ActionID", id);
- }
- if (!body) {
- return;
- }
-
- ast_manager_publish_event("SIPQualifyPeerDone", EVENT_FLAG_CALL, body);
-}
-
-/*! \brief Send qualify message to peer from cli or manager. Mostly for debugging. */
-static char *_sip_qualify_peer(int type, int fd, struct mansession *s, const struct message *m, int argc, const char *argv[])
-{
- struct sip_peer *peer;
- int load_realtime;
-
- if (argc < 4)
- return CLI_SHOWUSAGE;
-
- load_realtime = (argc == 5 && !strcmp(argv[4], "load")) ? TRUE : FALSE;
- if ((peer = sip_find_peer(argv[3], NULL, load_realtime, FINDPEERS, FALSE, 0))) {
- const char *id = astman_get_header(m,"ActionID");
-
- if (type != 0) {
- astman_send_ack(s, m, "SIP peer found - will qualify");
- }
-
- sip_poke_peer(peer, 1);
-
- publish_qualify_peer_done(id, argv[3]);
-
- sip_unref_peer(peer, "qualify: done with peer");
- } else if (type == 0) {
- ast_cli(fd, "Peer '%s' not found\n", argv[3]);
- } else {
- astman_send_error(s, m, "Peer not found");
- }
-
- return CLI_SUCCESS;
-}
-
-/*! \brief Qualify SIP peers in the manager API */
-static int manager_sip_qualify_peer(struct mansession *s, const struct message *m)
-{
- const char *a[4];
- const char *peer;
-
- peer = astman_get_header(m, "Peer");
- if (ast_strlen_zero(peer)) {
- astman_send_error(s, m, "Peer: missing.");
- return 0;
- }
- a[0] = "sip";
- a[1] = "qualify";
- a[2] = "peer";
- a[3] = peer;
-
- _sip_qualify_peer(1, -1, s, m, 4, a);
- return 0;
-}
-
-/*! \brief Send an OPTIONS packet to a SIP peer */
-static char *sip_qualify_peer(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
-{
- switch (cmd) {
- case CLI_INIT:
- e->command = "sip qualify peer";
- e->usage =
- "Usage: sip qualify peer [load]\n"
- " Requests a response from one SIP peer and the current status.\n"
- " Option \"load\" forces lookup of peer in realtime storage.\n";
- return NULL;
- case CLI_GENERATE:
- if (a->pos == 4) {
- static const char * const completions[] = { "load", NULL };
- return ast_cli_complete(a->word, completions, a->n);
- } else {
- return complete_sip_show_peer(a->line, a->word, a->pos, a->n);
- }
- }
- return _sip_qualify_peer(0, a->fd, NULL, NULL, a->argc, (const char **) a->argv);
-}
-
-/*! \brief list peer mailboxes to CLI */
-static void peer_mailboxes_to_str(struct ast_str **mailbox_str, struct sip_peer *peer)
-{
- struct sip_mailbox *mailbox;
-
- AST_LIST_TRAVERSE(&peer->mailboxes, mailbox, entry) {
- ast_str_append(mailbox_str, 0, "%s%s",
- mailbox->id,
- AST_LIST_NEXT(mailbox, entry) ? "," : "");
- }
-}
-
-static struct _map_x_s faxecmodes[] = {
- { SIP_PAGE2_T38SUPPORT_UDPTL, "None"},
- { SIP_PAGE2_T38SUPPORT_UDPTL_FEC, "FEC"},
- { SIP_PAGE2_T38SUPPORT_UDPTL_REDUNDANCY, "Redundancy"},
- { -1, NULL},
-};
-
-static const char *faxec2str(int faxec)
-{
- return map_x_s(faxecmodes, faxec, "Unknown");
-}
-
-/*! \brief Show one peer in detail (main function) */
-static char *_sip_show_peer(int type, int fd, struct mansession *s, const struct message *m, int argc, const char *argv[])
-{
- char status[30] = "";
- char cbuf[256];
- struct sip_peer *peer;
- struct ast_str *codec_buf = ast_str_alloca(AST_FORMAT_CAP_NAMES_LEN);
- struct ast_variable *v;
- int x = 0, load_realtime;
- int realtimepeers;
-
- realtimepeers = ast_check_realtime("sippeers");
-
- if (argc < 4)
- return CLI_SHOWUSAGE;
-
- load_realtime = (argc == 5 && !strcmp(argv[4], "load")) ? TRUE : FALSE;
- peer = sip_find_peer(argv[3], NULL, load_realtime, FINDPEERS, FALSE, 0);
-
- if (s) { /* Manager */
- if (peer) {
- const char *id = astman_get_header(m, "ActionID");
-
- astman_append(s, "Response: Success\r\n");
- if (!ast_strlen_zero(id))
- astman_append(s, "ActionID: %s\r\n", id);
- } else {
- snprintf (cbuf, sizeof(cbuf), "Peer %s not found.", argv[3]);
- astman_send_error(s, m, cbuf);
- return CLI_SUCCESS;
- }
- }
- if (peer && type==0 ) { /* Normal listing */
- struct ast_str *mailbox_str = ast_str_alloca(512);
- struct ast_str *path;
- struct sip_auth_container *credentials;
-
- ao2_lock(peer);
- credentials = peer->auth;
- if (credentials) {
- ao2_t_ref(credentials, +1, "Ref peer auth for show");
- }
- ao2_unlock(peer);
-
- ast_cli(fd, "\n\n");
- ast_cli(fd, " * Name : %s\n", peer->name);
- ast_cli(fd, " Description : %s\n", peer->description);
- if (realtimepeers) { /* Realtime is enabled */
- ast_cli(fd, " Realtime peer: %s\n", peer->is_realtime ? "Yes, cached" : "No");
- }
- ast_cli(fd, " Secret : %s\n", ast_strlen_zero(peer->secret)?"":"");
- ast_cli(fd, " MD5Secret : %s\n", ast_strlen_zero(peer->md5secret)?"":"");
- ast_cli(fd, " Remote Secret: %s\n", ast_strlen_zero(peer->remotesecret)?"":"");
- if (credentials) {
- struct sip_auth *auth;
-
- AST_LIST_TRAVERSE(&credentials->list, auth, node) {
- ast_cli(fd, " Realm-auth : Realm %-15.15s User %-10.20s %s\n",
- auth->realm,
- auth->username,
- !ast_strlen_zero(auth->secret)
- ? ""
- : (!ast_strlen_zero(auth->md5secret)
- ? "" : ""));
- }
- ao2_t_ref(credentials, -1, "Unref peer auth for show");
- }
- ast_cli(fd, " Context : %s\n", peer->context);
- ast_cli(fd, " Record On feature : %s\n", peer->record_on_feature);
- ast_cli(fd, " Record Off feature : %s\n", peer->record_off_feature);
- ast_cli(fd, " Subscr.Cont. : %s\n", S_OR(peer->subscribecontext, "") );
- ast_cli(fd, " Language : %s\n", peer->language);
- ast_cli(fd, " Tonezone : %s\n", peer->zone[0] != '\0' ? peer->zone : "");
- if (!ast_strlen_zero(peer->accountcode))
- ast_cli(fd, " Accountcode : %s\n", peer->accountcode);
- ast_cli(fd, " AMA flags : %s\n", ast_channel_amaflags2string(peer->amaflags));
- ast_cli(fd, " Transfer mode: %s\n", transfermode2str(peer->allowtransfer));
- ast_cli(fd, " CallingPres : %s\n", ast_describe_caller_presentation(peer->callingpres));
- if (!ast_strlen_zero(peer->fromuser))
- ast_cli(fd, " FromUser : %s\n", peer->fromuser);
- if (!ast_strlen_zero(peer->fromdomain))
- ast_cli(fd, " FromDomain : %s Port %d\n", peer->fromdomain, (peer->fromdomainport) ? peer->fromdomainport : STANDARD_SIP_PORT);
- ast_cli(fd, " Callgroup : ");
- print_group(fd, peer->callgroup, 0);
- ast_cli(fd, " Pickupgroup : ");
- print_group(fd, peer->pickupgroup, 0);
- ast_cli(fd, " Named Callgr : ");
- print_named_groups(fd, peer->named_callgroups, 0);
- ast_cli(fd, " Nam. Pickupgr: ");
- print_named_groups(fd, peer->named_pickupgroups, 0);
- peer_mailboxes_to_str(&mailbox_str, peer);
- ast_cli(fd, " MOH Suggest : %s\n", peer->mohsuggest);
- ast_cli(fd, " Mailbox : %s\n", ast_str_buffer(mailbox_str));
- ast_cli(fd, " VM Extension : %s\n", peer->vmexten);
- ast_cli(fd, " LastMsgsSent : %d/%d\n", (peer->lastmsgssent & 0x7fff0000) >> 16, peer->lastmsgssent & 0xffff);
- ast_cli(fd, " Call limit : %d\n", peer->call_limit);
- ast_cli(fd, " Max forwards : %d\n", peer->maxforwards);
- if (peer->busy_level)
- ast_cli(fd, " Busy level : %d\n", peer->busy_level);
- ast_cli(fd, " Dynamic : %s\n", AST_CLI_YESNO(peer->host_dynamic));
- ast_cli(fd, " Callerid : %s\n", ast_callerid_merge(cbuf, sizeof(cbuf), peer->cid_name, peer->cid_num, ""));
- ast_cli(fd, " MaxCallBR : %d kbps\n", peer->maxcallbitrate);
- ast_cli(fd, " Expire : %ld\n", ast_sched_when(sched, peer->expire));
- ast_cli(fd, " Insecure : %s\n", insecure2str(ast_test_flag(&peer->flags[0], SIP_INSECURE)));
- ast_cli(fd, " Force rport : %s\n", force_rport_string(peer->flags));
- ast_cli(fd, " Symmetric RTP: %s\n", comedia_string(peer->flags));
- ast_cli(fd, " ACL : %s\n", AST_CLI_YESNO(ast_acl_list_is_empty(peer->acl) == 0));
- ast_cli(fd, " ContactACL : %s\n", AST_CLI_YESNO(ast_acl_list_is_empty(peer->contactacl) == 0));
- ast_cli(fd, " DirectMedACL : %s\n", AST_CLI_YESNO(ast_acl_list_is_empty(peer->directmediaacl) == 0));
- ast_cli(fd, " T.38 support : %s\n", AST_CLI_YESNO(ast_test_flag(&peer->flags[1], SIP_PAGE2_T38SUPPORT)));
- ast_cli(fd, " T.38 EC mode : %s\n", faxec2str(ast_test_flag(&peer->flags[1], SIP_PAGE2_T38SUPPORT)));
- ast_cli(fd, " T.38 MaxDtgrm: %u\n", peer->t38_maxdatagram);
- ast_cli(fd, " DirectMedia : %s\n", AST_CLI_YESNO(ast_test_flag(&peer->flags[0], SIP_DIRECT_MEDIA)));
- ast_cli(fd, " PromiscRedir : %s\n", AST_CLI_YESNO(ast_test_flag(&peer->flags[0], SIP_PROMISCREDIR)));
- ast_cli(fd, " User=Phone : %s\n", AST_CLI_YESNO(ast_test_flag(&peer->flags[0], SIP_USEREQPHONE)));
- ast_cli(fd, " Video Support: %s\n", AST_CLI_YESNO(ast_test_flag(&peer->flags[1], SIP_PAGE2_VIDEOSUPPORT) || ast_test_flag(&peer->flags[1], SIP_PAGE2_VIDEOSUPPORT_ALWAYS)));
- ast_cli(fd, " Text Support : %s\n", AST_CLI_YESNO(ast_test_flag(&peer->flags[1], SIP_PAGE2_TEXTSUPPORT)));
- ast_cli(fd, " Ign SDP ver : %s\n", AST_CLI_YESNO(ast_test_flag(&peer->flags[1], SIP_PAGE2_IGNORESDPVERSION)));
- ast_cli(fd, " Trust RPID : %s\n", AST_CLI_YESNO(ast_test_flag(&peer->flags[0], SIP_TRUSTRPID)));
- ast_cli(fd, " Send RPID : %s\n", AST_CLI_YESNO(ast_test_flag(&peer->flags[0], SIP_SENDRPID)));
- ast_cli(fd, " Path support : %s\n", AST_CLI_YESNO(ast_test_flag(&peer->flags[0], SIP_USEPATH)));
- if ((path = sip_route_list(&peer->path, 1, 0))) {
- ast_cli(fd, " Path : %s\n", ast_str_buffer(path));
- ast_free(path);
- }
- ast_cli(fd, " TrustIDOutbnd: %s\n", trust_id_outbound2str(ast_test_flag(&peer->flags[1], SIP_PAGE2_TRUST_ID_OUTBOUND)));
- ast_cli(fd, " Subscriptions: %s\n", AST_CLI_YESNO(ast_test_flag(&peer->flags[1], SIP_PAGE2_ALLOWSUBSCRIBE)));
- ast_cli(fd, " Overlap dial : %s\n", allowoverlap2str(ast_test_flag(&peer->flags[1], SIP_PAGE2_ALLOWOVERLAP)));
- if (peer->outboundproxy)
- ast_cli(fd, " Outb. proxy : %s %s\n", ast_strlen_zero(peer->outboundproxy->name) ? "" : peer->outboundproxy->name,
- peer->outboundproxy->force ? "(forced)" : "");
-
- /* - is enumerated */
- ast_cli(fd, " DTMFmode : %s\n", dtmfmode2str(ast_test_flag(&peer->flags[0], SIP_DTMF)));
- ast_cli(fd, " Timer T1 : %d\n", peer->timer_t1);
- ast_cli(fd, " Timer B : %d\n", peer->timer_b);
- ast_cli(fd, " ToHost : %s\n", peer->tohost);
- ast_cli(fd, " Addr->IP : %s\n", ast_sockaddr_stringify(&peer->addr));
- ast_cli(fd, " Defaddr->IP : %s\n", ast_sockaddr_stringify(&peer->defaddr));
- ast_cli(fd, " Prim.Transp. : %s\n", sip_get_transport(peer->socket.type));
- ast_cli(fd, " Allowed.Trsp : %s\n", get_transport_list(peer->transports));
- if (!ast_strlen_zero(sip_cfg.regcontext))
- ast_cli(fd, " Reg. exten : %s\n", peer->regexten);
- ast_cli(fd, " Def. Username: %s\n", peer->username);
- ast_cli(fd, " SIP Options : ");
- if (peer->sipoptions) {
- int lastoption = -1;
- for (x = 0 ; x < ARRAY_LEN(sip_options); x++) {
- if (sip_options[x].id != lastoption) {
- if (peer->sipoptions & sip_options[x].id)
- ast_cli(fd, "%s ", sip_options[x].text);
- lastoption = x;
- }
- }
- } else
- ast_cli(fd, "(none)");
-
- ast_cli(fd, "\n");
- ast_cli(fd, " Codecs : %s\n", ast_format_cap_get_names(peer->caps, &codec_buf));
-
- ast_cli(fd, " Auto-Framing : %s\n", AST_CLI_YESNO(peer->autoframing));
- ast_cli(fd, " Status : ");
- peer_status(peer, status, sizeof(status));
- ast_cli(fd, "%s\n", status);
- ast_cli(fd, " Useragent : %s\n", peer->useragent);
- ast_cli(fd, " Reg. Contact : %s\n", peer->fullcontact);
- ast_cli(fd, " Qualify Freq : %d ms\n", peer->qualifyfreq);
- ast_cli(fd, " Keepalive : %d ms\n", peer->keepalive * 1000);
- if (peer->chanvars) {
- ast_cli(fd, " Variables :\n");
- for (v = peer->chanvars ; v ; v = v->next)
- ast_cli(fd, " %s = %s\n", v->name, v->value);
- }
-
- ast_cli(fd, " Sess-Timers : %s\n", stmode2str(peer->stimer.st_mode_oper));
- ast_cli(fd, " Sess-Refresh : %s\n", strefresherparam2str(peer->stimer.st_ref));
- ast_cli(fd, " Sess-Expires : %d secs\n", peer->stimer.st_max_se);
- ast_cli(fd, " Min-Sess : %d secs\n", peer->stimer.st_min_se);
- ast_cli(fd, " RTP Engine : %s\n", peer->engine);
- ast_cli(fd, " Parkinglot : %s\n", peer->parkinglot);
- ast_cli(fd, " Use Reason : %s\n", AST_CLI_YESNO(ast_test_flag(&peer->flags[1], SIP_PAGE2_Q850_REASON)));
- ast_cli(fd, " Encryption : %s\n", AST_CLI_YESNO(ast_test_flag(&peer->flags[1], SIP_PAGE2_USE_SRTP)));
- ast_cli(fd, " RTCP Mux : %s\n", AST_CLI_YESNO(ast_test_flag(&peer->flags[2], SIP_PAGE3_RTCP_MUX)));
- ast_cli(fd, "\n");
- peer = sip_unref_peer(peer, "sip_show_peer: sip_unref_peer: done with peer ptr");
- } else if (peer && type == 1) { /* manager listing */
- char buffer[256];
- struct ast_str *tmp_str = ast_str_alloca(512);
- astman_append(s, "Channeltype: SIP\r\n");
- astman_append(s, "ObjectName: %s\r\n", peer->name);
- astman_append(s, "ChanObjectType: peer\r\n");
- astman_append(s, "SecretExist: %s\r\n", ast_strlen_zero(peer->secret)?"N":"Y");
- astman_append(s, "RemoteSecretExist: %s\r\n", ast_strlen_zero(peer->remotesecret)?"N":"Y");
- astman_append(s, "MD5SecretExist: %s\r\n", ast_strlen_zero(peer->md5secret)?"N":"Y");
- astman_append(s, "Context: %s\r\n", peer->context);
- if (!ast_strlen_zero(peer->subscribecontext)) {
- astman_append(s, "SubscribeContext: %s\r\n", peer->subscribecontext);
- }
- astman_append(s, "Language: %s\r\n", peer->language);
- astman_append(s, "ToneZone: %s\r\n", peer->zone[0] != '\0' ? peer->zone : "");
- if (!ast_strlen_zero(peer->accountcode))
- astman_append(s, "Accountcode: %s\r\n", peer->accountcode);
- astman_append(s, "AMAflags: %s\r\n", ast_channel_amaflags2string(peer->amaflags));
- astman_append(s, "CID-CallingPres: %s\r\n", ast_describe_caller_presentation(peer->callingpres));
- if (!ast_strlen_zero(peer->fromuser))
- astman_append(s, "SIP-FromUser: %s\r\n", peer->fromuser);
- if (!ast_strlen_zero(peer->fromdomain))
- astman_append(s, "SIP-FromDomain: %s\r\nSip-FromDomain-Port: %d\r\n", peer->fromdomain, (peer->fromdomainport) ? peer->fromdomainport : STANDARD_SIP_PORT);
- astman_append(s, "Callgroup: ");
- astman_append(s, "%s\r\n", ast_print_group(buffer, sizeof(buffer), peer->callgroup));
- astman_append(s, "Pickupgroup: ");
- astman_append(s, "%s\r\n", ast_print_group(buffer, sizeof(buffer), peer->pickupgroup));
- astman_append(s, "Named Callgroup: ");
- astman_append(s, "%s\r\n", ast_print_namedgroups(&tmp_str, peer->named_callgroups));
- ast_str_reset(tmp_str);
- astman_append(s, "Named Pickupgroup: ");
- astman_append(s, "%s\r\n", ast_print_namedgroups(&tmp_str, peer->named_pickupgroups));
- ast_str_reset(tmp_str);
- astman_append(s, "MOHSuggest: %s\r\n", peer->mohsuggest);
- peer_mailboxes_to_str(&tmp_str, peer);
- astman_append(s, "VoiceMailbox: %s\r\n", ast_str_buffer(tmp_str));
- astman_append(s, "TransferMode: %s\r\n", transfermode2str(peer->allowtransfer));
- astman_append(s, "LastMsgsSent: %d\r\n", peer->lastmsgssent);
- astman_append(s, "Maxforwards: %d\r\n", peer->maxforwards);
- astman_append(s, "Call-limit: %d\r\n", peer->call_limit);
- astman_append(s, "Busy-level: %d\r\n", peer->busy_level);
- astman_append(s, "MaxCallBR: %d kbps\r\n", peer->maxcallbitrate);
- astman_append(s, "Dynamic: %s\r\n", peer->host_dynamic?"Y":"N");
- astman_append(s, "Callerid: %s\r\n", ast_callerid_merge(cbuf, sizeof(cbuf), peer->cid_name, peer->cid_num, ""));
- astman_append(s, "RegExpire: %ld seconds\r\n", ast_sched_when(sched, peer->expire));
- astman_append(s, "SIP-AuthInsecure: %s\r\n", insecure2str(ast_test_flag(&peer->flags[0], SIP_INSECURE)));
- astman_append(s, "SIP-Forcerport: %s\r\n", ast_test_flag(&peer->flags[2], SIP_PAGE3_NAT_AUTO_RPORT) ?
- (ast_test_flag(&peer->flags[0], SIP_NAT_FORCE_RPORT) ? "A" : "a") :
- (ast_test_flag(&peer->flags[0], SIP_NAT_FORCE_RPORT) ? "Y" : "N"));
- astman_append(s, "SIP-Comedia: %s\r\n", ast_test_flag(&peer->flags[2], SIP_PAGE3_NAT_AUTO_COMEDIA) ?
- (ast_test_flag(&peer->flags[1], SIP_PAGE2_SYMMETRICRTP) ? "A" : "a") :
- (ast_test_flag(&peer->flags[1], SIP_PAGE2_SYMMETRICRTP) ? "Y" : "N"));
- astman_append(s, "ACL: %s\r\n", (ast_acl_list_is_empty(peer->acl) ? "N" : "Y"));
- astman_append(s, "SIP-CanReinvite: %s\r\n", (ast_test_flag(&peer->flags[0], SIP_DIRECT_MEDIA)?"Y":"N"));
- astman_append(s, "SIP-DirectMedia: %s\r\n", (ast_test_flag(&peer->flags[0], SIP_DIRECT_MEDIA)?"Y":"N"));
- astman_append(s, "SIP-PromiscRedir: %s\r\n", (ast_test_flag(&peer->flags[0], SIP_PROMISCREDIR)?"Y":"N"));
- astman_append(s, "SIP-UserPhone: %s\r\n", (ast_test_flag(&peer->flags[0], SIP_USEREQPHONE)?"Y":"N"));
- astman_append(s, "SIP-VideoSupport: %s\r\n", (ast_test_flag(&peer->flags[1], SIP_PAGE2_VIDEOSUPPORT)?"Y":"N"));
- astman_append(s, "SIP-TextSupport: %s\r\n", (ast_test_flag(&peer->flags[1], SIP_PAGE2_TEXTSUPPORT)?"Y":"N"));
- astman_append(s, "SIP-T.38Support: %s\r\n", (ast_test_flag(&peer->flags[1], SIP_PAGE2_T38SUPPORT)?"Y":"N"));
- astman_append(s, "SIP-T.38EC: %s\r\n", faxec2str(ast_test_flag(&peer->flags[1], SIP_PAGE2_T38SUPPORT)));
- astman_append(s, "SIP-T.38MaxDtgrm: %u\r\n", peer->t38_maxdatagram);
- astman_append(s, "SIP-Sess-Timers: %s\r\n", stmode2str(peer->stimer.st_mode_oper));
- astman_append(s, "SIP-Sess-Refresh: %s\r\n", strefresherparam2str(peer->stimer.st_ref));
- astman_append(s, "SIP-Sess-Expires: %d\r\n", peer->stimer.st_max_se);
- astman_append(s, "SIP-Sess-Min: %d\r\n", peer->stimer.st_min_se);
- astman_append(s, "SIP-RTP-Engine: %s\r\n", peer->engine);
- astman_append(s, "SIP-Encryption: %s\r\n", ast_test_flag(&peer->flags[1], SIP_PAGE2_USE_SRTP) ? "Y" : "N");
- astman_append(s, "SIP-RTCP-Mux: %s\r\n", ast_test_flag(&peer->flags[2], SIP_PAGE3_RTCP_MUX) ? "Y" : "N");
-
- /* - is enumerated */
- astman_append(s, "SIP-DTMFmode: %s\r\n", dtmfmode2str(ast_test_flag(&peer->flags[0], SIP_DTMF)));
- astman_append(s, "ToHost: %s\r\n", peer->tohost);
- astman_append(s, "Address-IP: %s\r\nAddress-Port: %d\r\n", ast_sockaddr_stringify_addr(&peer->addr), ast_sockaddr_port(&peer->addr));
- astman_append(s, "Default-addr-IP: %s\r\nDefault-addr-port: %d\r\n", ast_sockaddr_stringify_addr(&peer->defaddr), ast_sockaddr_port(&peer->defaddr));
- astman_append(s, "Default-Username: %s\r\n", peer->username);
- if (!ast_strlen_zero(sip_cfg.regcontext))
- astman_append(s, "RegExtension: %s\r\n", peer->regexten);
- astman_append(s, "Codecs: %s\r\n", ast_format_cap_get_names(peer->caps, &codec_buf));
- astman_append(s, "Status: ");
- peer_status(peer, status, sizeof(status));
- astman_append(s, "%s\r\n", status);
- astman_append(s, "SIP-Useragent: %s\r\n", peer->useragent);
- astman_append(s, "Reg-Contact: %s\r\n", peer->fullcontact);
- astman_append(s, "QualifyFreq: %d ms\r\n", peer->qualifyfreq);
- astman_append(s, "Parkinglot: %s\r\n", peer->parkinglot);
- if (peer->chanvars) {
- for (v = peer->chanvars ; v ; v = v->next) {
- astman_append(s, "ChanVariable: %s=%s\r\n", v->name, v->value);
- }
- }
- astman_append(s, "SIP-Use-Reason-Header: %s\r\n", (ast_test_flag(&peer->flags[1], SIP_PAGE2_Q850_REASON)) ? "Y" : "N");
- astman_append(s, "Description: %s\r\n", peer->description);
-
- peer = sip_unref_peer(peer, "sip_show_peer: sip_unref_peer: done with peer");
-
- } else {
- ast_cli(fd, "Peer %s not found.\n", argv[3]);
- ast_cli(fd, "\n");
- }
-
- return CLI_SUCCESS;
-}
-
-/*! \brief Do completion on user name */
-static char *complete_sip_user(const char *word, int state)
-{
- char *result = NULL;
- int wordlen = strlen(word);
- int which = 0;
- struct ao2_iterator user_iter;
- struct sip_peer *user;
-
- user_iter = ao2_iterator_init(peers, 0);
- while ((user = ao2_t_iterator_next(&user_iter, "iterate thru peers table"))) {
- ao2_lock(user);
- if (!(user->type & SIP_TYPE_USER)) {
- ao2_unlock(user);
- sip_unref_peer(user, "complete sip user");
- continue;
- }
- /* locking of the object is not required because only the name and flags are being compared */
- if (!strncasecmp(word, user->name, wordlen) && ++which > state) {
- result = ast_strdup(user->name);
- }
- ao2_unlock(user);
- sip_unref_peer(user, "complete sip user");
- if (result) {
- break;
- }
- }
- ao2_iterator_destroy(&user_iter);
- return result;
-}
-/*! \brief Support routine for 'sip show user' CLI */
-static char *complete_sip_show_user(const char *line, const char *word, int pos, int state)
-{
- if (pos == 3)
- return complete_sip_user(word, state);
-
- return NULL;
-}
-
-/*! \brief Show one user in detail */
-static char *sip_show_user(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
-{
- char cbuf[256];
- struct sip_peer *user;
- struct ast_variable *v;
- int load_realtime;
-
- switch (cmd) {
- case CLI_INIT:
- e->command = "sip show user";
- e->usage =
- "Usage: sip show user [load]\n"
- " Shows all details on one SIP user and the current status.\n"
- " Option \"load\" forces lookup of peer in realtime storage.\n";
- return NULL;
- case CLI_GENERATE:
- if (a->pos == 4) {
- static const char * const completions[] = { "load", NULL };
- return ast_cli_complete(a->word, completions, a->n);
- } else {
- return complete_sip_show_user(a->line, a->word, a->pos, a->n);
- }
- }
-
- if (a->argc < 4)
- return CLI_SHOWUSAGE;
-
- /* Load from realtime storage? */
- load_realtime = (a->argc == 5 && !strcmp(a->argv[4], "load")) ? TRUE : FALSE;
-
- if ((user = sip_find_peer(a->argv[3], NULL, load_realtime, FINDUSERS, FALSE, 0))) {
- ao2_lock(user);
- ast_cli(a->fd, "\n\n");
- ast_cli(a->fd, " * Name : %s\n", user->name);
- ast_cli(a->fd, " Secret : %s\n", ast_strlen_zero(user->secret)?"":"");
- ast_cli(a->fd, " MD5Secret : %s\n", ast_strlen_zero(user->md5secret)?"":"");
- ast_cli(a->fd, " Context : %s\n", user->context);
- ast_cli(a->fd, " Language : %s\n", user->language);
- if (!ast_strlen_zero(user->accountcode))
- ast_cli(a->fd, " Accountcode : %s\n", user->accountcode);
- ast_cli(a->fd, " AMA flags : %s\n", ast_channel_amaflags2string(user->amaflags));
- ast_cli(a->fd, " Tonezone : %s\n", user->zone[0] != '\0' ? user->zone : "");
- ast_cli(a->fd, " Transfer mode: %s\n", transfermode2str(user->allowtransfer));
- ast_cli(a->fd, " MaxCallBR : %d kbps\n", user->maxcallbitrate);
- ast_cli(a->fd, " CallingPres : %s\n", ast_describe_caller_presentation(user->callingpres));
- ast_cli(a->fd, " Call limit : %d\n", user->call_limit);
- ast_cli(a->fd, " Callgroup : ");
- print_group(a->fd, user->callgroup, 0);
- ast_cli(a->fd, " Pickupgroup : ");
- print_group(a->fd, user->pickupgroup, 0);
- ast_cli(a->fd, " Named Callgr : ");
- print_named_groups(a->fd, user->named_callgroups, 0);
- ast_cli(a->fd, " Nam. Pickupgr: ");
- print_named_groups(a->fd, user->named_pickupgroups, 0);
- ast_cli(a->fd, " Callerid : %s\n", ast_callerid_merge(cbuf, sizeof(cbuf), user->cid_name, user->cid_num, ""));
- ast_cli(a->fd, " ACL : %s\n", AST_CLI_YESNO(ast_acl_list_is_empty(user->acl) == 0));
- ast_cli(a->fd, " Sess-Timers : %s\n", stmode2str(user->stimer.st_mode_oper));
- ast_cli(a->fd, " Sess-Refresh : %s\n", strefresherparam2str(user->stimer.st_ref));
- ast_cli(a->fd, " Sess-Expires : %d secs\n", user->stimer.st_max_se);
- ast_cli(a->fd, " Sess-Min-SE : %d secs\n", user->stimer.st_min_se);
- ast_cli(a->fd, " RTP Engine : %s\n", user->engine);
-
- ast_cli(a->fd, " Auto-Framing: %s \n", AST_CLI_YESNO(user->autoframing));
- if (user->chanvars) {
- ast_cli(a->fd, " Variables :\n");
- for (v = user->chanvars ; v ; v = v->next)
- ast_cli(a->fd, " %s = %s\n", v->name, v->value);
- }
-
- ast_cli(a->fd, "\n");
-
- ao2_unlock(user);
- sip_unref_peer(user, "sip show user");
- } else {
- ast_cli(a->fd, "User %s not found.\n", a->argv[3]);
- ast_cli(a->fd, "\n");
- }
-
- return CLI_SUCCESS;
-}
-
-
-static char *sip_show_sched(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
-{
- struct ast_str *cbuf;
- struct ast_cb_names cbnames = {
- 10,
- {
- "retrans_pkt",
- "__sip_autodestruct",
- "expire_register",
- "auto_congest",
- "sip_reg_timeout",
- "sip_poke_peer_s",
- "sip_poke_peer_now",
- "sip_poke_noanswer",
- "sip_reregister",
- "sip_reinvite_retry"
- },
- {
- retrans_pkt,
- __sip_autodestruct,
- expire_register,
- auto_congest,
- sip_reg_timeout,
- sip_poke_peer_s,
- sip_poke_peer_now,
- sip_poke_noanswer,
- sip_reregister,
- sip_reinvite_retry
- }
- };
-
- switch (cmd) {
- case CLI_INIT:
- e->command = "sip show sched";
- e->usage =
- "Usage: sip show sched\n"
- " Shows stats on what's in the sched queue at the moment\n";
- return NULL;
- case CLI_GENERATE:
- return NULL;
- }
-
- cbuf = ast_str_alloca(2048);
-
- ast_cli(a->fd, "\n");
- ast_sched_report(sched, &cbuf, &cbnames);
- ast_cli(a->fd, "%s", ast_str_buffer(cbuf));
-
- return CLI_SUCCESS;
-}
-
-/*! \brief Show SIP Registry (registrations with other SIP proxies */
-static char *sip_show_registry(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
-{
-#define FORMAT2 "%-39.39s %-6.6s %-12.12s %8.8s %-20.20s %-25.25s\n"
-#define FORMAT "%-39.39s %-6.6s %-12.12s %8d %-20.20s %-25.25s\n"
- char host[80];
- char user[80];
- char tmpdat[256];
- struct ast_tm tm;
- int counter = 0;
- struct ao2_iterator iter;
- struct sip_registry *iterator;
-
- switch (cmd) {
- case CLI_INIT:
- e->command = "sip show registry";
- e->usage =
- "Usage: sip show registry\n"
- " Lists all registration requests and status.\n";
- return NULL;
- case CLI_GENERATE:
- return NULL;
- }
-
- if (a->argc != 3)
- return CLI_SHOWUSAGE;
- ast_cli(a->fd, FORMAT2, "Host", "dnsmgr", "Username", "Refresh", "State", "Reg.Time");
-
- iter = ao2_iterator_init(registry_list, 0);
- while ((iterator = ao2_t_iterator_next(&iter, "sip_show_registry iter"))) {
- ao2_lock(iterator);
-
- snprintf(host, sizeof(host), "%s:%d", iterator->hostname, iterator->portno ? iterator->portno : STANDARD_SIP_PORT);
- snprintf(user, sizeof(user), "%s", iterator->username);
- if (!ast_strlen_zero(iterator->regdomain)) {
- snprintf(tmpdat, sizeof(tmpdat), "%s", user);
- snprintf(user, sizeof(user), "%s@%s", tmpdat, iterator->regdomain);}
- if (iterator->regdomainport) {
- snprintf(tmpdat, sizeof(tmpdat), "%s", user);
- snprintf(user, sizeof(user), "%s:%d", tmpdat, iterator->regdomainport);}
- if (iterator->regtime.tv_sec) {
- ast_localtime(&iterator->regtime, &tm, NULL);
- ast_strftime(tmpdat, sizeof(tmpdat), "%a, %d %b %Y %T", &tm);
- } else
- tmpdat[0] = '\0';
- ast_cli(a->fd, FORMAT, host, (iterator->dnsmgr) ? "Y" : "N", user, iterator->refresh, regstate2str(iterator->regstate), tmpdat);
-
- ao2_unlock(iterator);
- ao2_t_ref(iterator, -1, "sip_show_registry iter");
- counter++;
- }
- ao2_iterator_destroy(&iter);
-
- ast_cli(a->fd, "%d SIP registrations.\n", counter);
- return CLI_SUCCESS;
-#undef FORMAT
-#undef FORMAT2
-}
-
-/*! \brief Unregister (force expiration) a SIP peer in the registry via CLI
- \note This function does not tell the SIP device what's going on,
- so use it with great care.
-*/
-static char *sip_unregister(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
-{
- struct sip_peer *peer;
- int load_realtime = 0;
-
- switch (cmd) {
- case CLI_INIT:
- e->command = "sip unregister";
- e->usage =
- "Usage: sip unregister \n"
- " Unregister (force expiration) a SIP peer from the registry\n";
- return NULL;
- case CLI_GENERATE:
- return complete_sip_unregister(a->line, a->word, a->pos, a->n);
- }
-
- if (a->argc != 3)
- return CLI_SHOWUSAGE;
-
- if ((peer = sip_find_peer(a->argv[2], NULL, load_realtime, FINDPEERS, TRUE, 0))) {
- if (peer->expire > -1) {
- AST_SCHED_DEL_UNREF(sched, peer->expire,
- sip_unref_peer(peer, "remove register expire ref"));
- expire_register(sip_ref_peer(peer, "ref for expire_register"));
- ast_cli(a->fd, "Unregistered peer \'%s\'\n\n", a->argv[2]);
- } else {
- ast_cli(a->fd, "Peer %s not registered\n", a->argv[2]);
- }
- sip_unref_peer(peer, "sip_unregister: sip_unref_peer via sip_unregister: done with peer from sip_find_peer call");
- } else {
- ast_cli(a->fd, "Peer unknown: \'%s\'. Not unregistered.\n", a->argv[2]);
- }
-
- return CLI_SUCCESS;
-}
-
-/*! \brief Callback for show_chanstats */
-static int show_chanstats_cb(struct sip_pvt *cur, struct __show_chan_arg *arg)
-{
-#define FORMAT2 "%-15.15s %-11.11s %-8.8s %-10.10s %-10.10s ( %%) %-6.6s %-10.10s %-10.10s ( %%) %-6.6s\n"
-#define FORMAT "%-15.15s %-11.11s %-8.8s %-10.10u%-1.1s %-10.10u (%5.2f%%) %-6.4lf %-10.10u%-1.1s %-10.10u (%5.2f%%) %-6.4lf\n"
- struct ast_rtp_instance_stats stats;
- char durbuf[10];
- struct ast_channel *c;
- int fd = arg->fd;
-
- sip_pvt_lock(cur);
- c = cur->owner;
-
- if (cur->subscribed != NONE) {
- /* Subscriptions */
- sip_pvt_unlock(cur);
- return 0; /* don't care, we scan all channels */
- }
-
- if (!cur->rtp) {
- if (sipdebug) {
- ast_cli(fd, "%-15.15s %-11.11s (inv state: %s) -- %s\n",
- ast_sockaddr_stringify_addr(&cur->sa), cur->callid,
- invitestate2string[cur->invitestate].desc,
- "-- No RTP active");
- }
- sip_pvt_unlock(cur);
- return 0; /* don't care, we scan all channels */
- }
-
- if (ast_rtp_instance_get_stats(cur->rtp, &stats, AST_RTP_INSTANCE_STAT_ALL)) {
- sip_pvt_unlock(cur);
- ast_log(LOG_WARNING, "Could not get RTP stats.\n");
- return 0;
- }
-
- if (c) {
- ast_format_duration_hh_mm_ss(ast_channel_get_duration(c), durbuf, sizeof(durbuf));
- } else {
- durbuf[0] = '\0';
- }
-
- ast_cli(fd, FORMAT,
- ast_sockaddr_stringify_addr(&cur->sa),
- cur->callid,
- durbuf,
- stats.rxcount > (unsigned int) 100000 ? (unsigned int) (stats.rxcount)/(unsigned int) 1000 : stats.rxcount,
- stats.rxcount > (unsigned int) 100000 ? "K":" ",
- stats.rxploss,
- (stats.rxcount + stats.rxploss) > 0 ? (double) stats.rxploss / (stats.rxcount + stats.rxploss) * 100 : 0,
- stats.rxjitter,
- stats.txcount > (unsigned int) 100000 ? (unsigned int) (stats.txcount)/(unsigned int) 1000 : stats.txcount,
- stats.txcount > (unsigned int) 100000 ? "K":" ",
- stats.txploss,
- stats.txcount > 0 ? (double) stats.txploss / stats.txcount * 100 : 0,
- stats.txjitter
- );
- arg->numchans++;
- sip_pvt_unlock(cur);
-
- return 0; /* don't care, we scan all channels */
-}
-
-/*! \brief SIP show channelstats CLI (main function) */
-static char *sip_show_channelstats(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
-{
- struct __show_chan_arg arg = { .fd = a->fd, .numchans = 0 };
- struct sip_pvt *cur;
- struct ao2_iterator i;
-
- switch (cmd) {
- case CLI_INIT:
- e->command = "sip show channelstats";
- e->usage =
- "Usage: sip show channelstats\n"
- " Lists all currently active SIP channel's RTCP statistics.\n"
- " Note that calls in the much optimized RTP P2P bridge mode will not show any packets here.";
- return NULL;
- case CLI_GENERATE:
- return NULL;
- }
-
- if (a->argc != 3)
- return CLI_SHOWUSAGE;
-
- ast_cli(a->fd, FORMAT2, "Peer", "Call ID", "Duration", "Recv: Pack", "Lost", "Jitter", "Send: Pack", "Lost", "Jitter");
-
- /* iterate on the container and invoke the callback on each item */
- i = ao2_iterator_init(dialogs, 0);
- for (; (cur = ao2_iterator_next(&i)); ao2_ref(cur, -1)) {
- show_chanstats_cb(cur, &arg);
- }
- ao2_iterator_destroy(&i);
-
- ast_cli(a->fd, "%d active SIP channel%s\n", arg.numchans, (arg.numchans != 1) ? "s" : "");
- return CLI_SUCCESS;
-}
-#undef FORMAT
-#undef FORMAT2
-
-/*! \brief List global settings for the SIP channel */
-static char *sip_show_settings(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
-{
- int realtimepeers;
- int realtimeregs;
- struct ast_str *codec_buf = ast_str_alloca(AST_FORMAT_CAP_NAMES_LEN);
- const char *msg; /* temporary msg pointer */
- struct sip_auth_container *credentials;
-
- switch (cmd) {
- case CLI_INIT:
- e->command = "sip show settings";
- e->usage =
- "Usage: sip show settings\n"
- " Provides detailed list of the configuration of the SIP channel.\n";
- return NULL;
- case CLI_GENERATE:
- return NULL;
- }
-
- if (a->argc != 3)
- return CLI_SHOWUSAGE;
-
- realtimepeers = ast_check_realtime("sippeers");
- realtimeregs = ast_check_realtime("sipregs");
-
- ast_mutex_lock(&authl_lock);
- credentials = authl;
- if (credentials) {
- ao2_t_ref(credentials, +1, "Ref global auth for show");
- }
- ast_mutex_unlock(&authl_lock);
-
- ast_cli(a->fd, "\n\nGlobal Settings:\n");
- ast_cli(a->fd, "----------------\n");
- ast_cli(a->fd, " UDP Bindaddress: %s\n", ast_sockaddr_stringify(&bindaddr));
- if (ast_sockaddr_is_ipv6(&bindaddr) && ast_sockaddr_is_any(&bindaddr)) {
- ast_cli(a->fd, " ** Additional Info:\n");
- ast_cli(a->fd, " [::] may include IPv4 in addition to IPv6, if such a feature is enabled in the OS.\n");
- }
- ast_cli(a->fd, " TCP SIP Bindaddress: %s\n",
- sip_cfg.tcp_enabled != FALSE ?
- ast_sockaddr_stringify(&sip_tcp_desc.local_address) :
- "Disabled");
- ast_cli(a->fd, " TLS SIP Bindaddress: %s\n",
- default_tls_cfg.enabled != FALSE ?
- ast_sockaddr_stringify(&sip_tls_desc.local_address) :
- "Disabled");
- ast_cli(a->fd, " RTP Bindaddress: %s\n",
- !ast_sockaddr_isnull(&rtpbindaddr) ?
- ast_sockaddr_stringify_addr(&rtpbindaddr) :
- "Disabled");
- ast_cli(a->fd, " Videosupport: %s\n", AST_CLI_YESNO(ast_test_flag(&global_flags[1], SIP_PAGE2_VIDEOSUPPORT)));
- ast_cli(a->fd, " Textsupport: %s\n", AST_CLI_YESNO(ast_test_flag(&global_flags[1], SIP_PAGE2_TEXTSUPPORT)));
- ast_cli(a->fd, " Ignore SDP sess. ver.: %s\n", AST_CLI_YESNO(ast_test_flag(&global_flags[1], SIP_PAGE2_IGNORESDPVERSION)));
- ast_cli(a->fd, " AutoCreate Peer: %s\n", autocreatepeer2str(sip_cfg.autocreatepeer));
- ast_cli(a->fd, " Match Auth Username: %s\n", AST_CLI_YESNO(global_match_auth_username));
- ast_cli(a->fd, " Allow unknown access: %s\n", AST_CLI_YESNO(sip_cfg.allowguest));
- ast_cli(a->fd, " Allow subscriptions: %s\n", AST_CLI_YESNO(ast_test_flag(&global_flags[1], SIP_PAGE2_ALLOWSUBSCRIBE)));
- ast_cli(a->fd, " Allow overlap dialing: %s\n", allowoverlap2str(ast_test_flag(&global_flags[1], SIP_PAGE2_ALLOWOVERLAP)));
- ast_cli(a->fd, " Allow promisc. redir: %s\n", AST_CLI_YESNO(ast_test_flag(&global_flags[0], SIP_PROMISCREDIR)));
- ast_cli(a->fd, " Enable call counters: %s\n", AST_CLI_YESNO(global_callcounter));
- ast_cli(a->fd, " SIP domain support: %s\n", AST_CLI_YESNO(!AST_LIST_EMPTY(&domain_list)));
- ast_cli(a->fd, " Path support : %s\n", AST_CLI_YESNO(ast_test_flag(&global_flags[0], SIP_USEPATH)));
- ast_cli(a->fd, " Realm. auth: %s\n", AST_CLI_YESNO(credentials != NULL));
- if (credentials) {
- struct sip_auth *auth;
-
- AST_LIST_TRAVERSE(&credentials->list, auth, node) {
- ast_cli(a->fd, " Realm. auth entry: Realm %-15.15s User %-10.20s %s\n",
- auth->realm,
- auth->username,
- !ast_strlen_zero(auth->secret)
- ? ""
- : (!ast_strlen_zero(auth->md5secret)
- ? "" : ""));
- }
- ao2_t_ref(credentials, -1, "Unref global auth for show");
- }
- ast_cli(a->fd, " Our auth realm %s\n", sip_cfg.realm);
- ast_cli(a->fd, " Use domains as realms: %s\n", AST_CLI_YESNO(sip_cfg.domainsasrealm));
- ast_cli(a->fd, " Call to non-local dom.: %s\n", AST_CLI_YESNO(sip_cfg.allow_external_domains));
- ast_cli(a->fd, " URI user is phone no: %s\n", AST_CLI_YESNO(ast_test_flag(&global_flags[0], SIP_USEREQPHONE)));
- ast_cli(a->fd, " Always auth rejects: %s\n", AST_CLI_YESNO(sip_cfg.alwaysauthreject));
- ast_cli(a->fd, " Direct RTP setup: %s\n", AST_CLI_YESNO(sip_cfg.directrtpsetup));
- ast_cli(a->fd, " User Agent: %s\n", global_useragent);
- ast_cli(a->fd, " SDP Session Name: %s\n", ast_strlen_zero(global_sdpsession) ? "-" : global_sdpsession);
- ast_cli(a->fd, " SDP Owner Name: %s\n", ast_strlen_zero(global_sdpowner) ? "-" : global_sdpowner);
- ast_cli(a->fd, " Reg. context: %s\n", S_OR(sip_cfg.regcontext, "(not set)"));
- ast_cli(a->fd, " Regexten on Qualify: %s\n", AST_CLI_YESNO(sip_cfg.regextenonqualify));
- ast_cli(a->fd, " Trust RPID: %s\n", AST_CLI_YESNO(ast_test_flag(&global_flags[0], SIP_TRUSTRPID)));
- ast_cli(a->fd, " Send RPID: %s\n", AST_CLI_YESNO(ast_test_flag(&global_flags[0], SIP_SENDRPID)));
- ast_cli(a->fd, " Legacy userfield parse: %s\n", AST_CLI_YESNO(sip_cfg.legacy_useroption_parsing));
- ast_cli(a->fd, " Send Diversion: %s\n", AST_CLI_YESNO(sip_cfg.send_diversion));
- ast_cli(a->fd, " Caller ID: %s\n", default_callerid);
- if ((default_fromdomainport) && (default_fromdomainport != STANDARD_SIP_PORT)) {
- ast_cli(a->fd, " From: Domain: %s:%d\n", default_fromdomain, default_fromdomainport);
- } else {
- ast_cli(a->fd, " From: Domain: %s\n", default_fromdomain);
- }
- ast_cli(a->fd, " Record SIP history: %s\n", AST_CLI_ONOFF(recordhistory));
- ast_cli(a->fd, " Auth. Failure Events: %s\n", AST_CLI_ONOFF(global_authfailureevents));
-
- ast_cli(a->fd, " T.38 support: %s\n", AST_CLI_YESNO(ast_test_flag(&global_flags[1], SIP_PAGE2_T38SUPPORT)));
- ast_cli(a->fd, " T.38 EC mode: %s\n", faxec2str(ast_test_flag(&global_flags[1], SIP_PAGE2_T38SUPPORT)));
- ast_cli(a->fd, " T.38 MaxDtgrm: %u\n", global_t38_maxdatagram);
- if (!realtimepeers && !realtimeregs)
- ast_cli(a->fd, " SIP realtime: Disabled\n" );
- else
- ast_cli(a->fd, " SIP realtime: Enabled\n" );
- ast_cli(a->fd, " Qualify Freq : %d ms\n", global_qualifyfreq);
- ast_cli(a->fd, " Q.850 Reason header: %s\n", AST_CLI_YESNO(ast_test_flag(&global_flags[1], SIP_PAGE2_Q850_REASON)));
- ast_cli(a->fd, " Store SIP_CAUSE: %s\n", AST_CLI_YESNO(global_store_sip_cause));
- ast_cli(a->fd, "\nNetwork QoS Settings:\n");
- ast_cli(a->fd, "---------------------------\n");
- ast_cli(a->fd, " IP ToS SIP: %s\n", ast_tos2str(global_tos_sip));
- ast_cli(a->fd, " IP ToS RTP audio: %s\n", ast_tos2str(global_tos_audio));
- ast_cli(a->fd, " IP ToS RTP video: %s\n", ast_tos2str(global_tos_video));
- ast_cli(a->fd, " IP ToS RTP text: %s\n", ast_tos2str(global_tos_text));
- ast_cli(a->fd, " 802.1p CoS SIP: %u\n", global_cos_sip);
- ast_cli(a->fd, " 802.1p CoS RTP audio: %u\n", global_cos_audio);
- ast_cli(a->fd, " 802.1p CoS RTP video: %u\n", global_cos_video);
- ast_cli(a->fd, " 802.1p CoS RTP text: %u\n", global_cos_text);
- ast_cli(a->fd, " Jitterbuffer enabled: %s\n", AST_CLI_YESNO(ast_test_flag(&global_jbconf, AST_JB_ENABLED)));
- if (ast_test_flag(&global_jbconf, AST_JB_ENABLED)) {
- ast_cli(a->fd, " Jitterbuffer forced: %s\n", AST_CLI_YESNO(ast_test_flag(&global_jbconf, AST_JB_FORCED)));
- ast_cli(a->fd, " Jitterbuffer max size: %ld\n", global_jbconf.max_size);
- ast_cli(a->fd, " Jitterbuffer resync: %ld\n", global_jbconf.resync_threshold);
- ast_cli(a->fd, " Jitterbuffer impl: %s\n", global_jbconf.impl);
- if (!strcasecmp(global_jbconf.impl, "adaptive")) {
- ast_cli(a->fd, " Jitterbuffer tgt extra: %ld\n", global_jbconf.target_extra);
- }
- ast_cli(a->fd, " Jitterbuffer log: %s\n", AST_CLI_YESNO(ast_test_flag(&global_jbconf, AST_JB_LOG)));
- }
-
- ast_cli(a->fd, "\nNetwork Settings:\n");
- ast_cli(a->fd, "---------------------------\n");
- /* determine if/how SIP address can be remapped */
- if (localaddr == NULL)
- msg = "Disabled, no localnet list";
- else if (ast_sockaddr_isnull(&externaddr))
- msg = "Disabled";
- else if (!ast_strlen_zero(externhost))
- msg = "Enabled using externhost";
- else
- msg = "Enabled using externaddr";
- ast_cli(a->fd, " SIP address remapping: %s\n", msg);
- ast_cli(a->fd, " Externhost: %s\n", S_OR(externhost, ""));
- ast_cli(a->fd, " Externaddr: %s\n", ast_sockaddr_stringify(&externaddr));
- ast_cli(a->fd, " Externrefresh: %d\n", externrefresh);
- {
- struct ast_ha *d;
- const char *prefix = "Localnet:";
-
- for (d = localaddr; d ; prefix = "", d = d->next) {
- const char *addr = ast_strdupa(ast_sockaddr_stringify_addr(&d->addr));
- const char *mask = ast_strdupa(ast_sockaddr_stringify_addr(&d->netmask));
- ast_cli(a->fd, " %-24s%s/%s\n", prefix, addr, mask);
- }
- }
- ast_cli(a->fd, "\nGlobal Signalling Settings:\n");
- ast_cli(a->fd, "---------------------------\n");
- ast_cli(a->fd, " Codecs: %s\n", ast_format_cap_get_names(sip_cfg.caps, &codec_buf));
- ast_cli(a->fd, " Relax DTMF: %s\n", AST_CLI_YESNO(global_relaxdtmf));
- ast_cli(a->fd, " RFC2833 Compensation: %s\n", AST_CLI_YESNO(ast_test_flag(&global_flags[1], SIP_PAGE2_RFC2833_COMPENSATE)));
- ast_cli(a->fd, " Symmetric RTP: %s\n", comedia_string(global_flags));
- ast_cli(a->fd, " Compact SIP headers: %s\n", AST_CLI_YESNO(sip_cfg.compactheaders));
- ast_cli(a->fd, " RTP Keepalive: %d %s\n", global_rtpkeepalive, global_rtpkeepalive ? "" : "(Disabled)" );
- ast_cli(a->fd, " RTP Timeout: %d %s\n", global_rtptimeout, global_rtptimeout ? "" : "(Disabled)" );
- ast_cli(a->fd, " RTP Hold Timeout: %d %s\n", global_rtpholdtimeout, global_rtpholdtimeout ? "" : "(Disabled)");
- ast_cli(a->fd, " MWI NOTIFY mime type: %s\n", default_notifymime);
- ast_cli(a->fd, " DNS SRV lookup: %s\n", AST_CLI_YESNO(sip_cfg.srvlookup));
- ast_cli(a->fd, " Pedantic SIP support: %s\n", AST_CLI_YESNO(sip_cfg.pedanticsipchecking));
- ast_cli(a->fd, " Reg. min duration %d secs\n", min_expiry);
- ast_cli(a->fd, " Reg. max duration: %d secs\n", max_expiry);
- ast_cli(a->fd, " Reg. default duration: %d secs\n", default_expiry);
- ast_cli(a->fd, " Sub. min duration %d secs\n", min_subexpiry);
- ast_cli(a->fd, " Sub. max duration: %d secs\n", max_subexpiry);
- ast_cli(a->fd, " Outbound reg. timeout: %d secs\n", global_reg_timeout);
- ast_cli(a->fd, " Outbound reg. attempts: %d\n", global_regattempts_max);
- ast_cli(a->fd, " Outbound reg. retry 403:%s\n", AST_CLI_YESNO(global_reg_retry_403));
- ast_cli(a->fd, " Notify ringing state: %s%s\n", AST_CLI_YESNO(sip_cfg.notifyringing), sip_cfg.notifyringing == NOTIFYRINGING_NOTINUSE ? " (when not in use)" : "");
- if (sip_cfg.notifyringing) {
- ast_cli(a->fd, " Include CID: %s%s\n",
- AST_CLI_YESNO(sip_cfg.notifycid),
- sip_cfg.notifycid == IGNORE_CONTEXT ? " (Ignoring context)" : "");
- }
- ast_cli(a->fd, " Notify hold state: %s\n", AST_CLI_YESNO(sip_cfg.notifyhold));
- ast_cli(a->fd, " SIP Transfer mode: %s\n", transfermode2str(sip_cfg.allowtransfer));
- ast_cli(a->fd, " Max Call Bitrate: %d kbps\n", default_maxcallbitrate);
- ast_cli(a->fd, " Auto-Framing: %s\n", AST_CLI_YESNO(global_autoframing));
- ast_cli(a->fd, " Outb. proxy: %s %s\n", ast_strlen_zero(sip_cfg.outboundproxy.name) ? "" : sip_cfg.outboundproxy.name,
- sip_cfg.outboundproxy.force ? "(forced)" : "");
- ast_cli(a->fd, " Session Timers: %s\n", stmode2str(global_st_mode));
- ast_cli(a->fd, " Session Refresher: %s\n", strefresherparam2str(global_st_refresher));
- ast_cli(a->fd, " Session Expires: %d secs\n", global_max_se);
- ast_cli(a->fd, " Session Min-SE: %d secs\n", global_min_se);
- ast_cli(a->fd, " Timer T1: %d\n", global_t1);
- ast_cli(a->fd, " Timer T1 minimum: %d\n", global_t1min);
- ast_cli(a->fd, " Timer B: %d\n", global_timer_b);
- ast_cli(a->fd, " No premature media: %s\n", AST_CLI_YESNO(global_prematuremediafilter));
- ast_cli(a->fd, " Max forwards: %d\n", sip_cfg.default_max_forwards);
-
- ast_cli(a->fd, "\nDefault Settings:\n");
- ast_cli(a->fd, "-----------------\n");
- ast_cli(a->fd, " Allowed transports: %s\n", get_transport_list(default_transports));
- ast_cli(a->fd, " Outbound transport: %s\n", sip_get_transport(default_primary_transport));
- ast_cli(a->fd, " Context: %s\n", sip_cfg.default_context);
- ast_cli(a->fd, " Record on feature: %s\n", sip_cfg.default_record_on_feature);
- ast_cli(a->fd, " Record off feature: %s\n", sip_cfg.default_record_off_feature);
- ast_cli(a->fd, " Force rport: %s\n", force_rport_string(global_flags));
- ast_cli(a->fd, " DTMF: %s\n", dtmfmode2str(ast_test_flag(&global_flags[0], SIP_DTMF)));
- ast_cli(a->fd, " Qualify: %d\n", default_qualify);
- ast_cli(a->fd, " Keepalive: %d\n", default_keepalive);
- ast_cli(a->fd, " Use ClientCode: %s\n", AST_CLI_YESNO(ast_test_flag(&global_flags[0], SIP_USECLIENTCODE)));
- ast_cli(a->fd, " Progress inband: %s\n", (ast_test_flag(&global_flags[0], SIP_PROG_INBAND) == SIP_PROG_INBAND_NEVER) ? "Never" : (AST_CLI_YESNO(ast_test_flag(&global_flags[0], SIP_PROG_INBAND) != SIP_PROG_INBAND_NO)));
- ast_cli(a->fd, " Language: %s\n", default_language);
- ast_cli(a->fd, " Tone zone: %s\n", default_zone[0] != '\0' ? default_zone : "");
- ast_cli(a->fd, " MOH Interpret: %s\n", default_mohinterpret);
- ast_cli(a->fd, " MOH Suggest: %s\n", default_mohsuggest);
- ast_cli(a->fd, " Voice Mail Extension: %s\n", default_vmexten);
- ast_cli(a->fd, " RTCP Multiplexing: %s\n", AST_CLI_YESNO(ast_test_flag(&global_flags[2], SIP_PAGE3_RTCP_MUX)));
-
-
- if (realtimepeers || realtimeregs) {
- ast_cli(a->fd, "\nRealtime SIP Settings:\n");
- ast_cli(a->fd, "----------------------\n");
- ast_cli(a->fd, " Realtime Peers: %s\n", AST_CLI_YESNO(realtimepeers));
- ast_cli(a->fd, " Realtime Regs: %s\n", AST_CLI_YESNO(realtimeregs));
- ast_cli(a->fd, " Cache Friends: %s\n", AST_CLI_YESNO(ast_test_flag(&global_flags[1], SIP_PAGE2_RTCACHEFRIENDS)));
- ast_cli(a->fd, " Update: %s\n", AST_CLI_YESNO(sip_cfg.peer_rtupdate));
- ast_cli(a->fd, " Ignore Reg. Expire: %s\n", AST_CLI_YESNO(sip_cfg.ignore_regexpire));
- ast_cli(a->fd, " Save sys. name: %s\n", AST_CLI_YESNO(sip_cfg.rtsave_sysname));
- ast_cli(a->fd, " Save path header: %s\n", AST_CLI_YESNO(sip_cfg.rtsave_path));
- ast_cli(a->fd, " Auto Clear: %d (%s)\n", sip_cfg.rtautoclear, ast_test_flag(&global_flags[1], SIP_PAGE2_RTAUTOCLEAR) ? "Enabled" : "Disabled");
- }
- ast_cli(a->fd, "\n----\n");
- return CLI_SUCCESS;
-}
-
-static char *sip_show_mwi(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
-{
-#define FORMAT "%-30.30s %-12.12s %-10.10s %-10.10s\n"
- char host[80];
- struct ao2_iterator iter;
- struct sip_subscription_mwi *iterator;
-
- switch (cmd) {
- case CLI_INIT:
- e->command = "sip show mwi";
- e->usage =
- "Usage: sip show mwi\n"
- " Provides a list of MWI subscriptions and status.\n";
- return NULL;
- case CLI_GENERATE:
- return NULL;
- }
-
- ast_cli(a->fd, FORMAT, "Host", "Username", "Mailbox", "Subscribed");
-
- iter = ao2_iterator_init(subscription_mwi_list, 0);
- while ((iterator = ao2_t_iterator_next(&iter, "sip_show_mwi iter"))) {
- ao2_lock(iterator);
- snprintf(host, sizeof(host), "%s:%d", iterator->hostname, iterator->portno ? iterator->portno : STANDARD_SIP_PORT);
- ast_cli(a->fd, FORMAT, host, iterator->username, iterator->mailbox, AST_CLI_YESNO(iterator->subscribed));
- ao2_unlock(iterator);
- ao2_t_ref(iterator, -1, "sip_show_mwi iter");
- }
- ao2_iterator_destroy(&iter);
-
- return CLI_SUCCESS;
-#undef FORMAT
-}
-
-
-/*! \brief Show subscription type in string format */
-static const char *subscription_type2str(enum subscriptiontype subtype)
-{
- int i;
-
- for (i = 1; i < ARRAY_LEN(subscription_types); i++) {
- if (subscription_types[i].type == subtype) {
- return subscription_types[i].text;
- }
- }
- return subscription_types[0].text;
-}
-
-/*! \brief Find subscription type in array */
-static const struct cfsubscription_types *find_subscription_type(enum subscriptiontype subtype)
-{
- int i;
-
- for (i = 1; i < ARRAY_LEN(subscription_types); i++) {
- if (subscription_types[i].type == subtype) {
- return &subscription_types[i];
- }
- }
- return &subscription_types[0];
-}
-
-/*
- * We try to structure all functions that loop on data structures as
- * a handler for individual entries, and a mainloop that iterates
- * on the main data structure. This way, moving the code to containers
- * that support iteration through callbacks will be a lot easier.
- */
-
-#define FORMAT4 "%-15.15s %-15.15s %-15.15s %-15.15s %-13.13s %-15.15s %-10.10s %-6.6d\n"
-#define FORMAT3 "%-15.15s %-15.15s %-15.15s %-15.15s %-13.13s %-15.15s %-10.10s %-6.6s\n"
-#define FORMAT2 "%-15.15s %-15.15s %-15.15s %-15.15s %-7.7s %-15.15s %-10.10s %-10.10s\n"
-#define FORMAT "%-15.15s %-15.15s %-15.15s %-15.15s %-3.3s %-3.3s %-15.15s %-10.10s %-10.10s\n"
-
-/*! \brief callback for show channel|subscription */
-static int show_channels_cb(struct sip_pvt *cur, struct __show_chan_arg *arg)
-{
- const struct ast_sockaddr *dst;
-
- sip_pvt_lock(cur);
- dst = sip_real_dst(cur);
-
- /* XXX indentation preserved to reduce diff. Will be fixed later */
- if (cur->subscribed == NONE && !arg->subscriptions) {
- /* set if SIP transfer in progress */
- const char *referstatus = cur->refer ? referstatus2str(cur->refer->status) : "";
- struct ast_str *codec_buf = ast_str_alloca(AST_FORMAT_CAP_NAMES_LEN);
-
- ast_cli(arg->fd, FORMAT, ast_sockaddr_stringify_addr(dst),
- S_OR(cur->username, S_OR(cur->cid_num, "(None)")),
- cur->callid,
- cur->owner ? ast_format_cap_get_names(ast_channel_nativeformats(cur->owner), &codec_buf) : "(nothing)",
- AST_CLI_YESNO(ast_test_flag(&cur->flags[1], SIP_PAGE2_CALL_ONHOLD)),
- cur->needdestroy ? "(d)" : "",
- cur->lastmsg ,
- referstatus,
- cur->relatedpeer ? cur->relatedpeer->name : ""
- );
- arg->numchans++;
- }
- if (cur->subscribed != NONE && arg->subscriptions) {
- struct ast_str *mailbox_str = ast_str_alloca(512);
- if (cur->subscribed == MWI_NOTIFICATION && cur->relatedpeer)
- peer_mailboxes_to_str(&mailbox_str, cur->relatedpeer);
- ast_cli(arg->fd, FORMAT4, ast_sockaddr_stringify_addr(dst),
- S_OR(cur->username, S_OR(cur->cid_num, "(None)")),
- cur->callid,
- /* the 'complete' exten/context is hidden in the refer_to field for subscriptions */
- cur->subscribed == MWI_NOTIFICATION ? "--" : cur->subscribeuri,
- cur->subscribed == MWI_NOTIFICATION ? "" : ast_extension_state2str(cur->laststate),
- subscription_type2str(cur->subscribed),
- cur->subscribed == MWI_NOTIFICATION ? S_OR(ast_str_buffer(mailbox_str), "") : "",
- cur->expiry
- );
- arg->numchans++;
- }
- sip_pvt_unlock(cur);
- return 0; /* don't care, we scan all channels */
-}
-
-/*! \brief CLI for show channels or subscriptions.
- * This is a new-style CLI handler so a single function contains
- * the prototype for the function, the 'generator' to produce multiple
- * entries in case it is required, and the actual handler for the command.
- */
-static char *sip_show_channels(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
-{
- struct __show_chan_arg arg = { .fd = a->fd, .numchans = 0 };
- struct sip_pvt *cur;
- struct ao2_iterator i;
-
- if (cmd == CLI_INIT) {
- e->command = "sip show {channels|subscriptions}";
- e->usage =
- "Usage: sip show channels\n"
- " Lists all currently active SIP calls (dialogs).\n"
- "Usage: sip show subscriptions\n"
- " Lists active SIP subscriptions.\n";
- return NULL;
- } else if (cmd == CLI_GENERATE)
- return NULL;
-
- if (a->argc != e->args)
- return CLI_SHOWUSAGE;
- arg.subscriptions = !strcasecmp(a->argv[e->args - 1], "subscriptions");
- if (!arg.subscriptions)
- ast_cli(arg.fd, FORMAT2, "Peer", "User/ANR", "Call ID", "Format", "Hold", "Last Message", "Expiry", "Peer");
- else
- ast_cli(arg.fd, FORMAT3, "Peer", "User", "Call ID", "Extension", "Last state", "Type", "Mailbox", "Expiry");
-
- /* iterate on the container and invoke the callback on each item */
- i = ao2_iterator_init(dialogs, 0);
- for (; (cur = ao2_iterator_next(&i)); ao2_ref(cur, -1)) {
- show_channels_cb(cur, &arg);
- }
- ao2_iterator_destroy(&i);
-
- /* print summary information */
- ast_cli(arg.fd, "%d active SIP %s%s\n", arg.numchans,
- (arg.subscriptions ? "subscription" : "dialog"),
- ESS(arg.numchans)); /* ESS(n) returns an "s" if n>1 */
- return CLI_SUCCESS;
-#undef FORMAT
-#undef FORMAT2
-#undef FORMAT3
-}
-
-/*! \brief Support routine for 'sip show channel' and 'sip show history' CLI
- * This is in charge of generating all strings that match a prefix in the
- * given position. As many functions of this kind, each invokation has
- * O(state) time complexity so be careful in using it.
- */
-static char *complete_sipch(const char *line, const char *word, int pos, int state)
-{
- int which=0;
- struct sip_pvt *cur;
- char *c = NULL;
- int wordlen = strlen(word);
- struct ao2_iterator i;
-
- if (pos != 3) {
- return NULL;
- }
-
- i = ao2_iterator_init(dialogs, 0);
- while ((cur = ao2_t_iterator_next(&i, "iterate thru dialogs"))) {
- sip_pvt_lock(cur);
- if (!strncasecmp(word, cur->callid, wordlen) && ++which > state) {
- c = ast_strdup(cur->callid);
- sip_pvt_unlock(cur);
- dialog_unref(cur, "drop ref in iterator loop break");
- break;
- }
- sip_pvt_unlock(cur);
- dialog_unref(cur, "drop ref in iterator loop");
- }
- ao2_iterator_destroy(&i);
- return c;
-}
-
-
-/*! \brief Do completion on peer name */
-static char *complete_sip_peer(const char *word, int state, int flags2)
-{
- char *result = NULL;
- int wordlen = strlen(word);
- int which = 0;
- struct ao2_iterator i = ao2_iterator_init(peers, 0);
- struct sip_peer *peer;
-
- while ((peer = ao2_t_iterator_next(&i, "iterate thru peers table"))) {
- /* locking of the object is not required because only the name and flags are being compared */
- if (!strncasecmp(word, peer->name, wordlen) &&
- (!flags2 || ast_test_flag(&peer->flags[1], flags2)) &&
- ++which > state)
- result = ast_strdup(peer->name);
- sip_unref_peer(peer, "toss iterator peer ptr before break");
- if (result) {
- break;
- }
- }
- ao2_iterator_destroy(&i);
- return result;
-}
-
-/*! \brief Do completion on registered peer name */
-static char *complete_sip_registered_peer(const char *word, int state, int flags2)
-{
- char *result = NULL;
- int wordlen = strlen(word);
- int which = 0;
- struct ao2_iterator i;
- struct sip_peer *peer;
-
- i = ao2_iterator_init(peers, 0);
- while ((peer = ao2_t_iterator_next(&i, "iterate thru peers table"))) {
- if (!strncasecmp(word, peer->name, wordlen) &&
- (!flags2 || ast_test_flag(&peer->flags[1], flags2)) &&
- ++which > state && peer->expire > -1)
- result = ast_strdup(peer->name);
- if (result) {
- sip_unref_peer(peer, "toss iterator peer ptr before break");
- break;
- }
- sip_unref_peer(peer, "toss iterator peer ptr");
- }
- ao2_iterator_destroy(&i);
- return result;
-}
-
-/*! \brief Support routine for 'sip show history' CLI */
-static char *complete_sip_show_history(const char *line, const char *word, int pos, int state)
-{
- if (pos == 3)
- return complete_sipch(line, word, pos, state);
-
- return NULL;
-}
-
-/*! \brief Support routine for 'sip show peer' CLI */
-static char *complete_sip_show_peer(const char *line, const char *word, int pos, int state)
-{
- if (pos == 3) {
- return complete_sip_peer(word, state, 0);
- }
-
- return NULL;
-}
-
-/*! \brief Support routine for 'sip unregister' CLI */
-static char *complete_sip_unregister(const char *line, const char *word, int pos, int state)
-{
- if (pos == 2)
- return complete_sip_registered_peer(word, state, 0);
-
- return NULL;
-}
-
-/*! \brief Support routine for 'sip notify' CLI */
-static char *complete_sip_notify(const char *line, const char *word, int pos, int state)
-{
- char *c = NULL;
-
- if (pos == 2) {
- int which = 0;
- char *cat = NULL;
- int wordlen = strlen(word);
-
- /* do completion for notify type */
-
- if (!notify_types)
- return NULL;
-
- while ( (cat = ast_category_browse(notify_types, cat)) ) {
- if (!strncasecmp(word, cat, wordlen) && ++which > state) {
- c = ast_strdup(cat);
- break;
- }
- }
- return c;
- }
-
- if (pos > 2)
- return complete_sip_peer(word, state, 0);
-
- return NULL;
-}
-
-/*! \brief Show details of one active dialog */
-static char *sip_show_channel(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
-{
- struct sip_pvt *cur;
- size_t len;
- int found = 0;
- struct ao2_iterator i;
-
- switch (cmd) {
- case CLI_INIT:
- e->command = "sip show channel";
- e->usage =
- "Usage: sip show channel \n"
- " Provides detailed status on a given SIP dialog (identified by SIP call-id).\n";
- return NULL;
- case CLI_GENERATE:
- return complete_sipch(a->line, a->word, a->pos, a->n);
- }
-
- if (a->argc != 4)
- return CLI_SHOWUSAGE;
- len = strlen(a->argv[3]);
-
- i = ao2_iterator_init(dialogs, 0);
- while ((cur = ao2_t_iterator_next(&i, "iterate thru dialogs"))) {
- sip_pvt_lock(cur);
-
- if (!strncasecmp(cur->callid, a->argv[3], len)) {
- struct ast_str *strbuf;
- struct ast_str *codec_buf = ast_str_alloca(AST_FORMAT_CAP_NAMES_LEN);
-
- ast_cli(a->fd, "\n");
- if (cur->subscribed != NONE) {
- ast_cli(a->fd, " * Subscription (type: %s)\n", subscription_type2str(cur->subscribed));
- } else {
- ast_cli(a->fd, " * SIP Call\n");
- }
- ast_cli(a->fd, " Curr. trans. direction: %s\n", ast_test_flag(&cur->flags[0], SIP_OUTGOING) ? "Outgoing" : "Incoming");
- ast_cli(a->fd, " Call-ID: %s\n", cur->callid);
- ast_cli(a->fd, " Owner channel ID: %s\n", cur->owner ? ast_channel_name(cur->owner) : "");
- ast_cli(a->fd, " Our Codec Capability: %s\n", ast_format_cap_get_names(cur->caps, &codec_buf));
- ast_cli(a->fd, " Non-Codec Capability (DTMF): %d\n", cur->noncodeccapability);
- ast_cli(a->fd, " Their Codec Capability: %s\n", ast_format_cap_get_names(cur->peercaps, &codec_buf));
- ast_cli(a->fd, " Joint Codec Capability: %s\n", ast_format_cap_get_names(cur->jointcaps, &codec_buf));
- ast_cli(a->fd, " Format: %s\n", cur->owner ? ast_format_cap_get_names(ast_channel_nativeformats(cur->owner), &codec_buf) : "(nothing)" );
- ast_cli(a->fd, " T.38 support %s\n", AST_CLI_YESNO(cur->udptl != NULL));
- ast_cli(a->fd, " Video support %s\n", AST_CLI_YESNO(cur->vrtp != NULL));
- ast_cli(a->fd, " MaxCallBR: %d kbps\n", cur->maxcallbitrate);
- ast_cli(a->fd, " Theoretical Address: %s\n", ast_sockaddr_stringify(&cur->sa));
- ast_cli(a->fd, " Received Address: %s\n", ast_sockaddr_stringify(&cur->recv));
- ast_cli(a->fd, " SIP Transfer mode: %s\n", transfermode2str(cur->allowtransfer));
- ast_cli(a->fd, " Force rport: %s\n", force_rport_string(cur->flags));
- if (ast_sockaddr_isnull(&cur->redirip)) {
- ast_cli(a->fd,
- " Audio IP: %s (local)\n",
- ast_sockaddr_stringify_addr(&cur->ourip));
- } else {
- ast_cli(a->fd,
- " Audio IP: %s (Outside bridge)\n",
- ast_sockaddr_stringify_addr(&cur->redirip));
- }
- ast_cli(a->fd, " Our Tag: %s\n", cur->tag);
- ast_cli(a->fd, " Their Tag: %s\n", cur->theirtag);
- ast_cli(a->fd, " SIP User agent: %s\n", cur->useragent);
- if (!ast_strlen_zero(cur->username)) {
- ast_cli(a->fd, " Username: %s\n", cur->username);
- }
- if (!ast_strlen_zero(cur->peername)) {
- ast_cli(a->fd, " Peername: %s\n", cur->peername);
- }
- if (!ast_strlen_zero(cur->uri)) {
- ast_cli(a->fd, " Original uri: %s\n", cur->uri);
- }
- if (!ast_strlen_zero(cur->cid_num)) {
- ast_cli(a->fd, " Caller-ID: %s\n", cur->cid_num);
- }
- ast_cli(a->fd, " Need Destroy: %s\n", AST_CLI_YESNO(cur->needdestroy));
- ast_cli(a->fd, " Last Message: %s\n", cur->lastmsg);
- ast_cli(a->fd, " Promiscuous Redir: %s\n", AST_CLI_YESNO(ast_test_flag(&cur->flags[0], SIP_PROMISCREDIR)));
- if ((strbuf = sip_route_list(&cur->route, 1, 0))) {
- ast_cli(a->fd, " Route: %s\n", ast_str_buffer(strbuf));
- ast_free(strbuf);
- }
- ast_cli(a->fd, " DTMF Mode: %s\n", dtmfmode2str(ast_test_flag(&cur->flags[0], SIP_DTMF)));
- ast_cli(a->fd, " SIP Options: ");
- if (cur->sipoptions) {
- int x;
- for (x = 0 ; x < ARRAY_LEN(sip_options); x++) {
- if (cur->sipoptions & sip_options[x].id)
- ast_cli(a->fd, "%s ", sip_options[x].text);
- }
- ast_cli(a->fd, "\n");
- } else {
- ast_cli(a->fd, "(none)\n");
- }
-
- if (!cur->stimer) {
- ast_cli(a->fd, " Session-Timer: Uninitiallized\n");
- } else {
- ast_cli(a->fd, " Session-Timer: %s\n", cur->stimer->st_active ? "Active" : "Inactive");
- if (cur->stimer->st_active == TRUE) {
- ast_cli(a->fd, " S-Timer Interval: %d\n", cur->stimer->st_interval);
- ast_cli(a->fd, " S-Timer Refresher: %s\n", strefresher2str(cur->stimer->st_ref));
- ast_cli(a->fd, " S-Timer Sched Id: %d\n", cur->stimer->st_schedid);
- ast_cli(a->fd, " S-Timer Peer Sts: %s\n", cur->stimer->st_active_peer_ua ? "Active" : "Inactive");
- ast_cli(a->fd, " S-Timer Cached Min-SE: %d\n", cur->stimer->st_cached_min_se);
- ast_cli(a->fd, " S-Timer Cached SE: %d\n", cur->stimer->st_cached_max_se);
- ast_cli(a->fd, " S-Timer Cached Ref: %s\n", strefresher2str(cur->stimer->st_cached_ref));
- ast_cli(a->fd, " S-Timer Cached Mode: %s\n", stmode2str(cur->stimer->st_cached_mode));
- }
- }
-
- /* add transport and media types */
- ast_cli(a->fd, " Transport: %s\n", ast_transport2str(cur->socket.type));
- ast_cli(a->fd, " Media: %s\n", cur->srtp ? "SRTP" : cur->rtp ? "RTP" : "None");
-
- ast_cli(a->fd, "\n\n");
-
- found++;
- }
-
- sip_pvt_unlock(cur);
-
- ao2_t_ref(cur, -1, "toss dialog ptr set by iterator_next");
- }
- ao2_iterator_destroy(&i);
-
- if (!found) {
- ast_cli(a->fd, "No such SIP Call ID starting with '%s'\n", a->argv[3]);
- }
-
- return CLI_SUCCESS;
-}
-
-/*! \brief Show history details of one dialog */
-static char *sip_show_history(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
-{
- struct sip_pvt *cur;
- size_t len;
- int found = 0;
- struct ao2_iterator i;
-
- switch (cmd) {
- case CLI_INIT:
- e->command = "sip show history";
- e->usage =
- "Usage: sip show history \n"
- " Provides detailed dialog history on a given SIP call (specified by call-id).\n";
- return NULL;
- case CLI_GENERATE:
- return complete_sip_show_history(a->line, a->word, a->pos, a->n);
- }
-
- if (a->argc != 4) {
- return CLI_SHOWUSAGE;
- }
-
- if (!recordhistory) {
- ast_cli(a->fd, "\n***Note: History recording is currently DISABLED. Use 'sip set history on' to ENABLE.\n");
- }
-
- len = strlen(a->argv[3]);
-
- i = ao2_iterator_init(dialogs, 0);
- while ((cur = ao2_t_iterator_next(&i, "iterate thru dialogs"))) {
- sip_pvt_lock(cur);
- if (!strncasecmp(cur->callid, a->argv[3], len)) {
- struct sip_history *hist;
- int x = 0;
-
- ast_cli(a->fd, "\n");
- if (cur->subscribed != NONE) {
- ast_cli(a->fd, " * Subscription\n");
- } else {
- ast_cli(a->fd, " * SIP Call\n");
- }
- if (cur->history) {
- AST_LIST_TRAVERSE(cur->history, hist, list)
- ast_cli(a->fd, "%d. %s\n", ++x, hist->event);
- }
- if (x == 0) {
- ast_cli(a->fd, "Call '%s' has no history\n", cur->callid);
- }
- found++;
- }
- sip_pvt_unlock(cur);
- ao2_t_ref(cur, -1, "toss dialog ptr from iterator_next");
- }
- ao2_iterator_destroy(&i);
-
- if (!found) {
- ast_cli(a->fd, "No such SIP Call ID starting with '%s'\n", a->argv[3]);
- }
-
- return CLI_SUCCESS;
-}
-
-/*! \brief Dump SIP history to debug log file at end of lifespan for SIP dialog */
-static void sip_dump_history(struct sip_pvt *dialog)
-{
- int x = 0;
- struct sip_history *hist;
- static int errmsg = 0;
-
- if (!dialog) {
- return;
- }
-
- if (!sipdebug && !DEBUG_ATLEAST(1)) {
- if (!errmsg) {
- ast_log(LOG_NOTICE, "You must have debugging enabled (SIP or Asterisk) in order to dump SIP history.\n");
- errmsg = 1;
- }
- return;
- }
-
- ast_log(LOG_DEBUG, "\n---------- SIP HISTORY for '%s' \n", dialog->callid);
- if (dialog->subscribed) {
- ast_log(LOG_DEBUG, " * Subscription\n");
- } else {
- ast_log(LOG_DEBUG, " * SIP Call\n");
- }
- if (dialog->history) {
- AST_LIST_TRAVERSE(dialog->history, hist, list)
- ast_log(LOG_DEBUG, " %-3.3d. %s\n", ++x, hist->event);
- }
- if (!x) {
- ast_log(LOG_DEBUG, "Call '%s' has no history\n", dialog->callid);
- }
- ast_log(LOG_DEBUG, "\n---------- END SIP HISTORY for '%s' \n", dialog->callid);
-}
-
-
-/*! \brief Receive SIP INFO Message */
-static void handle_request_info(struct sip_pvt *p, struct sip_request *req)
-{
- const char *buf = "";
- unsigned int event;
- const char *c = sip_get_header(req, "Content-Type");
-
- /* Need to check the media/type */
-
- if (!strcasecmp(c, "application/hook-flash")) {
- /* send a FLASH event, for ATAs that send flash as hook-flash not dtmf */
- struct ast_frame f = { AST_FRAME_CONTROL, { AST_CONTROL_FLASH, } };
- ast_queue_frame(p->owner, &f);
- if (sipdebug) {
- ast_verbose("* DTMF-relay event received: FLASH\n");
- }
- transmit_response(p, "200 OK", req);
- return;
- }
-
- if (!strcasecmp(c, "application/dtmf-relay") ||
- !strcasecmp(c, "application/vnd.nortelnetworks.digits") ||
- !strcasecmp(c, "application/dtmf")) {
- unsigned int duration = 0;
-
- if (!p->owner) { /* not a PBX call */
- transmit_response(p, "481 Call leg/transaction does not exist", req);
- sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
- return;
- }
-
- /* If dtmf-relay or vnd.nortelnetworks.digits, parse the signal and duration;
- * otherwise use the body as the signal */
- if (strcasecmp(c, "application/dtmf")) {
- const char *tmp;
-
- if (ast_strlen_zero(buf = get_content_line(req, "Signal", '='))
- && ast_strlen_zero(buf = get_content_line(req, "d", '='))) {
- ast_log(LOG_WARNING, "Unable to retrieve DTMF signal for INFO message on "
- "call %s\n", p->callid);
- transmit_response(p, "200 OK", req);
- return;
- }
- if (!ast_strlen_zero((tmp = get_content_line(req, "Duration", '=')))) {
- sscanf(tmp, "%30u", &duration);
- }
- } else {
- /* Type is application/dtmf, simply use what's in the message body */
- buf = get_content(req);
- }
-
- /* An empty message body requires us to send a 200 OK */
- if (ast_strlen_zero(buf)) {
- transmit_response(p, "200 OK", req);
- return;
- }
-
- if (!duration) {
- duration = 100; /* 100 ms */
- }
-
- if (buf[0] == '*') {
- event = 10;
- } else if (buf[0] == '#') {
- event = 11;
- } else if (buf[0] == '!') {
- event = 16;
- } else if ('A' <= buf[0] && buf[0] <= 'D') {
- event = 12 + buf[0] - 'A';
- } else if ('a' <= buf[0] && buf[0] <= 'd') {
- event = 12 + buf[0] - 'a';
- } else if ((sscanf(buf, "%30u", &event) != 1) || event > 16) {
- ast_log(AST_LOG_WARNING, "Unable to convert DTMF event signal code to a valid "
- "value for INFO message on call %s\n", p->callid);
- transmit_response(p, "200 OK", req);
- return;
- }
-
- if (event == 16) {
- /* send a FLASH event */
- struct ast_frame f = { AST_FRAME_CONTROL, { AST_CONTROL_FLASH, } };
- ast_queue_frame(p->owner, &f);
- if (sipdebug) {
- ast_verbose("* DTMF-relay event received: FLASH\n");
- }
- } else {
- /* send a DTMF event */
- struct ast_frame f = { AST_FRAME_DTMF, };
- if (event < 10) {
- f.subclass.integer = '0' + event;
- } else if (event == 10) {
- f.subclass.integer = '*';
- } else if (event == 11) {
- f.subclass.integer = '#';
- } else {
- f.subclass.integer = 'A' + (event - 12);
- }
- f.len = duration;
- ast_queue_frame(p->owner, &f);
- if (sipdebug) {
- ast_verbose("* DTMF-relay event received: %c\n", (int) f.subclass.integer);
- }
- }
- transmit_response(p, "200 OK", req);
- return;
- } else if (!strcasecmp(c, "application/media_control+xml")) {
- /* Eh, we'll just assume it's a fast picture update for now */
- if (p->owner) {
- ast_queue_control(p->owner, AST_CONTROL_VIDUPDATE);
- }
- transmit_response(p, "200 OK", req);
- return;
- } else if (!ast_strlen_zero(c = sip_get_header(req, "X-ClientCode"))) {
- /* Client code (from SNOM phone) */
- if (ast_test_flag(&p->flags[0], SIP_USECLIENTCODE)) {
- if (p->owner) {
- ast_cdr_setuserfield(ast_channel_name(p->owner), c);
- }
- transmit_response(p, "200 OK", req);
- } else {
- transmit_response(p, "403 Forbidden", req);
- }
- return;
- } else if (!ast_strlen_zero(c = sip_get_header(req, "Record"))) {
- /* INFO messages generated by some phones to start/stop recording
- * on phone calls.
- */
-
- char feat[AST_FEATURE_MAX_LEN];
- int feat_res = -1;
- int j;
- struct ast_frame f = { AST_FRAME_DTMF, };
- int suppress_warning = 0; /* Supress warning if the feature is blank */
-
- if (!p->owner) { /* not a PBX call */
- transmit_response(p, "481 Call leg/transaction does not exist", req);
- sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
- return;
- }
-
- /* first, get the feature string, if it exists */
- if (p->relatedpeer) {
- if (!strcasecmp(c, "on")) {
- if (ast_strlen_zero(p->relatedpeer->record_on_feature)) {
- suppress_warning = 1;
- } else {
- feat_res = ast_get_feature(p->owner, p->relatedpeer->record_on_feature, feat, sizeof(feat));
- }
- } else if (!strcasecmp(c, "off")) {
- if (ast_strlen_zero(p->relatedpeer->record_off_feature)) {
- suppress_warning = 1;
- } else {
- feat_res = ast_get_feature(p->owner, p->relatedpeer->record_off_feature, feat, sizeof(feat));
- }
- } else {
- ast_log(LOG_ERROR, "Received INFO requesting to record with invalid value: %s\n", c);
- }
- }
- if (feat_res || ast_strlen_zero(feat)) {
- if (!suppress_warning) {
- ast_log(LOG_WARNING, "Recording requested, but no One Touch Monitor registered. (See features.conf)\n");
- }
- /* 403 means that we don't support this feature, so don't request it again */
- transmit_response(p, "403 Forbidden", req);
- return;
- }
- /* Send the feature code to the PBX as DTMF, just like the handset had sent it */
- f.len = 100;
- for (j = 0; j < strlen(feat); j++) {
- f.subclass.integer = feat[j];
- ast_queue_frame(p->owner, &f);
- if (sipdebug) {
- ast_verbose("* DTMF-relay event faked: %c\n", f.subclass.integer);
- }
- }
-
- ast_debug(1, "Got a Request to Record the channel, state %s\n", c);
- transmit_response(p, "200 OK", req);
- return;
- } else if (ast_strlen_zero(c = sip_get_header(req, "Content-Length")) || !strcasecmp(c, "0")) {
- /* This is probably just a packet making sure the signalling is still up, just send back a 200 OK */
- transmit_response(p, "200 OK", req);
- return;
- }
-
- /* Other type of INFO message, not really understood by Asterisk */
-
- ast_log(LOG_WARNING, "Unable to parse INFO message from %s. Content %s\n", p->callid, buf);
- transmit_response(p, "415 Unsupported media type", req);
- return;
-}
-
-/*! \brief Enable SIP Debugging for a single IP */
-static char *sip_do_debug_ip(int fd, const char *arg)
-{
- if (ast_sockaddr_resolve_first_af(&debugaddr, arg, 0, 0)) {
- return CLI_SHOWUSAGE;
- }
-
- ast_cli(fd, "SIP Debugging Enabled for IP: %s\n", ast_sockaddr_stringify_addr(&debugaddr));
- sipdebug |= sip_debug_console;
-
- return CLI_SUCCESS;
-}
-
-/*! \brief Turn on SIP debugging for a given peer */
-static char *sip_do_debug_peer(int fd, const char *arg)
-{
- struct sip_peer *peer = sip_find_peer(arg, NULL, TRUE, FINDPEERS, FALSE, 0);
- if (!peer) {
- ast_cli(fd, "No such peer '%s'\n", arg);
- } else if (ast_sockaddr_isnull(&peer->addr)) {
- ast_cli(fd, "Unable to get IP address of peer '%s'\n", arg);
- } else {
- ast_sockaddr_copy(&debugaddr, &peer->addr);
- ast_cli(fd, "SIP Debugging Enabled for IP: %s\n", ast_sockaddr_stringify_addr(&debugaddr));
- sipdebug |= sip_debug_console;
- }
- if (peer) {
- sip_unref_peer(peer, "sip_do_debug_peer: sip_unref_peer, from sip_find_peer call");
- }
- return CLI_SUCCESS;
-}
-
-/*! \brief Turn on SIP debugging (CLI command) */
-static char *sip_do_debug(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
-{
- int oldsipdebug = sipdebug & sip_debug_console;
- const char *what;
-
- if (cmd == CLI_INIT) {
- e->command = "sip set debug {on|off|ip|peer}";
- e->usage =
- "Usage: sip set debug {off|on|ip addr[:port]|peer peername}\n"
- " Globally disables dumping of SIP packets,\n"
- " or enables it either globally or for a (single)\n"
- " IP address or registered peer.\n";
- return NULL;
- } else if (cmd == CLI_GENERATE) {
- if (a->pos == 4 && !strcasecmp(a->argv[3], "peer"))
- return complete_sip_peer(a->word, a->n, 0);
- return NULL;
- }
-
- what = a->argv[e->args-1]; /* guaranteed to exist */
- if (a->argc == e->args) { /* on/off */
- if (!strcasecmp(what, "on")) {
- sipdebug |= sip_debug_console;
- sipdebug_text = 1; /*! \note this can be a special debug command - "sip debug text" or something */
- memset(&debugaddr, 0, sizeof(debugaddr));
- ast_cli(a->fd, "SIP Debugging %senabled\n", oldsipdebug ? "re-" : "");
- return CLI_SUCCESS;
- } else if (!strcasecmp(what, "off")) {
- sipdebug &= ~sip_debug_console;
- sipdebug_text = 0;
- if (sipdebug == sip_debug_none) {
- ast_cli(a->fd, "SIP Debugging Disabled\n");
- } else {
- ast_cli(a->fd, "SIP Debugging still enabled due to configuration.\n");
- ast_cli(a->fd, "Set sipdebug=no in sip.conf and reload to actually disable.\n");
- }
- return CLI_SUCCESS;
- }
- } else if (a->argc == e->args + 1) { /* ip/peer */
- if (!strcasecmp(what, "ip"))
- return sip_do_debug_ip(a->fd, a->argv[e->args]);
- else if (!strcasecmp(what, "peer"))
- return sip_do_debug_peer(a->fd, a->argv[e->args]);
- }
- return CLI_SHOWUSAGE; /* default, failure */
-}
-
-/*! \brief Cli command to send SIP notify to peer */
-static char *sip_cli_notify(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
-{
- struct ast_variable *varlist;
- int i;
-
- switch (cmd) {
- case CLI_INIT:
- e->command = "sip notify";
- e->usage =
- "Usage: sip notify [...]\n"
- " Send a NOTIFY message to a SIP peer or peers\n"
- " Message types are defined in sip_notify.conf\n";
- return NULL;
- case CLI_GENERATE:
- return complete_sip_notify(a->line, a->word, a->pos, a->n);
- }
-
- if (a->argc < 4)
- return CLI_SHOWUSAGE;
-
- if (!notify_types) {
- ast_cli(a->fd, "No %s file found, or no types listed there\n", notify_config);
- return CLI_FAILURE;
- }
-
- varlist = ast_variable_browse(notify_types, a->argv[2]);
-
- if (!varlist) {
- ast_cli(a->fd, "Unable to find notify type '%s'\n", a->argv[2]);
- return CLI_FAILURE;
- }
-
- for (i = 3; i < a->argc; i++) {
- struct sip_pvt *p;
- char buf[512];
- struct ast_variable *header, *var;
-
- if (!(p = sip_alloc(NULL, NULL, 0, SIP_NOTIFY, NULL, 0))) {
- ast_log(LOG_WARNING, "Unable to build sip pvt data for notify (memory/socket error)\n");
- return CLI_FAILURE;
- }
-
- if (create_addr(p, a->argv[i], NULL, 1)) {
- /* Maybe they're not registered, etc. */
- dialog_unlink_all(p);
- dialog_unref(p, "unref dialog inside for loop" );
- /* sip_destroy(p); */
- ast_cli(a->fd, "Could not create address for '%s'\n", a->argv[i]);
- continue;
- }
-
- /* Notify is outgoing call */
- ast_set_flag(&p->flags[0], SIP_OUTGOING);
- sip_notify_alloc(p);
- p->notify->headers = header = ast_variable_new("Subscription-State", "terminated", "");
-
- for (var = varlist; var; var = var->next) {
- ast_copy_string(buf, var->value, sizeof(buf));
- ast_unescape_semicolon(buf);
-
- if (!strcasecmp(var->name, "Content")) {
- if (ast_str_strlen(p->notify->content))
- ast_str_append(&p->notify->content, 0, "\r\n");
- ast_str_append(&p->notify->content, 0, "%s", buf);
- } else if (!strcasecmp(var->name, "Content-Length")) {
- ast_log(LOG_WARNING, "it is not necessary to specify Content-Length in sip_notify.conf, ignoring\n");
- } else {
- header->next = ast_variable_new(var->name, buf, "");
- header = header->next;
- }
- }
-
- /* Now that we have the peer's address, set our ip and change callid */
- ast_sip_ouraddrfor(&p->sa, &p->ourip, p);
- build_via(p);
-
- change_callid_pvt(p, NULL);
-
- ast_cli(a->fd, "Sending NOTIFY of type '%s' to '%s'\n", a->argv[2], a->argv[i]);
- sip_scheddestroy(p, SIP_TRANS_TIMEOUT);
- transmit_invite(p, SIP_NOTIFY, 0, 2, NULL);
- dialog_unref(p, "bump down the count of p since we're done with it.");
- }
-
- return CLI_SUCCESS;
-}
-
-/*! \brief Enable/Disable SIP History logging (CLI) */
-static char *sip_set_history(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
-{
- switch (cmd) {
- case CLI_INIT:
- e->command = "sip set history {on|off}";
- e->usage =
- "Usage: sip set history {on|off}\n"
- " Enables/Disables recording of SIP dialog history for debugging purposes.\n"
- " Use 'sip show history' to view the history of a call number.\n";
- return NULL;
- case CLI_GENERATE:
- return NULL;
- }
-
- if (a->argc != e->args)
- return CLI_SHOWUSAGE;
-
- if (!strncasecmp(a->argv[e->args - 1], "on", 2)) {
- recordhistory = TRUE;
- ast_cli(a->fd, "SIP History Recording Enabled (use 'sip show history')\n");
- } else if (!strncasecmp(a->argv[e->args - 1], "off", 3)) {
- recordhistory = FALSE;
- ast_cli(a->fd, "SIP History Recording Disabled\n");
- } else {
- return CLI_SHOWUSAGE;
- }
- return CLI_SUCCESS;
-}
-
-/*! \brief Authenticate for outbound registration */
-static int do_register_auth(struct sip_pvt *p, struct sip_request *req, enum sip_auth_type code)
-{
- char *header, *respheader;
- char digest[1024];
-
- p->authtries++;
- sip_auth_headers(code, &header, &respheader);
- memset(digest, 0, sizeof(digest));
- if (reply_digest(p, req, header, SIP_REGISTER, digest, sizeof(digest))) {
- /* There's nothing to use for authentication */
- /* No digest challenge in request */
- if (sip_debug_test_pvt(p) && p->registry)
- ast_verbose("No authentication challenge, sending blank registration to domain/host name %s\n", p->registry->hostname);
- /* No old challenge */
- return -1;
- }
- if (p->do_history)
- append_history(p, "RegistryAuth", "Try: %d", p->authtries);
- if (sip_debug_test_pvt(p) && p->registry)
- ast_verbose("Responding to challenge, registration to domain/host name %s\n", p->registry->hostname);
- return transmit_register(p->registry, SIP_REGISTER, digest, respheader);
-}
-
-/*! \brief Add authentication on outbound SIP packet */
-static int do_proxy_auth(struct sip_pvt *p, struct sip_request *req, enum sip_auth_type code, int sipmethod, int init)
-{
- char *header, *respheader;
- char digest[1024];
-
- if (!p->options && !(p->options = ast_calloc(1, sizeof(*p->options))))
- return -2;
-
- p->authtries++;
- sip_auth_headers(code, &header, &respheader);
- ast_debug(2, "Auth attempt %d on %s\n", p->authtries, sip_methods[sipmethod].text);
- memset(digest, 0, sizeof(digest));
- if (reply_digest(p, req, header, sipmethod, digest, sizeof(digest) )) {
- /* No way to authenticate */
- return -1;
- }
- /* Now we have a reply digest */
- p->options->auth = digest;
- p->options->authheader = respheader;
- return transmit_invite(p, sipmethod, sipmethod == SIP_INVITE, init, NULL);
-}
-
-/*! \brief reply to authentication for outbound registrations
-\retval -1 if we have no auth
-\note This is used for register= servers in sip.conf, SIP proxies we register
- with for receiving calls from. */
-static int reply_digest(struct sip_pvt *p, struct sip_request *req, char *header, int sipmethod, char *digest, int digest_len)
-{
- char tmp[512];
- char *c;
- char oldnonce[256];
- int start = 0;
-
- /* table of recognised keywords, and places where they should be copied */
- const struct x {
- const char *key;
- const ast_string_field *field;
- } *i, keys[] = {
- { "realm=", &p->realm },
- { "nonce=", &p->nonce },
- { "opaque=", &p->opaque },
- { "qop=", &p->qop },
- { "domain=", &p->domain },
- { NULL, 0 },
- };
-
- do {
- ast_copy_string(tmp, __get_header(req, header, &start), sizeof(tmp));
- if (ast_strlen_zero(tmp))
- return -1;
- } while (strcasestr(tmp, "algorithm=") && !strcasestr(tmp, "algorithm=MD5"));
- if (strncasecmp(tmp, "Digest ", strlen("Digest "))) {
- ast_log(LOG_WARNING, "missing Digest.\n");
- return -1;
- }
- c = tmp + strlen("Digest ");
- ast_copy_string(oldnonce, p->nonce, sizeof(oldnonce));
- while (c && *(c = ast_skip_blanks(c))) { /* lookup for keys */
- for (i = keys; i->key != NULL; i++) {
- char *src, *separator;
- if (strncasecmp(c, i->key, strlen(i->key)) != 0)
- continue;
- /* Found. Skip keyword, take text in quotes or up to the separator. */
- c += strlen(i->key);
- if (*c == '"') {
- src = ++c;
- separator = "\"";
- } else {
- src = c;
- separator = ",";
- }
- strsep(&c, separator); /* clear separator and move ptr */
- ast_string_field_ptr_set(p, i->field, src);
- break;
- }
- if (i->key == NULL) /* not found, try ',' */
- strsep(&c, ",");
- }
- /* Reset nonce count */
- if (strcmp(p->nonce, oldnonce))
- p->noncecount = 0;
-
- /* Save auth data for following registrations */
- if (p->registry) {
- struct sip_registry *r = p->registry;
-
- if (strcmp(r->nonce, p->nonce)) {
- ast_string_field_set(r, realm, p->realm);
- ast_string_field_set(r, nonce, p->nonce);
- ast_string_field_set(r, authdomain, p->domain);
- ast_string_field_set(r, opaque, p->opaque);
- ast_string_field_set(r, qop, p->qop);
- r->noncecount = 0;
- }
- }
- return build_reply_digest(p, sipmethod, digest, digest_len);
-}
-
-/*! \brief Build reply digest
-\retval -1 if we have no auth
-\note Build digest challenge for authentication of registrations and calls
- Also used for authentication of BYE
-*/
-static int build_reply_digest(struct sip_pvt *p, int method, char* digest, int digest_len)
-{
- char a1[256];
- char a2[256];
- char a1_hash[256];
- char a2_hash[256];
- char resp[256];
- char resp_hash[256];
- char uri[256];
- char opaque[256] = "";
- char cnonce[80];
- const char *username;
- const char *secret;
- const char *md5secret;
- struct sip_auth *auth; /* Realm authentication credential */
- struct sip_auth_container *credentials;
-
- if (!ast_strlen_zero(p->domain))
- snprintf(uri, sizeof(uri), "%s:%s", p->socket.type == AST_TRANSPORT_TLS ? "sips" : "sip", p->domain);
- else if (!ast_strlen_zero(p->uri))
- ast_copy_string(uri, p->uri, sizeof(uri));
- else
- snprintf(uri, sizeof(uri), "%s:%s@%s", p->socket.type == AST_TRANSPORT_TLS ? "sips" : "sip", p->username, ast_sockaddr_stringify_host_remote(&p->sa));
-
- snprintf(cnonce, sizeof(cnonce), "%08lx", (unsigned long)ast_random());
-
- /* Check if we have peer credentials */
- ao2_lock(p);
- credentials = p->peerauth;
- if (credentials) {
- ao2_t_ref(credentials, +1, "Ref peer auth for digest");
- }
- ao2_unlock(p);
- auth = find_realm_authentication(credentials, p->realm);
- if (!auth) {
- /* If not, check global credentials */
- if (credentials) {
- ao2_t_ref(credentials, -1, "Unref peer auth for digest");
- }
- ast_mutex_lock(&authl_lock);
- credentials = authl;
- if (credentials) {
- ao2_t_ref(credentials, +1, "Ref global auth for digest");
- }
- ast_mutex_unlock(&authl_lock);
- auth = find_realm_authentication(credentials, p->realm);
- }
-
- if (auth) {
- ast_debug(3, "use realm [%s] from peer [%s][%s]\n", auth->username, p->peername, p->username);
- username = auth->username;
- secret = auth->secret;
- md5secret = auth->md5secret;
- if (sipdebug)
- ast_debug(1, "Using realm %s authentication for call %s\n", p->realm, p->callid);
- } else {
- /* No authentication, use peer or register= config */
- username = p->authname;
- secret = p->relatedpeer
- && !ast_strlen_zero(p->relatedpeer->remotesecret)
- ? p->relatedpeer->remotesecret : p->peersecret;
- md5secret = p->peermd5secret;
- }
- if (ast_strlen_zero(username)) {
- /* We have no authentication */
- if (credentials) {
- ao2_t_ref(credentials, -1, "Unref auth for digest");
- }
- return -1;
- }
-
- /* Calculate SIP digest response */
- snprintf(a1, sizeof(a1), "%s:%s:%s", username, p->realm, secret);
- snprintf(a2, sizeof(a2), "%s:%s", sip_methods[method].text, uri);
- if (!ast_strlen_zero(md5secret))
- ast_copy_string(a1_hash, md5secret, sizeof(a1_hash));
- else
- ast_md5_hash(a1_hash, a1);
- ast_md5_hash(a2_hash, a2);
-
- p->noncecount++;
- if (!ast_strlen_zero(p->qop))
- snprintf(resp, sizeof(resp), "%s:%s:%08x:%s:%s:%s", a1_hash, p->nonce, (unsigned)p->noncecount, cnonce, "auth", a2_hash);
- else
- snprintf(resp, sizeof(resp), "%s:%s:%s", a1_hash, p->nonce, a2_hash);
- ast_md5_hash(resp_hash, resp);
-
- /* only include the opaque string if it's set */
- if (!ast_strlen_zero(p->opaque)) {
- snprintf(opaque, sizeof(opaque), ", opaque=\"%s\"", p->opaque);
- }
-
- /* XXX We hard code our qop to "auth" for now. XXX */
- if (!ast_strlen_zero(p->qop))
- snprintf(digest, digest_len, "Digest username=\"%s\", realm=\"%s\", algorithm=MD5, uri=\"%s\", nonce=\"%s\", response=\"%s\"%s, qop=auth, cnonce=\"%s\", nc=%08x", username, p->realm, uri, p->nonce, resp_hash, opaque, cnonce, (unsigned)p->noncecount);
- else
- snprintf(digest, digest_len, "Digest username=\"%s\", realm=\"%s\", algorithm=MD5, uri=\"%s\", nonce=\"%s\", response=\"%s\"%s", username, p->realm, uri, p->nonce, resp_hash, opaque);
-
- append_history(p, "AuthResp", "Auth response sent for %s in realm %s - nc %d", username, p->realm, p->noncecount);
-
- if (credentials) {
- ao2_t_ref(credentials, -1, "Unref auth for digest");
- }
- return 0;
-}
-
-/*! \brief Read SIP header (dialplan function) */
-static int func_header_read(struct ast_channel *chan, const char *function, char *data, char *buf, size_t len)
-{
- struct sip_pvt *p;
- const char *content = NULL;
- char *mutable_data = ast_strdupa(data);
- AST_DECLARE_APP_ARGS(args,
- AST_APP_ARG(header);
- AST_APP_ARG(number);
- );
- int i, number, start = 0;
-
- if (!chan) {
- ast_log(LOG_WARNING, "No channel was provided to %s function.\n", function);
- return -1;
- }
-
- if (ast_strlen_zero(data)) {
- ast_log(LOG_WARNING, "This function requires a header name.\n");
- return -1;
- }
-
- ast_channel_lock(chan);
- if (!IS_SIP_TECH(ast_channel_tech(chan))) {
- ast_log(LOG_WARNING, "This function can only be used on SIP channels.\n");
- ast_channel_unlock(chan);
- return -1;
- }
-
- AST_STANDARD_APP_ARGS(args, mutable_data);
- if (!args.number) {
- number = 1;
- } else {
- sscanf(args.number, "%30d", &number);
- if (number < 1)
- number = 1;
- }
-
- p = ast_channel_tech_pvt(chan);
-
- /* If there is no private structure, this channel is no longer alive */
- if (!p) {
- ast_channel_unlock(chan);
- return -1;
- }
-
- for (i = 0; i < number; i++)
- content = __get_header(&p->initreq, args.header, &start);
-
- if (ast_strlen_zero(content)) {
- ast_channel_unlock(chan);
- return -1;
- }
-
- ast_copy_string(buf, content, len);
- ast_channel_unlock(chan);
-
- return 0;
-}
-
-static struct ast_custom_function sip_header_function = {
- .name = "SIP_HEADER",
- .read = func_header_read,
-};
-
-/*! \brief Read unique list of SIP headers (dialplan function) */
-static int func_headers_read2(struct ast_channel *chan, const char *function, char *data, struct ast_str **buf, ssize_t maxlen)
-{
- int i;
- struct sip_pvt *pvt;
- char *mutable_data = ast_strdupa(data);
- struct ast_str *token = ast_str_alloca(100);
- AST_DECLARE_APP_ARGS(args,
- AST_APP_ARG(pattern);
- );
-
- if (!chan) {
- return -1;
- }
-
- ast_channel_lock(chan);
-
- if (!IS_SIP_TECH(ast_channel_tech(chan))) {
- ast_log(LOG_WARNING, "This function can only be used on SIP channels.\n");
- ast_channel_unlock(chan);
- return -1;
- }
-
- pvt = ast_channel_tech_pvt(chan);
- if (!pvt) {
- ast_channel_unlock(chan);
- return -1;
- }
-
- AST_STANDARD_APP_ARGS(args, mutable_data);
- if (!args.pattern || strcmp(args.pattern, "*") == 0) {
- args.pattern = "";
- }
-
- for (i = 0; i < pvt->initreq.headers; i++) {
- const char *header = REQ_OFFSET_TO_STR(&pvt->initreq, header[i]);
- if (ast_begins_with(header, args.pattern)) {
- int hdrlen = strcspn(header, " \t:,"); /* Comma will break our logic, and illegal per RFC. */
- const char *term = ast_skip_blanks(header + hdrlen);
- if (hdrlen > 0 && *term == ':') { /* Header is malformed otherwise! */
- const char *s = NULL;
-
- /* Return short headers in full form always. */
- if (hdrlen == 1) {
- char short_hdr[2] = { header[0], '\0' };
- s = find_full_alias(short_hdr, NULL);
- }
- if (s) {
- /* Short header was found and expanded. */
- ast_str_set(&token, -1, "%s,", s);
- } else {
- /* Return the header as is, whether 1-character or not. */
- ast_str_set(&token, -1, "%.*s,", hdrlen, header);
- }
-
- /* Has the same header been already added? */
- s = ast_str_buffer(*buf);
- while ((s = strstr(s, ast_str_buffer(token))) != NULL) {
- /* Found suffix, but is it the full token? */
- if (s == ast_str_buffer(*buf) || s[-1] == ',')
- break;
- /* Only suffix matched, go on with the search after the comma. */
- s += hdrlen + 1;
- }
-
- /* s is null iff not broken from the loop, hence header not yet added. */
- if (s == NULL) {
- ast_str_append(buf, maxlen, "%s", ast_str_buffer(token));
- }
- }
- }
- }
-
- ast_str_truncate(*buf, -1); /* Trim the last comma. Safe if empty. */
-
- ast_channel_unlock(chan);
- return 0;
-}
-
-static struct ast_custom_function sip_headers_function = {
- .name = "SIP_HEADERS",
- .read2 = func_headers_read2,
-};
-
-
-/*! \brief Dial plan function to check if domain is local */
-static int func_check_sipdomain(struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)
-{
- if (ast_strlen_zero(data)) {
- ast_log(LOG_WARNING, "CHECKSIPDOMAIN requires an argument - A domain name\n");
- return -1;
- }
- if (check_sip_domain(data, NULL, 0))
- ast_copy_string(buf, data, len);
- else
- buf[0] = '\0';
- return 0;
-}
-
-static struct ast_custom_function checksipdomain_function = {
- .name = "CHECKSIPDOMAIN",
- .read = func_check_sipdomain,
-};
-
-/*! \brief ${SIPPEER()} Dialplan function - reads peer data */
-static int function_sippeer(struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)
-{
- struct sip_peer *peer;
- char *colname;
-
- if ((colname = strchr(data, ','))) {
- *colname++ = '\0';
- } else {
- colname = "ip";
- }
-
- if (!(peer = sip_find_peer(data, NULL, TRUE, FINDPEERS, FALSE, 0)))
- return -1;
-
- if (!strcasecmp(colname, "ip")) {
- ast_copy_string(buf, ast_sockaddr_stringify_addr(&peer->addr), len);
- } else if (!strcasecmp(colname, "port")) {
- snprintf(buf, len, "%d", ast_sockaddr_port(&peer->addr));
- } else if (!strcasecmp(colname, "status")) {
- peer_status(peer, buf, len);
- } else if (!strcasecmp(colname, "language")) {
- ast_copy_string(buf, peer->language, len);
- } else if (!strcasecmp(colname, "regexten")) {
- ast_copy_string(buf, peer->regexten, len);
- } else if (!strcasecmp(colname, "limit")) {
- snprintf(buf, len, "%d", peer->call_limit);
- } else if (!strcasecmp(colname, "busylevel")) {
- snprintf(buf, len, "%d", peer->busy_level);
- } else if (!strcasecmp(colname, "curcalls")) {
- snprintf(buf, len, "%d", peer->inuse);
- } else if (!strcasecmp(colname, "maxforwards")) {
- snprintf(buf, len, "%d", peer->maxforwards);
- } else if (!strcasecmp(colname, "accountcode")) {
- ast_copy_string(buf, peer->accountcode, len);
- } else if (!strcasecmp(colname, "callgroup")) {
- ast_print_group(buf, len, peer->callgroup);
- } else if (!strcasecmp(colname, "pickupgroup")) {
- ast_print_group(buf, len, peer->pickupgroup);
- } else if (!strcasecmp(colname, "namedcallgroup")) {
- struct ast_str *tmp_str = ast_str_create(1024);
- if (tmp_str) {
- ast_copy_string(buf, ast_print_namedgroups(&tmp_str, peer->named_callgroups), len);
- ast_free(tmp_str);
- }
- } else if (!strcasecmp(colname, "namedpickupgroup")) {
- struct ast_str *tmp_str = ast_str_create(1024);
- if (tmp_str) {
- ast_copy_string(buf, ast_print_namedgroups(&tmp_str, peer->named_pickupgroups), len);
- ast_free(tmp_str);
- }
- } else if (!strcasecmp(colname, "useragent")) {
- ast_copy_string(buf, peer->useragent, len);
- } else if (!strcasecmp(colname, "mailbox")) {
- struct ast_str *mailbox_str = ast_str_alloca(512);
- peer_mailboxes_to_str(&mailbox_str, peer);
- ast_copy_string(buf, ast_str_buffer(mailbox_str), len);
- } else if (!strcasecmp(colname, "context")) {
- ast_copy_string(buf, peer->context, len);
- } else if (!strcasecmp(colname, "expire")) {
- snprintf(buf, len, "%d", peer->expire);
- } else if (!strcasecmp(colname, "dynamic")) {
- ast_copy_string(buf, peer->host_dynamic ? "yes" : "no", len);
- } else if (!strcasecmp(colname, "callerid_name")) {
- ast_copy_string(buf, peer->cid_name, len);
- } else if (!strcasecmp(colname, "callerid_num")) {
- ast_copy_string(buf, peer->cid_num, len);
- } else if (!strcasecmp(colname, "codecs")) {
- struct ast_str *codec_buf = ast_str_alloca(AST_FORMAT_CAP_NAMES_LEN);
- ast_format_cap_get_names(peer->caps, &codec_buf);
- ast_copy_string(buf, ast_str_buffer(codec_buf), len);
- } else if (!strcasecmp(colname, "encryption")) {
- snprintf(buf, len, "%u", ast_test_flag(&peer->flags[1], SIP_PAGE2_USE_SRTP));
- } else if (!strncasecmp(colname, "chanvar[", 8)) {
- char *chanvar=colname + 8;
- struct ast_variable *v;
-
- chanvar = strsep(&chanvar, "]");
- for (v = peer->chanvars ; v ; v = v->next) {
- if (!strcasecmp(v->name, chanvar)) {
- ast_copy_string(buf, v->value, len);
- }
- }
- } else if (!strncasecmp(colname, "codec[", 6)) {
- char *codecnum;
- struct ast_format *codec;
-
- codecnum = colname + 6; /* move past the '[' */
- codecnum = strsep(&codecnum, "]"); /* trim trailing ']' if any */
- codec = ast_format_cap_get_format(peer->caps, atoi(codecnum));
- if (codec) {
- ast_copy_string(buf, ast_format_get_name(codec), len);
- ao2_ref(codec, -1);
- } else {
- buf[0] = '\0';
- }
- } else {
- buf[0] = '\0';
- }
-
- sip_unref_peer(peer, "sip_unref_peer from function_sippeer, just before return");
-
- return 0;
-}
-
-/*! \brief Structure to declare a dialplan function: SIPPEER */
-static struct ast_custom_function sippeer_function = {
- .name = "SIPPEER",
- .read = function_sippeer,
-};
-
-/*! \brief update redirecting information for a channel based on headers
- *
- */
-static void change_redirecting_information(struct sip_pvt *p, struct sip_request *req,
- struct ast_party_redirecting *redirecting,
- struct ast_set_party_redirecting *update_redirecting, int set_call_forward)
-{
- char *redirecting_from_name = NULL;
- char *redirecting_from_number = NULL;
- char *redirecting_to_name = NULL;
- char *redirecting_to_number = NULL;
- char *reason_str = NULL;
- int reason = AST_REDIRECTING_REASON_UNCONDITIONAL;
- int is_response = req->method == SIP_RESPONSE;
- int res = 0;
-
- res = get_rdnis(p, req, &redirecting_from_name, &redirecting_from_number, &reason, &reason_str);
- if (res == -1) {
- if (is_response) {
- get_name_and_number(sip_get_header(req, "TO"), &redirecting_from_name, &redirecting_from_number);
- } else {
- return;
- }
- }
-
- /* At this point, all redirecting "from" info should be filled in appropriately
- * on to the "to" info
- */
-
- if (is_response) {
- parse_moved_contact(p, req, &redirecting_to_name, &redirecting_to_number, set_call_forward);
- } else {
- get_name_and_number(sip_get_header(req, "TO"), &redirecting_to_name, &redirecting_to_number);
- }
-
- if (!ast_strlen_zero(redirecting_from_number)) {
- ast_debug(3, "Got redirecting from number %s\n", redirecting_from_number);
- update_redirecting->from.number = 1;
- redirecting->from.number.valid = 1;
- ast_free(redirecting->from.number.str);
- redirecting->from.number.str = redirecting_from_number;
- } else {
- ast_free(redirecting_from_number);
- }
- if (!ast_strlen_zero(redirecting_from_name)) {
- ast_debug(3, "Got redirecting from name %s\n", redirecting_from_name);
- update_redirecting->from.name = 1;
- redirecting->from.name.valid = 1;
- ast_free(redirecting->from.name.str);
- redirecting->from.name.str = redirecting_from_name;
- } else {
- ast_free(redirecting_from_name);
- }
- if (!ast_strlen_zero(p->cid_tag)) {
- ast_free(redirecting->from.tag);
- redirecting->from.tag = ast_strdup(p->cid_tag);
- ast_free(redirecting->to.tag);
- redirecting->to.tag = ast_strdup(p->cid_tag);
- }
- if (!ast_strlen_zero(redirecting_to_number)) {
- ast_debug(3, "Got redirecting to number %s\n", redirecting_to_number);
- update_redirecting->to.number = 1;
- redirecting->to.number.valid = 1;
- ast_free(redirecting->to.number.str);
- redirecting->to.number.str = redirecting_to_number;
- } else {
- ast_free(redirecting_to_number);
- }
- if (!ast_strlen_zero(redirecting_to_name)) {
- ast_debug(3, "Got redirecting to name %s\n", redirecting_to_name);
- update_redirecting->to.name = 1;
- redirecting->to.name.valid = 1;
- ast_free(redirecting->to.name.str);
- redirecting->to.name.str = redirecting_to_name;
- } else {
- ast_free(redirecting_to_name);
- }
- redirecting->reason.code = reason;
- ast_free(redirecting->reason.str);
- redirecting->reason.str = reason_str;
- if (reason_str) {
- ast_debug(3, "Got redirecting reason %s\n", ast_strlen_zero(reason_str)
- ? sip_reason_code_to_str(&redirecting->reason) : reason_str);
- }
-}
-
-/*! \brief Parse 302 Moved temporalily response
- \todo XXX Doesn't redirect over TLS on sips: uri's.
- If we get a redirect to a SIPS: uri, this needs to be going back to the
- dialplan (this is a request for a secure signalling path).
- Note that transport=tls is deprecated, but we need to support it on incoming requests.
-*/
-static void parse_moved_contact(struct sip_pvt *p, struct sip_request *req, char **name, char **number, int set_call_forward)
-{
- char contact[SIPBUFSIZE];
- char *contact_name = NULL;
- char *contact_number = NULL;
- char *separator, *trans;
- char *domain;
- enum ast_transport transport = AST_TRANSPORT_UDP;
-
- ast_copy_string(contact, sip_get_header(req, "Contact"), sizeof(contact));
- if ((separator = strchr(contact, ',')))
- *separator = '\0';
-
- contact_number = get_in_brackets(contact);
- if ((trans = strcasestr(contact_number, ";transport="))) {
- trans += 11;
-
- if ((separator = strchr(trans, ';')))
- *separator = '\0';
-
- if (!strncasecmp(trans, "tcp", 3))
- transport = AST_TRANSPORT_TCP;
- else if (!strncasecmp(trans, "tls", 3))
- transport = AST_TRANSPORT_TLS;
- else {
- if (strncasecmp(trans, "udp", 3))
- ast_debug(1, "received contact with an invalid transport, '%s'\n", contact_number);
- /* This will assume UDP for all unknown transports */
- transport = AST_TRANSPORT_UDP;
- }
- }
- contact_number = remove_uri_parameters(contact_number);
-
- if (p->socket.tcptls_session) {
- ao2_ref(p->socket.tcptls_session, -1);
- p->socket.tcptls_session = NULL;
- } else if (p->socket.ws_session) {
- ast_websocket_unref(p->socket.ws_session);
- p->socket.ws_session = NULL;
- }
-
- set_socket_transport(&p->socket, transport);
-
- if (set_call_forward && ast_test_flag(&p->flags[0], SIP_PROMISCREDIR)) {
- char *host = NULL;
- if (!strncasecmp(contact_number, "sip:", 4))
- contact_number += 4;
- else if (!strncasecmp(contact_number, "sips:", 5))
- contact_number += 5;
- separator = strchr(contact_number, '/');
- if (separator)
- *separator = '\0';
- if ((host = strchr(contact_number, '@'))) {
- *host++ = '\0';
- ast_debug(2, "Found promiscuous redirection to 'SIP/%s::::%s@%s'\n", contact_number, sip_get_transport(transport), host);
- if (p->owner)
- ast_channel_call_forward_build(p->owner, "SIP/%s::::%s@%s", contact_number, sip_get_transport(transport), host);
- } else {
- ast_debug(2, "Found promiscuous redirection to 'SIP/::::%s@%s'\n", sip_get_transport(transport), contact_number);
- if (p->owner)
- ast_channel_call_forward_build(p->owner, "SIP/::::%s@%s", sip_get_transport(transport), contact_number);
- }
- } else {
- separator = strchr(contact, '@');
- if (separator) {
- *separator++ = '\0';
- domain = separator;
- } else {
- /* No username part */
- domain = contact;
- }
- separator = strchr(contact, '/'); /* WHEN do we hae a forward slash in the URI? */
- if (separator)
- *separator = '\0';
-
- if (!strncasecmp(contact_number, "sip:", 4))
- contact_number += 4;
- else if (!strncasecmp(contact_number, "sips:", 5))
- contact_number += 5;
- separator = strchr(contact_number, ';'); /* And username ; parameters? */
- if (separator)
- *separator = '\0';
- ast_uri_decode(contact_number, ast_uri_sip_user);
- if (set_call_forward) {
- ast_debug(2, "Received 302 Redirect to extension '%s' (domain %s)\n", contact_number, domain);
- if (p->owner) {
- pbx_builtin_setvar_helper(p->owner, "SIPDOMAIN", domain);
- ast_channel_call_forward_set(p->owner, contact_number);
- }
- }
- }
-
- /* We've gotten the number for the contact, now get the name */
-
- if (*contact == '\"') {
- contact_name = contact + 1;
- if (!(separator = (char *)find_closing_quote(contact_name, NULL))) {
- ast_log(LOG_NOTICE, "No closing quote on name in Contact header? %s\n", contact);
- }
- *separator = '\0';
- }
-
- if (name && !ast_strlen_zero(contact_name)) {
- *name = ast_strdup(contact_name);
- }
- if (number) {
- *number = ast_strdup(contact_number);
- }
-}
-
-/*!
- * \brief Check pending actions on SIP call
- *
- * \note both sip_pvt and sip_pvt's owner channel (if present)
- * must be locked for this function.
- *
- * \note Run by the sched thread.
- */
-static void check_pendings(struct sip_pvt *p)
-{
- if (ast_test_flag(&p->flags[0], SIP_PENDINGBYE)) {
- if (p->reinviteid > -1) {
- /* Outstanding p->reinviteid timeout, so wait... */
- return;
- }
- if (p->invitestate == INV_PROCEEDING || p->invitestate == INV_EARLY_MEDIA) {
- /* if we can't BYE, then this is really a pending CANCEL */
- p->invitestate = INV_CANCELLED;
- transmit_request(p, SIP_CANCEL, p->lastinvite, XMIT_RELIABLE, FALSE);
- /* If the cancel occurred on an initial invite, cancel the pending BYE */
- if (!ast_test_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED)) {
- ast_clear_flag(&p->flags[0], SIP_PENDINGBYE | SIP_NEEDREINVITE);
- }
- /* Actually don't destroy us yet, wait for the 487 on our original
- INVITE, but do set an autodestruct just in case we never get it. */
- } else {
- /* We have a pending outbound invite, don't send something
- * new in-transaction, unless it is a pending reinvite, then
- * by the time we are called here, we should probably just hang up. */
- if (p->pendinginvite && !p->ongoing_reinvite)
- return;
-
- if (p->owner) {
- ast_softhangup_nolock(p->owner, AST_SOFTHANGUP_DEV);
- }
- /* Perhaps there is an SD change INVITE outstanding */
- transmit_request_with_auth(p, SIP_BYE, 0, XMIT_RELIABLE, TRUE);
- ast_clear_flag(&p->flags[0], SIP_PENDINGBYE | SIP_NEEDREINVITE);
- }
- sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
- } else if (ast_test_flag(&p->flags[0], SIP_NEEDREINVITE)) {
- /* if we can't REINVITE, hold it for later */
- if (p->pendinginvite
- || p->invitestate == INV_CALLING
- || p->invitestate == INV_PROCEEDING
- || p->invitestate == INV_EARLY_MEDIA
- || p->waitid > -1) {
- ast_debug(2, "NOT Sending pending reinvite (yet) on '%s'\n", p->callid);
- } else {
- ast_debug(2, "Sending pending reinvite on '%s'\n", p->callid);
- /* Didn't get to reinvite yet, so do it now */
- transmit_reinvite_with_sdp(p, (p->t38.state == T38_LOCAL_REINVITE ? TRUE : FALSE), FALSE);
- ast_clear_flag(&p->flags[0], SIP_NEEDREINVITE);
- }
- }
-}
-
-/* Run by the sched thread. */
-static int __sched_check_pendings(const void *data)
-{
- struct sip_pvt *pvt = (void *) data;
- struct ast_channel *owner;
-
- owner = sip_pvt_lock_full(pvt);
- check_pendings(pvt);
- if (owner) {
- ast_channel_unlock(owner);
- ast_channel_unref(owner);
- }
- sip_pvt_unlock(pvt);
-
- dialog_unref(pvt, "Check pending actions action");
- return 0;
-}
-
-static void sched_check_pendings(struct sip_pvt *pvt)
-{
- dialog_ref(pvt, "Check pending actions action");
- if (ast_sched_add(sched, 0, __sched_check_pendings, pvt) < 0) {
- /* Uh Oh. Expect bad behavior. */
- dialog_unref(pvt, "Failed to schedule check pending actions action");
- }
-}
-
-/*!
- * \brief Reset the NEEDREINVITE flag after waiting when we get 491 on a Re-invite
- * to avoid race conditions between asterisk servers.
- *
- * \note Run by the sched thread.
- */
-static int sip_reinvite_retry(const void *data)
-{
- struct sip_pvt *p = (struct sip_pvt *) data;
- struct ast_channel *owner;
-
- owner = sip_pvt_lock_full(p);
- ast_set_flag(&p->flags[0], SIP_NEEDREINVITE);
- p->waitid = -1;
- check_pendings(p);
- sip_pvt_unlock(p);
- if (owner) {
- ast_channel_unlock(owner);
- ast_channel_unref(owner);
- }
- dialog_unref(p, "Schedule waitid complete");
- return 0;
-}
-
-/* Run by the sched thread. */
-static int __stop_reinvite_retry(const void *data)
-{
- struct sip_pvt *pvt = (void *) data;
-
- AST_SCHED_DEL_UNREF(sched, pvt->waitid,
- dialog_unref(pvt, "Stop scheduled waitid"));
- dialog_unref(pvt, "Stop reinvite retry action");
- return 0;
-}
-
-static void stop_reinvite_retry(struct sip_pvt *pvt)
-{
- dialog_ref(pvt, "Stop reinvite retry action");
- if (ast_sched_add(sched, 0, __stop_reinvite_retry, pvt) < 0) {
- /* Uh Oh. Expect bad behavior. */
- dialog_unref(pvt, "Failed to schedule stop reinvite retry action");
- }
-}
-
-/*!
- * \brief Handle authentication challenge for SIP UPDATE
- *
- * This function is only called upon the receipt of a 401/407 response to an UPDATE.
- */
-static void handle_response_update(struct sip_pvt *p, int resp, const char *rest, struct sip_request *req, uint32_t seqno)
-{
- if (p->options) {
- p->options->auth_type = (resp == 401 ? WWW_AUTH : PROXY_AUTH);
- }
- if ((p->authtries == MAX_AUTHTRIES) || do_proxy_auth(p, req, resp, SIP_UPDATE, 1)) {
- ast_log(LOG_NOTICE, "Failed to authenticate on UPDATE to '%s'\n", sip_get_header(&p->initreq, "From"));
- }
-}
-
-static void cc_handle_publish_error(struct sip_pvt *pvt, const int resp, struct sip_request *req, struct sip_epa_entry *epa_entry)
-{
- struct cc_epa_entry *cc_entry = epa_entry->instance_data;
- struct sip_monitor_instance *monitor_instance = ao2_callback(sip_monitor_instances, 0,
- find_sip_monitor_instance_by_suspension_entry, epa_entry);
- const char *min_expires;
-
- if (!monitor_instance) {
- ast_log(LOG_WARNING, "Can't find monitor_instance corresponding to epa_entry %p.\n", epa_entry);
- return;
- }
-
- if (resp != 423) {
- ast_cc_monitor_failed(cc_entry->core_id, monitor_instance->device_name,
- "Received error response to our PUBLISH");
- ao2_ref(monitor_instance, -1);
- return;
- }
-
- /* Allrighty, the other end doesn't like our Expires value. They think it's
- * too small, so let's see if they've provided a more sensible value. If they
- * haven't, then we'll just double our Expires value and see if they like that
- * instead.
- *
- * XXX Ideally this logic could be placed into its own function so that SUBSCRIBE,
- * PUBLISH, and REGISTER could all benefit from the same shared code.
- */
- min_expires = sip_get_header(req, "Min-Expires");
- if (ast_strlen_zero(min_expires)) {
- pvt->expiry *= 2;
- if (pvt->expiry < 0) {
- /* You dork! You overflowed! */
- ast_cc_monitor_failed(cc_entry->core_id, monitor_instance->device_name,
- "PUBLISH expiry overflowed");
- ao2_ref(monitor_instance, -1);
- return;
- }
- } else if (sscanf(min_expires, "%30d", &pvt->expiry) != 1) {
- ast_cc_monitor_failed(cc_entry->core_id, monitor_instance->device_name,
- "Min-Expires has non-numeric value");
- ao2_ref(monitor_instance, -1);
- return;
- }
- /* At this point, we have most certainly changed pvt->expiry, so try transmitting
- * again
- */
- transmit_invite(pvt, SIP_PUBLISH, FALSE, 0, NULL);
- ao2_ref(monitor_instance, -1);
-}
-
-static void handle_response_publish(struct sip_pvt *p, int resp, const char *rest, struct sip_request *req, uint32_t seqno)
-{
- struct sip_epa_entry *epa_entry = p->epa_entry;
- const char *etag = sip_get_header(req, "Sip-ETag");
-
- ast_assert(epa_entry != NULL);
-
- if (resp == 401 || resp == 407) {
- ast_string_field_set(p, theirtag, NULL);
- if (p->options) {
- p->options->auth_type = (resp == 401 ? WWW_AUTH : PROXY_AUTH);
- }
- if ((p->authtries == MAX_AUTHTRIES) || do_proxy_auth(p, req, resp, SIP_PUBLISH, 0)) {
- ast_log(LOG_NOTICE, "Failed to authenticate on PUBLISH to '%s'\n", sip_get_header(&p->initreq, "From"));
- pvt_set_needdestroy(p, "Failed to authenticate on PUBLISH");
- sip_alreadygone(p);
- }
- return;
- }
-
- if (resp == 501 || resp == 405) {
- mark_method_unallowed(&p->allowed_methods, SIP_PUBLISH);
- }
-
- if (resp == 200) {
- p->authtries = 0;
- /* If I've read section 6, item 6 of RFC 3903 correctly,
- * an ESC will only generate a new etag when it sends a 200 OK
- */
- if (!ast_strlen_zero(etag)) {
- ast_copy_string(epa_entry->entity_tag, etag, sizeof(epa_entry->entity_tag));
- }
- /* The nominal case. Everything went well. Everybody is happy.
- * Each EPA will have a specific action to take as a result of this
- * development, so ... callbacks!
- */
- if (epa_entry->static_data->handle_ok) {
- epa_entry->static_data->handle_ok(p, req, epa_entry);
- }
- } else {
- /* Rather than try to make individual callbacks for each error
- * type, there is just a single error callback. The callback
- * can distinguish between error messages and do what it needs to
- */
- if (epa_entry->static_data->handle_error) {
- epa_entry->static_data->handle_error(p, resp, req, epa_entry);
- }
- }
-}
-
-/*!
- * \internal
- * \brief Set hangup source and cause.
- *
- * \param p SIP private.
- * \param cause Hangup cause to queue. Zero if no cause.
- *
- * \pre p and p->owner are locked.
- */
-static void sip_queue_hangup_cause(struct sip_pvt *p, int cause)
-{
- struct ast_channel *owner = p->owner;
- const char *name = ast_strdupa(ast_channel_name(owner));
-
- /* Cannot hold any channel/private locks when calling. */
- ast_channel_ref(owner);
- ast_channel_unlock(owner);
- sip_pvt_unlock(p);
- ast_set_hangupsource(owner, name, 0);
- if (cause) {
- ast_queue_hangup_with_cause(owner, cause);
- } else {
- ast_queue_hangup(owner);
- }
- ast_channel_unref(owner);
-
- /* Relock things. */
- owner = sip_pvt_lock_full(p);
- if (owner) {
- ast_channel_unref(owner);
- }
-}
-
-/*! \brief Handle SIP response to INVITE dialogue */
-static void handle_response_invite(struct sip_pvt *p, int resp, const char *rest, struct sip_request *req, uint32_t seqno)
-{
- int outgoing = ast_test_flag(&p->flags[0], SIP_OUTGOING);
- int res = 0;
- int xmitres = 0;
- int reinvite = ast_test_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
- char *p_hdrval;
- int rtn;
- struct ast_party_connected_line connected;
- struct ast_set_party_connected_line update_connected;
-
- if (reinvite) {
- ast_debug(4, "SIP response %d to RE-invite on %s call %s\n", resp, outgoing ? "outgoing" : "incoming", p->callid);
- } else {
- ast_debug(4, "SIP response %d to standard invite\n", resp);
- }
-
- if (p->alreadygone) { /* This call is already gone */
- ast_debug(1, "Got response on call that is already terminated: %s (ignoring)\n", p->callid);
- return;
- }
-
- /* Acknowledge sequence number - This only happens on INVITE from SIP-call */
- /* Don't auto congest anymore since we've gotten something useful back */
- AST_SCHED_DEL_UNREF(sched, p->initid, dialog_unref(p, "when you delete the initid sched, you should dec the refcount for the stored dialog ptr"));
-
- /* RFC3261 says we must treat every 1xx response (but not 100)
- that we don't recognize as if it was 183.
- */
- if (resp > 100 && resp < 200 && resp!=101 && resp != 180 && resp != 181 && resp != 182 && resp != 183) {
- resp = 183;
- }
-
- /* For INVITE, treat all 2XX responses as we would a 200 response */
- if ((resp >= 200) && (resp < 300)) {
- resp = 200;
- }
-
- /* Any response between 100 and 199 is PROCEEDING */
- if (resp >= 100 && resp < 200 && p->invitestate == INV_CALLING) {
- p->invitestate = INV_PROCEEDING;
- }
-
- /* Final response, not 200 ? */
- if (resp >= 300 && (p->invitestate == INV_CALLING || p->invitestate == INV_PROCEEDING || p->invitestate == INV_EARLY_MEDIA )) {
- p->invitestate = INV_COMPLETED;
- }
-
- if ((resp >= 200 && reinvite)) {
- p->ongoing_reinvite = 0;
- stop_reinviteid(p);
- }
-
- /* Final response, clear out pending invite */
- if ((resp == 200 || resp >= 300) && p->pendinginvite && seqno == p->pendinginvite) {
- p->pendinginvite = 0;
- }
-
- /* If this is a response to our initial INVITE, we need to set what we can use
- * for this peer.
- */
- if (!reinvite) {
- set_pvt_allowed_methods(p, req);
- }
-
- switch (resp) {
- case 100: /* Trying */
- case 101: /* Dialog establishment */
- if (!req->ignore && p->invitestate != INV_CANCELLED) {
- sip_cancel_destroy(p);
- }
- sched_check_pendings(p);
- break;
-
- case 180: /* 180 Ringing */
- case 182: /* 182 Queued */
- if (!req->ignore && p->invitestate != INV_CANCELLED) {
- sip_cancel_destroy(p);
- }
- /* Store Route-set from provisional SIP responses so
- * early-dialog request can be routed properly
- * */
- parse_ok_contact(p, req);
- if (!reinvite) {
- build_route(p, req, 1, resp);
- }
- if (!req->ignore && p->owner) {
- if (get_rpid(p, req)) {
- /* Queue a connected line update */
- ast_party_connected_line_init(&connected);
- memset(&update_connected, 0, sizeof(update_connected));
-
- update_connected.id.number = 1;
- connected.id.number.valid = 1;
- connected.id.number.str = (char *) p->cid_num;
- connected.id.number.presentation = p->callingpres;
-
- update_connected.id.name = 1;
- connected.id.name.valid = 1;
- connected.id.name.str = (char *) p->cid_name;
- connected.id.name.presentation = p->callingpres;
-
- /* Invalidate any earlier private connected id representation */
- ast_set_party_id_all(&update_connected.priv);
-
- connected.id.tag = (char *) p->cid_tag;
- connected.source = AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER;
- ast_channel_queue_connected_line_update(p->owner, &connected,
- &update_connected);
- }
- sip_handle_cc(p, req, AST_CC_CCNR);
- ast_queue_control(p->owner, AST_CONTROL_RINGING);
- if (ast_channel_state(p->owner) != AST_STATE_UP) {
- ast_setstate(p->owner, AST_STATE_RINGING);
- if (p->relatedpeer) {
- ast_devstate_changed(AST_DEVICE_UNKNOWN, AST_DEVSTATE_NOT_CACHABLE, "SIP/%s", p->relatedpeer->name);
- }
- }
- }
- if (find_sdp(req)) {
- if (p->invitestate != INV_CANCELLED) {
- p->invitestate = INV_EARLY_MEDIA;
- }
- res = process_sdp(p, req, SDP_T38_NONE, FALSE);
- if (!req->ignore && p->owner) {
- /* Queue a progress frame only if we have SDP in 180 or 182 */
- ast_queue_control(p->owner, AST_CONTROL_PROGRESS);
- /* We have not sent progress, but we have been sent progress so enable early media */
- ast_set_flag(&p->flags[0], SIP_PROGRESS_SENT);
- }
- ast_rtp_instance_activate(p->rtp);
- }
- sched_check_pendings(p);
- break;
-
- case 181: /* Call Is Being Forwarded */
- if (!req->ignore && p->invitestate != INV_CANCELLED) {
- sip_cancel_destroy(p);
- }
- /* Store Route-set from provisional SIP responses so
- * early-dialog request can be routed properly
- * */
- parse_ok_contact(p, req);
- if (!reinvite) {
- build_route(p, req, 1, resp);
- }
- if (!req->ignore && p->owner) {
- struct ast_party_redirecting redirecting;
- struct ast_set_party_redirecting update_redirecting;
-
- ast_party_redirecting_init(&redirecting);
- memset(&update_redirecting, 0, sizeof(update_redirecting));
- change_redirecting_information(p, req, &redirecting, &update_redirecting,
- FALSE);
-
- /* Invalidate any earlier private redirecting id representations */
- ast_set_party_id_all(&update_redirecting.priv_orig);
- ast_set_party_id_all(&update_redirecting.priv_from);
- ast_set_party_id_all(&update_redirecting.priv_to);
-
- ast_channel_queue_redirecting_update(p->owner, &redirecting,
- &update_redirecting);
- ast_party_redirecting_free(&redirecting);
- sip_handle_cc(p, req, AST_CC_CCNR);
- }
- sched_check_pendings(p);
- break;
-
- case 183: /* Session progress */
- if (!req->ignore && p->invitestate != INV_CANCELLED) {
- sip_cancel_destroy(p);
- }
- /* Store Route-set from provisional SIP responses so
- * early-dialog request can be routed properly
- * */
- parse_ok_contact(p, req);
- if (!reinvite) {
- build_route(p, req, 1, resp);
- }
- if (!req->ignore && p->owner) {
- if (get_rpid(p, req)) {
- /* Queue a connected line update */
- ast_party_connected_line_init(&connected);
- memset(&update_connected, 0, sizeof(update_connected));
-
- update_connected.id.number = 1;
- connected.id.number.valid = 1;
- connected.id.number.str = (char *) p->cid_num;
- connected.id.number.presentation = p->callingpres;
-
- update_connected.id.name = 1;
- connected.id.name.valid = 1;
- connected.id.name.str = (char *) p->cid_name;
- connected.id.name.presentation = p->callingpres;
-
- /* Invalidate any earlier private connected id representation */
- ast_set_party_id_all(&update_connected.priv);
-
- connected.id.tag = (char *) p->cid_tag;
- connected.source = AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER;
- ast_channel_queue_connected_line_update(p->owner, &connected,
- &update_connected);
- }
- sip_handle_cc(p, req, AST_CC_CCNR);
- }
- if (find_sdp(req)) {
- if (p->invitestate != INV_CANCELLED) {
- p->invitestate = INV_EARLY_MEDIA;
- }
- res = process_sdp(p, req, SDP_T38_NONE, FALSE);
- if (!req->ignore && p->owner) {
- /* Queue a progress frame */
- ast_queue_control(p->owner, AST_CONTROL_PROGRESS);
- /* We have not sent progress, but we have been sent progress so enable early media */
- ast_set_flag(&p->flags[0], SIP_PROGRESS_SENT);
- }
- ast_rtp_instance_activate(p->rtp);
- } else {
- /* Alcatel PBXs are known to send 183s with no SDP after sending
- * a 100 Trying response. We're just going to treat this sort of thing
- * the same as we would treat a 180 Ringing
- */
- if (!req->ignore && p->owner) {
- ast_queue_control(p->owner, AST_CONTROL_RINGING);
- }
- }
- sched_check_pendings(p);
- break;
-
- case 200: /* 200 OK on invite - someone's answering our call */
- if (!req->ignore && p->invitestate != INV_CANCELLED) {
- sip_cancel_destroy(p);
- }
- p->authtries = 0;
- if (find_sdp(req)) {
- res = process_sdp(p, req, SDP_T38_ACCEPT, FALSE);
- if (res && !req->ignore) {
- if (!reinvite) {
- /* This 200 OK's SDP is not acceptable, so we need to ack, then hangup */
- /* For re-invites, we try to recover */
- ast_set_flag(&p->flags[0], SIP_PENDINGBYE);
- p->hangupcause = AST_CAUSE_BEARERCAPABILITY_NOTAVAIL;
- if (p->owner) {
- ast_channel_hangupcause_set(p->owner, AST_CAUSE_BEARERCAPABILITY_NOTAVAIL);
- sip_queue_hangup_cause(p, AST_CAUSE_BEARERCAPABILITY_NOTAVAIL);
- }
- }
- }
- ast_rtp_instance_activate(p->rtp);
- } else if (!reinvite) {
- struct ast_sockaddr remote_address = {{0,}};
-
- ast_rtp_instance_get_requested_target_address(p->rtp, &remote_address);
- if (ast_sockaddr_isnull(&remote_address) || (!ast_strlen_zero(p->theirprovtag) && strcmp(p->theirtag, p->theirprovtag))) {
- ast_log(LOG_WARNING, "Received response: \"200 OK\" from '%s' without SDP\n", p->relatedpeer->name);
- ast_set_flag(&p->flags[0], SIP_PENDINGBYE);
- ast_rtp_instance_activate(p->rtp);
- }
- }
-
- if (!req->ignore && p->owner) {
- int rpid_changed;
-
- rpid_changed = get_rpid(p, req);
- if (rpid_changed || !reinvite) {
- /* Queue a connected line update */
- ast_party_connected_line_init(&connected);
- memset(&update_connected, 0, sizeof(update_connected));
- if (rpid_changed
- || !ast_strlen_zero(p->cid_num)
- || (p->callingpres & AST_PRES_RESTRICTION) != AST_PRES_ALLOWED) {
- update_connected.id.number = 1;
- connected.id.number.valid = 1;
- connected.id.number.str = (char *) p->cid_num;
- connected.id.number.presentation = p->callingpres;
- }
- if (rpid_changed
- || !ast_strlen_zero(p->cid_name)
- || (p->callingpres & AST_PRES_RESTRICTION) != AST_PRES_ALLOWED) {
- update_connected.id.name = 1;
- connected.id.name.valid = 1;
- connected.id.name.str = (char *) p->cid_name;
- connected.id.name.presentation = p->callingpres;
- }
- if (update_connected.id.number || update_connected.id.name) {
- /* Invalidate any earlier private connected id representation */
- ast_set_party_id_all(&update_connected.priv);
-
- connected.id.tag = (char *) p->cid_tag;
- connected.source = AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER;
- ast_channel_queue_connected_line_update(p->owner, &connected,
- &update_connected);
- }
- }
- }
-
- /* Parse contact header for continued conversation */
- /* When we get 200 OK, we know which device (and IP) to contact for this call */
- /* This is important when we have a SIP proxy between us and the phone */
- if (outgoing) {
- update_call_counter(p, DEC_CALL_RINGING);
- parse_ok_contact(p, req);
- /* Save Record-Route for any later requests we make on this dialogue */
- if (!reinvite) {
- build_route(p, req, 1, resp);
- }
- if(set_address_from_contact(p)) {
- /* Bad contact - we don't know how to reach this device */
- /* We need to ACK, but then send a bye */
- if (sip_route_empty(&p->route) && !req->ignore) {
- ast_set_flag(&p->flags[0], SIP_PENDINGBYE);
- }
- }
-
- }
-
- if (!req->ignore && p->owner) {
- if (!reinvite && !res) {
- ast_queue_control(p->owner, AST_CONTROL_ANSWER);
- } else { /* RE-invite */
- if (p->t38.state == T38_DISABLED || p->t38.state == T38_REJECTED) {
- ast_queue_control(p->owner, AST_CONTROL_UPDATE_RTP_PEER);
- } else {
- ast_queue_frame(p->owner, &ast_null_frame);
- }
- }
- } else {
- /* It's possible we're getting an 200 OK after we've tried to disconnect
- by sending CANCEL */
- /* First send ACK, then send bye */
- if (!req->ignore) {
- ast_set_flag(&p->flags[0], SIP_PENDINGBYE);
- }
- }
-
- /* Check for Session-Timers related headers */
- if (st_get_mode(p, 0) != SESSION_TIMER_MODE_REFUSE) {
- p_hdrval = (char*)sip_get_header(req, "Session-Expires");
- if (!ast_strlen_zero(p_hdrval)) {
- /* UAS supports Session-Timers */
- enum st_refresher_param st_ref_param;
- int tmp_st_interval = 0;
- rtn = parse_session_expires(p_hdrval, &tmp_st_interval, &st_ref_param);
- if (rtn != 0) {
- ast_set_flag(&p->flags[0], SIP_PENDINGBYE);
- } else if (tmp_st_interval < st_get_se(p, FALSE)) {
- ast_log(LOG_WARNING, "Got Session-Expires less than local Min-SE in 200 OK, tearing down call\n");
- ast_set_flag(&p->flags[0], SIP_PENDINGBYE);
- }
- if (st_ref_param == SESSION_TIMER_REFRESHER_PARAM_UAC) {
- p->stimer->st_ref = SESSION_TIMER_REFRESHER_US;
- } else if (st_ref_param == SESSION_TIMER_REFRESHER_PARAM_UAS) {
- p->stimer->st_ref = SESSION_TIMER_REFRESHER_THEM;
- } else {
- ast_log(LOG_WARNING, "Unknown refresher on %s\n", p->callid);
- }
- if (tmp_st_interval) {
- p->stimer->st_interval = tmp_st_interval;
- }
- p->stimer->st_active = TRUE;
- p->stimer->st_active_peer_ua = TRUE;
- start_session_timer(p);
- } else {
- /* UAS doesn't support Session-Timers */
- if (st_get_mode(p, 0) == SESSION_TIMER_MODE_ORIGINATE) {
- p->stimer->st_ref = SESSION_TIMER_REFRESHER_US;
- p->stimer->st_active_peer_ua = FALSE;
- start_session_timer(p);
- }
- }
- }
-
-
- /* If I understand this right, the branch is different for a non-200 ACK only */
- p->invitestate = INV_TERMINATED;
- ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
- xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, TRUE);
- sched_check_pendings(p);
- break;
-
- case 407: /* Proxy authentication */
- case 401: /* Www auth */
- /* First we ACK */
- xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
- if (p->options) {
- p->options->auth_type = resp;
- }
-
- /* Then we AUTH */
- ast_string_field_set(p, theirtag, NULL); /* forget their old tag, so we don't match tags when getting response */
- if (!req->ignore) {
- if (p->authtries < MAX_AUTHTRIES) {
- p->invitestate = INV_CALLING;
- }
- if (p->authtries == MAX_AUTHTRIES || do_proxy_auth(p, req, resp, SIP_INVITE, 1)) {
- ast_log(LOG_NOTICE, "Failed to authenticate on INVITE to '%s'\n", sip_get_header(&p->initreq, "From"));
- pvt_set_needdestroy(p, "failed to authenticate on INVITE");
- sip_alreadygone(p);
- if (p->owner) {
- ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
- }
- }
- }
- break;
-
- case 403: /* Forbidden */
- /* First we ACK */
- xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
- ast_log(LOG_WARNING, "Received response: \"Forbidden\" from '%s'\n", sip_get_header(&p->initreq, "From"));
- if (!req->ignore && p->owner) {
- sip_queue_hangup_cause(p, hangup_sip2cause(resp));
- }
- break;
-
- case 400: /* Bad Request */
- case 414: /* Bad request URI */
- case 493: /* Undecipherable */
- case 404: /* Not found */
- xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
- if (p->owner && !req->ignore) {
- sip_queue_hangup_cause(p, hangup_sip2cause(resp));
- }
- break;
-
- case 481: /* Call leg does not exist */
- /* Could be REFER caused INVITE with replaces */
- ast_log(LOG_WARNING, "Re-invite to non-existing call leg on other UA. SIP dialog '%s'. Giving up.\n", p->callid);
- xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
- if (p->owner) {
- ast_queue_hangup_with_cause(p->owner, hangup_sip2cause(resp));
- }
- break;
-
- case 422: /* Session-Timers: Session interval too small */
- xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
- ast_string_field_set(p, theirtag, NULL);
- p->invitestate = INV_CALLING;
- proc_422_rsp(p, req);
- break;
-
- case 428: /* Use identity header - rfc 4474 - not supported by Asterisk yet */
- xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
- append_history(p, "Identity", "SIP identity is required. Not supported by Asterisk.");
- ast_log(LOG_WARNING, "SIP identity required by proxy. SIP dialog '%s'. Giving up.\n", p->callid);
- if (p->owner && !req->ignore) {
- ast_queue_hangup_with_cause(p->owner, hangup_sip2cause(resp));
- }
- break;
-
- case 480: /* Temporarily unavailable. */
- /* RFC 3261 encourages setting the reason phrase to something indicative
- * of why the endpoint is not available. We will make this readable via the
- * redirecting reason.
- */
- xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
- append_history(p, "TempUnavailable", "Endpoint is temporarily unavailable.");
- if (p->owner && !req->ignore) {
- struct ast_party_redirecting redirecting;
- struct ast_set_party_redirecting update_redirecting;
- char *quoted_rest = ast_alloca(strlen(rest) + 3);
-
- ast_party_redirecting_set_init(&redirecting, ast_channel_redirecting(p->owner));
- memset(&update_redirecting, 0, sizeof(update_redirecting));
-
- redirecting.reason.code = ast_redirecting_reason_parse(rest);
- if (redirecting.reason.code < 0) {
- sprintf(quoted_rest, "\"%s\"", rest);/* Safe */
-
- redirecting.reason.code = AST_REDIRECTING_REASON_UNKNOWN;
- redirecting.reason.str = quoted_rest;
- } else {
- redirecting.reason.str = "";
- }
-
- ast_channel_queue_redirecting_update(p->owner, &redirecting, &update_redirecting);
-
- ast_queue_control(p->owner, AST_CONTROL_BUSY);
- }
- break;
- case 487: /* Cancelled transaction */
- /* We have sent CANCEL on an outbound INVITE
- This transaction is already scheduled to be killed by sip_hangup().
- */
- xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
- if (p->owner && !req->ignore) {
- ast_queue_hangup_with_cause(p->owner, AST_CAUSE_NORMAL_CLEARING);
- append_history(p, "Hangup", "Got 487 on CANCEL request from us. Queued AST hangup request");
- } else if (!req->ignore) {
- update_call_counter(p, DEC_CALL_LIMIT);
- append_history(p, "Hangup", "Got 487 on CANCEL request from us on call without owner. Killing this dialog.");
- }
- sched_check_pendings(p);
- sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
- break;
- case 415: /* Unsupported media type */
- case 488: /* Not acceptable here */
- case 606: /* Not Acceptable */
- xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
- if (p->udptl && p->t38.state == T38_LOCAL_REINVITE) {
- change_t38_state(p, T38_REJECTED);
- /* Try to reset RTP timers */
- /* XXX Why is this commented away??? */
- //ast_rtp_set_rtptimers_onhold(p->rtp);
-
- /* Trigger a reinvite back to audio */
- transmit_reinvite_with_sdp(p, FALSE, FALSE);
- } else {
- /* We can't set up this call, so give up */
- if (p->owner && !req->ignore) {
- ast_queue_hangup_with_cause(p->owner, hangup_sip2cause(resp));
- }
- }
- break;
- case 491: /* Pending */
- xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
- if (p->owner && !req->ignore) {
- if (ast_channel_state(p->owner) != AST_STATE_UP) {
- ast_queue_hangup_with_cause(p->owner, hangup_sip2cause(resp));
- } else {
- /* This is a re-invite that failed. */
- /* Reset the flag after a while
- */
- int wait;
-
- /* RFC 3261, if owner of call, wait between 2.1 to 4 seconds,
- * if not owner of call, wait 0 to 2 seconds */
- if (p->outgoing_call) {
- wait = 2100 + ast_random() % 2000;
- } else {
- wait = ast_random() % 2000;
- }
- dialog_ref(p, "Schedule waitid for sip_reinvite_retry.");
- p->waitid = ast_sched_add(sched, wait, sip_reinvite_retry, p);
- if (p->waitid < 0) {
- /* Uh Oh. Expect bad behavior. */
- dialog_ref(p, "Failed to schedule waitid");
- }
- ast_debug(2, "Reinvite race. Scheduled sip_reinvite_retry in %d secs in handle_response_invite (waitid %d, dialog '%s')\n",
- wait, p->waitid, p->callid);
- }
- }
- break;
-
- case 408: /* Request timeout */
- case 405: /* Not allowed */
- case 501: /* Not implemented */
- xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
- if (p->owner) {
- ast_queue_hangup_with_cause(p->owner, hangup_sip2cause(resp));
- }
- break;
- }
- if (xmitres == XMIT_ERROR) {
- ast_log(LOG_WARNING, "Could not transmit message in dialog %s\n", p->callid);
- }
-}
-
-/*! \brief Handle SIP response in NOTIFY transaction
- We've sent a NOTIFY, now handle responses to it
- */
-static void handle_response_notify(struct sip_pvt *p, int resp, const char *rest, struct sip_request *req, uint32_t seqno)
-{
- switch (resp) {
- case 200: /* Notify accepted */
- /* They got the notify, this is the end */
- if (p->owner) {
- if (p->refer) {
- ast_log(LOG_NOTICE, "Got OK on REFER Notify message\n");
- } else {
- ast_log(LOG_WARNING, "Notify answer on an owned channel? - %s\n", ast_channel_name(p->owner));
- }
- } else {
- if (p->subscribed == NONE && !p->refer) {
- ast_debug(4, "Got 200 accepted on NOTIFY %s\n", p->callid);
- pvt_set_needdestroy(p, "received 200 response");
- }
- if (ast_test_flag(&p->flags[1], SIP_PAGE2_STATECHANGEQUEUE)) {
- struct state_notify_data data = {
- .state = p->laststate,
- .device_state_info = p->last_device_state_info,
- .presence_state = p->last_presence_state,
- .presence_subtype = p->last_presence_subtype,
- .presence_message = p->last_presence_message,
- };
- /* Ready to send the next state we have on queue */
- ast_clear_flag(&p->flags[1], SIP_PAGE2_STATECHANGEQUEUE);
- extensionstate_update(p->context, p->exten, &data, p, TRUE);
- }
- }
- break;
- case 401: /* Not www-authorized on SIP method */
- case 407: /* Proxy auth */
- if (!p->notify) {
- break; /* Only device notify can use NOTIFY auth */
- }
- ast_string_field_set(p, theirtag, NULL);
- if (ast_strlen_zero(p->authname)) {
- ast_log(LOG_WARNING, "Asked to authenticate NOTIFY to %s but we have no matching peer or realm auth!\n", ast_sockaddr_stringify(&p->recv));
- pvt_set_needdestroy(p, "unable to authenticate NOTIFY");
- }
- if (p->authtries > 1 || do_proxy_auth(p, req, resp, SIP_NOTIFY, 0)) {
- ast_log(LOG_NOTICE, "Failed to authenticate on NOTIFY to '%s'\n", sip_get_header(&p->initreq, "From"));
- pvt_set_needdestroy(p, "failed to authenticate NOTIFY");
- }
- break;
- case 481: /* Call leg does not exist */
- pvt_set_needdestroy(p, "Received 481 response for NOTIFY");
- break;
- }
-}
-
-/*! \brief Handle SIP response in SUBSCRIBE transaction */
-static void handle_response_subscribe(struct sip_pvt *p, int resp, const char *rest, struct sip_request *req, uint32_t seqno)
-{
- if (p->subscribed == CALL_COMPLETION) {
- struct sip_monitor_instance *monitor_instance;
-
- if (resp < 300) {
- return;
- }
-
- /* Final failure response received. */
- monitor_instance = ao2_callback(sip_monitor_instances, 0,
- find_sip_monitor_instance_by_subscription_pvt, p);
- if (monitor_instance) {
- ast_cc_monitor_failed(monitor_instance->core_id,
- monitor_instance->device_name,
- "Received error response to our SUBSCRIBE");
- ao2_ref(monitor_instance, -1);
- }
- return;
- }
-
- if (p->subscribed != MWI_NOTIFICATION) {
- return;
- }
- if (!p->mwi) {
- return;
- }
-
- switch (resp) {
- case 200: /* Subscription accepted */
- ast_debug(3, "Got 200 OK on subscription for MWI\n");
- set_pvt_allowed_methods(p, req);
- if (p->options) {
- if (p->options->outboundproxy) {
- ao2_ref(p->options->outboundproxy, -1);
- }
- ast_free(p->options);
- p->options = NULL;
- }
- p->mwi->subscribed = 1;
- start_mwi_subscription(p->mwi, mwi_expiry * 1000);
- break;
- case 401:
- case 407:
- ast_string_field_set(p, theirtag, NULL);
- if (p->authtries > 1 || do_proxy_auth(p, req, resp, SIP_SUBSCRIBE, 0)) {
- ast_log(LOG_NOTICE, "Failed to authenticate on SUBSCRIBE to '%s'\n", sip_get_header(&p->initreq, "From"));
- p->mwi->call = NULL;
- ao2_t_ref(p->mwi, -1, "failed to authenticate SUBSCRIBE");
- pvt_set_needdestroy(p, "failed to authenticate SUBSCRIBE");
- }
- break;
- case 403:
- transmit_response_with_date(p, "200 OK", req);
- ast_log(LOG_WARNING, "Authentication failed while trying to subscribe for MWI.\n");
- p->mwi->call = NULL;
- ao2_t_ref(p->mwi, -1, "received 403 response");
- pvt_set_needdestroy(p, "received 403 response");
- sip_alreadygone(p);
- break;
- case 404:
- ast_log(LOG_WARNING, "Subscription failed for MWI. The remote side said that a mailbox may not have been configured.\n");
- p->mwi->call = NULL;
- ao2_t_ref(p->mwi, -1, "received 404 response");
- pvt_set_needdestroy(p, "received 404 response");
- break;
- case 481:
- ast_log(LOG_WARNING, "Subscription failed for MWI. The remote side said that our dialog did not exist.\n");
- p->mwi->call = NULL;
- ao2_t_ref(p->mwi, -1, "received 481 response");
- pvt_set_needdestroy(p, "received 481 response");
- break;
-
- case 400: /* Bad Request */
- case 414: /* Request URI too long */
- case 493: /* Undecipherable */
- case 500:
- case 501:
- ast_log(LOG_WARNING, "Subscription failed for MWI. The remote side may have suffered a heart attack.\n");
- p->mwi->call = NULL;
- ao2_t_ref(p->mwi, -1, "received 500/501 response");
- pvt_set_needdestroy(p, "received serious error (500/501/493/414/400) response");
- break;
- }
-}
-
-/*! \brief Handle SIP response in REFER transaction
- We've sent a REFER, now handle responses to it
- */
-static void handle_response_refer(struct sip_pvt *p, int resp, const char *rest, struct sip_request *req, uint32_t seqno)
-{
- enum ast_control_transfer message = AST_TRANSFER_FAILED;
-
- /* If no refer structure exists, then do nothing */
- if (!p->refer)
- return;
-
- switch (resp) {
- case 202: /* Transfer accepted */
- /* We need to do something here */
- /* The transferee is now sending INVITE to target */
- p->refer->status = REFER_ACCEPTED;
- /* Now wait for next message */
- ast_debug(3, "Got 202 accepted on transfer\n");
- /* We should hang along, waiting for NOTIFY's here */
- break;
-
- case 401: /* Not www-authorized on SIP method */
- case 407: /* Proxy auth */
- if (ast_strlen_zero(p->authname)) {
- ast_log(LOG_WARNING, "Asked to authenticate REFER to %s but we have no matching peer or realm auth!\n",
- ast_sockaddr_stringify(&p->recv));
- if (p->owner) {
- ast_queue_control_data(p->owner, AST_CONTROL_TRANSFER, &message, sizeof(message));
- }
- pvt_set_needdestroy(p, "unable to authenticate REFER");
- }
- if (p->authtries > 1 || do_proxy_auth(p, req, resp, SIP_REFER, 0)) {
- ast_log(LOG_NOTICE, "Failed to authenticate on REFER to '%s'\n", sip_get_header(&p->initreq, "From"));
- p->refer->status = REFER_NOAUTH;
- if (p->owner) {
- ast_queue_control_data(p->owner, AST_CONTROL_TRANSFER, &message, sizeof(message));
- }
- pvt_set_needdestroy(p, "failed to authenticate REFER");
- }
- break;
-
- case 405: /* Method not allowed */
- /* Return to the current call onhold */
- /* Status flag needed to be reset */
- ast_log(LOG_NOTICE, "SIP transfer to %s failed, REFER not allowed. \n", p->refer->refer_to);
- pvt_set_needdestroy(p, "received 405 response");
- p->refer->status = REFER_FAILED;
- if (p->owner) {
- ast_queue_control_data(p->owner, AST_CONTROL_TRANSFER, &message, sizeof(message));
- }
- break;
-
- case 481: /* Call leg does not exist */
-
- /* A transfer with Replaces did not work */
- /* OEJ: We should Set flag, cancel the REFER, go back
- to original call - but right now we can't */
- ast_log(LOG_WARNING, "Remote host can't match REFER request to call '%s'. Giving up.\n", p->callid);
- if (p->owner)
- ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
- pvt_set_needdestroy(p, "received 481 response");
- break;
-
- case 500: /* Server error */
- case 501: /* Method not implemented */
- /* Return to the current call onhold */
- /* Status flag needed to be reset */
- ast_log(LOG_NOTICE, "SIP transfer to %s failed, call miserably fails. \n", p->refer->refer_to);
- pvt_set_needdestroy(p, "received 500/501 response");
- p->refer->status = REFER_FAILED;
- if (p->owner) {
- ast_queue_control_data(p->owner, AST_CONTROL_TRANSFER, &message, sizeof(message));
- }
- break;
- case 603: /* Transfer declined */
- ast_log(LOG_NOTICE, "SIP transfer to %s declined, call miserably fails. \n", p->refer->refer_to);
- p->refer->status = REFER_FAILED;
- pvt_set_needdestroy(p, "received 603 response");
- if (p->owner) {
- ast_queue_control_data(p->owner, AST_CONTROL_TRANSFER, &message, sizeof(message));
- }
- break;
- default:
- /* We should treat unrecognized 9xx as 900. 400 is actually
- specified as a possible response, but any 4-6xx is
- theoretically possible. */
-
- if (resp < 299) { /* 1xx cases don't get here */
- ast_log(LOG_WARNING, "SIP transfer to %s had unexpected 2xx response (%d), confusion is possible. \n", p->refer->refer_to, resp);
- } else {
- ast_log(LOG_WARNING, "SIP transfer to %s with response (%d). \n", p->refer->refer_to, resp);
- }
-
- p->refer->status = REFER_FAILED;
- pvt_set_needdestroy(p, "received failure response");
- if (p->owner) {
- ast_queue_control_data(p->owner, AST_CONTROL_TRANSFER, &message, sizeof(message));
- }
- break;
- }
-}
-
-/*! \brief Handle responses on REGISTER to services */
-static int handle_response_register(struct sip_pvt *p, int resp, const char *rest, struct sip_request *req, uint32_t seqno)
-{
- int expires, expires_ms;
- struct sip_registry *r;
- r = p->registry;
-
- switch (resp) {
- case 401: /* Unauthorized */
- if (p->authtries == MAX_AUTHTRIES || do_register_auth(p, req, resp)) {
- ast_log(LOG_NOTICE, "Failed to authenticate on REGISTER to '%s@%s' (Tries %d)\n", p->registry->username, p->registry->hostname, p->authtries);
- pvt_set_needdestroy(p, "failed to authenticate REGISTER");
- }
- break;
- case 403: /* Forbidden */
- if (global_reg_retry_403) {
- ast_log(LOG_NOTICE, "Treating 403 response to REGISTER as non-fatal for %s@%s\n",
- p->registry->username, p->registry->hostname);
- ast_string_field_set(r, nonce, "");
- ast_string_field_set(p, nonce, "");
- break;
- }
- ast_log(LOG_WARNING, "Forbidden - wrong password on authentication for REGISTER for '%s' to '%s'\n", p->registry->username, p->registry->hostname);
- r->regstate = REG_STATE_NOAUTH;
- stop_register_timeout(r);
- sip_publish_registry(r->username, r->hostname, regstate2str(r->regstate));
- pvt_set_needdestroy(p, "received 403 response");
- break;
- case 404: /* Not found */
- ast_log(LOG_WARNING, "Got 404 Not found on SIP register to service %s@%s, giving up\n", p->registry->username, p->registry->hostname);
- pvt_set_needdestroy(p, "received 404 response");
- if (r->call)
- r->call = dialog_unref(r->call, "unsetting registry->call pointer-- case 404");
- r->regstate = REG_STATE_REJECTED;
- stop_register_timeout(r);
- sip_publish_registry(r->username, r->hostname, regstate2str(r->regstate));
- break;
- case 407: /* Proxy auth */
- if (p->authtries == MAX_AUTHTRIES || do_register_auth(p, req, resp)) {
- ast_log(LOG_NOTICE, "Failed to authenticate on REGISTER to '%s' (tries '%d')\n", sip_get_header(&p->initreq, "From"), p->authtries);
- pvt_set_needdestroy(p, "failed to authenticate REGISTER");
- }
- break;
- case 408: /* Request timeout */
- /* Got a timeout response, so reset the counter of failed responses */
- if (r) {
- r->regattempts = 0;
- } else {
- ast_log(LOG_WARNING, "Got a 408 response to our REGISTER on call %s after we had destroyed the registry object\n", p->callid);
- }
- break;
- case 423: /* Interval too brief */
- r->expiry = atoi(sip_get_header(req, "Min-Expires"));
- ast_log(LOG_WARNING, "Got 423 Interval too brief for service %s@%s, minimum is %d seconds\n", p->registry->username, p->registry->hostname, r->expiry);
- if (r->call) {
- r->call = dialog_unref(r->call, "unsetting registry->call pointer-- case 423");
- pvt_set_needdestroy(p, "received 423 response");
- }
- if (r->expiry > max_expiry) {
- ast_log(LOG_WARNING, "Required expiration time from %s@%s is too high, giving up\n", p->registry->username, p->registry->hostname);
- r->expiry = r->configured_expiry;
- r->regstate = REG_STATE_REJECTED;
- stop_register_timeout(r);
- } else {
- r->regstate = REG_STATE_UNREGISTERED;
- transmit_register(r, SIP_REGISTER, NULL, NULL);
- }
- sip_publish_registry(r->username, r->hostname, regstate2str(r->regstate));
- break;
- case 400: /* Bad request */
- case 414: /* Request URI too long */
- case 493: /* Undecipherable */
- case 479: /* Kamailio/OpenSIPS: Not able to process the URI - address is wrong in register*/
- ast_log(LOG_WARNING, "Got error %d on register to %s@%s, giving up (check config)\n", resp, p->registry->username, p->registry->hostname);
- pvt_set_needdestroy(p, "received 4xx response");
- if (r->call)
- r->call = dialog_unref(r->call, "unsetting registry->call pointer-- case 4xx");
- r->regstate = REG_STATE_REJECTED;
- stop_register_timeout(r);
- sip_publish_registry(r->username, r->hostname, regstate2str(r->regstate));
- break;
- case 200: /* 200 OK */
- if (!r) {
- ast_log(LOG_WARNING, "Got 200 OK on REGISTER, but there isn't a registry entry for '%s' (we probably already got the OK)\n", S_OR(p->peername, p->username));
- pvt_set_needdestroy(p, "received erroneous 200 response");
- return 0;
- }
-
- ast_debug(1, "Registration successful\n");
- if (r->timeout > -1) {
- ast_debug(1, "Cancelling timeout %d\n", r->timeout);
- }
- r->regstate = REG_STATE_REGISTERED;
- stop_register_timeout(r);
- r->regtime = ast_tvnow(); /* Reset time of last successful registration */
- sip_publish_registry(r->username, r->hostname, regstate2str(r->regstate));
- r->regattempts = 0;
- if (r->call)
- r->call = dialog_unref(r->call, "unsetting registry->call pointer-- case 200");
- ao2_t_replace(p->registry, NULL, "unref registry entry p->registry");
-
- /* destroy dialog now to avoid interference with next register */
- pvt_set_needdestroy(p, "Registration successfull");
-
- /* set us up for re-registering
- * figure out how long we got registered for
- * according to section 6.13 of RFC, contact headers override
- * expires headers, so check those first */
- expires = 0;
-
- /* XXX todo: try to save the extra call */
- if (!ast_strlen_zero(sip_get_header(req, "Contact"))) {
- const char *contact = NULL;
- const char *tmptmp = NULL;
- int start = 0;
- for(;;) {
- contact = __get_header(req, "Contact", &start);
- /* this loop ensures we get a contact header about our register request */
- if(!ast_strlen_zero(contact)) {
- if( (tmptmp=strstr(contact, p->our_contact))) {
- contact=tmptmp;
- break;
- }
- } else
- break;
- }
- tmptmp = strcasestr(contact, "expires=");
- if (tmptmp) {
- if (sscanf(tmptmp + 8, "%30d", &expires) != 1) {
- expires = 0;
- }
- }
-
- }
- if (!expires)
- expires=atoi(sip_get_header(req, "expires"));
- if (!expires)
- expires=default_expiry;
-
- expires_ms = expires * 1000;
- if (expires <= EXPIRY_GUARD_LIMIT)
- expires_ms -= MAX((expires_ms * EXPIRY_GUARD_PCT), EXPIRY_GUARD_MIN);
- else
- expires_ms -= EXPIRY_GUARD_SECS * 1000;
- if (sipdebug)
- ast_log(LOG_NOTICE, "Outbound Registration: Expiry for %s is %d sec (Scheduling reregistration in %d s)\n", r->hostname, expires, expires_ms/1000);
-
- r->refresh= (int) expires_ms / 1000;
-
- /* Schedule re-registration before we expire */
- start_reregister_timeout(r, expires_ms);
- }
- return 1;
-}
-
-/*! \brief Handle qualification responses (OPTIONS) */
-static void handle_response_peerpoke(struct sip_pvt *p, int resp, struct sip_request *req)
-{
- struct sip_peer *peer = /* sip_ref_peer( */ p->relatedpeer /* , "bump refcount on p, as it is being used in this function(handle_response_peerpoke)")*/ ; /* hope this is already refcounted! */
- int statechanged, is_reachable, was_reachable;
- int pingtime = ast_tvdiff_ms(ast_tvnow(), peer->ps);
-
- /*
- * Compute the response time to a ping (goes in peer->lastms.)
- * -1 means did not respond, 0 means unknown,
- * 1..maxms is a valid response, >maxms means late response.
- */
- if (pingtime < 1) { /* zero = unknown, so round up to 1 */
- pingtime = 1;
- }
-
- if (!peer->maxms) { /* this should never happens */
- pvt_set_needdestroy(p, "got OPTIONS response but qualify is not enabled");
- return;
- }
-
- /* Now determine new state and whether it has changed.
- * Use some helper variables to simplify the writing
- * of the expressions.
- */
- was_reachable = peer->lastms > 0 && peer->lastms <= peer->maxms;
- is_reachable = pingtime <= peer->maxms;
- statechanged = peer->lastms == 0 /* yes, unknown before */
- || was_reachable != is_reachable;
-
- peer->lastms = pingtime;
- peer->call = dialog_unref(peer->call, "unref dialog peer->call");
- if (statechanged) {
- const char *s = is_reachable ? "Reachable" : "Lagged";
- char str_lastms[20];
-
- snprintf(str_lastms, sizeof(str_lastms), "%d", pingtime);
-
- ast_log(LOG_NOTICE, "Peer '%s' is now %s. (%dms / %dms)\n",
- peer->name, s, pingtime, peer->maxms);
- ast_devstate_changed(AST_DEVICE_UNKNOWN, AST_DEVSTATE_CACHABLE, "SIP/%s", peer->name);
- if (sip_cfg.peer_rtupdate) {
- ast_update_realtime(ast_check_realtime("sipregs") ? "sipregs" : "sippeers", "name", peer->name, "lastms", str_lastms, SENTINEL);
- }
- if (peer->endpoint) {
- RAII_VAR(struct ast_json *, blob, NULL, ast_json_unref);
- ast_endpoint_set_state(peer->endpoint, AST_ENDPOINT_ONLINE);
- blob = ast_json_pack("{s: s, s: i}",
- "peer_status", s,
- "time", pingtime);
- ast_endpoint_blob_publish(peer->endpoint, ast_endpoint_state_type(), blob);
- }
-
- if (is_reachable && sip_cfg.regextenonqualify) {
- register_peer_exten(peer, TRUE);
- }
- }
-
- pvt_set_needdestroy(p, "got OPTIONS response");
-
- /* Try again eventually */
- AST_SCHED_REPLACE_UNREF(peer->pokeexpire, sched,
- is_reachable ? peer->qualifyfreq : DEFAULT_FREQ_NOTOK,
- sip_poke_peer_s, peer,
- sip_unref_peer(_data, "removing poke peer ref"),
- sip_unref_peer(peer, "removing poke peer ref"),
- sip_ref_peer(peer, "adding poke peer ref"));
-}
-
-/*!
- * \internal
- * \brief Handle responses to INFO messages
- *
- * \note The INFO method MUST NOT change the state of calls or
- * related sessions (RFC 2976).
- */
-static void handle_response_info(struct sip_pvt *p, int resp, const char *rest, struct sip_request *req, uint32_t seqno)
-{
- int sipmethod = SIP_INFO;
-
- switch (resp) {
- case 401: /* Not www-authorized on SIP method */
- case 407: /* Proxy auth required */
- ast_log(LOG_WARNING, "Host '%s' requests authentication (%d) for '%s'\n",
- ast_sockaddr_stringify(&p->sa), resp, sip_methods[sipmethod].text);
- break;
- case 405: /* Method not allowed */
- case 501: /* Not Implemented */
- mark_method_unallowed(&p->allowed_methods, sipmethod);
- if (p->relatedpeer) {
- mark_method_allowed(&p->relatedpeer->disallowed_methods, sipmethod);
- }
- ast_log(LOG_WARNING, "Host '%s' does not implement '%s'\n",
- ast_sockaddr_stringify(&p->sa), sip_methods[sipmethod].text);
- break;
- default:
- if (300 <= resp && resp < 700) {
- ast_verb(3, "Got SIP %s response %d \"%s\" back from host '%s'\n",
- sip_methods[sipmethod].text, resp, rest, ast_sockaddr_stringify(&p->sa));
- }
- break;
- }
-}
-
-/*!
- * \internal
- * \brief Handle auth requests to a MESSAGE request
- * \retval TRUE if authentication failed.
- */
-static int do_message_auth(struct sip_pvt *p, int resp, const char *rest, struct sip_request *req, uint32_t seqno)
-{
- char *header;
- char *respheader;
- char digest[1024];
-
- if (p->options) {
- p->options->auth_type = (resp == 401 ? WWW_AUTH : PROXY_AUTH);
- }
-
- if (p->authtries == MAX_AUTHTRIES) {
- ast_log(LOG_NOTICE, "Failed to authenticate MESSAGE with host '%s'\n",
- ast_sockaddr_stringify(&p->sa));
- return -1;
- }
-
- ++p->authtries;
- sip_auth_headers((resp == 401 ? WWW_AUTH : PROXY_AUTH), &header, &respheader);
- memset(digest, 0, sizeof(digest));
- if (reply_digest(p, req, header, SIP_MESSAGE, digest, sizeof(digest))) {
- /* There's nothing to use for authentication */
- ast_debug(1, "Nothing to use for MESSAGE authentication\n");
- return -1;
- }
-
- if (p->do_history) {
- append_history(p, "MessageAuth", "Try: %d", p->authtries);
- }
-
- transmit_message(p, 0, 1);
- return 0;
-}
-
-/*!
- * \internal
- * \brief Handle responses to MESSAGE messages
- *
- * \note The MESSAGE method should not change the state of calls
- * or related sessions if associated with a dialog. (Implied by
- * RFC 3428 Section 2).
- */
-static void handle_response_message(struct sip_pvt *p, int resp, const char *rest, struct sip_request *req, uint32_t seqno)
-{
- int sipmethod = SIP_MESSAGE;
- int in_dialog = ast_test_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
-
- switch (resp) {
- case 401: /* Not www-authorized on SIP method */
- case 407: /* Proxy auth required */
- if (do_message_auth(p, resp, rest, req, seqno) && !in_dialog) {
- pvt_set_needdestroy(p, "MESSAGE authentication failed");
- }
- break;
- case 405: /* Method not allowed */
- case 501: /* Not Implemented */
- mark_method_unallowed(&p->allowed_methods, sipmethod);
- if (p->relatedpeer) {
- mark_method_allowed(&p->relatedpeer->disallowed_methods, sipmethod);
- }
- ast_log(LOG_WARNING, "Host '%s' does not implement '%s'\n",
- ast_sockaddr_stringify(&p->sa), sip_methods[sipmethod].text);
- if (!in_dialog) {
- pvt_set_needdestroy(p, "MESSAGE not implemented or allowed");
- }
- break;
- default:
- if (100 <= resp && resp < 200) {
- /* Must allow provisional responses for out-of-dialog requests. */
- } else if (200 <= resp && resp < 300) {
- p->authtries = 0; /* Reset authentication counter */
- if (!in_dialog) {
- pvt_set_needdestroy(p, "MESSAGE delivery accepted");
- }
- } else if (300 <= resp && resp < 700) {
- ast_verb(3, "Got SIP %s response %d \"%s\" back from host '%s'\n",
- sip_methods[sipmethod].text, resp, rest, ast_sockaddr_stringify(&p->sa));
- if (!in_dialog) {
- pvt_set_needdestroy(p, (300 <= resp && resp < 600)
- ? "MESSAGE delivery failed" : "MESSAGE delivery refused");
- }
- }
- break;
- }
-}
-
-/*! \brief Immediately stop RTP, VRTP and UDPTL as applicable */
-static void stop_media_flows(struct sip_pvt *p)
-{
- /* Immediately stop RTP, VRTP and UDPTL as applicable */
- if (p->rtp)
- ast_rtp_instance_stop(p->rtp);
- if (p->vrtp)
- ast_rtp_instance_stop(p->vrtp);
- if (p->trtp)
- ast_rtp_instance_stop(p->trtp);
- if (p->udptl)
- ast_udptl_stop(p->udptl);
-}
-
-/*! \brief Handle SIP response in dialogue
- \note only called by handle_incoming */
-static void handle_response(struct sip_pvt *p, int resp, const char *rest, struct sip_request *req, uint32_t seqno)
-{
- struct ast_channel *owner;
- int sipmethod;
- const char *c = sip_get_header(req, "Cseq");
- /* GCC 4.2 complains if I try to cast c as a char * when passing it to ast_skip_nonblanks, so make a copy of it */
- char *c_copy = ast_strdupa(c);
- /* Skip the Cseq and its subsequent spaces */
- const char *msg = ast_skip_blanks(ast_skip_nonblanks(c_copy));
- int ack_res = FALSE;
-
- if (!msg)
- msg = "";
-
- sipmethod = find_sip_method(msg);
- owner = p->owner;
- if (owner) {
- ast_channel_hangupcause_set(owner, 0);
- if (use_reason_header(p, req)) {
- /* Use the SIP cause */
- ast_channel_hangupcause_set(owner, hangup_sip2cause(resp));
- }
- }
-
- /* Acknowledge whatever it is destined for */
- if ((resp >= 100) && (resp <= 199)) {
- /* NON-INVITE messages do not ack a 1XX response. RFC 3261 section 17.1.2.2 */
- if (sipmethod == SIP_INVITE) {
- ack_res = __sip_semi_ack(p, seqno, 0, sipmethod);
- }
- } else {
- ack_res = __sip_ack(p, seqno, 0, sipmethod);
- }
-
- if (ack_res == FALSE) {
- /* RFC 3261 13.2.2.4 and 17.1.1.2 - We must re-send ACKs to re-transmitted final responses */
- if (sipmethod == SIP_INVITE && resp >= 200) {
- transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, resp < 300 ? TRUE: FALSE);
- }
-
- append_history(p, "Ignore", "Ignoring this retransmit\n");
- return;
- }
-
- /* If this is a NOTIFY for a subscription clear the flag that indicates that we have a NOTIFY pending */
- if (!p->owner && sipmethod == SIP_NOTIFY && p->pendinginvite) {
- p->pendinginvite = 0;
- }
-
- /* Get their tag if we haven't already */
- if (ast_strlen_zero(p->theirtag) || (resp >= 200)) {
- char tag[128];
-
- gettag(req, "To", tag, sizeof(tag));
- ast_string_field_set(p, theirtag, tag);
- } else {
- /* Store theirtag to track for changes when 200 responses to invites are received without SDP */
- ast_string_field_set(p, theirprovtag, p->theirtag);
- }
-
- /* This needs to be configurable on a channel/peer level,
- not mandatory for all communication. Sadly enough, NAT implementations
- are not so stable so we can always rely on these headers.
- Temporarily disabled, while waiting for fix.
- Fix assigned to Rizzo :-)
- */
- /* check_via_response(p, req); */
-
- /* RFC 3261 Section 15 specifies that if we receive a 408 or 481
- * in response to a BYE, then we should end the current dialog
- * and session. It is known that at least one phone manufacturer
- * potentially will send a 404 in response to a BYE, so we'll be
- * liberal in what we accept and end the dialog and session if we
- * receive any of those responses to a BYE.
- */
- if ((resp == 404 || resp == 408 || resp == 481) && sipmethod == SIP_BYE) {
- pvt_set_needdestroy(p, "received 4XX response to a BYE");
- return;
- }
-
- if (p->relatedpeer && sipmethod == SIP_OPTIONS) {
- /* We don't really care what the response is, just that it replied back.
- Well, as long as it's not a 100 response... since we might
- need to hang around for something more "definitive" */
- if (resp != 100)
- handle_response_peerpoke(p, resp, req);
- } else if (sipmethod == SIP_REFER && resp >= 200) {
- handle_response_refer(p, resp, rest, req, seqno);
- } else if (sipmethod == SIP_PUBLISH) {
- /* SIP PUBLISH transcends this morass of doodoo and instead
- * we just always call the response handler. Good gravy!
- */
- handle_response_publish(p, resp, rest, req, seqno);
- } else if (sipmethod == SIP_INFO) {
- /* More good gravy! */
- handle_response_info(p, resp, rest, req, seqno);
- } else if (sipmethod == SIP_MESSAGE) {
- /* More good gravy! */
- handle_response_message(p, resp, rest, req, seqno);
- } else if (sipmethod == SIP_NOTIFY) {
- /* The gravy train continues to roll */
- handle_response_notify(p, resp, rest, req, seqno);
- } else if (ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
- switch(resp) {
- case 100: /* 100 Trying */
- case 101: /* 101 Dialog establishment */
- case 183: /* 183 Session Progress */
- case 180: /* 180 Ringing */
- case 182: /* 182 Queued */
- case 181: /* 181 Call Is Being Forwarded */
- if (sipmethod == SIP_INVITE)
- handle_response_invite(p, resp, rest, req, seqno);
- break;
- case 200: /* 200 OK */
- p->authtries = 0; /* Reset authentication counter */
- if (sipmethod == SIP_INVITE) {
- handle_response_invite(p, resp, rest, req, seqno);
- } else if (sipmethod == SIP_REGISTER) {
- handle_response_register(p, resp, rest, req, seqno);
- } else if (sipmethod == SIP_SUBSCRIBE) {
- ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
- handle_response_subscribe(p, resp, rest, req, seqno);
- } else if (sipmethod == SIP_BYE) { /* Ok, we're ready to go */
- pvt_set_needdestroy(p, "received 200 response");
- ast_clear_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
- }
- break;
- case 401: /* Not www-authorized on SIP method */
- case 407: /* Proxy auth required */
- if (sipmethod == SIP_INVITE)
- handle_response_invite(p, resp, rest, req, seqno);
- else if (sipmethod == SIP_SUBSCRIBE)
- handle_response_subscribe(p, resp, rest, req, seqno);
- else if (p->registry && sipmethod == SIP_REGISTER)
- handle_response_register(p, resp, rest, req, seqno);
- else if (sipmethod == SIP_UPDATE) {
- handle_response_update(p, resp, rest, req, seqno);
- } else if (sipmethod == SIP_BYE) {
- if (p->options)
- p->options->auth_type = resp;
- if (ast_strlen_zero(p->authname)) {
- ast_log(LOG_WARNING, "Asked to authenticate %s, to %s but we have no matching peer!\n",
- msg, ast_sockaddr_stringify(&p->recv));
- pvt_set_needdestroy(p, "unable to authenticate BYE");
- } else if ((p->authtries == MAX_AUTHTRIES) || do_proxy_auth(p, req, resp, sipmethod, 0)) {
- ast_log(LOG_NOTICE, "Failed to authenticate on %s to '%s'\n", msg, sip_get_header(&p->initreq, "From"));
- pvt_set_needdestroy(p, "failed to authenticate BYE");
- }
- } else {
- ast_log(LOG_WARNING, "Got authentication request (%d) on %s to '%s'\n", resp, sip_methods[sipmethod].text, sip_get_header(req, "To"));
- pvt_set_needdestroy(p, "received 407 response");
- }
- break;
- case 403: /* Forbidden - we failed authentication */
- if (sipmethod == SIP_INVITE)
- handle_response_invite(p, resp, rest, req, seqno);
- else if (sipmethod == SIP_SUBSCRIBE)
- handle_response_subscribe(p, resp, rest, req, seqno);
- else if (p->registry && sipmethod == SIP_REGISTER)
- handle_response_register(p, resp, rest, req, seqno);
- else {
- ast_log(LOG_WARNING, "Forbidden - maybe wrong password on authentication for %s\n", msg);
- pvt_set_needdestroy(p, "received 403 response");
- }
- break;
- case 400: /* Bad Request */
- case 414: /* Request URI too long */
- case 493: /* Undecipherable */
- case 404: /* Not found */
- if (p->registry && sipmethod == SIP_REGISTER)
- handle_response_register(p, resp, rest, req, seqno);
- else if (sipmethod == SIP_INVITE)
- handle_response_invite(p, resp, rest, req, seqno);
- else if (sipmethod == SIP_SUBSCRIBE)
- handle_response_subscribe(p, resp, rest, req, seqno);
- else if (owner)
- ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
- break;
- case 423: /* Interval too brief */
- if (sipmethod == SIP_REGISTER)
- handle_response_register(p, resp, rest, req, seqno);
- break;
- case 408: /* Request timeout - terminate dialog */
- if (sipmethod == SIP_INVITE)
- handle_response_invite(p, resp, rest, req, seqno);
- else if (sipmethod == SIP_REGISTER)
- handle_response_register(p, resp, rest, req, seqno);
- else if (sipmethod == SIP_BYE) {
- pvt_set_needdestroy(p, "received 408 response");
- ast_debug(4, "Got timeout on bye. Thanks for the answer. Now, kill this call\n");
- } else {
- if (owner)
- ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
- pvt_set_needdestroy(p, "received 408 response");
- }
- break;
-
- case 428:
- case 422: /* Session-Timers: Session Interval Too Small */
- if (sipmethod == SIP_INVITE) {
- handle_response_invite(p, resp, rest, req, seqno);
- }
- break;
- case 480:
- if (sipmethod == SIP_INVITE) {
- handle_response_invite(p, resp, rest, req, seqno);
- } else if (sipmethod == SIP_SUBSCRIBE) {
- handle_response_subscribe(p, resp, rest, req, seqno);
- } else if (owner) {
- /* No specific handler. Default to congestion */
- ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
- }
- break;
- case 481: /* Call leg does not exist */
- if (sipmethod == SIP_INVITE) {
- handle_response_invite(p, resp, rest, req, seqno);
- } else if (sipmethod == SIP_SUBSCRIBE) {
- handle_response_subscribe(p, resp, rest, req, seqno);
- } else if (sipmethod == SIP_BYE) {
- /* The other side has no transaction to bye,
- just assume it's all right then */
- ast_log(LOG_WARNING, "Remote host can't match request %s to call '%s'. Giving up.\n", sip_methods[sipmethod].text, p->callid);
- } else if (sipmethod == SIP_CANCEL) {
- /* The other side has no transaction to cancel,
- just assume it's all right then */
- ast_log(LOG_WARNING, "Remote host can't match request %s to call '%s'. Giving up.\n", sip_methods[sipmethod].text, p->callid);
- } else {
- ast_log(LOG_WARNING, "Remote host can't match request %s to call '%s'. Giving up.\n", sip_methods[sipmethod].text, p->callid);
- /* Guessing that this is not an important request */
- }
- break;
- case 487:
- if (sipmethod == SIP_INVITE)
- handle_response_invite(p, resp, rest, req, seqno);
- break;
- case 415: /* Unsupported media type */
- case 488: /* Not acceptable here - codec error */
- case 606: /* Not Acceptable */
- if (sipmethod == SIP_INVITE)
- handle_response_invite(p, resp, rest, req, seqno);
- break;
- case 491: /* Pending */
- if (sipmethod == SIP_INVITE)
- handle_response_invite(p, resp, rest, req, seqno);
- else {
- ast_debug(1, "Got 491 on %s, unsupported. Call ID %s\n", sip_methods[sipmethod].text, p->callid);
- pvt_set_needdestroy(p, "received 491 response");
- }
- break;
- case 405: /* Method not allowed */
- case 501: /* Not Implemented */
- mark_method_unallowed(&p->allowed_methods, sipmethod);
- if (p->relatedpeer) {
- mark_method_allowed(&p->relatedpeer->disallowed_methods, sipmethod);
- }
- if (sipmethod == SIP_INVITE)
- handle_response_invite(p, resp, rest, req, seqno);
- else
- ast_log(LOG_WARNING, "Host '%s' does not implement '%s'\n", ast_sockaddr_stringify(&p->sa), msg);
- break;
- default:
- if ((resp >= 200) && (resp < 300)) { /* on any 2XX response do the following */
- if (sipmethod == SIP_INVITE) {
- handle_response_invite(p, resp, rest, req, seqno);
- }
- } else if ((resp >= 300) && (resp < 700)) {
- /* Fatal response */
- if ((resp != 487))
- ast_verb(3, "Got SIP response %d \"%s\" back from %s\n", resp, rest, ast_sockaddr_stringify(&p->sa));
-
- if (sipmethod == SIP_INVITE)
- stop_media_flows(p); /* Immediately stop RTP, VRTP and UDPTL as applicable */
-
- /* XXX Locking issues?? XXX */
- switch(resp) {
- case 300: /* Multiple Choices */
- case 301: /* Moved permanently */
- case 302: /* Moved temporarily */
- case 305: /* Use Proxy */
- if (p->owner) {
- struct ast_party_redirecting redirecting;
- struct ast_set_party_redirecting update_redirecting;
-
- ast_party_redirecting_init(&redirecting);
- memset(&update_redirecting, 0, sizeof(update_redirecting));
- change_redirecting_information(p, req, &redirecting,
- &update_redirecting, TRUE);
- ast_channel_set_redirecting(p->owner, &redirecting,
- &update_redirecting);
- ast_party_redirecting_free(&redirecting);
- }
- /* Fall through */
- case 486: /* Busy here */
- case 600: /* Busy everywhere */
- case 603: /* Decline */
- if (p->owner) {
- sip_handle_cc(p, req, AST_CC_CCBS);
- ast_queue_control(p->owner, AST_CONTROL_BUSY);
- }
- break;
- case 482: /* Loop Detected */
- case 404: /* Not Found */
- case 410: /* Gone */
- case 400: /* Bad Request */
- case 500: /* Server error */
- if (sipmethod == SIP_SUBSCRIBE) {
- handle_response_subscribe(p, resp, rest, req, seqno);
- break;
- }
- /* Fall through */
- case 502: /* Bad gateway */
- case 503: /* Service Unavailable */
- case 504: /* Server Timeout */
- if (owner)
- ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
- break;
- case 484: /* Address Incomplete */
- if (owner && sipmethod != SIP_BYE) {
- switch (ast_test_flag(&p->flags[1], SIP_PAGE2_ALLOWOVERLAP)) {
- case SIP_PAGE2_ALLOWOVERLAP_YES:
- ast_queue_hangup_with_cause(p->owner, hangup_sip2cause(resp));
- break;
- default:
- ast_queue_hangup_with_cause(p->owner, hangup_sip2cause(404));
- break;
- }
- }
- break;
- default:
- /* Send hangup */
- if (owner && sipmethod != SIP_BYE)
- ast_queue_hangup_with_cause(p->owner, hangup_sip2cause(resp));
- break;
- }
- /* ACK on invite */
- if (sipmethod == SIP_INVITE)
- transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
- sip_alreadygone(p);
- if (!p->owner) {
- pvt_set_needdestroy(p, "transaction completed");
- }
- } else if ((resp >= 100) && (resp < 200)) {
- if (sipmethod == SIP_INVITE) {
- if (!req->ignore) {
- sip_cancel_destroy(p);
- }
- if (find_sdp(req))
- process_sdp(p, req, SDP_T38_NONE, FALSE);
- if (p->owner) {
- /* Queue a progress frame */
- ast_queue_control(p->owner, AST_CONTROL_PROGRESS);
- }
- }
- } else
- ast_log(LOG_NOTICE, "Don't know how to handle a %d %s response from %s\n", resp, rest, p->owner ? ast_channel_name(p->owner) : ast_sockaddr_stringify(&p->sa));
- }
- } else {
- /* Responses to OUTGOING SIP requests on INCOMING calls
- get handled here. As well as out-of-call message responses */
- if (req->debug)
- ast_verbose("SIP Response message for INCOMING dialog %s arrived\n", msg);
-
- if (sipmethod == SIP_INVITE && resp == 200) {
- /* Tags in early session is replaced by the tag in 200 OK, which is
- the final reply to our INVITE */
- char tag[128];
-
- gettag(req, "To", tag, sizeof(tag));
- ast_string_field_set(p, theirtag, tag);
- }
-
- switch(resp) {
- case 200:
- if (sipmethod == SIP_INVITE) {
- handle_response_invite(p, resp, rest, req, seqno);
- } else if (sipmethod == SIP_CANCEL) {
- ast_debug(1, "Got 200 OK on CANCEL\n");
-
- /* Wait for 487, then destroy */
- } else if (sipmethod == SIP_BYE) {
- pvt_set_needdestroy(p, "transaction completed");
- }
- break;
- case 401: /* www-auth */
- case 407:
- if (sipmethod == SIP_INVITE)
- handle_response_invite(p, resp, rest, req, seqno);
- else if (sipmethod == SIP_BYE) {
- if (p->authtries == MAX_AUTHTRIES || do_proxy_auth(p, req, resp, sipmethod, 0)) {
- ast_log(LOG_NOTICE, "Failed to authenticate on %s to '%s'\n", msg, sip_get_header(&p->initreq, "From"));
- pvt_set_needdestroy(p, "failed to authenticate BYE");
- }
- }
- break;
- case 481: /* Call leg does not exist */
- if (sipmethod == SIP_INVITE) {
- /* Re-invite failed */
- handle_response_invite(p, resp, rest, req, seqno);
- } else if (sipmethod == SIP_BYE) {
- pvt_set_needdestroy(p, "received 481 response");
- } else if (sipdebug) {
- ast_debug(1, "Remote host can't match request %s to call '%s'. Giving up\n", sip_methods[sipmethod].text, p->callid);
- }
- break;
- case 501: /* Not Implemented */
- if (sipmethod == SIP_INVITE)
- handle_response_invite(p, resp, rest, req, seqno);
- break;
- default: /* Errors without handlers */
- if ((resp >= 100) && (resp < 200)) {
- if (sipmethod == SIP_INVITE) { /* re-invite */
- if (!req->ignore) {
- sip_cancel_destroy(p);
- }
- }
- } else if ((resp >= 200) && (resp < 300)) { /* on any unrecognized 2XX response do the following */
- if (sipmethod == SIP_INVITE) {
- handle_response_invite(p, resp, rest, req, seqno);
- }
- } else if ((resp >= 300) && (resp < 700)) {
- if ((resp != 487))
- ast_verb(3, "Incoming call: Got SIP response %d \"%s\" back from %s\n", resp, rest, ast_sockaddr_stringify(&p->sa));
- switch(resp) {
- case 415: /* Unsupported media type */
- case 488: /* Not acceptable here - codec error */
- case 603: /* Decline */
- case 500: /* Server error */
- case 502: /* Bad gateway */
- case 503: /* Service Unavailable */
- case 504: /* Server timeout */
- /* re-invite failed */
- if (sipmethod == SIP_INVITE) {
- sip_cancel_destroy(p);
- }
- break;
- }
- }
- break;
- }
- }
-}
-
-/*! \brief SIP pickup support function
- * Starts in a new thread, then pickup the call
- */
-static void *sip_pickup_thread(void *stuff)
-{
- struct ast_channel *chan;
- chan = stuff;
-
- ast_channel_hangupcause_set(chan, AST_CAUSE_NORMAL_CLEARING);
- if (ast_pickup_call(chan)) {
- ast_channel_hangupcause_set(chan, AST_CAUSE_CALL_REJECTED);
- }
- ast_hangup(chan);
- ast_channel_unref(chan);
- chan = NULL;
- return NULL;
-}
-
-/*! \brief Pickup a call using the subsystem in features.c
- * This is executed in a separate thread
- */
-static int sip_pickup(struct ast_channel *chan)
-{
- pthread_t threadid;
-
- ast_channel_ref(chan);
-
- if (ast_pthread_create_detached_background(&threadid, NULL, sip_pickup_thread, chan)) {
- ast_debug(1, "Unable to start Group pickup thread on channel %s\n", ast_channel_name(chan));
- ast_channel_unref(chan);
- return -1;
- }
- ast_debug(1, "Started Group pickup thread on channel %s\n", ast_channel_name(chan));
- return 0;
-}
-
-/*! \brief Get tag from packet
- *
- * \return pointer to the provided tag buffer.
- * \retval NULL if the tag was not found.
- */
-static const char *gettag(const struct sip_request *req, const char *header, char *tagbuf, int tagbufsize)
-{
- const char *thetag;
-
- if (!tagbuf)
- return NULL;
- tagbuf[0] = '\0'; /* reset the buffer */
- thetag = sip_get_header(req, header);
- thetag = strcasestr(thetag, ";tag=");
- if (thetag) {
- thetag += 5;
- ast_copy_string(tagbuf, thetag, tagbufsize);
- return strsep(&tagbuf, ";");
- }
- return NULL;
-}
-
-static int handle_cc_notify(struct sip_pvt *pvt, struct sip_request *req)
-{
- struct sip_monitor_instance *monitor_instance = ao2_callback(sip_monitor_instances, 0,
- find_sip_monitor_instance_by_subscription_pvt, pvt);
- const char *status = get_content_line(req, "cc-state", ':');
- struct cc_epa_entry *cc_entry;
- char *uri;
-
- if (!monitor_instance) {
- transmit_response(pvt, "400 Bad Request", req);
- return -1;
- }
-
- if (ast_strlen_zero(status)) {
- ao2_ref(monitor_instance, -1);
- transmit_response(pvt, "400 Bad Request", req);
- return -1;
- }
-
- if (!strcmp(status, "queued")) {
- /* We've been told that we're queued. This is the endpoint's way of telling
- * us that it has accepted our CC request. We need to alert the core of this
- * development
- */
- ast_cc_monitor_request_acked(monitor_instance->core_id, "SIP endpoint %s accepted request", monitor_instance->device_name);
- transmit_response(pvt, "200 OK", req);
- ao2_ref(monitor_instance, -1);
- return 0;
- }
-
- /* It's open! Yay! */
- uri = get_content_line(req, "cc-URI", ':');
- if (ast_strlen_zero(uri)) {
- uri = get_in_brackets((char *)sip_get_header(req, "From"));
- }
-
- ast_string_field_set(monitor_instance, notify_uri, uri);
- if (monitor_instance->suspension_entry) {
- cc_entry = monitor_instance->suspension_entry->instance_data;
- if (cc_entry->current_state == CC_CLOSED) {
- /* If we've created a suspension entry and the current state is closed, then that means
- * we got a notice from the CC core earlier to suspend monitoring, but because this particular
- * call leg had not yet notified us that it was ready for recall, it meant that we
- * could not yet send a PUBLISH. Now, however, we can.
- */
- construct_pidf_body(CC_CLOSED, monitor_instance->suspension_entry->body,
- sizeof(monitor_instance->suspension_entry->body), monitor_instance->peername);
- transmit_publish(monitor_instance->suspension_entry, SIP_PUBLISH_INITIAL, monitor_instance->notify_uri);
- } else {
- ast_cc_monitor_callee_available(monitor_instance->core_id, "SIP monitored callee has become available");
- }
- } else {
- ast_cc_monitor_callee_available(monitor_instance->core_id, "SIP monitored callee has become available");
- }
- ao2_ref(monitor_instance, -1);
- transmit_response(pvt, "200 OK", req);
-
- return 0;
-}
-
-/*! \brief Handle incoming notifications */
-static int handle_request_notify(struct sip_pvt *p, struct sip_request *req, struct ast_sockaddr *addr, uint32_t seqno, const char *e)
-{
- /* This is mostly a skeleton for future improvements */
- /* Mostly created to return proper answers on notifications on outbound REFER's */
- int res = 0;
- const char *event = sip_get_header(req, "Event");
- char *sep;
-
- if( (sep = strchr(event, ';')) ) { /* XXX bug here - overwriting string ? */
- *sep++ = '\0';
- }
-
- if (sipdebug)
- ast_debug(2, "Got NOTIFY Event: %s\n", event);
-
- if (!strcmp(event, "refer")) {
- /* Save nesting depth for now, since there might be other events we will
- support in the future */
-
- /* Handle REFER notifications */
- char *buf, *cmd, *code;
- int respcode;
- int success = TRUE;
-
- /* EventID for each transfer... EventID is basically the REFER cseq
-
- We are getting notifications on a call that we transferred
- We should hangup when we are getting a 200 OK in a sipfrag
- Check if we have an owner of this event */
-
- /* Check the content type */
- if (strncasecmp(sip_get_header(req, "Content-Type"), "message/sipfrag", strlen("message/sipfrag"))) {
- /* We need a sipfrag */
- transmit_response(p, "400 Bad request", req);
- sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
- return -1;
- }
-
- /* Get the text of the attachment */
- if (ast_strlen_zero(buf = get_content(req))) {
- ast_log(LOG_WARNING, "Unable to retrieve attachment from NOTIFY %s\n", p->callid);
- transmit_response(p, "400 Bad request", req);
- sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
- return -1;
- }
-
- /*
- From the RFC...
- A minimal, but complete, implementation can respond with a single
- NOTIFY containing either the body:
- SIP/2.0 100 Trying
-
- if the subscription is pending, the body:
- SIP/2.0 200 OK
- if the reference was successful, the body:
- SIP/2.0 503 Service Unavailable
- if the reference failed, or the body:
- SIP/2.0 603 Declined
-
- if the REFER request was accepted before approval to follow the
- reference could be obtained and that approval was subsequently denied
- (see Section 2.4.7).
-
- If there are several REFERs in the same dialog, we need to
- match the ID of the event header...
- */
- ast_debug(3, "* SIP Transfer NOTIFY Attachment: \n---%s\n---\n", buf);
- cmd = ast_skip_blanks(buf);
- code = cmd;
- /* We are at SIP/2.0 */
- while(*code && (*code > 32)) { /* Search white space */
- code++;
- }
- *code++ = '\0';
- code = ast_skip_blanks(code);
- sep = code;
- sep++;
- while(*sep && (*sep > 32)) { /* Search white space */
- sep++;
- }
- *sep++ = '\0'; /* Response string */
- respcode = atoi(code);
- switch (respcode) {
- case 200: /* OK: The new call is up, hangup this call */
- /* Hangup the call that we are replacing */
- break;
- case 301: /* Moved permanently */
- case 302: /* Moved temporarily */
- /* Do we get the header in the packet in this case? */
- success = FALSE;
- break;
- case 503: /* Service Unavailable: The new call failed */
- case 603: /* Declined: Not accepted */
- /* Cancel transfer, continue the current call */
- success = FALSE;
- break;
- case 0: /* Parse error */
- /* Cancel transfer, continue the current call */
- ast_log(LOG_NOTICE, "Error parsing sipfrag in NOTIFY in response to REFER.\n");
- success = FALSE;
- break;
- default:
- if (respcode < 200) {
- /* ignore provisional responses */
- success = -1;
- } else {
- ast_log(LOG_NOTICE, "Got unknown code '%d' in NOTIFY in response to REFER.\n", respcode);
- success = FALSE;
- }
- break;
- }
- if (success == FALSE) {
- ast_log(LOG_NOTICE, "Transfer failed. Sorry. Nothing further to do with this call\n");
- }
-
- if (p->owner && success != -1) {
- enum ast_control_transfer message = success ? AST_TRANSFER_SUCCESS : AST_TRANSFER_FAILED;
- ast_queue_control_data(p->owner, AST_CONTROL_TRANSFER, &message, sizeof(message));
- }
- /* Confirm that we received this packet */
- transmit_response(p, "200 OK", req);
- } else if (!strcmp(event, "message-summary")) {
- const char *mailbox = NULL;
- char *c = ast_strdupa(get_content_line(req, "Voice-Message", ':'));
-
- if (!p->mwi) {
- struct sip_peer *peer = sip_find_peer(NULL, &p->recv, TRUE, FINDPEERS, FALSE, p->socket.type);
-
- if (peer) {
- mailbox = ast_strdupa(peer->unsolicited_mailbox);
- sip_unref_peer(peer, "removing unsolicited mwi ref");
- }
- } else {
- mailbox = p->mwi->mailbox;
- }
-
- if (!ast_strlen_zero(mailbox) && !ast_strlen_zero(c)) {
- char *old = strsep(&c, " ");
- char *new = strsep(&old, "/");
-
- ast_publish_mwi_state(mailbox, "SIP_Remote", atoi(new), atoi(old));
-
- transmit_response(p, "200 OK", req);
- } else {
- transmit_response(p, "489 Bad event", req);
- res = -1;
- }
- } else if (!strcmp(event, "keep-alive")) {
- /* Used by Sipura/Linksys for NAT pinhole,
- * just confirm that we received the packet. */
- transmit_response(p, "200 OK", req);
- } else if (!strcmp(event, "call-completion")) {
- res = handle_cc_notify(p, req);
- } else {
- /* We don't understand this event. */
- transmit_response(p, "489 Bad event", req);
- res = -1;
- }
-
- if (!p->lastinvite)
- sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
-
- return res;
-}
-
-/*! \brief Handle incoming OPTIONS request
- An OPTIONS request should be answered like an INVITE from the same UA, including SDP
-*/
-static int handle_request_options(struct sip_pvt *p, struct sip_request *req, struct ast_sockaddr *addr, const char *e)
-{
- const char *msg;
- enum sip_get_dest_result gotdest;
- int res;
-
- if (p->lastinvite) {
- /* if this is a request in an active dialog, just confirm that the dialog exists. */
- transmit_response_with_allow(p, "200 OK", req, 0);
- return 0;
- }
-
- if (sip_cfg.auth_options_requests) {
- /* Do authentication if this OPTIONS request began the dialog */
- copy_request(&p->initreq, req);
- set_pvt_allowed_methods(p, req);
- res = check_user(p, req, SIP_OPTIONS, e, XMIT_UNRELIABLE, addr);
- if (res == AUTH_CHALLENGE_SENT) {
- sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
- return 0;
- }
- if (res < 0) { /* Something failed in authentication */
- send_check_user_failure_response(p, req, res, XMIT_UNRELIABLE);
- sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
- return 0;
- }
- }
-
- /* must go through authentication before getting here */
- gotdest = get_destination(p, req, NULL);
- build_contact(p, req, 1);
-
- if (ast_strlen_zero(p->context))
- ast_string_field_set(p, context, sip_cfg.default_context);
-
- if (ast_shutting_down()) {
- /*
- * Not taking any new calls at this time.
- * Likely a server availability OPTIONS poll.
- */
- msg = "503 Unavailable";
- } else {
- msg = "404 Not Found";
- switch (gotdest) {
- case SIP_GET_DEST_INVALID_URI:
- msg = "416 Unsupported URI scheme";
- break;
- case SIP_GET_DEST_EXTEN_MATCHMORE:
- case SIP_GET_DEST_REFUSED:
- case SIP_GET_DEST_EXTEN_NOT_FOUND:
- //msg = "404 Not Found";
- break;
- case SIP_GET_DEST_EXTEN_FOUND:
- msg = "200 OK";
- break;
- }
- }
- transmit_response_with_allow(p, msg, req, 0);
-
- /* Destroy if this OPTIONS was the opening request, but not if
- it's in the middle of a normal call flow. */
- sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
-
- return 0;
-}
-
-/*! \brief Handle the transfer part of INVITE with a replaces: header,
- *
- * This is used for call-pickup and for attended transfers initiated on
- * remote endpoints (i.e. a REFER received on a remote server).
- *
- * \note p and p->owner are locked upon entering this function. If the
- * call pickup or attended transfer is successful, then p->owner will
- * be unlocked upon exiting this function. This is communicated to the
- * caller through the nounlock parameter.
- *
- * \param p The sip_pvt where the INVITE with Replaces was received
- * \param req The incoming INVITE
- * \param[out] nounlock Indicator if p->owner should remained locked. On successful transfer, this will be set true.
- * \param replaces_pvt sip_pvt referenced by Replaces header
- * \param replaces_chan replaces_pvt's owner channel
- * \retval 0 Success
- * \retval non-zero Failure
- */
-static int handle_invite_replaces(struct sip_pvt *p, struct sip_request *req,
- int *nounlock, struct sip_pvt *replaces_pvt, struct ast_channel *replaces_chan)
-{
- struct ast_channel *c;
- struct ast_bridge *bridge;
-
- if (req->ignore) {
- return 0;
- }
-
- if (!p->owner) {
- /* What to do if no channel ??? */
- ast_log(LOG_ERROR, "Unable to create new channel. Invite/replace failed.\n");
- transmit_response_reliable(p, "503 Service Unavailable", req);
- append_history(p, "Xfer", "INVITE/Replace Failed. No new channel.");
- sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
- return 1;
- }
- append_history(p, "Xfer", "INVITE/Replace received");
-
- /* Get a ref to ensure the channel cannot go away on us. */
- c = ast_channel_ref(p->owner);
-
- /* Fake call progress */
- transmit_response(p, "100 Trying", req);
- ast_setstate(c, AST_STATE_RING);
-
- ast_debug(4, "Invite/Replaces: preparing to replace %s with %s\n", ast_channel_name(replaces_chan), ast_channel_name(c));
-
- *nounlock = 1;
- ast_channel_unlock(c);
- sip_pvt_unlock(p);
-
- ast_raw_answer(c);
-
- bridge = ast_bridge_transfer_acquire_bridge(replaces_chan);
- if (bridge) {
- /*
- * We have two refs of the channel. One is held in c and the other
- * is notionally represented by p->owner. The impart is "stealing"
- * the p->owner ref on success so the bridging system can have
- * control of when the channel is hung up.
- */
- if (ast_bridge_impart(bridge, c, replaces_chan, NULL,
- AST_BRIDGE_IMPART_CHAN_INDEPENDENT)) {
- ast_hangup(c);
- }
- ao2_ref(bridge, -1);
- } else {
- int pickedup;
- ast_channel_lock(replaces_chan);
- pickedup = ast_can_pickup(replaces_chan) && !ast_do_pickup(c, replaces_chan);
- ast_channel_unlock(replaces_chan);
- if (!pickedup) {
- ast_channel_move(replaces_chan, c);
- }
- ast_hangup(c);
- }
- ast_channel_unref(c);
- sip_pvt_lock(p);
- return 0;
-}
-
-/*! \note No channel or pvt locks should be held while calling this function. */
-static int do_magic_pickup(struct ast_channel *channel, const char *extension, const char *context)
-{
- struct ast_str *str = ast_str_alloca(AST_MAX_EXTENSION + AST_MAX_CONTEXT + 2);
- struct ast_app *pickup = pbx_findapp("Pickup");
-
- if (!pickup) {
- ast_log(LOG_ERROR, "Unable to perform pickup: Application 'Pickup' not loaded (app_directed_pickup.so).\n");
- return -1;
- }
-
- ast_str_set(&str, 0, "%s@%s", extension, sip_cfg.notifycid == IGNORE_CONTEXT ? "PICKUPMARK" : context);
-
- ast_debug(2, "About to call Pickup(%s)\n", ast_str_buffer(str));
-
- /* There is no point in capturing the return value since pickup_exec
- doesn't return anything meaningful unless the passed data is an empty
- string (which in our case it will not be) */
- pbx_exec(channel, pickup, ast_str_buffer(str));
-
- return 0;
-}
-
-/*!
- * \brief Called to deny a T38 reinvite if the core does not respond to our request
- *
- * \note Run by the sched thread.
- */
-static int sip_t38_abort(const void *data)
-{
- struct sip_pvt *pvt = (struct sip_pvt *) data;
- struct ast_channel *owner;
-
- owner = sip_pvt_lock_full(pvt);
- pvt->t38id = -1;
-
- /*
- * An application may have taken ownership of the T.38 negotiation
- * on the channel while we were waiting to grab the lock. If it
- * did, the T.38 state will have been changed. This is our
- * indication that we do *not* want to abort the negotiation
- * process.
- */
- if (pvt->t38.state == T38_PEER_REINVITE) {
- /* Still waiting for a response on timeout so reject the offer. */
- change_t38_state(pvt, T38_REJECTED);
- transmit_response_reliable(pvt, "488 Not acceptable here", &pvt->initreq);
- }
-
- if (owner) {
- ast_channel_unlock(owner);
- ast_channel_unref(owner);
- }
- sip_pvt_unlock(pvt);
- dialog_unref(pvt, "t38id complete");
- return 0;
-}
-
-/* Run by the sched thread. */
-static int __stop_t38_abort_timer(const void *data)
-{
- struct sip_pvt *pvt = (void *) data;
-
- AST_SCHED_DEL_UNREF(sched, pvt->t38id,
- dialog_unref(pvt, "Stop scheduled t38id"));
- dialog_unref(pvt, "Stop t38id action");
- return 0;
-}
-
-static void stop_t38_abort_timer(struct sip_pvt *pvt)
-{
- dialog_ref(pvt, "Stop t38id action");
- if (ast_sched_add(sched, 0, __stop_t38_abort_timer, pvt) < 0) {
- /* Uh Oh. Expect bad behavior. */
- dialog_unref(pvt, "Failed to schedule stop t38id action");
- }
-}
-
-/* Run by the sched thread. */
-static int __start_t38_abort_timer(const void *data)
-{
- struct sip_pvt *pvt = (void *) data;
-
- AST_SCHED_DEL_UNREF(sched, pvt->t38id,
- dialog_unref(pvt, "Stop scheduled t38id"));
-
- dialog_ref(pvt, "Schedule t38id");
- pvt->t38id = ast_sched_add(sched, 5000, sip_t38_abort, pvt);
- if (pvt->t38id < 0) {
- /* Uh Oh. Expect bad behavior. */
- dialog_unref(pvt, "Failed to schedule t38id");
- }
-
- dialog_unref(pvt, "Start t38id action");
- return 0;
-}
-
-static void start_t38_abort_timer(struct sip_pvt *pvt)
-{
- dialog_ref(pvt, "Start t38id action");
- if (ast_sched_add(sched, 0, __start_t38_abort_timer, pvt) < 0) {
- /* Uh Oh. Expect bad behavior. */
- dialog_unref(pvt, "Failed to schedule start t38id action");
- }
-}
-
-/*!
- * \brief bare-bones support for SIP UPDATE
- *
- * XXX This is not even close to being RFC 3311-compliant. We don't advertise
- * that we support the UPDATE method, so no one should ever try sending us
- * an UPDATE anyway. However, Asterisk can send an UPDATE to change connected
- * line information, so we need to be prepared to handle this. The way we distinguish
- * such an UPDATE is through the X-Asterisk-rpid-update header.
- *
- * Actually updating the media session may be some future work.
- */
-static int handle_request_update(struct sip_pvt *p, struct sip_request *req)
-{
- if (ast_strlen_zero(sip_get_header(req, "X-Asterisk-rpid-update"))) {
- transmit_response(p, "501 Method Not Implemented", req);
- return 0;
- }
- if (!p->owner) {
- transmit_response(p, "481 Call/Transaction Does Not Exist", req);
- return 0;
- }
- if (get_rpid(p, req)) {
- struct ast_party_connected_line connected;
- struct ast_set_party_connected_line update_connected;
-
- ast_party_connected_line_init(&connected);
- memset(&update_connected, 0, sizeof(update_connected));
-
- update_connected.id.number = 1;
- connected.id.number.valid = 1;
- connected.id.number.str = (char *) p->cid_num;
- connected.id.number.presentation = p->callingpres;
-
- update_connected.id.name = 1;
- connected.id.name.valid = 1;
- connected.id.name.str = (char *) p->cid_name;
- connected.id.name.presentation = p->callingpres;
-
- /* Invalidate any earlier private connected id representation */
- ast_set_party_id_all(&update_connected.priv);
-
- connected.id.tag = (char *) p->cid_tag;
- connected.source = AST_CONNECTED_LINE_UPDATE_SOURCE_TRANSFER;
- ast_channel_queue_connected_line_update(p->owner, &connected, &update_connected);
- }
- transmit_response(p, "200 OK", req);
- return 0;
-}
-
-/*!
- * \internal
- * \brief Check Session Timers for an INVITE request
- *
- * \retval 0 ok
- * \retval -1 failure
- */
-static int handle_request_invite_st(struct sip_pvt *p, struct sip_request *req, int reinvite)
-{
- const char *p_uac_se_hdr; /* UAC's Session-Expires header string */
- const char *p_uac_min_se; /* UAC's requested Min-SE interval (char string) */
- int uac_max_se = -1; /* UAC's Session-Expires in integer format */
- int uac_min_se = -1; /* UAC's Min-SE in integer format */
- int st_active = FALSE; /* Session-Timer on/off boolean */
- int st_interval = 0; /* Session-Timer negotiated refresh interval */
- enum st_refresher tmp_st_ref = SESSION_TIMER_REFRESHER_AUTO; /* Session-Timer refresher */
- int dlg_min_se = -1;
- int dlg_max_se = global_max_se;
- int rtn;
-
- /* Session-Timers */
- if ((p->sipoptions & SIP_OPT_TIMER)) {
- enum st_refresher_param st_ref_param = SESSION_TIMER_REFRESHER_PARAM_UNKNOWN;
-
- /* The UAC has requested session-timers for this session. Negotiate
- the session refresh interval and who will be the refresher */
- ast_debug(2, "Incoming INVITE with 'timer' option supported\n");
-
- /* Allocate Session-Timers struct w/in the dialog */
- if (!p->stimer) {
- sip_st_alloc(p);
- }
-
- /* Parse the Session-Expires header */
- p_uac_se_hdr = sip_get_header(req, "Session-Expires");
- if (!ast_strlen_zero(p_uac_se_hdr)) {
- ast_debug(2, "INVITE also has \"Session-Expires\" header.\n");
- rtn = parse_session_expires(p_uac_se_hdr, &uac_max_se, &st_ref_param);
- tmp_st_ref = (st_ref_param == SESSION_TIMER_REFRESHER_PARAM_UAC) ? SESSION_TIMER_REFRESHER_THEM : SESSION_TIMER_REFRESHER_US;
- if (rtn != 0) {
- transmit_response_reliable(p, "400 Session-Expires Invalid Syntax", req);
- return -1;
- }
- }
-
- /* Parse the Min-SE header */
- p_uac_min_se = sip_get_header(req, "Min-SE");
- if (!ast_strlen_zero(p_uac_min_se)) {
- ast_debug(2, "INVITE also has \"Min-SE\" header.\n");
- rtn = parse_minse(p_uac_min_se, &uac_min_se);
- if (rtn != 0) {
- transmit_response_reliable(p, "400 Min-SE Invalid Syntax", req);
- return -1;
- }
- }
-
- dlg_min_se = st_get_se(p, FALSE);
- switch (st_get_mode(p, 1)) {
- case SESSION_TIMER_MODE_ACCEPT:
- case SESSION_TIMER_MODE_ORIGINATE:
- if (uac_max_se > 0 && uac_max_se < dlg_min_se) {
- transmit_response_with_minse(p, "422 Session Interval Too Small", req, dlg_min_se);
- return -1;
- }
-
- p->stimer->st_active_peer_ua = TRUE;
- st_active = TRUE;
- if (st_ref_param == SESSION_TIMER_REFRESHER_PARAM_UNKNOWN) {
- tmp_st_ref = st_get_refresher(p);
- }
-
- dlg_max_se = st_get_se(p, TRUE);
- if (uac_max_se > 0) {
- if (dlg_max_se >= uac_min_se) {
- st_interval = (uac_max_se < dlg_max_se) ? uac_max_se : dlg_max_se;
- } else {
- st_interval = uac_max_se;
- }
- } else if (uac_min_se > 0) {
- st_interval = MAX(dlg_max_se, uac_min_se);
- } else {
- st_interval = dlg_max_se;
- }
- break;
-
- case SESSION_TIMER_MODE_REFUSE:
- if (p->reqsipoptions & SIP_OPT_TIMER) {
- transmit_response_with_unsupported(p, "420 Option Disabled", req, "timer");
- ast_log(LOG_WARNING, "Received SIP INVITE with supported but disabled option: timer\n");
- return -1;
- }
- break;
-
- default:
- ast_log(LOG_ERROR, "Internal Error %u at %s:%d\n", st_get_mode(p, 1), __FILE__, __LINE__);
- break;
- }
- } else {
- /* The UAC did not request session-timers. Asterisk (UAS), will now decide
- (based on session-timer-mode in sip.conf) whether to run session-timers for
- this session or not. */
- switch (st_get_mode(p, 1)) {
- case SESSION_TIMER_MODE_ORIGINATE:
- st_active = TRUE;
- st_interval = st_get_se(p, TRUE);
- tmp_st_ref = SESSION_TIMER_REFRESHER_US;
- p->stimer->st_active_peer_ua = (p->sipoptions & SIP_OPT_TIMER) ? TRUE : FALSE;
- break;
-
- default:
- break;
- }
- }
-
- if (reinvite == 0) {
- /* Session-Timers: Start session refresh timer based on negotiation/config */
- if (st_active == TRUE) {
- p->stimer->st_active = TRUE;
- p->stimer->st_interval = st_interval;
- p->stimer->st_ref = tmp_st_ref;
- }
- } else {
- if (p->stimer->st_active == TRUE) {
- /* Session-Timers: A re-invite request sent within a dialog will serve as
- a refresh request, no matter whether the re-invite was sent for refreshing
- the session or modifying it.*/
- ast_debug (2, "Restarting session-timers on a refresh - %s\n", p->callid);
-
- /* The UAC may be adjusting the session-timers mid-session */
- if (st_interval > 0) {
- p->stimer->st_interval = st_interval;
- p->stimer->st_ref = tmp_st_ref;
- }
- }
- }
-
- return 0;
-}
-
-/*!
- * \brief Handle incoming INVITE request
- * \note If the INVITE has a Replaces header, it is part of an
- * attended transfer. If so, we do not go through the dial
- * plan but try to find the active call and masquerade
- * into it
- */
-static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, struct ast_sockaddr *addr, uint32_t seqno, int *recount, const char *e, int *nounlock)
-{
- int res = INV_REQ_SUCCESS;
- int gotdest;
- const char *p_replaces;
- char *replace_id = NULL;
- const char *required;
- unsigned int required_profile = 0;
- struct ast_channel *c = NULL; /* New channel */
- struct sip_peer *authpeer = NULL; /* Matching Peer */
- int reinvite = 0;
- struct ast_party_redirecting redirecting;
- struct ast_set_party_redirecting update_redirecting;
- int supported_start = 0;
- int require_start = 0;
- char unsupported[256] = { 0, };
- struct {
- char exten[AST_MAX_EXTENSION];
- char context[AST_MAX_CONTEXT];
- } pickup = {
- .exten = "",
- };
- RAII_VAR(struct sip_pvt *, replaces_pvt, NULL, ao2_cleanup);
- RAII_VAR(struct ast_channel *, replaces_chan, NULL, ao2_cleanup);
-
- /* Find out what they support */
- if (!p->sipoptions) {
- const char *supported = NULL;
- do {
- supported = __get_header(req, "Supported", &supported_start);
- if (!ast_strlen_zero(supported)) {
- p->sipoptions |= parse_sip_options(supported, NULL, 0);
- }
- } while (!ast_strlen_zero(supported));
- }
-
- /* Find out what they require */
- do {
- required = __get_header(req, "Require", &require_start);
- if (!ast_strlen_zero(required)) {
- required_profile |= parse_sip_options(required, unsupported, ARRAY_LEN(unsupported));
- }
- } while (!ast_strlen_zero(required));
-
- /* If there are any options required that we do not support,
- * then send a 420 with only those unsupported options listed */
- if (!ast_strlen_zero(unsupported)) {
- transmit_response_with_unsupported(p, "420 Bad extension (unsupported)", req, unsupported);
- ast_log(LOG_WARNING, "Received SIP INVITE with unsupported required extension: %s\n", unsupported);
- p->invitestate = INV_COMPLETED;
- if (!p->lastinvite) {
- sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
- }
- res = -1;
- goto request_invite_cleanup;
- }
-
-
- /* The option tags may be present in Supported: or Require: headers.
- Include the Require: option tags for further processing as well */
- p->sipoptions |= required_profile;
- p->reqsipoptions = required_profile;
-
- /* Check if this is a loop */
- if (ast_test_flag(&p->flags[0], SIP_OUTGOING) && p->owner && (p->invitestate != INV_TERMINATED && p->invitestate != INV_CONFIRMED) && ast_channel_state(p->owner) != AST_STATE_UP) {
- /* This is a call to ourself. Send ourselves an error code and stop
- processing immediately, as SIP really has no good mechanism for
- being able to call yourself */
- /* If pedantic is on, we need to check the tags. If they're different, this is
- in fact a forked call through a SIP proxy somewhere. */
- int different;
- const char *initial_rlpart2 = REQ_OFFSET_TO_STR(&p->initreq, rlpart2);
- const char *this_rlpart2 = REQ_OFFSET_TO_STR(req, rlpart2);
- if (sip_cfg.pedanticsipchecking)
- different = sip_uri_cmp(initial_rlpart2, this_rlpart2);
- else
- different = strcmp(initial_rlpart2, this_rlpart2);
- if (!different) {
- transmit_response(p, "482 Loop Detected", req);
- p->invitestate = INV_COMPLETED;
- sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
- res = INV_REQ_FAILED;
- goto request_invite_cleanup;
- } else {
- /*! This is a spiral. What we need to do is to just change the outgoing INVITE
- * so that it now routes to the new Request URI. Since we created the INVITE ourselves
- * that should be all we need to do.
- *
- * \todo XXX This needs to be reviewed. YOu don't change the request URI really, you route the packet
- * correctly instead...
- */
- char *uri = ast_strdupa(this_rlpart2);
- char *at = strchr(uri, '@');
- char *peerorhost;
- ast_debug(2, "Potential spiral detected. Original RURI was %s, new RURI is %s\n", initial_rlpart2, this_rlpart2);
- transmit_response(p, "100 Trying", req);
- if (at) {
- *at = '\0';
- }
- /* Parse out "sip:" */
- if ((peerorhost = strchr(uri, ':'))) {
- *peerorhost++ = '\0';
- }
- ast_string_field_set(p, theirtag, NULL);
- /* Treat this as if there were a call forward instead...
- */
- ast_channel_call_forward_set(p->owner, peerorhost);
- ast_queue_control(p->owner, AST_CONTROL_BUSY);
- res = INV_REQ_FAILED;
- goto request_invite_cleanup;
- }
- }
-
- if (!req->ignore && p->pendinginvite) {
- if (!ast_test_flag(&p->flags[0], SIP_OUTGOING) && (p->invitestate == INV_COMPLETED || p->invitestate == INV_TERMINATED)) {
- /* What do these circumstances mean? We have received an INVITE for an "incoming" dialog for which we
- * have sent a final response. We have not yet received an ACK, though (which is why p->pendinginvite is non-zero).
- * We also know that the INVITE is not a retransmission, because otherwise the "ignore" flag would be set.
- * This means that either we are receiving a reinvite for a terminated dialog, or we are receiving an INVITE with
- * credentials based on one we challenged earlier.
- *
- * The action to take in either case is to treat the INVITE as though it contains an implicit ACK for the previous
- * transaction. Calling __sip_ack will take care of this by clearing the p->pendinginvite and removing the response
- * from the previous transaction from the list of outstanding packets.
- */
- __sip_ack(p, p->pendinginvite, 1, 0);
- } else {
- /* We already have a pending invite. Sorry. You are on hold. */
- p->glareinvite = seqno;
- transmit_response_reliable(p, "491 Request Pending", req);
- check_via(p, req);
- ast_debug(1, "Got INVITE on call where we already have pending INVITE, deferring that - %s\n", p->callid);
- /* Don't destroy dialog here */
- res = INV_REQ_FAILED;
- goto request_invite_cleanup;
- }
- }
-
- p_replaces = sip_get_header(req, "Replaces");
- if (!ast_strlen_zero(p_replaces)) {
- /* We have a replaces header */
- char *ptr;
- char *fromtag = NULL;
- char *totag = NULL;
- char *start, *to;
- int error = 0;
-
- if (p->owner) {
- ast_debug(3, "INVITE w Replaces on existing call? Refusing action. [%s]\n", p->callid);
- transmit_response_reliable(p, "400 Bad request", req); /* The best way to not accept the transfer */
- check_via(p, req);
- copy_request(&p->initreq, req);
- /* Do not destroy existing call */
- res = INV_REQ_ERROR;
- goto request_invite_cleanup;
- }
-
- if (sipdebug)
- ast_debug(3, "INVITE part of call transfer. Replaces [%s]\n", p_replaces);
- /* Create a buffer we can manipulate */
- replace_id = ast_strdupa(p_replaces);
- ast_uri_decode(replace_id, ast_uri_sip_user);
-
- if (!sip_refer_alloc(p)) {
- transmit_response_reliable(p, "500 Server Internal Error", req);
- append_history(p, "Xfer", "INVITE/Replace Failed. Out of memory.");
- sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
- p->invitestate = INV_COMPLETED;
- res = INV_REQ_ERROR;
- check_via(p, req);
- copy_request(&p->initreq, req);
- goto request_invite_cleanup;
- }
-
- /* Todo: (When we find phones that support this)
- if the replaces header contains ";early-only"
- we can only replace the call in early
- stage, not after it's up.
-
- If it's not in early mode, 486 Busy.
- */
-
- /* Skip leading whitespace */
- replace_id = ast_skip_blanks(replace_id);
-
- start = replace_id;
- while ( (ptr = strsep(&start, ";")) ) {
- ptr = ast_skip_blanks(ptr); /* XXX maybe unnecessary ? */
- if ( (to = strcasestr(ptr, "to-tag=") ) )
- totag = to + 7; /* skip the keyword */
- else if ( (to = strcasestr(ptr, "from-tag=") ) ) {
- fromtag = to + 9; /* skip the keyword */
- fromtag = strsep(&fromtag, "&"); /* trim what ? */
- }
- }
-
- if (sipdebug)
- ast_debug(4, "Invite/replaces: Will use Replace-Call-ID : %s Fromtag: %s Totag: %s\n",
- replace_id,
- fromtag ? fromtag : "",
- totag ? totag : "");
-
- /* Try to find call that we are replacing.
- If we have a Replaces header, we need to cancel that call if we succeed with this call.
- First we cheat a little and look for a magic call-id from phones that support
- dialog-info+xml so we can do technology independent pickup... */
- if (strncmp(replace_id, "pickup-", 7) == 0) {
- RAII_VAR(struct sip_pvt *, subscription, NULL, ao2_cleanup);
- RAII_VAR(struct ast_channel *, subscription_chan, NULL, ao2_cleanup);
-
- replace_id += 7; /* Worst case we are looking at \0 */
-
- if (get_sip_pvt_from_replaces(replace_id, totag, fromtag, &subscription, &subscription_chan)) {
- ast_log(LOG_NOTICE, "Unable to find subscription with call-id: %s\n", replace_id);
- transmit_response_reliable(p, "481 Call Leg Does Not Exist (Replaces)", req);
- error = 1;
- } else {
- SCOPED_LOCK(lock, subscription, sip_pvt_lock, sip_pvt_unlock);
- ast_log(LOG_NOTICE, "Trying to pick up %s@%s\n", subscription->exten, subscription->context);
- ast_copy_string(pickup.exten, subscription->exten, sizeof(pickup.exten));
- ast_copy_string(pickup.context, subscription->context, sizeof(pickup.context));
- }
- }
-
- if (!error && ast_strlen_zero(pickup.exten) && get_sip_pvt_from_replaces(replace_id,
- totag, fromtag, &replaces_pvt, &replaces_chan)) {
- ast_log(LOG_NOTICE, "Supervised transfer attempted to replace non-existent call id (%s)!\n", replace_id);
- transmit_response_reliable(p, "481 Call Leg Does Not Exist (Replaces)", req);
- error = 1;
- }
-
- /* The matched call is the call from the transferer to Asterisk .
- We want to bridge the bridged part of the call to the
- incoming invite, thus taking over the refered call */
-
- if (replaces_pvt == p) {
- ast_log(LOG_NOTICE, "INVITE with replaces into it's own call id (%s == %s)!\n", replace_id, p->callid);
- transmit_response_reliable(p, "400 Bad request", req); /* The best way to not accept the transfer */
- error = 1;
- }
-
- if (!error && ast_strlen_zero(pickup.exten) && !replaces_chan) {
- /* Oops, someting wrong anyway, no owner, no call */
- ast_log(LOG_NOTICE, "Supervised transfer attempted to replace non-existing call id (%s)!\n", replace_id);
- /* Check for better return code */
- transmit_response_reliable(p, "481 Call Leg Does Not Exist (Replace)", req);
- error = 1;
- }
-
- if (!error && ast_strlen_zero(pickup.exten) &&
- ast_channel_state(replaces_chan) != AST_STATE_RINGING &&
- ast_channel_state(replaces_chan) != AST_STATE_RING &&
- ast_channel_state(replaces_chan) != AST_STATE_UP &&
- /*
- * Check the down state as well because some SIP devices do not
- * give 180 ringing when they can just give 183 session progress
- * instead. same fix the one in ast_can_pickup
- * git show 0a8f9d2cf08
- */
- ast_channel_state(replaces_chan) != AST_STATE_DOWN) {
- ast_log(LOG_NOTICE, "Supervised transfer attempted to replace non-ringing or active call id (%s)!\n", replace_id);
- transmit_response_reliable(p, "603 Declined (Replaces)", req);
- error = 1;
- }
-
- if (error) { /* Give up this dialog */
- append_history(p, "Xfer", "INVITE/Replace Failed.");
- sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
- p->invitestate = INV_COMPLETED;
- res = INV_REQ_ERROR;
- check_via(p, req);
- copy_request(&p->initreq, req);
- goto request_invite_cleanup;
- }
- }
-
- /* Check if this is an INVITE that sets up a new dialog or
- a re-invite in an existing dialog */
-
- if (!req->ignore) {
- int newcall = (p->initreq.headers ? TRUE : FALSE);
-
- sip_cancel_destroy(p);
-
- /* This also counts as a pending invite */
- p->pendinginvite = seqno;
- check_via(p, req);
-
- copy_request(&p->initreq, req); /* Save this INVITE as the transaction basis */
- if (sipdebug)
- ast_debug(1, "Initializing initreq for method %s - callid %s\n", sip_methods[req->method].text, p->callid);
-
- /* Parse new contact both for existing (re-invite) and new calls. */
- parse_ok_contact(p, req);
-
- if (!p->owner) { /* Not a re-invite */
- if (req->debug)
- ast_verbose("Using INVITE request as basis request - %s\n", p->callid);
- if (newcall)
- append_history(p, "Invite", "New call: %s", p->callid);
- } else { /* Re-invite on existing call */
- ast_clear_flag(&p->flags[0], SIP_OUTGOING); /* This is now an inbound dialog */
- if (get_rpid(p, req)) {
- struct ast_party_connected_line connected;
- struct ast_set_party_connected_line update_connected;
-
- ast_party_connected_line_init(&connected);
- memset(&update_connected, 0, sizeof(update_connected));
-
- update_connected.id.number = 1;
- connected.id.number.valid = 1;
- connected.id.number.str = (char *) p->cid_num;
- connected.id.number.presentation = p->callingpres;
-
- update_connected.id.name = 1;
- connected.id.name.valid = 1;
- connected.id.name.str = (char *) p->cid_name;
- connected.id.name.presentation = p->callingpres;
-
- /* Invalidate any earlier private connected id representation */
- ast_set_party_id_all(&update_connected.priv);
-
- connected.id.tag = (char *) p->cid_tag;
- connected.source = AST_CONNECTED_LINE_UPDATE_SOURCE_TRANSFER;
- ast_channel_queue_connected_line_update(p->owner, &connected,
- &update_connected);
- }
- /* Handle SDP here if we already have an owner */
- if (find_sdp(req)) {
- if (process_sdp(p, req, SDP_T38_INITIATE, TRUE)) {
- if (!ast_strlen_zero(sip_get_header(req, "Content-Encoding"))) {
- /* Asterisk does not yet support any Content-Encoding methods. Always
- * attempt to process the sdp, but return a 415 if a Content-Encoding header
- * was present after processing failed. */
- transmit_response_reliable(p, "415 Unsupported Media type", req);
- } else {
- transmit_response_reliable(p, "488 Not acceptable here", req);
- }
- if (!p->lastinvite)
- sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
- res = INV_REQ_ERROR;
- goto request_invite_cleanup;
- }
- ast_queue_control(p->owner, AST_CONTROL_SRCUPDATE);
- } else {
- ast_format_cap_remove_by_type(p->jointcaps, AST_MEDIA_TYPE_UNKNOWN);
- ast_format_cap_append_from_cap(p->jointcaps, p->caps, AST_MEDIA_TYPE_UNKNOWN);
- ast_debug(1, "Hm.... No sdp for the moment\n");
- /* Some devices signal they want to be put off hold by sending a re-invite
- *without* an SDP, which is supposed to mean "Go back to your state"
- and since they put os on remote hold, we go back to off hold */
- if (ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD)) {
- ast_queue_unhold(p->owner);
- /* Activate a re-invite */
- ast_queue_frame(p->owner, &ast_null_frame);
- change_hold_state(p, req, FALSE, 0);
- }
- }
- if (p->do_history) /* This is a response, note what it was for */
- append_history(p, "ReInv", "Re-invite received");
- }
- } else if (req->debug)
- ast_verbose("Ignoring this INVITE request\n");
-
- if (!p->lastinvite && !req->ignore && !p->owner) {
- /* This is a new invite */
- /* Handle authentication if this is our first invite */
- int cc_recall_core_id = -1;
- set_pvt_allowed_methods(p, req);
- res = check_user_full(p, req, SIP_INVITE, e, XMIT_RELIABLE, addr, &authpeer);
- if (res == AUTH_CHALLENGE_SENT) {
- p->invitestate = INV_COMPLETED; /* Needs to restart in another INVITE transaction */
- goto request_invite_cleanup;
- }
- if (res < 0) { /* Something failed in authentication */
- send_check_user_failure_response(p, req, res, XMIT_RELIABLE);
- p->invitestate = INV_COMPLETED;
- sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
- goto request_invite_cleanup;
- }
-
- /* Successful authentication and peer matching so record the peer related to this pvt (for easy access to peer settings) */
- if (p->relatedpeer) {
- p->relatedpeer = sip_unref_peer(p->relatedpeer,"unsetting the relatedpeer field in the dialog, before it is set to something else.");
- }
- if (authpeer) {
- p->relatedpeer = sip_ref_peer(authpeer, "setting dialog's relatedpeer pointer");
- }
-
- req->authenticated = 1;
-
- /* We have a successful authentication, process the SDP portion if there is one */
- if (find_sdp(req)) {
- if (process_sdp(p, req, SDP_T38_INITIATE, TRUE)) {
- /* Asterisk does not yet support any Content-Encoding methods. Always
- * attempt to process the sdp, but return a 415 if a Content-Encoding header
- * was present after processing fails. */
- if (!ast_strlen_zero(sip_get_header(req, "Content-Encoding"))) {
- transmit_response_reliable(p, "415 Unsupported Media type", req);
- } else {
- /* Unacceptable codecs */
- transmit_response_reliable(p, "488 Not acceptable here", req);
- }
- p->invitestate = INV_COMPLETED;
- sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
- ast_debug(1, "No compatible codecs for this SIP call.\n");
- res = INV_REQ_ERROR;
- goto request_invite_cleanup;
- }
- } else { /* No SDP in invite, call control session */
- ast_format_cap_remove_by_type(p->jointcaps, AST_MEDIA_TYPE_UNKNOWN);
- ast_format_cap_append_from_cap(p->jointcaps, p->caps, AST_MEDIA_TYPE_UNKNOWN);
- ast_debug(2, "No SDP in Invite, third party call control\n");
- }
-
- /* Initialize the context if it hasn't been already */
- if (ast_strlen_zero(p->context))
- ast_string_field_set(p, context, sip_cfg.default_context);
-
-
- /* Check number of concurrent calls -vs- incoming limit HERE */
- ast_debug(1, "Checking SIP call limits for device %s\n", p->username);
- if ((res = update_call_counter(p, INC_CALL_LIMIT))) {
- if (res < 0) {
- ast_log(LOG_NOTICE, "Failed to place call for device %s, too many calls\n", p->username);
- transmit_response_reliable(p, "480 Temporarily Unavailable (Call limit) ", req);
- sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
- p->invitestate = INV_COMPLETED;
-
- res = AUTH_SESSION_LIMIT;
- }
-
- goto request_invite_cleanup;
- }
- gotdest = get_destination(p, NULL, &cc_recall_core_id); /* Get destination right away */
- extract_uri(p, req); /* Get the Contact URI */
- build_contact(p, req, 1); /* Build our contact header */
-
- if (p->rtp) {
- ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_DTMF, ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833);
- ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_DTMF_COMPENSATE, ast_test_flag(&p->flags[1], SIP_PAGE2_RFC2833_COMPENSATE));
- }
-
- if (!replace_id && (gotdest != SIP_GET_DEST_EXTEN_FOUND)) { /* No matching extension found */
- switch(gotdest) {
- case SIP_GET_DEST_INVALID_URI:
- transmit_response_reliable(p, "416 Unsupported URI scheme", req);
- break;
- case SIP_GET_DEST_EXTEN_MATCHMORE:
- if (ast_test_flag(&p->flags[1], SIP_PAGE2_ALLOWOVERLAP)
- == SIP_PAGE2_ALLOWOVERLAP_YES) {
- transmit_response_reliable(p, "484 Address Incomplete", req);
- break;
- }
- /*
- * XXX We would have to implement collecting more digits in
- * chan_sip for any other schemes of overlap dialing.
- *
- * For SIP_PAGE2_ALLOWOVERLAP_DTMF it is better to do this in
- * the dialplan using the Incomplete application rather than
- * having the channel driver do it.
- */
- /* Fall through */
- case SIP_GET_DEST_EXTEN_NOT_FOUND:
- {
- char *decoded_exten = ast_strdupa(p->exten);
- transmit_response_reliable(p, "404 Not Found", req);
- ast_uri_decode(decoded_exten, ast_uri_sip_user);
- ast_log(LOG_NOTICE, "Call from '%s' (%s) to extension"
- " '%s' rejected because extension not found in context '%s'.\n",
- S_OR(p->username, p->peername), ast_sockaddr_stringify(&p->recv), decoded_exten, p->context);
- sip_report_failed_acl(p, "no_extension_match");
- }
- break;
- case SIP_GET_DEST_REFUSED:
- default:
- transmit_response_reliable(p, "403 Forbidden", req);
- } /* end switch */
-
- p->invitestate = INV_COMPLETED;
- update_call_counter(p, DEC_CALL_LIMIT);
- sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
- res = INV_REQ_FAILED;
- goto request_invite_cleanup;
- } else {
- /* If no extension was specified, use the s one */
- /* Basically for calling to IP/Host name only */
- if (ast_strlen_zero(p->exten))
- ast_string_field_set(p, exten, "s");
- /* Initialize our tag */
-
- make_our_tag(p);
-
- if (handle_request_invite_st(p, req, reinvite)) {
- p->invitestate = INV_COMPLETED;
- sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
- res = INV_REQ_ERROR;
- goto request_invite_cleanup;
- }
-
- /* First invitation - create the channel. Allocation
- * failures are handled below. */
-
- c = sip_new(p, AST_STATE_DOWN, S_OR(p->peername, NULL), NULL, NULL, p->logger_callid);
-
- if (cc_recall_core_id != -1) {
- ast_setup_cc_recall_datastore(c, cc_recall_core_id);
- ast_cc_agent_set_interfaces_chanvar(c);
- }
- *recount = 1;
-
- /* Save Record-Route for any later requests we make on this dialogue */
- build_route(p, req, 0, 0);
-
- if (c) {
- ast_party_redirecting_init(&redirecting);
- memset(&update_redirecting, 0, sizeof(update_redirecting));
- change_redirecting_information(p, req, &redirecting, &update_redirecting,
- FALSE); /*Will return immediately if no Diversion header is present */
- ast_channel_set_redirecting(c, &redirecting, &update_redirecting);
- ast_party_redirecting_free(&redirecting);
- }
- }
- } else {
- ast_party_redirecting_init(&redirecting);
- memset(&update_redirecting, 0, sizeof(update_redirecting));
- if (sipdebug) {
- if (!req->ignore)
- ast_debug(2, "Got a SIP re-invite for call %s\n", p->callid);
- else
- ast_debug(2, "Got a SIP re-transmit of INVITE for call %s\n", p->callid);
- }
- if (!req->ignore)
- reinvite = 1;
-
- if (handle_request_invite_st(p, req, reinvite)) {
- p->invitestate = INV_COMPLETED;
- if (!p->lastinvite) {
- sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
- }
- res = INV_REQ_ERROR;
- goto request_invite_cleanup;
- }
-
- c = p->owner;
- change_redirecting_information(p, req, &redirecting, &update_redirecting, FALSE); /*Will return immediately if no Diversion header is present */
- if (c) {
- ast_channel_set_redirecting(c, &redirecting, &update_redirecting);
- }
- ast_party_redirecting_free(&redirecting);
- }
-
- /* Check if OLI/ANI-II is present in From: */
- parse_oli(req, p->owner);
-
- if (reinvite && p->stimer) {
- restart_session_timer(p);
- }
-
- if (!req->ignore && p)
- p->lastinvite = seqno;
-
- if (c && replace_id) { /* Attended transfer or call pickup - we're the target */
- if (!ast_strlen_zero(pickup.exten)) {
- append_history(p, "Xfer", "INVITE/Replace received");
-
- /* Let the caller know we're giving it a shot */
- transmit_response(p, "100 Trying", req);
- p->invitestate = INV_PROCEEDING;
- ast_setstate(c, AST_STATE_RING);
-
- /* Do the pickup itself */
- ast_channel_unlock(c);
- *nounlock = 1;
-
- /* since p->owner (c) is unlocked, we need to go ahead and unlock pvt for both
- * magic pickup and ast_hangup. Both of these functions will attempt to lock
- * p->owner again, which can cause a deadlock if we already hold a lock on p.
- * Locking order is, channel then pvt. Dead lock avoidance must be used if
- * called the other way around. */
- sip_pvt_unlock(p);
- do_magic_pickup(c, pickup.exten, pickup.context);
- /* Now we're either masqueraded or we failed to pickup, in either case we... */
- ast_hangup(c);
- sip_pvt_lock(p); /* pvt is expected to remain locked on return, so re-lock it */
-
- res = INV_REQ_FAILED;
- goto request_invite_cleanup;
- } else {
- /* Go and take over the target call */
- if (sipdebug)
- ast_debug(4, "Sending this call to the invite/replaces handler %s\n", p->callid);
- res = handle_invite_replaces(p, req, nounlock, replaces_pvt, replaces_chan);
- goto request_invite_cleanup;
- }
- }
-
-
- if (c) { /* We have a call -either a new call or an old one (RE-INVITE) */
- enum ast_channel_state c_state = ast_channel_state(c);
- RAII_VAR(struct ast_features_pickup_config *, pickup_cfg, ast_get_chan_features_pickup_config(c), ao2_cleanup);
- const char *pickupexten;
-
- if (!pickup_cfg) {
- ast_log(LOG_ERROR, "Unable to retrieve pickup configuration options. Unable to detect call pickup extension\n");
- pickupexten = "";
- } else {
- pickupexten = ast_strdupa(pickup_cfg->pickupexten);
- }
-
- if (c_state != AST_STATE_UP && reinvite &&
- (p->invitestate == INV_TERMINATED || p->invitestate == INV_CONFIRMED)) {
- /* If these conditions are true, and the channel is still in the 'ringing'
- * state, then this likely means that we have a situation where the initial
- * INVITE transaction has completed *but* the channel's state has not yet been
- * changed to UP. The reason this could happen is if the reinvite is received
- * on the SIP socket prior to an application calling ast_read on this channel
- * to read the answer frame we earlier queued on it. In this case, the reinvite
- * is completely legitimate so we need to handle this the same as if the channel
- * were already UP. Thus we are purposely falling through to the AST_STATE_UP case.
- */
- c_state = AST_STATE_UP;
- }
-
- switch(c_state) {
- case AST_STATE_DOWN:
- ast_debug(2, "%s: New call is still down.... Trying... \n", ast_channel_name(c));
- transmit_provisional_response(p, "100 Trying", req, 0);
- p->invitestate = INV_PROCEEDING;
- ast_setstate(c, AST_STATE_RING);
- if (strcmp(p->exten, pickupexten)) { /* Call to extension -start pbx on this call */
- enum ast_pbx_result result;
-
- result = ast_pbx_start(c);
-
- switch(result) {
- case AST_PBX_FAILED:
- sip_alreadygone(p);
- ast_log(LOG_WARNING, "Failed to start PBX :(\n");
- p->invitestate = INV_COMPLETED;
- transmit_response_reliable(p, "503 Unavailable", req);
- break;
- case AST_PBX_CALL_LIMIT:
- sip_alreadygone(p);
- ast_log(LOG_WARNING, "Failed to start PBX (call limit reached) \n");
- p->invitestate = INV_COMPLETED;
- transmit_response_reliable(p, "480 Temporarily Unavailable", req);
- res = AUTH_SESSION_LIMIT;
- break;
- case AST_PBX_SUCCESS:
- /* nothing to do */
- break;
- }
-
- if (result) {
-
- /* Unlock locks so ast_hangup can do its magic */
- ast_channel_unlock(c);
- *nounlock = 1;
- sip_pvt_unlock(p);
- ast_hangup(c);
- sip_pvt_lock(p);
- c = NULL;
- }
- } else { /* Pickup call in call group */
- if (sip_pickup(c)) {
- ast_log(LOG_WARNING, "Failed to start Group pickup by %s\n", ast_channel_name(c));
- transmit_response_reliable(p, "480 Temporarily Unavailable", req);
- sip_alreadygone(p);
- ast_channel_hangupcause_set(c, AST_CAUSE_FAILURE);
-
- /* Unlock locks so ast_hangup can do its magic */
- ast_channel_unlock(c);
- *nounlock = 1;
-
- p->invitestate = INV_COMPLETED;
- sip_pvt_unlock(p);
- ast_hangup(c);
- sip_pvt_lock(p);
- c = NULL;
- }
- }
- break;
- case AST_STATE_RING:
- transmit_provisional_response(p, "100 Trying", req, 0);
- p->invitestate = INV_PROCEEDING;
- break;
- case AST_STATE_RINGING:
- transmit_provisional_response(p, "180 Ringing", req, 0);
- p->invitestate = INV_PROCEEDING;
- break;
- case AST_STATE_UP:
- ast_debug(2, "%s: This call is UP.... \n", ast_channel_name(c));
-
- transmit_response(p, "100 Trying", req);
-
- if (p->t38.state == T38_PEER_REINVITE) {
- start_t38_abort_timer(p);
- } else if (p->t38.state == T38_ENABLED) {
- ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
- transmit_response_with_t38_sdp(p, "200 OK", req, (reinvite ? XMIT_RELIABLE : (req->ignore ? XMIT_UNRELIABLE : XMIT_CRITICAL)));
- } else if ((p->t38.state == T38_DISABLED) || (p->t38.state == T38_REJECTED)) {
- /* If this is not a re-invite or something to ignore - it's critical */
- if (p->srtp && !ast_test_flag(p->srtp, AST_SRTP_CRYPTO_OFFER_OK)) {
- ast_log(LOG_WARNING, "Target does not support required crypto\n");
- transmit_response_reliable(p, "488 Not Acceptable Here (crypto)", req);
- } else {
- ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
- transmit_response_with_sdp(p, "200 OK", req, (reinvite ? XMIT_RELIABLE : (req->ignore ? XMIT_UNRELIABLE : XMIT_CRITICAL)), p->session_modify == TRUE ? FALSE : TRUE, FALSE);
- ast_queue_control(p->owner, AST_CONTROL_UPDATE_RTP_PEER);
- }
- }
-
- p->invitestate = INV_TERMINATED;
- break;
- default:
- ast_log(LOG_WARNING, "Don't know how to handle INVITE in state %u\n", ast_channel_state(c));
- transmit_response(p, "100 Trying", req);
- break;
- }
- } else {
- if (!req->ignore && p && (p->autokillid == -1)) {
- const char *msg;
-
- if ((!ast_format_cap_count(p->jointcaps)))
- msg = "488 Not Acceptable Here (codec error)";
- else {
- ast_log(LOG_NOTICE, "Unable to create/find SIP channel for this INVITE\n");
- msg = "503 Unavailable";
- }
- transmit_response_reliable(p, msg, req);
- p->invitestate = INV_COMPLETED;
- sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
- }
- }
-
-request_invite_cleanup:
-
- if (authpeer) {
- authpeer = sip_unref_peer(authpeer, "sip_unref_peer, from handle_request_invite authpeer");
- }
-
- return res;
-}
-
-/*! \brief Check for the presence of OLI tag(s) in the From header and set on the channel
- */
-static void parse_oli(struct sip_request *req, struct ast_channel *chan)
-{
- const char *from = NULL;
- const char *s = NULL;
- int ani2 = 0;
-
- if (!chan || !req) {
- /* null pointers are not helpful */
- return;
- }
-
- from = sip_get_header(req, "From");
- if (ast_strlen_zero(from)) {
- /* no From header */
- return;
- }
-
- /* Look for the possible OLI tags. */
- if ((s = strcasestr(from, ";isup-oli="))) {
- s += 10;
- } else if ((s = strcasestr(from, ";ss7-oli="))) {
- s += 9;
- } else if ((s = strcasestr(from, ";oli="))) {
- s += 5;
- }
-
- if (ast_strlen_zero(s)) {
- /* OLI tag is missing, or present with nothing following the '=' sign */
- return;
- }
-
- /* just in case OLI is quoted */
- if (*s == '\"') {
- s++;
- }
-
- if (sscanf(s, "%d", &ani2)) {
- ast_channel_caller(chan)->ani2 = ani2;
- }
-
- return;
-}
-
-/*! \brief Find all call legs and bridge transferee with target
- * called from handle_request_refer
- *
- * \note this function assumes two locks to begin with, sip_pvt transferer and current.chan1 (the pvt's owner)...
- * 2 additional locks are held at the beginning of the function, targetcall_pvt, and targetcall_pvt's owner
- * channel (which is stored in target.chan1). These 2 locks _MUST_ be let go by the end of the function. Do
- * not be confused into thinking a pvt's owner is the same thing as the channels locked at the beginning of
- * this function, after the masquerade this may not be true. Be consistent and unlock only the exact same
- * pointers that were locked to begin with.
- *
- * If this function is successful, only the transferer pvt lock will remain on return. Setting nounlock indicates
- * to handle_request_do() that the pvt's owner it locked does not require an unlock.
- */
-static int local_attended_transfer(struct sip_pvt *transferer, struct ast_channel *transferer_chan, uint32_t seqno, int *nounlock)
-{
- RAII_VAR(struct sip_pvt *, targetcall_pvt, NULL, ao2_cleanup);
- RAII_VAR(struct ast_channel *, targetcall_chan, NULL, ao2_cleanup);
- enum ast_transfer_result transfer_res;
-
- /* Check if the call ID of the replaces header does exist locally */
- if (get_sip_pvt_from_replaces(transferer->refer->replaces_callid,
- transferer->refer->replaces_callid_totag,
- transferer->refer->replaces_callid_fromtag,
- &targetcall_pvt, &targetcall_chan)) {
- if (transferer->refer->localtransfer) {
- /* We did not find the refered call. Sorry, can't accept then */
- /* Let's fake a response from someone else in order
- to follow the standard */
- transmit_notify_with_sipfrag(transferer, seqno, "481 Call leg/transaction does not exist", TRUE);
- append_history(transferer, "Xfer", "Refer failed");
- ast_clear_flag(&transferer->flags[0], SIP_GOTREFER);
- transferer->refer->status = REFER_FAILED;
- return -1;
- }
- /* Fall through for remote transfers that we did not find locally */
- ast_debug(3, "SIP attended transfer: Not our call - generating INVITE with replaces\n");
- return 0;
- }
-
- if (!targetcall_chan) { /* No active channel */
- ast_debug(4, "SIP attended transfer: Error: No owner of target call\n");
- /* Cancel transfer */
- transmit_notify_with_sipfrag(transferer, seqno, "503 Service Unavailable", TRUE);
- append_history(transferer, "Xfer", "Refer failed");
- ast_clear_flag(&transferer->flags[0], SIP_GOTREFER);
- transferer->refer->status = REFER_FAILED;
- return -1;
- }
-
- ast_set_flag(&transferer->flags[0], SIP_DEFER_BYE_ON_TRANSFER); /* Delay hangup */
-
- sip_pvt_unlock(transferer);
- ast_channel_unlock(transferer_chan);
- *nounlock = 1;
-
- transfer_res = ast_bridge_transfer_attended(transferer_chan, targetcall_chan);
-
- sip_pvt_lock(transferer);
-
- switch (transfer_res) {
- case AST_BRIDGE_TRANSFER_SUCCESS:
- transferer->refer->status = REFER_200OK;
- transmit_notify_with_sipfrag(transferer, seqno, "200 OK", TRUE);
- append_history(transferer, "Xfer", "Refer succeeded");
- return 1;
- case AST_BRIDGE_TRANSFER_FAIL:
- transferer->refer->status = REFER_FAILED;
- transmit_notify_with_sipfrag(transferer, seqno, "500 Internal Server Error", TRUE);
- append_history(transferer, "Xfer", "Refer failed (internal error)");
- ast_clear_flag(&transferer->flags[0], SIP_DEFER_BYE_ON_TRANSFER);
- return -1;
- case AST_BRIDGE_TRANSFER_INVALID:
- transferer->refer->status = REFER_FAILED;
- transmit_notify_with_sipfrag(transferer, seqno, "503 Service Unavailable", TRUE);
- append_history(transferer, "Xfer", "Refer failed (invalid bridge state)");
- ast_clear_flag(&transferer->flags[0], SIP_DEFER_BYE_ON_TRANSFER);
- return -1;
- case AST_BRIDGE_TRANSFER_NOT_PERMITTED:
- transferer->refer->status = REFER_FAILED;
- transmit_notify_with_sipfrag(transferer, seqno, "403 Forbidden", TRUE);
- append_history(transferer, "Xfer", "Refer failed (operation not permitted)");
- ast_clear_flag(&transferer->flags[0], SIP_DEFER_BYE_ON_TRANSFER);
- return -1;
- default:
- break;
- }
-
- return 1;
-}
-
-/*!
- * Data to set on a channel that runs dialplan
- * at the completion of a blind transfer
- */
-struct blind_transfer_cb_data {
- /*! Contents of the REFER's Referred-by header */
- const char *referred_by;
- /*! Domain of the URI in the REFER's Refer-To header */
- const char *domain;
- /*! Contents of what to place in a Replaces header of an INVITE */
- const char *replaces;
- /*! Redirecting information to set on the channel */
- struct ast_party_redirecting redirecting;
- /*! Parts of the redirecting structure that are to be updated */
- struct ast_set_party_redirecting update_redirecting;
-};
-
-/*!
- * \internal
- * \brief Callback called on new outbound channel during blind transfer
- *
- * We use this opportunity to populate the channel with data from the REFER
- * so that, if necessary, we can include proper information on any new INVITE
- * we may send out.
- *
- * \param chan The new outbound channel
- * \param user_data_wrapper A blind_transfer_cb_data struct
- * \param transfer_type Unused
- */
-static void blind_transfer_cb(struct ast_channel *chan, struct transfer_channel_data *user_data_wrapper,
- enum ast_transfer_type transfer_type)
-{
- struct blind_transfer_cb_data *cb_data = user_data_wrapper->data;
-
- pbx_builtin_setvar_helper(chan, "SIPTRANSFER", "yes");
- pbx_builtin_setvar_helper(chan, "SIPTRANSFER_REFERER", cb_data->referred_by);
- pbx_builtin_setvar_helper(chan, "SIPTRANSFER_REPLACES", cb_data->replaces);
- pbx_builtin_setvar_helper(chan, "SIPDOMAIN", cb_data->domain);
- ast_channel_update_redirecting(chan, &cb_data->redirecting, &cb_data->update_redirecting);
-}
-
-/*! \brief Handle incoming REFER request */
-/*! \page SIP_REFER SIP transfer Support (REFER)
-
- REFER is used for call transfer in SIP. We get a REFER
- to place a new call with an INVITE somwhere and then
- keep the transferor up-to-date of the transfer. If the
- transfer fails, get back on line with the orginal call.
-
- - REFER can be sent outside or inside of a dialog.
- Asterisk only accepts REFER inside of a dialog.
-
- - If we get a replaces header, it is an attended transfer
-
- \par Blind transfers
- The transferor provides the transferee
- with the transfer targets contact. The signalling between
- transferer or transferee should not be cancelled, so the
- call is recoverable if the transfer target can not be reached
- by the transferee.
-
- In this case, Asterisk receives a TRANSFER from
- the transferor, thus is the transferee. We should
- try to set up a call to the contact provided
- and if that fails, re-connect the current session.
- If the new call is set up, we issue a hangup.
- In this scenario, we are following section 5.2
- in the SIP CC Transfer draft. (Transfer without
- a GRUU)
-
- \par Transfer with consultation hold
- In this case, the transferor
- talks to the transfer target before the transfer takes place.
- This is implemented with SIP hold and transfer.
- Note: The invite From: string could indicate a transfer.
- (Section 6. Transfer with consultation hold)
- The transferor places the transferee on hold, starts a call
- with the transfer target to alert them to the impending
- transfer, terminates the connection with the target, then
- proceeds with the transfer (as in Blind transfer above)
-
- \par Attended transfer
- The transferor places the transferee
- on hold, calls the transfer target to alert them,
- places the target on hold, then proceeds with the transfer
- using a Replaces header field in the Refer-to header. This
- will force the transfee to send an Invite to the target,
- with a replaces header that instructs the target to
- hangup the call between the transferor and the target.
- In this case, the Refer/to: uses the AOR address. (The same
- URI that the transferee used to establish the session with
- the transfer target (To: ). The Require: replaces header should
- be in the INVITE to avoid the wrong UA in a forked SIP proxy
- scenario to answer and have no call to replace with.
-
- The referred-by header is *NOT* required, but if we get it,
- can be copied into the INVITE to the transfer target to
- inform the target about the transferor
-
- "Any REFER request has to be appropriately authenticated.".
-
- We can't destroy dialogs, since we want the call to continue.
-
- */
-static int handle_request_refer(struct sip_pvt *p, struct sip_request *req, uint32_t seqno, int *nounlock)
-{
- char *refer_to = NULL;
- char *refer_to_context = NULL;
- int res = 0;
- struct blind_transfer_cb_data cb_data;
- enum ast_transfer_result transfer_res;
- RAII_VAR(struct ast_channel *, transferer, NULL, ast_channel_cleanup);
- RAII_VAR(struct ast_str *, replaces_str, NULL, ast_free_ptr);
-
- if (req->debug) {
- ast_verbose("Call %s got a SIP call transfer from %s: (REFER)!\n",
- p->callid,
- ast_test_flag(&p->flags[0], SIP_OUTGOING) ? "callee" : "caller");
- }
-
- if (!p->owner) {
- /* This is a REFER outside of an existing SIP dialog */
- /* We can't handle that, so decline it */
- ast_debug(3, "Call %s: Declined REFER, outside of dialog...\n", p->callid);
- transmit_response(p, "603 Declined (No dialog)", req);
- if (!req->ignore) {
- append_history(p, "Xfer", "Refer failed. Outside of dialog.");
- sip_alreadygone(p);
- pvt_set_needdestroy(p, "outside of dialog");
- }
- return 0;
- }
-
- /* Check if transfer is allowed from this device */
- if (p->allowtransfer == TRANSFER_CLOSED ) {
- /* Transfer not allowed, decline */
- transmit_response(p, "603 Declined (policy)", req);
- append_history(p, "Xfer", "Refer failed. Allowtransfer == closed.");
- /* Do not destroy SIP session */
- return 0;
- }
-
- if (!req->ignore && ast_test_flag(&p->flags[0], SIP_GOTREFER)) {
- /* Already have a pending REFER */
- transmit_response(p, "491 Request pending", req);
- append_history(p, "Xfer", "Refer failed. Request pending.");
- return 0;
- }
-
- /* Allocate memory for call transfer data */
- if (!sip_refer_alloc(p)) {
- transmit_response(p, "500 Internal Server Error", req);
- append_history(p, "Xfer", "Refer failed. Memory allocation error.");
- return -3;
- }
-
- res = get_refer_info(p, req); /* Extract headers */
-
- p->refer->status = REFER_SENT;
-
- if (res != 0) {
- switch (res) {
- case -2: /* Syntax error */
- transmit_response(p, "400 Bad Request (Refer-to missing)", req);
- append_history(p, "Xfer", "Refer failed. Refer-to missing.");
- if (req->debug) {
- ast_debug(1, "SIP transfer to black hole can't be handled (no refer-to: )\n");
- }
- break;
- case -3:
- transmit_response(p, "603 Declined (Non sip: uri)", req);
- append_history(p, "Xfer", "Refer failed. Non SIP uri");
- if (req->debug) {
- ast_debug(1, "SIP transfer to non-SIP uri denied\n");
- }
- break;
- default:
- /* Refer-to extension not found, fake a failed transfer */
- transmit_response(p, "202 Accepted", req);
- append_history(p, "Xfer", "Refer failed. Bad extension.");
- transmit_notify_with_sipfrag(p, seqno, "404 Not found", TRUE);
- ast_clear_flag(&p->flags[0], SIP_GOTREFER);
- if (req->debug) {
- ast_debug(1, "SIP transfer to bad extension: %s\n", p->refer->refer_to);
- }
- break;
- }
- return 0;
- }
-
- if (ast_strlen_zero(p->context)) {
- ast_string_field_set(p, context, sip_cfg.default_context);
- }
-
- /* If we do not support SIP domains, all transfers are local */
- if (sip_cfg.allow_external_domains && check_sip_domain(p->refer->refer_to_domain, NULL, 0)) {
- p->refer->localtransfer = 1;
- if (sipdebug) {
- ast_debug(3, "This SIP transfer is local : %s\n", p->refer->refer_to_domain);
- }
- } else if (AST_LIST_EMPTY(&domain_list) || check_sip_domain(p->refer->refer_to_domain, NULL, 0)) {
- /* This PBX doesn't bother with SIP domains or domain is local, so this transfer is local */
- p->refer->localtransfer = 1;
- } else if (sipdebug) {
- ast_debug(3, "This SIP transfer is to a remote SIP extension (remote domain %s)\n", p->refer->refer_to_domain);
- }
-
- /* Is this a repeat of a current request? Ignore it */
- /* Don't know what else to do right now. */
- if (req->ignore) {
- return 0;
- }
-
- /* Get the transferer's channel */
- transferer = ast_channel_ref(p->owner);
-
- if (sipdebug) {
- ast_debug(3, "SIP %s transfer: Transferer channel %s\n",
- p->refer->attendedtransfer ? "attended" : "blind",
- ast_channel_name(transferer));
- }
-
- ast_set_flag(&p->flags[0], SIP_GOTREFER);
-
- /* From here on failures will be indicated with NOTIFY requests */
- transmit_response(p, "202 Accepted", req);
-
- /* Attended transfer: Find all call legs and bridge transferee with target*/
- if (p->refer->attendedtransfer) {
- /* both p and p->owner _MUST_ be locked while calling local_attended_transfer */
- if ((res = local_attended_transfer(p, transferer, seqno, nounlock))) {
- ast_clear_flag(&p->flags[0], SIP_GOTREFER);
- return res;
- }
- /* Fall through for remote transfers that we did not find locally */
- if (sipdebug) {
- ast_debug(4, "SIP attended transfer: Still not our call - generating INVITE with replaces\n");
- }
- /* Fallthrough if we can't find the call leg internally */
- }
-
- /* Copy data we can not safely access after letting the pvt lock go. */
- refer_to = ast_strdupa(p->refer->refer_to);
- refer_to_context = ast_strdupa(p->refer->refer_to_context);
-
- ast_party_redirecting_init(&cb_data.redirecting);
- memset(&cb_data.update_redirecting, 0, sizeof(cb_data.update_redirecting));
- change_redirecting_information(p, req, &cb_data.redirecting, &cb_data.update_redirecting, 0);
-
- cb_data.domain = ast_strdupa(p->refer->refer_to_domain);
- cb_data.referred_by = ast_strdupa(p->refer->referred_by);
-
- if (!ast_strlen_zero(p->refer->replaces_callid)) {
- replaces_str = ast_str_create(128);
- if (!replaces_str) {
- ast_log(LOG_NOTICE, "Unable to create Replaces string for remote attended transfer. Transfer failed\n");
- ast_clear_flag(&p->flags[0], SIP_GOTREFER);
- ast_party_redirecting_free(&cb_data.redirecting);
- return -1;
- }
- ast_str_append(&replaces_str, 0, "%s%s%s%s%s", p->refer->replaces_callid,
- !ast_strlen_zero(p->refer->replaces_callid_totag) ? ";to-tag=" : "",
- S_OR(p->refer->replaces_callid_totag, ""),
- !ast_strlen_zero(p->refer->replaces_callid_fromtag) ? ";from-tag=" : "",
- S_OR(p->refer->replaces_callid_fromtag, ""));
- cb_data.replaces = ast_str_buffer(replaces_str);
- } else {
- cb_data.replaces = NULL;
- }
-
- if (!*nounlock) {
- ast_channel_unlock(p->owner);
- *nounlock = 1;
- }
-
- ast_set_flag(&p->flags[0], SIP_DEFER_BYE_ON_TRANSFER);
- sip_pvt_unlock(p);
- transfer_res = ast_bridge_transfer_blind(1, transferer, refer_to, refer_to_context, blind_transfer_cb, &cb_data);
- sip_pvt_lock(p);
-
- switch (transfer_res) {
- case AST_BRIDGE_TRANSFER_INVALID:
- res = -1;
- p->refer->status = REFER_FAILED;
- transmit_notify_with_sipfrag(p, seqno, "503 Service Unavailable (can't handle one-legged xfers)", TRUE);
- append_history(p, "Xfer", "Refer failed (only bridged calls).");
- ast_clear_flag(&p->flags[0], SIP_DEFER_BYE_ON_TRANSFER);
- break;
- case AST_BRIDGE_TRANSFER_NOT_PERMITTED:
- res = -1;
- p->refer->status = REFER_FAILED;
- transmit_notify_with_sipfrag(p, seqno, "403 Forbidden", TRUE);
- append_history(p, "Xfer", "Refer failed (bridge does not permit transfers)");
- ast_clear_flag(&p->flags[0], SIP_DEFER_BYE_ON_TRANSFER);
- break;
- case AST_BRIDGE_TRANSFER_FAIL:
- res = -1;
- p->refer->status = REFER_FAILED;
- transmit_notify_with_sipfrag(p, seqno, "500 Internal Server Error", TRUE);
- append_history(p, "Xfer", "Refer failed (internal error)");
- ast_clear_flag(&p->flags[0], SIP_DEFER_BYE_ON_TRANSFER);
- break;
- case AST_BRIDGE_TRANSFER_SUCCESS:
- res = 0;
- p->refer->status = REFER_200OK;
- transmit_notify_with_sipfrag(p, seqno, "200 OK", TRUE);
- append_history(p, "Xfer", "Refer succeeded.");
- break;
- default:
- break;
- }
-
- ast_clear_flag(&p->flags[0], SIP_GOTREFER);
- ast_party_redirecting_free(&cb_data.redirecting);
- return res;
-}
-
-/*! \brief Handle incoming CANCEL request */
-static int handle_request_cancel(struct sip_pvt *p, struct sip_request *req)
-{
-
- check_via(p, req);
- sip_alreadygone(p);
-
- if (p->owner && ast_channel_state(p->owner) == AST_STATE_UP) {
- /* This call is up, cancel is ignored, we need a bye */
- transmit_response(p, "200 OK", req);
- ast_debug(1, "Got CANCEL on an answered call. Ignoring... \n");
- return 0;
- }
-
- use_reason_header(p, req);
-
- /* At this point, we could have cancelled the invite at the same time
- as the other side sends a CANCEL. Our final reply with error code
- might not have been received by the other side before the CANCEL
- was sent, so let's just give up retransmissions and waiting for
- ACK on our error code. The call is hanging up any way. */
- if (p->invitestate == INV_TERMINATED || p->invitestate == INV_COMPLETED) {
- __sip_pretend_ack(p);
- }
- if (p->invitestate != INV_TERMINATED)
- p->invitestate = INV_CANCELLED;
-
- if (ast_test_flag(&p->flags[0], SIP_INC_COUNT) || ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD))
- update_call_counter(p, DEC_CALL_LIMIT);
-
- stop_media_flows(p); /* Immediately stop RTP, VRTP and UDPTL as applicable */
- if (p->owner) {
- sip_queue_hangup_cause(p, ast_channel_hangupcause(p->owner));
- } else {
- sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
- }
- if (p->initreq.data && ast_str_strlen(p->initreq.data) > 0) {
- struct sip_pkt *pkt, *prev_pkt;
- /* If the CANCEL we are receiving is a retransmission, and we already have scheduled
- * a reliable 487, then we don't want to schedule another one on top of the previous
- * one.
- *
- * As odd as this may sound, we can't rely on the previously-transmitted "reliable"
- * response in this situation. What if we've sent all of our reliable responses
- * already and now all of a sudden, we get this second CANCEL?
- *
- * The only way to do this correctly is to cancel our previously-scheduled reliably-
- * transmitted response and send a new one in its place.
- */
- for (pkt = p->packets, prev_pkt = NULL; pkt; prev_pkt = pkt, pkt = pkt->next) {
- if (pkt->seqno == p->lastinvite && pkt->response_code == 487) {
- /* Unlink and destroy the packet object. */
- UNLINK(pkt, p->packets, prev_pkt);
- stop_retrans_pkt(pkt);
- ao2_t_ref(pkt, -1, "Packet retransmission list");
- break;
- }
- }
- transmit_response_reliable(p, "487 Request Terminated", &p->initreq);
- transmit_response(p, "200 OK", req);
- return 1;
- } else {
- transmit_response(p, "481 Call Leg Does Not Exist", req);
- return 0;
- }
-}
-
-/*! \brief Handle incoming BYE request */
-static int handle_request_bye(struct sip_pvt *p, struct sip_request *req)
-{
- struct ast_channel *c=NULL;
- int res;
- const char *required;
- RAII_VAR(struct ast_channel *, peer_channel, NULL, ast_channel_cleanup);
- char quality_buf[AST_MAX_USER_FIELD], *quality;
-
- /* If we have an INCOMING invite that we haven't answered, terminate that transaction */
- if (p->pendinginvite && !ast_test_flag(&p->flags[0], SIP_OUTGOING) && !req->ignore) {
- transmit_response_reliable(p, "487 Request Terminated", &p->initreq);
- }
-
- __sip_pretend_ack(p);
-
- p->invitestate = INV_TERMINATED;
-
- copy_request(&p->initreq, req);
- if (sipdebug)
- ast_debug(1, "Initializing initreq for method %s - callid %s\n", sip_methods[req->method].text, p->callid);
- check_via(p, req);
- sip_alreadygone(p);
-
- if (p->owner) {
- RAII_VAR(struct ast_channel *, owner_relock, NULL, ast_channel_cleanup);
- RAII_VAR(struct ast_channel *, owner_ref, NULL, ast_channel_cleanup);
-
- /* Grab a reference to p->owner to prevent it from going away */
- owner_ref = ast_channel_ref(p->owner);
-
- /* Established locking order here is bridge, channel, pvt
- * and the bridge will be locked during ast_channel_bridge_peer */
- ast_channel_unlock(owner_ref);
- sip_pvt_unlock(p);
-
- peer_channel = ast_channel_bridge_peer(owner_ref);
-
- owner_relock = sip_pvt_lock_full(p);
- if (!owner_relock) {
- ast_debug(3, "Unable to reacquire owner channel lock, channel is gone\n");
- return 0;
- }
- }
-
- /* Get RTCP quality before end of call */
- if (p->rtp) {
- if (p->do_history) {
- if ((quality = ast_rtp_instance_get_quality(p->rtp, AST_RTP_INSTANCE_STAT_FIELD_QUALITY, quality_buf, sizeof(quality_buf)))) {
- append_history(p, "RTCPaudio", "Quality:%s", quality);
- }
- if ((quality = ast_rtp_instance_get_quality(p->rtp, AST_RTP_INSTANCE_STAT_FIELD_QUALITY_JITTER, quality_buf, sizeof(quality_buf)))) {
- append_history(p, "RTCPaudioJitter", "Quality:%s", quality);
- }
- if ((quality = ast_rtp_instance_get_quality(p->rtp, AST_RTP_INSTANCE_STAT_FIELD_QUALITY_LOSS, quality_buf, sizeof(quality_buf)))) {
- append_history(p, "RTCPaudioLoss", "Quality:%s", quality);
- }
- if ((quality = ast_rtp_instance_get_quality(p->rtp, AST_RTP_INSTANCE_STAT_FIELD_QUALITY_RTT, quality_buf, sizeof(quality_buf)))) {
- append_history(p, "RTCPaudioRTT", "Quality:%s", quality);
- }
- }
-
- if (p->owner) {
- RAII_VAR(struct ast_channel *, owner_relock, NULL, ast_channel_cleanup);
- RAII_VAR(struct ast_channel *, owner_ref, NULL, ast_channel_cleanup);
- struct ast_rtp_instance *p_rtp;
-
- /* Grab a reference to p->owner to prevent it from going away */
- owner_ref = ast_channel_ref(p->owner);
-
- p_rtp = p->rtp;
- ao2_ref(p_rtp, +1);
-
- /* Established locking order here is bridge, channel, pvt
- * and the bridge and channel will be locked during
- * ast_rtp_instance_set_stats_vars */
- ast_channel_unlock(owner_ref);
- sip_pvt_unlock(p);
-
- ast_rtp_instance_set_stats_vars(owner_ref, p_rtp);
- ao2_ref(p_rtp, -1);
-
- if (peer_channel) {
- ast_channel_lock(peer_channel);
- if (IS_SIP_TECH(ast_channel_tech(peer_channel))) {
- struct sip_pvt *peer_pvt;
-
- peer_pvt = ast_channel_tech_pvt(peer_channel);
- if (peer_pvt) {
- ao2_ref(peer_pvt, +1);
- sip_pvt_lock(peer_pvt);
- if (peer_pvt->rtp) {
- struct ast_rtp_instance *peer_rtp;
-
- peer_rtp = peer_pvt->rtp;
- ao2_ref(peer_rtp, +1);
- ast_channel_unlock(peer_channel);
- sip_pvt_unlock(peer_pvt);
- ast_rtp_instance_set_stats_vars(peer_channel, peer_rtp);
- ao2_ref(peer_rtp, -1);
- ast_channel_lock(peer_channel);
- sip_pvt_lock(peer_pvt);
- }
- sip_pvt_unlock(peer_pvt);
- ao2_ref(peer_pvt, -1);
- }
- }
- ast_channel_unlock(peer_channel);
- }
-
- owner_relock = sip_pvt_lock_full(p);
- if (!owner_relock) {
- ast_debug(3, "Unable to reacquire owner channel lock, channel is gone\n");
- return 0;
- }
- }
- }
-
- if (p->vrtp && (quality = ast_rtp_instance_get_quality(p->vrtp, AST_RTP_INSTANCE_STAT_FIELD_QUALITY, quality_buf, sizeof(quality_buf)))) {
- if (p->do_history) {
- append_history(p, "RTCPvideo", "Quality:%s", quality);
- }
- if (p->owner) {
- pbx_builtin_setvar_helper(p->owner, "RTPVIDEOQOS", quality);
- }
- }
-
- if (p->trtp && (quality = ast_rtp_instance_get_quality(p->trtp, AST_RTP_INSTANCE_STAT_FIELD_QUALITY, quality_buf, sizeof(quality_buf)))) {
- if (p->do_history) {
- append_history(p, "RTCPtext", "Quality:%s", quality);
- }
- if (p->owner) {
- pbx_builtin_setvar_helper(p->owner, "RTPTEXTQOS", quality);
- }
- }
-
- stop_media_flows(p); /* Immediately stop RTP, VRTP and UDPTL as applicable */
- if (p->stimer) {
- stop_session_timer(p); /* Stop Session-Timer */
- }
-
- use_reason_header(p, req);
- if (!ast_strlen_zero(sip_get_header(req, "Also"))) {
- ast_log(LOG_NOTICE, "Client '%s' using deprecated BYE/Also transfer method. Ask vendor to support REFER instead\n",
- ast_sockaddr_stringify(&p->recv));
- if (ast_strlen_zero(p->context))
- ast_string_field_set(p, context, sip_cfg.default_context);
- res = get_also_info(p, req);
- if (!res) {
- c = p->owner;
- if (c) {
- if (peer_channel) {
- RAII_VAR(struct ast_channel *, owner_relock, NULL, ast_channel_cleanup);
- char *local_context = ast_strdupa(p->context);
- char *local_refer_to = ast_strdupa(p->refer->refer_to);
-
- /* Grab a reference to p->owner to prevent it from going away */
- ast_channel_ref(c);
-
- /* Don't actually hangup here... */
- ast_queue_unhold(c);
- ast_channel_unlock(c); /* async_goto can do a masquerade, no locks can be held during a masq */
- sip_pvt_unlock(p);
-
- ast_async_goto(peer_channel, local_context, local_refer_to, 1);
-
- owner_relock = sip_pvt_lock_full(p);
- ast_channel_cleanup(c);
- if (!owner_relock) {
- ast_debug(3, "Unable to reacquire owner channel lock, channel is gone\n");
- return 0;
- }
- } else {
- ast_queue_hangup(p->owner);
- }
- }
- } else {
- ast_log(LOG_WARNING, "Invalid transfer information from '%s'\n", ast_sockaddr_stringify(&p->recv));
- if (p->owner)
- ast_queue_hangup_with_cause(p->owner, AST_CAUSE_PROTOCOL_ERROR);
- }
- } else if (p->owner) {
- sip_queue_hangup_cause(p, ast_channel_hangupcause(p->owner));
- sip_scheddestroy_final(p, DEFAULT_TRANS_TIMEOUT);
- ast_debug(3, "Received bye, issuing owner hangup\n");
- } else {
- sip_scheddestroy_final(p, DEFAULT_TRANS_TIMEOUT);
- ast_debug(3, "Received bye, no owner, selfdestruct soon.\n");
- }
- ast_clear_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
-
- /* Find out what they require */
- required = sip_get_header(req, "Require");
- if (!ast_strlen_zero(required)) {
- char unsupported[256] = { 0, };
- parse_sip_options(required, unsupported, ARRAY_LEN(unsupported));
- /* If there are any options required that we do not support,
- * then send a 420 with only those unsupported options listed */
- if (!ast_strlen_zero(unsupported)) {
- transmit_response_with_unsupported(p, "420 Bad extension (unsupported)", req, unsupported);
- ast_log(LOG_WARNING, "Received SIP BYE with unsupported required extension: required:%s unsupported:%s\n", required, unsupported);
- } else {
- transmit_response(p, "200 OK", req);
- }
- } else {
- transmit_response(p, "200 OK", req);
- }
-
- /* Destroy any pending invites so we won't try to do another
- * scheduled reINVITE. */
- stop_reinvite_retry(p);
-
- return 1;
-}
-
-/*! \brief Handle incoming MESSAGE request */
-static int handle_request_message(struct sip_pvt *p, struct sip_request *req, struct ast_sockaddr *addr, const char *e)
-{
- if (!req->ignore) {
- if (req->debug)
- ast_verbose("Receiving message!\n");
- receive_message(p, req, addr, e);
- } else
- transmit_response(p, "202 Accepted", req);
- return 1;
-}
-
-static int sip_msg_send(const struct ast_msg *msg, const char *to, const char *from);
-
-static const struct ast_msg_tech sip_msg_tech = {
- .name = "sip",
- .msg_send = sip_msg_send,
-};
-
-/*!
- * \internal
- * \brief Check if the given header name is blocked.
- *
- * \details Determine if the given header name from the user is
- * blocked for outgoing MESSAGE packets.
- *
- * \param header_name Name of header to see if it is blocked.
- *
- * \retval TRUE if the given header is blocked.
- */
-static int block_msg_header(const char *header_name)
-{
- int idx;
-
- /*
- * Don't block Content-Type or Max-Forwards headers because the
- * user can override them.
- */
- static const char *hdr[] = {
- "To",
- "From",
- "Via",
- "Route",
- "Contact",
- "Call-ID",
- "CSeq",
- "Allow",
- "Content-Length",
- "Request-URI",
- };
-
- for (idx = 0; idx < ARRAY_LEN(hdr); ++idx) {
- if (!strcasecmp(header_name, hdr[idx])) {
- /* Block addition of this header. */
- return 1;
- }
- }
- return 0;
-}
-
-static int sip_msg_send(const struct ast_msg *msg, const char *to, const char *from)
-{
- struct sip_pvt *pvt;
- int res;
- char *to_uri;
- char *to_host;
- char *to_user;
- const char *var;
- const char *val;
- struct ast_msg_var_iterator *iter;
- struct sip_peer *peer_ptr;
-
- if (!(pvt = sip_alloc(NULL, NULL, 0, SIP_MESSAGE, NULL, 0))) {
- return -1;
- }
-
- for (iter = ast_msg_var_iterator_init(msg);
- ast_msg_var_iterator_next(msg, iter, &var, &val);
- ast_msg_var_unref_current(iter)) {
- if (!strcasecmp(var, "Request-URI")) {
- ast_string_field_set(pvt, fullcontact, val);
- break;
- }
- }
- ast_msg_var_iterator_destroy(iter);
-
- to_uri = ast_strdupa(to);
- to_uri = get_in_brackets(to_uri);
- parse_uri(to_uri, "sip:,sips:", &to_user, NULL, &to_host, NULL);
-
- if (ast_strlen_zero(to_host)) {
- ast_log(LOG_WARNING, "MESSAGE(to) is invalid for SIP - '%s'\n", to);
- dialog_unlink_all(pvt);
- dialog_unref(pvt, "MESSAGE(to) is invalid for SIP");
- return -1;
- }
-
- if (!ast_strlen_zero(from)) {
- if ((peer_ptr = sip_find_peer(from, NULL, 0, 1, 0, 0))) {
- ast_string_field_set(pvt, fromname, S_OR(peer_ptr->cid_name, peer_ptr->name));
- ast_string_field_set(pvt, fromuser, S_OR(peer_ptr->cid_num, peer_ptr->name));
- sip_unref_peer(peer_ptr, "sip_unref_peer, from sip_msg_send, sip_find_peer");
- } else if (strchr(from, '<')) { /* from is callerid-style */
- char *sender;
- char *name = NULL, *location = NULL, *user = NULL, *domain = NULL;
-
- sender = ast_strdupa(from);
- ast_callerid_parse(sender, &name, &location);
- if (ast_strlen_zero(location)) {
- /* This can occur if either
- * 1) A name-addr style From header does not close the angle brackets
- * properly.
- * 2) The From header is not in name-addr style and the content of the
- * From contains characters other than 0-9, *, #, or +.
- *
- * In both cases, ast_callerid_parse() should have parsed the From header
- * as a name rather than a number. So we just need to set the location
- * to what was parsed as a name, and set the name NULL since there was
- * no name present.
- */
- location = name;
- name = NULL;
- }
- ast_string_field_set(pvt, fromname, name);
- if (strchr(location, ':')) { /* Must be a URI */
- parse_uri(location, "sip:,sips:", &user, NULL, &domain, NULL);
- SIP_PEDANTIC_DECODE(user);
- SIP_PEDANTIC_DECODE(domain);
- extract_host_from_hostport(&domain);
- ast_string_field_set(pvt, fromuser, user);
- ast_string_field_set(pvt, fromdomain, domain);
- } else { /* Treat it as an exten/user */
- ast_string_field_set(pvt, fromuser, location);
- }
- } else { /* assume we just have the name, use defaults for the rest */
- ast_string_field_set(pvt, fromname, from);
- }
- }
-
- sip_pvt_lock(pvt);
-
- /* Look up the host to contact */
- if (create_addr(pvt, to_host, NULL, TRUE)) {
- sip_pvt_unlock(pvt);
- dialog_unlink_all(pvt);
- dialog_unref(pvt, "create_addr failed sending a MESSAGE");
- return -1;
- }
-
- if (!ast_strlen_zero(to_user)) {
- ast_string_field_set(pvt, username, to_user);
- }
- ast_sip_ouraddrfor(&pvt->sa, &pvt->ourip, pvt);
- build_via(pvt);
- ast_set_flag(&pvt->flags[0], SIP_OUTGOING);
-
- /* XXX Does pvt->expiry need to be set? */
-
- /* Save additional MESSAGE headers in case of authentication request. */
- for (iter = ast_msg_var_iterator_init(msg);
- ast_msg_var_iterator_next(msg, iter, &var, &val);
- ast_msg_var_unref_current(iter)) {
- if (!strcasecmp(var, "Max-Forwards")) {
- /* Decrement Max-Forwards for SIP loop prevention. */
- if (sscanf(val, "%30d", &pvt->maxforwards) != 1 || pvt->maxforwards < 1) {
- ast_msg_var_iterator_destroy(iter);
- sip_pvt_unlock(pvt);
- dialog_unlink_all(pvt);
- dialog_unref(pvt, "MESSAGE(Max-Forwards) reached zero.");
- ast_log(LOG_NOTICE,
- "MESSAGE(Max-Forwards) reached zero. MESSAGE not sent.\n");
- return -1;
- }
- --pvt->maxforwards;
- continue;
- }
- if (block_msg_header(var)) {
- /* Block addition of this header. */
- continue;
- }
- add_msg_header(pvt, var, val);
- }
- ast_msg_var_iterator_destroy(iter);
-
- ast_string_field_set(pvt, msg_body, ast_msg_get_body(msg));
- res = transmit_message(pvt, 1, 0);
-
- sip_pvt_unlock(pvt);
- sip_scheddestroy(pvt, DEFAULT_TRANS_TIMEOUT);
- dialog_unref(pvt, "sent a MESSAGE");
-
- return res;
-}
-
-static enum sip_publish_type determine_sip_publish_type(struct sip_request *req, const char * const event, const char * const etag, const char * const expires, int *expires_int)
-{
- int etag_present = !ast_strlen_zero(etag);
- int body_present = req->lines > 0;
-
- ast_assert(expires_int != NULL);
-
- if (ast_strlen_zero(expires)) {
- /* Section 6, item 4, second bullet point of RFC 3903 says to
- * use a locally-configured default expiration if none is provided
- * in the request
- */
- *expires_int = DEFAULT_PUBLISH_EXPIRES;
- } else if (sscanf(expires, "%30d", expires_int) != 1) {
- return SIP_PUBLISH_UNKNOWN;
- }
-
- if (*expires_int == 0) {
- return SIP_PUBLISH_REMOVE;
- } else if (!etag_present && body_present) {
- return SIP_PUBLISH_INITIAL;
- } else if (etag_present && !body_present) {
- return SIP_PUBLISH_REFRESH;
- } else if (etag_present && body_present) {
- return SIP_PUBLISH_MODIFY;
- }
-
- return SIP_PUBLISH_UNKNOWN;
-}
-
-#ifdef HAVE_LIBXML2
-static int pidf_validate_tuple(struct ast_xml_node *tuple_node)
-{
- const char *id;
- int status_found = FALSE;
- struct ast_xml_node *tuple_children;
- struct ast_xml_node *tuple_children_iterator;
- /* Tuples have to have an id attribute or they're invalid */
- if (!(id = ast_xml_get_attribute(tuple_node, "id"))) {
- ast_log(LOG_WARNING, "Tuple XML element has no attribute 'id'\n");
- return FALSE;
- }
- /* We don't care what it actually is, just that it's there */
- ast_xml_free_attr(id);
- /* This is a tuple. It must have a status element */
- if (!(tuple_children = ast_xml_node_get_children(tuple_node))) {
- /* The tuple has no children. It sucks */
- ast_log(LOG_WARNING, "Tuple XML element has no child elements\n");
- return FALSE;
- }
- for (tuple_children_iterator = tuple_children; tuple_children_iterator;
- tuple_children_iterator = ast_xml_node_get_next(tuple_children_iterator)) {
- /* Similar to the wording used regarding tuples, the status element should appear
- * first. However, we will once again relax things and accept the status at any
- * position. We will enforce that only a single status element can be present.
- */
- if (strcmp(ast_xml_node_get_name(tuple_children_iterator), "status")) {
- /* Not the status, we don't care */
- continue;
- }
- if (status_found == TRUE) {
- /* THERE CAN BE ONLY ONE!!! */
- ast_log(LOG_WARNING, "Multiple status elements found in tuple. Only one allowed\n");
- return FALSE;
- }
- status_found = TRUE;
- }
- return status_found;
-}
-
-
-static int pidf_validate_presence(struct ast_xml_doc *doc)
-{
- struct ast_xml_node *presence_node = ast_xml_get_root(doc);
- struct ast_xml_node *child_nodes;
- struct ast_xml_node *node_iterator;
- struct ast_xml_ns *ns;
- const char *entity;
- const char *namespace;
- const char presence_namespace[] = "urn:ietf:params:xml:ns:pidf";
-
- if (!presence_node) {
- ast_log(LOG_WARNING, "Unable to retrieve root node of the XML document\n");
- return FALSE;
- }
- /* Okay, we managed to open the document! YAY! Now, let's start making sure it's all PIDF-ified
- * correctly.
- */
- if (strcmp(ast_xml_node_get_name(presence_node), "presence")) {
- ast_log(LOG_WARNING, "Root node of PIDF document is not 'presence'. Invalid\n");
- return FALSE;
- }
-
- /* The presence element must have an entity attribute and an xmlns attribute. Furthermore
- * the xmlns attribute must be "urn:ietf:params:xml:ns:pidf"
- */
- if (!(entity = ast_xml_get_attribute(presence_node, "entity"))) {
- ast_log(LOG_WARNING, "Presence element of PIDF document has no 'entity' attribute\n");
- return FALSE;
- }
- /* We're not interested in what the entity is, just that it exists */
- ast_xml_free_attr(entity);
-
- if (!(ns = ast_xml_find_namespace(doc, presence_node, NULL))) {
- ast_log(LOG_WARNING, "Couldn't find default namespace...\n");
- return FALSE;
- }
-
- namespace = ast_xml_get_ns_href(ns);
- if (ast_strlen_zero(namespace) || strcmp(namespace, presence_namespace)) {
- ast_log(LOG_WARNING, "PIDF document has invalid namespace value %s\n", namespace);
- return FALSE;
- }
-
- if (!(child_nodes = ast_xml_node_get_children(presence_node))) {
- ast_log(LOG_WARNING, "PIDF document has no elements as children of 'presence'. Invalid\n");
- return FALSE;
- }
-
- /* Check for tuple elements. RFC 3863 says that PIDF documents can have any number of
- * tuples, including 0. The big thing here is that if there are tuple elements present,
- * they have to have a single status element within.
- *
- * The RFC is worded such that tuples should appear as the first elements as children of
- * the presence element. However, we'll be accepting of documents which may place other elements
- * before the tuple(s).
- */
- for (node_iterator = child_nodes; node_iterator;
- node_iterator = ast_xml_node_get_next(node_iterator)) {
- if (strcmp(ast_xml_node_get_name(node_iterator), "tuple")) {
- /* Not a tuple. We don't give a rat's hind quarters */
- continue;
- }
- if (pidf_validate_tuple(node_iterator) == FALSE) {
- ast_log(LOG_WARNING, "Unable to validate tuple\n");
- return FALSE;
- }
- }
-
- return TRUE;
-}
-
-/*!
- * \brief Makes sure that body is properly formatted PIDF
- *
- * Specifically, we check that the document has a "presence" element
- * at the root and that within that, there is at least one "tuple" element
- * that contains a "status" element.
- *
- * XXX This function currently assumes a default namespace is used. Of course
- * if you're not using a default namespace, you're probably a stupid jerk anyway.
- *
- * \param req The SIP request to check
- * \param[out] pidf_doc The validated PIDF doc.
- * \retval FALSE The XML was malformed or the basic PIDF structure was marred
- * \retval TRUE The PIDF document is of a valid format
- */
-static int sip_pidf_validate(struct sip_request *req, struct ast_xml_doc **pidf_doc)
-{
- struct ast_xml_doc *doc;
- const char *content_type = sip_get_header(req, "Content-Type");
- char *pidf_body;
- int res;
-
- if (ast_strlen_zero(content_type) || strcmp(content_type, "application/pidf+xml")) {
- ast_log(LOG_WARNING, "Content type is not PIDF\n");
- return FALSE;
- }
-
- if (!(pidf_body = get_content(req))) {
- ast_log(LOG_WARNING, "Unable to get PIDF body\n");
- return FALSE;
- }
-
- if (!(doc = ast_xml_read_memory(pidf_body, strlen(pidf_body)))) {
- ast_log(LOG_WARNING, "Unable to open XML PIDF document. Is it malformed?\n");
- return FALSE;
- }
-
- res = pidf_validate_presence(doc);
- if (res == TRUE) {
- *pidf_doc = doc;
- } else {
- ast_xml_close(doc);
- }
- return res;
-}
-
-static int cc_esc_publish_handler(struct sip_pvt *pvt, struct sip_request *req, struct event_state_compositor *esc, struct sip_esc_entry *esc_entry)
-{
- const char *uri = REQ_OFFSET_TO_STR(req, rlpart2);
- struct ast_cc_agent *agent;
- struct sip_cc_agent_pvt *agent_pvt;
- struct ast_xml_doc *pidf_doc = NULL;
- const char *basic_status = NULL;
- struct ast_xml_node *presence_node;
- struct ast_xml_node *presence_children;
- struct ast_xml_node *tuple_node;
- struct ast_xml_node *tuple_children;
- struct ast_xml_node *status_node;
- struct ast_xml_node *status_children;
- struct ast_xml_node *basic_node;
- int res = 0;
-
- if (!((agent = find_sip_cc_agent_by_notify_uri(uri)) || (agent = find_sip_cc_agent_by_subscribe_uri(uri)))) {
- ast_log(LOG_WARNING, "Could not find agent using uri '%s'\n", uri);
- transmit_response(pvt, "412 Conditional Request Failed", req);
- return -1;
- }
-
- agent_pvt = agent->private_data;
-
- if (sip_pidf_validate(req, &pidf_doc) == FALSE) {
- res = -1;
- goto cc_publish_cleanup;
- }
-
- /* It's important to note that the PIDF validation routine has no knowledge
- * of what we specifically want in this instance. A valid PIDF document could
- * have no tuples, or it could have tuples whose status element has no basic
- * element contained within. While not violating the PIDF spec, these are
- * insufficient for our needs in this situation
- */
- presence_node = ast_xml_get_root(pidf_doc);
- if (!(presence_children = ast_xml_node_get_children(presence_node))) {
- ast_log(LOG_WARNING, "No tuples within presence element.\n");
- res = -1;
- goto cc_publish_cleanup;
- }
-
- if (!(tuple_node = ast_xml_find_element(presence_children, "tuple", NULL, NULL))) {
- ast_log(LOG_NOTICE, "Couldn't find tuple node?\n");
- res = -1;
- goto cc_publish_cleanup;
- }
-
- /* We already made sure that the tuple has a status node when we validated the PIDF
- * document earlier. So there's no need to enclose this operation in an if statement.
- */
- tuple_children = ast_xml_node_get_children(tuple_node);
- /* coverity[null_returns: FALSE] */
- status_node = ast_xml_find_element(tuple_children, "status", NULL, NULL);
-
- if (!(status_children = ast_xml_node_get_children(status_node))) {
- ast_log(LOG_WARNING, "No basic elements within status element.\n");
- res = -1;
- goto cc_publish_cleanup;
- }
-
- if (!(basic_node = ast_xml_find_element(status_children, "basic", NULL, NULL))) {
- ast_log(LOG_WARNING, "Couldn't find basic node?\n");
- res = -1;
- goto cc_publish_cleanup;
- }
-
- basic_status = ast_xml_get_text(basic_node);
-
- if (ast_strlen_zero(basic_status)) {
- ast_log(LOG_NOTICE, "NOthing in basic node?\n");
- res = -1;
- goto cc_publish_cleanup;
- }
-
- if (!strcmp(basic_status, "open")) {
- agent_pvt->is_available = TRUE;
- ast_cc_agent_caller_available(agent->core_id, "Received PUBLISH stating SIP caller %s is available",
- agent->device_name);
- } else if (!strcmp(basic_status, "closed")) {
- agent_pvt->is_available = FALSE;
- ast_cc_agent_caller_busy(agent->core_id, "Received PUBLISH stating SIP caller %s is busy",
- agent->device_name);
- } else {
- ast_log(LOG_NOTICE, "Invalid content in basic element: %s\n", basic_status);
- }
-
-cc_publish_cleanup:
- if (basic_status) {
- ast_xml_free_text(basic_status);
- }
- if (pidf_doc) {
- ast_xml_close(pidf_doc);
- }
- ao2_ref(agent, -1);
- if (res) {
- transmit_response(pvt, "400 Bad Request", req);
- }
- return res;
-}
-
-#endif /* HAVE_LIBXML2 */
-
-static int handle_sip_publish_initial(struct sip_pvt *p, struct sip_request *req, struct event_state_compositor *esc, const int expires)
-{
- struct sip_esc_entry *esc_entry = create_esc_entry(esc, req, expires);
- int res = 0;
-
- if (!esc_entry) {
- transmit_response(p, "503 Internal Server Failure", req);
- return -1;
- }
-
- if (esc->callbacks->initial_handler) {
- res = esc->callbacks->initial_handler(p, req, esc, esc_entry);
- }
-
- if (!res) {
- transmit_response_with_sip_etag(p, "200 OK", req, esc_entry, 0);
- }
-
- ao2_ref(esc_entry, -1);
- return res;
-}
-
-static int handle_sip_publish_refresh(struct sip_pvt *p, struct sip_request *req, struct event_state_compositor *esc, const char * const etag, const int expires)
-{
- struct sip_esc_entry *esc_entry = get_esc_entry(etag, esc);
- int expires_ms = expires * 1000;
- int res = 0;
-
- if (!esc_entry) {
- transmit_response(p, "412 Conditional Request Failed", req);
- return -1;
- }
-
- AST_SCHED_REPLACE_UNREF(esc_entry->sched_id, sched, expires_ms, publish_expire, esc_entry,
- ao2_ref(_data, -1),
- ao2_ref(esc_entry, -1),
- ao2_ref(esc_entry, +1));
-
- if (esc->callbacks->refresh_handler) {
- res = esc->callbacks->refresh_handler(p, req, esc, esc_entry);
- }
-
- if (!res) {
- transmit_response_with_sip_etag(p, "200 OK", req, esc_entry, 1);
- }
-
- ao2_ref(esc_entry, -1);
- return res;
-}
-
-static int handle_sip_publish_modify(struct sip_pvt *p, struct sip_request *req, struct event_state_compositor *esc, const char * const etag, const int expires)
-{
- struct sip_esc_entry *esc_entry = get_esc_entry(etag, esc);
- int expires_ms = expires * 1000;
- int res = 0;
-
- if (!esc_entry) {
- transmit_response(p, "412 Conditional Request Failed", req);
- return -1;
- }
-
- AST_SCHED_REPLACE_UNREF(esc_entry->sched_id, sched, expires_ms, publish_expire, esc_entry,
- ao2_ref(_data, -1),
- ao2_ref(esc_entry, -1),
- ao2_ref(esc_entry, +1));
-
- if (esc->callbacks->modify_handler) {
- res = esc->callbacks->modify_handler(p, req, esc, esc_entry);
- }
-
- if (!res) {
- transmit_response_with_sip_etag(p, "200 OK", req, esc_entry, 1);
- }
-
- ao2_ref(esc_entry, -1);
- return res;
-}
-
-static int handle_sip_publish_remove(struct sip_pvt *p, struct sip_request *req, struct event_state_compositor *esc, const char * const etag)
-{
- struct sip_esc_entry *esc_entry = get_esc_entry(etag, esc);
- int res = 0;
-
- if (!esc_entry) {
- transmit_response(p, "412 Conditional Request Failed", req);
- return -1;
- }
-
- AST_SCHED_DEL(sched, esc_entry->sched_id);
- /* Scheduler's ref of the esc_entry */
- ao2_ref(esc_entry, -1);
-
- if (esc->callbacks->remove_handler) {
- res = esc->callbacks->remove_handler(p, req, esc, esc_entry);
- }
-
- if (!res) {
- transmit_response_with_sip_etag(p, "200 OK", req, esc_entry, 1);
- }
-
- /* Ref from finding the esc_entry earlier in function */
- ao2_unlink(esc->compositor, esc_entry);
- ao2_ref(esc_entry, -1);
- return res;
-}
-
-static int handle_request_publish(struct sip_pvt *p, struct sip_request *req, struct ast_sockaddr *addr, const uint32_t seqno, const char *uri)
-{
- const char *etag = sip_get_header(req, "SIP-If-Match");
- const char *event = sip_get_header(req, "Event");
- struct event_state_compositor *esc;
- enum sip_publish_type publish_type;
- const char *expires_str = sip_get_header(req, "Expires");
- int expires_int;
- int auth_result;
- int handler_result = -1;
-
- if (ast_strlen_zero(event)) {
- transmit_response(p, "489 Bad Event", req);
- pvt_set_needdestroy(p, "missing Event: header");
- return -1;
- }
-
- if (!(esc = get_esc(event))) {
- transmit_response(p, "489 Bad Event", req);
- pvt_set_needdestroy(p, "unknown event package in publish");
- return -1;
- }
-
- auth_result = check_user(p, req, SIP_PUBLISH, uri, XMIT_UNRELIABLE, addr);
- if (auth_result == AUTH_CHALLENGE_SENT) {
- p->lastinvite = seqno;
- return 0;
- } else if (auth_result < 0) {
- send_check_user_failure_response(p, req, auth_result, XMIT_UNRELIABLE);
- sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
- ast_string_field_set(p, theirtag, NULL);
- return 0;
- } else if (auth_result == AUTH_SUCCESSFUL && p->lastinvite) {
- /* We need to stop retransmitting the 401 */
- __sip_ack(p, p->lastinvite, 1, 0);
- }
-
- publish_type = determine_sip_publish_type(req, event, etag, expires_str, &expires_int);
-
- if (expires_int > max_expiry) {
- expires_int = max_expiry;
- } else if (expires_int < min_expiry && expires_int > 0) {
- transmit_response_with_minexpires(p, "423 Interval too small", req, min_expiry);
- pvt_set_needdestroy(p, "Expires is less that the min expires allowed.");
- return 0;
- }
- p->expiry = expires_int;
-
- /* It is the responsibility of these handlers to formulate any response
- * sent for a PUBLISH
- */
- switch (publish_type) {
- case SIP_PUBLISH_UNKNOWN:
- transmit_response(p, "400 Bad Request", req);
- break;
- case SIP_PUBLISH_INITIAL:
- handler_result = handle_sip_publish_initial(p, req, esc, expires_int);
- break;
- case SIP_PUBLISH_REFRESH:
- handler_result = handle_sip_publish_refresh(p, req, esc, etag, expires_int);
- break;
- case SIP_PUBLISH_MODIFY:
- handler_result = handle_sip_publish_modify(p, req, esc, etag, expires_int);
- break;
- case SIP_PUBLISH_REMOVE:
- handler_result = handle_sip_publish_remove(p, req, esc, etag);
- break;
- default:
- transmit_response(p, "400 Impossible Condition", req);
- break;
- }
- if (!handler_result && p->expiry > 0) {
- sip_scheddestroy(p, (p->expiry + 10) * 1000);
- } else {
- pvt_set_needdestroy(p, "forcing expiration");
- }
-
- return handler_result;
-}
-
-/*!
- * \internal
- * \brief Subscribe to MWI events for the specified peer
- *
- * \note The peer cannot be locked during this method. sip_send_mwi_peer will
- * attempt to lock the peer after the event subscription lock is held; if the peer is locked during
- * this method then we will attempt to lock the event subscription lock but after the peer, creating
- * a locking inversion.
- */
-static void add_peer_mwi_subs(struct sip_peer *peer)
-{
- struct sip_mailbox *mailbox;
-
- AST_LIST_TRAVERSE(&peer->mailboxes, mailbox, entry) {
- if (mailbox->status != SIP_MAILBOX_STATUS_NEW) {
- continue;
- }
- mailbox->event_sub = ast_mwi_subscribe_pool(mailbox->id, mwi_event_cb, peer);
- if (mailbox->event_sub) {
- stasis_subscription_accept_message_type(
- ast_mwi_subscriber_subscription(mailbox->event_sub),
- stasis_subscription_change_type());
- }
- }
-}
-
-static int handle_cc_subscribe(struct sip_pvt *p, struct sip_request *req)
-{
- const char *uri = REQ_OFFSET_TO_STR(req, rlpart2);
- char *param_separator;
- struct ast_cc_agent *agent;
- struct sip_cc_agent_pvt *agent_pvt;
- const char *expires_str = sip_get_header(req, "Expires");
- int expires = -1; /* Just need it to be non-zero */
-
- if (!ast_strlen_zero(expires_str)) {
- sscanf(expires_str, "%30d", &expires);
- }
-
- if ((param_separator = strchr(uri, ';'))) {
- *param_separator = '\0';
- }
-
- p->subscribed = CALL_COMPLETION;
-
- if (!(agent = find_sip_cc_agent_by_subscribe_uri(uri))) {
- if (!expires) {
- /* Typically, if a 0 Expires reaches us and we can't find
- * the corresponding agent, it means that the CC transaction
- * has completed and so the calling side is just trying to
- * clean up its subscription. We'll just respond with a
- * 200 OK and be done with it
- */
- transmit_response(p, "200 OK", req);
- return 0;
- }
- ast_log(LOG_WARNING, "Invalid URI '%s' in CC subscribe\n", uri);
- transmit_response(p, "404 Not Found", req);
- return -1;
- }
-
- agent_pvt = agent->private_data;
-
- if (!expires) {
- /* We got sent a SUBSCRIBE and found an agent. This means that CC
- * is being canceled.
- */
- ast_cc_failed(agent->core_id, "CC is being canceled by %s", agent->device_name);
- transmit_response(p, "200 OK", req);
- ao2_ref(agent, -1);
- return 0;
- }
-
- agent_pvt->subscribe_pvt = dialog_ref(p, "SIP CC agent gains reference to subscription dialog");
- ast_cc_agent_accept_request(agent->core_id, "SIP caller %s has requested CC via SUBSCRIBE",
- agent->device_name);
-
- /* We don't send a response here. That is done in the agent's ack callback or in the
- * agent destructor, should a failure occur before we have responded
- */
- ao2_ref(agent, -1);
- return 0;
-}
-
-/*! \brief Handle incoming SUBSCRIBE request */
-static int handle_request_subscribe(struct sip_pvt *p, struct sip_request *req, struct ast_sockaddr *addr, uint32_t seqno, const char *e)
-{
- int res = 0;
- struct sip_peer *authpeer = NULL;
- char *event = ast_strdupa(sip_get_header(req, "Event")); /* Get Event package name */
- int resubscribe = (p->subscribed != NONE) && !req->ignore;
- char *options;
-
- if (p->initreq.headers) {
- /* We already have a dialog */
- if (p->initreq.method != SIP_SUBSCRIBE) {
- /* This is a SUBSCRIBE within another SIP dialog, which we do not support */
- /* For transfers, this could happen, but since we haven't seen it happening, let us just refuse this */
- transmit_response(p, "403 Forbidden (within dialog)", req);
- /* Do not destroy session, since we will break the call if we do */
- ast_debug(1, "Got a subscription within the context of another call, can't handle that - %s (Method %s)\n", p->callid, sip_methods[p->initreq.method].text);
- return 0;
- } else if (req->debug) {
- if (resubscribe)
- ast_debug(1, "Got a re-subscribe on existing subscription %s\n", p->callid);
- else
- ast_debug(1, "Got a new subscription %s (possibly with auth) or retransmission\n", p->callid);
- }
- }
-
- /* Check if we have a global disallow setting on subscriptions.
- if so, we don't have to check peer settings after auth, which saves a lot of processing
- */
- if (!sip_cfg.allowsubscribe) {
- transmit_response(p, "403 Forbidden (policy)", req);
- pvt_set_needdestroy(p, "forbidden");
- return 0;
- }
-
- if (!req->ignore && !resubscribe) { /* Set up dialog, new subscription */
- const char *to = sip_get_header(req, "To");
- char totag[128];
- set_pvt_allowed_methods(p, req);
-
- /* Check to see if a tag was provided, if so this is actually a resubscription of a dialog we no longer know about */
- if (!ast_strlen_zero(to) && gettag(req, "To", totag, sizeof(totag))) {
- if (req->debug)
- ast_verbose("Received resubscription for a dialog we no longer know about. Telling remote side to subscribe again.\n");
- transmit_response(p, "481 Subscription does not exist", req);
- pvt_set_needdestroy(p, "subscription does not exist");
- return 0;
- }
-
- /* Use this as the basis */
- if (req->debug)
- ast_verbose("Creating new subscription\n");
-
- copy_request(&p->initreq, req);
- if (sipdebug)
- ast_debug(4, "Initializing initreq for method %s - callid %s\n", sip_methods[req->method].text, p->callid);
- check_via(p, req);
- build_route(p, req, 0, 0);
- } else if (req->debug && req->ignore)
- ast_verbose("Ignoring this SUBSCRIBE request\n");
-
- /* Find parameters to Event: header value and remove them for now */
- if (ast_strlen_zero(event)) {
- transmit_response(p, "489 Bad Event", req);
- ast_debug(2, "Received SIP subscribe for unknown event package: \n");
- pvt_set_needdestroy(p, "unknown event package in subscribe");
- return 0;
- }
- if ((options = strchr(event, ';')) != NULL) {
- *options++ = '\0';
- }
-
- /* Handle authentication if we're new and not a retransmission. We can't just
- * use if !req->ignore, because then we'll end up sending
- * a 200 OK if someone retransmits without sending auth */
- if (p->subscribed == NONE || resubscribe) {
- res = check_user_full(p, req, SIP_SUBSCRIBE, e, XMIT_UNRELIABLE, addr, &authpeer);
-
- /* if an authentication response was sent, we are done here */
- if (res == AUTH_CHALLENGE_SENT) /* authpeer = NULL here */
- return 0;
- if (res != AUTH_SUCCESSFUL) {
- send_check_user_failure_response(p, req, res, XMIT_UNRELIABLE);
- pvt_set_needdestroy(p, "authentication failed");
- return 0;
- }
- }
-
- /* At this point, we hold a reference to authpeer (if not NULL). It
- * must be released when done.
- */
-
- /* Check if this device is allowed to subscribe at all */
- if (!ast_test_flag(&p->flags[1], SIP_PAGE2_ALLOWSUBSCRIBE)) {
- transmit_response(p, "403 Forbidden (policy)", req);
- pvt_set_needdestroy(p, "subscription not allowed");
- if (authpeer) {
- sip_unref_peer(authpeer, "sip_unref_peer, from handle_request_subscribe (authpeer 1)");
- }
- return 0;
- }
-
- /* Get full contact header - this needs to be used as a request URI in NOTIFY's */
- parse_ok_contact(p, req);
- build_contact(p, req, 1);
-
- /* Initialize tag for new subscriptions */
- if (ast_strlen_zero(p->tag)) {
- make_our_tag(p);
- }
-
- if (!strcmp(event, "presence") || !strcmp(event, "dialog")) { /* Presence, RFC 3842 */
- int gotdest;
- const char *accept;
- int start = 0;
- enum subscriptiontype subscribed = NONE;
- const char *unknown_accept = NULL;
-
- /* Get destination right away */
- gotdest = get_destination(p, NULL, NULL);
- if (gotdest != SIP_GET_DEST_EXTEN_FOUND) {
- if (gotdest == SIP_GET_DEST_INVALID_URI) {
- transmit_response(p, "416 Unsupported URI scheme", req);
- } else {
- transmit_response(p, "404 Not Found", req);
- }
- pvt_set_needdestroy(p, "subscription target not found");
- if (authpeer) {
- sip_unref_peer(authpeer, "sip_unref_peer, from handle_request_subscribe (authpeer 2)");
- }
- return 0;
- }
-
- /* Header from Xten Eye-beam Accept: multipart/related, application/rlmi+xml, application/pidf+xml, application/xpidf+xml */
- accept = __get_header(req, "Accept", &start);
- while ((subscribed == NONE) && !ast_strlen_zero(accept)) {
- if (strstr(accept, "application/pidf+xml")) {
- if (strstr(p->useragent, "Polycom")) {
- subscribed = XPIDF_XML; /* Older versions of Polycom firmware will claim pidf+xml, but really they only support xpidf+xml */
- } else {
- subscribed = PIDF_XML; /* RFC 3863 format */
- }
- } else if (strstr(accept, "application/dialog-info+xml")) {
- subscribed = DIALOG_INFO_XML;
- /* IETF draft: draft-ietf-sipping-dialog-package-05.txt */
- } else if (strstr(accept, "application/cpim-pidf+xml")) {
- subscribed = CPIM_PIDF_XML; /* RFC 3863 format */
- } else if (strstr(accept, "application/xpidf+xml")) {
- subscribed = XPIDF_XML; /* Early pre-RFC 3863 format with MSN additions (Microsoft Messenger) */
- } else {
- unknown_accept = accept;
- }
- /* check to see if there is another Accept header present */
- accept = __get_header(req, "Accept", &start);
- }
-
- if (!start) {
- if (p->subscribed == NONE) { /* if the subscribed field is not already set, and there is no accept header... */
- transmit_response(p, "489 Bad Event", req);
- ast_log(LOG_WARNING,"SUBSCRIBE failure: no Accept header: pvt: "
- "stateid: %d, laststate: %d, dialogver: %u, subscribecont: "
- "'%s', subscribeuri: '%s'\n",
- p->stateid,
- p->laststate,
- p->dialogver,
- p->subscribecontext,
- p->subscribeuri);
- pvt_set_needdestroy(p, "no Accept header");
- if (authpeer) {
- sip_unref_peer(authpeer, "sip_unref_peer, from handle_request_subscribe (authpeer 2)");
- }
- return 0;
- }
- /* if p->subscribed is non-zero, then accept is not obligatory; according to rfc 3265 section 3.1.3, at least.
- so, we'll just let it ride, keeping the value from a previous subscription, and not abort the subscription */
- } else if (subscribed == NONE) {
- /* Can't find a format for events that we know about */
- char buf[200];
-
- if (!ast_strlen_zero(unknown_accept)) {
- snprintf(buf, sizeof(buf), "489 Bad Event (format %s)", unknown_accept);
- } else {
- snprintf(buf, sizeof(buf), "489 Bad Event");
- }
- transmit_response(p, buf, req);
- ast_log(LOG_WARNING,"SUBSCRIBE failure: unrecognized format:"
- "'%s' pvt: subscribed: %d, stateid: %d, laststate: %d,"
- "dialogver: %u, subscribecont: '%s', subscribeuri: '%s'\n",
- unknown_accept,
- (int)p->subscribed,
- p->stateid,
- p->laststate,
- p->dialogver,
- p->subscribecontext,
- p->subscribeuri);
- pvt_set_needdestroy(p, "unrecognized format");
- if (authpeer) {
- sip_unref_peer(authpeer, "sip_unref_peer, from handle_request_subscribe (authpeer 2)");
- }
- return 0;
- } else {
- p->subscribed = subscribed;
- }
- } else if (!strcmp(event, "message-summary")) {
- int start = 0;
- int found_supported = 0;
- const char *accept;
-
- accept = __get_header(req, "Accept", &start);
- while (!found_supported && !ast_strlen_zero(accept)) {
- found_supported = strcmp(accept, "application/simple-message-summary") ? 0 : 1;
- if (!found_supported) {
- ast_debug(3, "Received SIP mailbox subscription for unknown format: %s\n", accept);
- }
- accept = __get_header(req, "Accept", &start);
- }
- /* If !start, there is no Accept header at all */
- if (start && !found_supported) {
- /* Format requested that we do not support */
- transmit_response(p, "406 Not Acceptable", req);
- ast_debug(2, "Received SIP mailbox subscription for unknown format\n");
- pvt_set_needdestroy(p, "unknown format");
- if (authpeer) {
- sip_unref_peer(authpeer, "sip_unref_peer, from handle_request_subscribe (authpeer 3)");
- }
- return 0;
- }
- /* Looks like they actually want a mailbox status
- This version of Asterisk supports mailbox subscriptions
- The subscribed URI needs to exist in the dial plan
- In most devices, this is configurable to the voicemailmain extension you use
- */
- if (!authpeer || AST_LIST_EMPTY(&authpeer->mailboxes)) {
- if (!authpeer) {
- transmit_response(p, "404 Not found", req);
- } else {
- transmit_response(p, "404 Not found (no mailbox)", req);
- ast_log(LOG_NOTICE, "Received SIP subscribe for peer without mailbox: %s\n", S_OR(authpeer->name, ""));
- }
- pvt_set_needdestroy(p, "received 404 response");
-
- if (authpeer) {
- sip_unref_peer(authpeer, "sip_unref_peer, from handle_request_subscribe (authpeer 3)");
- }
- return 0;
- }
-
- p->subscribed = MWI_NOTIFICATION;
- if (ast_test_flag(&authpeer->flags[1], SIP_PAGE2_SUBSCRIBEMWIONLY)) {
- ao2_unlock(p);
- add_peer_mwi_subs(authpeer);
- ao2_lock(p);
- }
- if (authpeer->mwipvt != p) { /* Destroy old PVT if this is a new one */
- /* We only allow one subscription per peer */
- if (authpeer->mwipvt) {
- dialog_unlink_all(authpeer->mwipvt);
- authpeer->mwipvt = dialog_unref(authpeer->mwipvt, "unref dialog authpeer->mwipvt");
- }
- authpeer->mwipvt = dialog_ref(p, "setting peers' mwipvt to p");
- }
-
- if (p->relatedpeer != authpeer) {
- if (p->relatedpeer) {
- sip_unref_peer(p->relatedpeer, "Unref previously stored relatedpeer ptr");
- }
- p->relatedpeer = sip_ref_peer(authpeer, "setting dialog's relatedpeer pointer");
- }
- /* Do not release authpeer here */
- } else if (!strcmp(event, "call-completion")) {
- handle_cc_subscribe(p, req);
- } else { /* At this point, Asterisk does not understand the specified event */
- transmit_response(p, "489 Bad Event", req);
- ast_debug(2, "Received SIP subscribe for unknown event package: %s\n", event);
- pvt_set_needdestroy(p, "unknown event package");
- if (authpeer) {
- sip_unref_peer(authpeer, "sip_unref_peer, from handle_request_subscribe (authpeer 5)");
- }
- return 0;
- }
-
- if (!req->ignore) {
- p->lastinvite = seqno;
- }
- if (!p->needdestroy) {
- const char *expires_str = sip_get_header(req, "Expires");
-
- if (ast_strlen_zero(expires_str)) {
- p->expiry = default_expiry;
- } else {
- p->expiry = atoi(expires_str);
- }
-
- /* check if the requested expiry-time is within the approved limits from sip.conf */
- if (p->expiry > max_subexpiry) {
- p->expiry = max_subexpiry;
- } else if (p->expiry < min_subexpiry && p->expiry > 0) {
- transmit_response_with_minexpires(p, "423 Interval too small", req, min_subexpiry);
- ast_log(LOG_WARNING, "Received subscription for extension \"%s\" context \"%s\" "
- "with Expire header less than 'subminexpire' limit. Received \"Expire: %d\" min is %d\n",
- p->exten, p->context, p->expiry, min_subexpiry);
- pvt_set_needdestroy(p, "Expires is less that the min expires allowed.");
- if (authpeer) {
- sip_unref_peer(authpeer, "sip_unref_peer, from handle_request_subscribe (authpeer 6)");
- }
- return 0;
- }
-
- if (sipdebug) {
- const char *action = p->expiry > 0 ? "Adding" : "Removing";
- if (p->subscribed == MWI_NOTIFICATION && p->relatedpeer) {
- ast_debug(2, "%s subscription for mailbox notification - peer %s\n",
- action, p->relatedpeer->name);
- } else if (p->subscribed == CALL_COMPLETION) {
- ast_debug(2, "%s CC subscription for peer %s\n", action, p->username);
- } else {
- ast_debug(2, "%s subscription for extension %s context %s for peer %s\n",
- action, p->exten, p->context, p->username);
- }
- }
-
- /* Remove subscription expiry for renewals */
- sip_cancel_destroy(p);
- if (p->expiry > 0) {
- /* Set timer for destruction of call at expiration */
- sip_scheddestroy(p, (p->expiry + 10) * 1000);
- }
-
- if (p->subscribed == MWI_NOTIFICATION) {
- ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
- transmit_response(p, "200 OK", req);
- if (p->relatedpeer) { /* Send first notification */
- struct sip_peer *peer = p->relatedpeer;
- sip_ref_peer(peer, "ensure a peer ref is held during MWI sending");
- ao2_unlock(p);
- sip_send_mwi_to_peer(peer, 0);
- ao2_lock(p);
- sip_unref_peer(peer, "release a peer ref now that MWI is sent");
- }
- } else if (p->subscribed != CALL_COMPLETION) {
- struct state_notify_data data = { 0, };
- char *subtype = NULL;
- char *message = NULL;
- struct ao2_container *device_state_info = NULL;
-
- if (p->expiry > 0 && !resubscribe) {
- /* Add subscription for extension state from the PBX core */
- if (p->stateid != -1) {
- ast_extension_state_del(p->stateid, cb_extensionstate);
- }
- dialog_ref(p, "copying dialog ptr into extension state struct");
- p->stateid = ast_extension_state_add_destroy_extended(p->context, p->exten, cb_extensionstate, cb_extensionstate_destroy, p);
- if (p->stateid == -1) {
- dialog_unref(p, "copying dialog ptr into extension state struct failed");
- }
- }
-
- sip_pvt_unlock(p);
- data.state = ast_extension_state_extended(NULL, p->context, p->exten, &device_state_info);
- sip_pvt_lock(p);
-
- if (data.state < 0) {
- ao2_cleanup(device_state_info);
- if (p->expiry > 0) {
- ast_log(LOG_NOTICE, "Got SUBSCRIBE for extension %s@%s from %s, but there is no hint for that extension.\n", p->exten, p->context, ast_sockaddr_stringify(&p->sa));
- }
- transmit_response(p, "404 Not found", req);
- pvt_set_needdestroy(p, "no extension for SUBSCRIBE");
- if (authpeer) {
- sip_unref_peer(authpeer, "sip_unref_peer, from handle_request_subscribe (authpeer 6)");
- }
- return 0;
- }
- if (allow_notify_user_presence(p)) {
- data.presence_state = ast_hint_presence_state(NULL, p->context, p->exten, &subtype, &message);
- data.presence_subtype = subtype;
- data.presence_message = message;
- }
- ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
- transmit_response(p, "200 OK", req);
- /* RFC 3265: A notification must be sent on every subscribe, so force it */
- data.device_state_info = device_state_info;
- if (data.state & AST_EXTENSION_RINGING) {
- /* save last_ringing_channel_time if this state really contains a ringing channel
- * because extensionstate_update() doesn't do it if forced
- */
- struct ast_channel *ringing = find_ringing_channel(data.device_state_info, p);
- if (ringing) {
- p->last_ringing_channel_time = ast_channel_creationtime(ringing);
- ao2_ref(ringing, -1);
- }
- /* If there is no channel, this likely indicates that the ringing indication
- * is due to a custom device state. These do not have associated channels.
- */
- }
- extensionstate_update(p->context, p->exten, &data, p, TRUE);
- append_history(p, "Subscribestatus", "%s", ast_extension_state2str(data.state));
- /* hide the 'complete' exten/context in the refer_to field for later display */
- ast_string_field_build(p, subscribeuri, "%s@%s", p->exten, p->context);
- /* Deleted the slow iteration of all sip dialogs to find old subscribes from this peer for exten@context */
-
- ao2_cleanup(device_state_info);
- ast_free(subtype);
- ast_free(message);
- }
- if (!p->expiry) {
- pvt_set_needdestroy(p, "forcing expiration");
- }
- }
-
- if (authpeer) {
- sip_unref_peer(authpeer, "unref pointer into (*authpeer)");
- }
- return 1;
-}
-
-/*! \brief Handle incoming REGISTER request */
-static int handle_request_register(struct sip_pvt *p, struct sip_request *req, struct ast_sockaddr *addr, const char *e)
-{
- enum check_auth_result res;
-
- /* If this is not the intial request, and the initial request isn't
- * a register, something screwy happened, so bail */
- if (p->initreq.headers && p->initreq.method != SIP_REGISTER) {
- ast_log(LOG_WARNING, "Ignoring spurious REGISTER with Call-ID: %s\n", p->callid);
- return -1;
- }
-
- /* Use this as the basis */
- copy_request(&p->initreq, req);
- if (sipdebug)
- ast_debug(4, "Initializing initreq for method %s - callid %s\n", sip_methods[req->method].text, p->callid);
- check_via(p, req);
-
- if ((res = register_verify(p, addr, req, e)) < 0) {
- const char *reason;
-
- switch (res) {
- case AUTH_SECRET_FAILED:
- reason = "Wrong password";
- break;
- case AUTH_USERNAME_MISMATCH:
- reason = "Username/auth name mismatch";
- break;
- case AUTH_NOT_FOUND:
- reason = "No matching peer found";
- break;
- case AUTH_UNKNOWN_DOMAIN:
- reason = "Not a local domain";
- break;
- case AUTH_PEER_NOT_DYNAMIC:
- reason = "Peer is not supposed to register";
- break;
- case AUTH_ACL_FAILED:
- reason = "Device does not match ACL";
- break;
- case AUTH_BAD_TRANSPORT:
- reason = "Device not configured to use this transport type";
- break;
- case AUTH_RTP_FAILED:
- reason = "RTP initialization failed";
- break;
- default:
- reason = "Unknown failure";
- break;
- }
- ast_log(LOG_NOTICE, "Registration from '%s' failed for '%s' - %s\n",
- sip_get_header(req, "To"), ast_sockaddr_stringify(addr),
- reason);
- append_history(p, "RegRequest", "Failed : Account %s : %s", sip_get_header(req, "To"), reason);
- } else {
- req->authenticated = 1;
- append_history(p, "RegRequest", "Succeeded : Account %s", sip_get_header(req, "To"));
- }
-
- if (res != AUTH_CHALLENGE_SENT) {
- /* Destroy the session, but keep us around for just a bit in case they don't
- get our 200 OK */
- sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
- }
-
- return res;
-}
-
-/*!
- * \brief Handle incoming SIP requests (methods)
- * \note
- * This is where all incoming requests go first.
- * \note
- * called with p and p->owner locked
- */
-static int handle_incoming(struct sip_pvt *p, struct sip_request *req, struct ast_sockaddr *addr, int *recount, int *nounlock)
-{
- /* Called with p->lock held, as well as p->owner->lock if appropriate, keeping things
- relatively static */
- const char *cmd;
- const char *cseq;
- const char *useragent;
- const char *via;
- const char *callid;
- int via_pos = 0;
- uint32_t seqno;
- int len;
- int respid;
- int res = 0;
- const char *e;
- int error = 0;
- int oldmethod = p->method;
- int acked = 0;
-
- /* RFC 3261 - 8.1.1 A valid SIP request must contain To, From, CSeq, Call-ID and Via.
- * 8.2.6.2 Response must have To, From, Call-ID CSeq, and Via related to the request,
- * so we can check to make sure these fields exist for all requests and responses */
- cseq = sip_get_header(req, "Cseq");
- cmd = REQ_OFFSET_TO_STR(req, header[0]);
- /* Save the via_pos so we can check later that responses only have 1 Via header */
- via = __get_header(req, "Via", &via_pos);
- /* This must exist already because we've called find_call by now */
- callid = sip_get_header(req, "Call-ID");
-
- /* Must have Cseq */
- if (ast_strlen_zero(cmd) || ast_strlen_zero(cseq) || ast_strlen_zero(via)) {
- ast_log(LOG_ERROR, "Dropping this SIP message with Call-ID '%s', it's incomplete.\n", callid);
- error = 1;
- }
- if (!error && sscanf(cseq, "%30u%n", &seqno, &len) != 1) {
- ast_log(LOG_ERROR, "No seqno in '%s'. Dropping incomplete message.\n", cmd);
- error = 1;
- }
- if (error) {
- if (!p->initreq.headers) { /* New call */
- pvt_set_needdestroy(p, "no headers");
- }
- return -1;
- }
- /* Get the command XXX */
-
- cmd = REQ_OFFSET_TO_STR(req, rlpart1);
- e = ast_skip_blanks(REQ_OFFSET_TO_STR(req, rlpart2));
-
- /* Save useragent of the client */
- useragent = sip_get_header(req, "User-Agent");
- if (!ast_strlen_zero(useragent))
- ast_string_field_set(p, useragent, useragent);
-
- /* Find out SIP method for incoming request */
- if (req->method == SIP_RESPONSE) { /* Response to our request */
- /* ignore means "don't do anything with it" but still have to
- * respond appropriately.
- * But in this case this is a response already, so we really
- * have nothing to do with this message, and even setting the
- * ignore flag is pointless.
- */
- if (ast_strlen_zero(e)) {
- return 0;
- }
- if (sscanf(e, "%30d %n", &respid, &len) != 1) {
- ast_log(LOG_WARNING, "Invalid response: '%s'\n", e);
- return 0;
- }
- if (respid <= 0) {
- ast_log(LOG_WARNING, "Invalid SIP response code: '%d'\n", respid);
- return 0;
- }
- /* RFC 3261 - 8.1.3.3 If more than one Via header field value is present in a reponse
- * the UAC SHOULD discard the message. This is not perfect, as it will not catch multiple
- * headers joined with a comma. Fixing that would pretty much involve writing a new parser */
- if (!ast_strlen_zero(__get_header(req, "via", &via_pos))) {
- ast_log(LOG_WARNING, "Misrouted SIP response '%s' with Call-ID '%s', too many vias\n", e, callid);
- return 0;
- }
- if (p->ocseq && (p->ocseq < seqno)) {
- ast_debug(1, "Ignoring out of order response %u (expecting %u)\n", seqno, p->ocseq);
- return -1;
- } else {
- if ((respid == 200) || ((respid >= 300) && (respid <= 399))) {
- extract_uri(p, req);
- }
-
- if (p->owner) {
- struct ast_control_pvt_cause_code *cause_code;
- int data_size = sizeof(*cause_code);
- /* size of the string making up the cause code is "SIP " + cause length */
- data_size += 4 + strlen(REQ_OFFSET_TO_STR(req, rlpart2));
- cause_code = ast_alloca(data_size);
- memset(cause_code, 0, data_size);
-
- ast_copy_string(cause_code->chan_name, ast_channel_name(p->owner), AST_CHANNEL_NAME);
-
- snprintf(cause_code->code, data_size - sizeof(*cause_code) + 1, "SIP %s", REQ_OFFSET_TO_STR(req, rlpart2));
-
- cause_code->ast_cause = hangup_sip2cause(respid);
- if (global_store_sip_cause) {
- cause_code->emulate_sip_cause = 1;
- }
-
- ast_queue_control_data(p->owner, AST_CONTROL_PVT_CAUSE_CODE, cause_code, data_size);
- ast_channel_hangupcause_hash_set(p->owner, cause_code, data_size);
- }
-
- handle_response(p, respid, e + len, req, seqno);
- }
- return 0;
- }
-
- /* New SIP request coming in
- (could be new request in existing SIP dialog as well...)
- */
- p->method = req->method; /* Find out which SIP method they are using */
- ast_debug(4, "**** Received %s (%u) - Command in SIP %s\n", sip_methods[p->method].text, sip_methods[p->method].id, cmd);
-
- if (p->icseq && (p->icseq > seqno) ) {
- if (p->pendinginvite && seqno == p->pendinginvite && (req->method == SIP_ACK || req->method == SIP_CANCEL)) {
- ast_debug(2, "Got CANCEL or ACK on INVITE with transactions in between.\n");
- } else {
- ast_debug(1, "Ignoring too old SIP packet packet %u (expecting >= %u)\n", seqno, p->icseq);
- if (req->method == SIP_INVITE) {
- unsigned int ran = (ast_random() % 10) + 1;
- char seconds[4];
- snprintf(seconds, sizeof(seconds), "%u", ran);
- transmit_response_with_retry_after(p, "500 Server error", req, seconds); /* respond according to RFC 3261 14.2 with Retry-After betwewn 0 and 10 */
- } else if (req->method != SIP_ACK) {
- transmit_response(p, "500 Server error", req); /* We must respond according to RFC 3261 sec 12.2 */
- }
- return -1;
- }
- } else if (p->icseq &&
- p->icseq == seqno &&
- req->method != SIP_ACK &&
- (p->method != SIP_CANCEL || p->alreadygone)) {
- /* ignore means "don't do anything with it" but still have to
- respond appropriately. We do this if we receive a repeat of
- the last sequence number */
- req->ignore = 1;
- ast_debug(3, "Ignoring SIP message because of retransmit (%s Seqno %u, ours %u)\n", sip_methods[p->method].text, p->icseq, seqno);
- }
-
- /* RFC 3261 section 9. "CANCEL has no effect on a request to which a UAS has
- * already given a final response." */
- if (!p->pendinginvite && (req->method == SIP_CANCEL)) {
- transmit_response(p, "481 Call/Transaction Does Not Exist", req);
- return res;
- }
-
- if (seqno >= p->icseq)
- /* Next should follow monotonically (but not necessarily
- incrementally -- thanks again to the genius authors of SIP --
- increasing */
- p->icseq = seqno;
-
- /* Find their tag if we haven't got it */
- if (ast_strlen_zero(p->theirtag)) {
- char tag[128];
-
- gettag(req, "From", tag, sizeof(tag));
- ast_string_field_set(p, theirtag, tag);
- }
- snprintf(p->lastmsg, sizeof(p->lastmsg), "Rx: %s", cmd);
-
- if (sip_cfg.pedanticsipchecking) {
- /* If this is a request packet without a from tag, it's not
- correct according to RFC 3261 */
- /* Check if this a new request in a new dialog with a totag already attached to it,
- RFC 3261 - section 12.2 - and we don't want to mess with recovery */
- if (!p->initreq.headers && req->has_to_tag) {
- /* If this is a first request and it got a to-tag, it is not for us */
- if (!req->ignore && req->method == SIP_INVITE) {
- /* Just because we think this is a dialog-starting INVITE with a to-tag
- * doesn't mean it actually is. It could be a reinvite for an established, but
- * unknown dialog. In such a case, we need to change our tag to the
- * incoming INVITE's to-tag so that they will recognize the 481 we send and
- * so that we will properly match their incoming ACK.
- */
- char totag[128];
- gettag(req, "To", totag, sizeof(totag));
- ast_string_field_set(p, tag, totag);
- p->pendinginvite = p->icseq;
- transmit_response_reliable(p, "481 Call/Transaction Does Not Exist", req);
- /* Will cease to exist after ACK */
- return res;
- } else if (req->method != SIP_ACK) {
- transmit_response(p, "481 Call/Transaction Does Not Exist", req);
- sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
- return res;
- }
- /* Otherwise, this is an ACK. It will always have a to-tag */
- }
- }
-
- if (!e && (p->method == SIP_INVITE || p->method == SIP_SUBSCRIBE || p->method == SIP_REGISTER || p->method == SIP_NOTIFY || p->method == SIP_PUBLISH)) {
- transmit_response(p, "400 Bad request", req);
- sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
- return -1;
- }
-
- /* Handle various incoming SIP methods in requests */
- switch (p->method) {
- case SIP_OPTIONS:
- res = handle_request_options(p, req, addr, e);
- break;
- case SIP_INVITE:
- res = handle_request_invite(p, req, addr, seqno, recount, e, nounlock);
-
- if (res < 9) {
- sip_report_security_event(NULL, &p->recv, p, req, res);
- }
-
- switch (res) {
- case INV_REQ_SUCCESS:
- res = 1;
- break;
- case INV_REQ_FAILED:
- res = 0;
- break;
- case INV_REQ_ERROR:
- res = -1;
- break;
- default:
- res = 0;
- break;
- }
-
- break;
- case SIP_REFER:
- res = handle_request_refer(p, req, seqno, nounlock);
- break;
- case SIP_CANCEL:
- res = handle_request_cancel(p, req);
- break;
- case SIP_BYE:
- res = handle_request_bye(p, req);
- break;
- case SIP_MESSAGE:
- res = handle_request_message(p, req, addr, e);
- break;
- case SIP_PUBLISH:
- res = handle_request_publish(p, req, addr, seqno, e);
- break;
- case SIP_SUBSCRIBE:
- res = handle_request_subscribe(p, req, addr, seqno, e);
- break;
- case SIP_REGISTER:
- res = handle_request_register(p, req, addr, e);
- sip_report_security_event(p->exten, NULL, p, req, res);
- break;
- case SIP_INFO:
- if (req->debug)
- ast_verbose("Receiving INFO!\n");
- if (!req->ignore)
- handle_request_info(p, req);
- else /* if ignoring, transmit response */
- transmit_response(p, "200 OK", req);
- break;
- case SIP_NOTIFY:
- res = handle_request_notify(p, req, addr, seqno, e);
- break;
- case SIP_UPDATE:
- res = handle_request_update(p, req);
- break;
- case SIP_ACK:
- /* Make sure we don't ignore this */
- if (seqno == p->pendinginvite) {
- p->invitestate = INV_TERMINATED;
- p->pendinginvite = 0;
- acked = __sip_ack(p, seqno, 1 /* response */, 0);
- if (p->owner && find_sdp(req)) {
- if (process_sdp(p, req, SDP_T38_NONE, FALSE)) {
- return -1;
- }
- if (ast_test_flag(&p->flags[0], SIP_DIRECT_MEDIA)) {
- ast_queue_control(p->owner, AST_CONTROL_UPDATE_RTP_PEER);
- }
- }
- sched_check_pendings(p);
- } else if (p->glareinvite == seqno) {
- /* handle ack for the 491 pending sent for glareinvite */
- p->glareinvite = 0;
- acked = __sip_ack(p, seqno, 1, 0);
- }
- if (!acked) {
- /* Got an ACK that did not match anything. Ignore
- * silently and restore previous method */
- p->method = oldmethod;
- }
- if (!p->lastinvite && ast_strlen_zero(p->nonce)) {
- pvt_set_needdestroy(p, "unmatched ACK");
- }
- break;
- default:
- transmit_response_with_allow(p, "501 Method Not Implemented", req, 0);
- ast_log(LOG_NOTICE, "Unknown SIP command '%s' from '%s'\n",
- cmd, ast_sockaddr_stringify(&p->sa));
- /* If this is some new method, and we don't have a call, destroy it now */
- if (!p->initreq.headers) {
- pvt_set_needdestroy(p, "unimplemented method");
- }
- break;
- }
- return res;
-}
-
-/*! \brief Read data from SIP UDP socket
-\note sipsock_read locks the owner channel while we are processing the SIP message
-\retval 1 on error.
-\retval 0 on success.
-\note Successful messages is connected to SIP call and forwarded to handle_incoming()
-*/
-static int sipsock_read(int *id, int fd, short events, void *ignore)
-{
- struct sip_request req;
- struct ast_sockaddr addr;
- int res;
- static char readbuf[65535];
-
- memset(&req, 0, sizeof(req));
- res = ast_recvfrom(fd, readbuf, sizeof(readbuf) - 1, 0, &addr);
- if (res < 0) {
-#if !defined(__FreeBSD__)
- if (errno == EAGAIN)
- ast_log(LOG_NOTICE, "SIP: Received packet with bad UDP checksum\n");
- else
-#endif
- if (errno != ECONNREFUSED)
- ast_log(LOG_WARNING, "Recv error: %s\n", strerror(errno));
- return 1;
- }
-
- readbuf[res] = '\0';
-
- if (!(req.data = ast_str_create(SIP_MIN_PACKET))) {
- return 1;
- }
-
- if (ast_str_set(&req.data, 0, "%s", readbuf) == AST_DYNSTR_BUILD_FAILED) {
- return -1;
- }
-
- req.socket.fd = sipsock;
- set_socket_transport(&req.socket, AST_TRANSPORT_UDP);
- req.socket.tcptls_session = NULL;
-
- handle_request_do(&req, &addr);
- deinit_req(&req);
-
- return 1;
-}
-
-/*! \brief Handle incoming SIP message - request or response
-
- This is used for all transports (udp, tcp and tcp/tls)
-*/
-static int handle_request_do(struct sip_request *req, struct ast_sockaddr *addr)
-{
- struct sip_pvt *p;
- struct ast_channel *owner_chan_ref = NULL;
- int recount = 0;
- int nounlock = 0;
-
- if (sip_debug_test_addr(addr)) /* Set the debug flag early on packet level */
- req->debug = 1;
- if (sip_cfg.pedanticsipchecking)
- lws2sws(req->data); /* Fix multiline headers */
- if (req->debug) {
- ast_verbose("\n<--- SIP read from %s:%s --->\n%s\n<------------->\n",
- sip_get_transport(req->socket.type), ast_sockaddr_stringify(addr), ast_str_buffer(req->data));
- }
-
- if (parse_request(req) == -1) { /* Bad packet, can't parse */
- ast_str_reset(req->data); /* nulling this out is NOT a good idea here. */
- return 1;
- }
- req->method = find_sip_method(REQ_OFFSET_TO_STR(req, rlpart1));
-
- if (req->debug)
- ast_verbose("--- (%d headers %d lines)%s ---\n", req->headers, req->lines, (req->headers + req->lines == 0) ? " Nat keepalive" : "");
-
- if (req->headers < 2) { /* Must have at least two headers */
- ast_str_reset(req->data); /* nulling this out is NOT a good idea here. */
- return 1;
- }
- ast_mutex_lock(&netlock);
-
- /* Find the active SIP dialog or create a new one */
- p = find_call(req, addr, req->method); /* returns p with a reference only. _NOT_ locked*/
- if (p == NULL) {
- ast_debug(1, "Invalid SIP message - rejected , no callid, len %zu\n", ast_str_strlen(req->data));
- ast_mutex_unlock(&netlock);
- return 1;
- }
-
- if (p->logger_callid) {
- ast_callid_threadassoc_add(p->logger_callid);
- }
-
- /* Lock both the pvt and the owner if owner is present. This will
- * not fail. */
- owner_chan_ref = sip_pvt_lock_full(p);
-
- copy_socket_data(&p->socket, &req->socket);
-
- ast_sockaddr_copy(&p->recv, addr);
-
- /* if we have an owner, then this request has been authenticated */
- if (p->owner) {
- req->authenticated = 1;
- }
-
- if (p->do_history) /* This is a request or response, note what it was for */
- append_history(p, "Rx", "%s / %s / %s", ast_str_buffer(req->data), sip_get_header(req, "CSeq"), REQ_OFFSET_TO_STR(req, rlpart2));
-
- if (handle_incoming(p, req, addr, &recount, &nounlock) == -1) {
- /* Request failed */
- ast_debug(1, "SIP message could not be handled, bad request: %-70.70s\n", p->callid[0] ? p->callid : "");
- }
-
- if (recount) {
- ast_update_use_count();
- }
-
- if (p->owner && !nounlock) {
- ast_channel_unlock(p->owner);
- }
- if (owner_chan_ref) {
- ast_channel_unref(owner_chan_ref);
- }
- sip_pvt_unlock(p);
- ast_mutex_unlock(&netlock);
-
- if (p->logger_callid) {
- ast_callid_threadassoc_remove();
- }
- ao2_t_ref(p, -1, "throw away dialog ptr from find_call at end of routine"); /* p is gone after the return */
-
- return 1;
-}
-
-/*! \brief Returns the port to use for this socket
- *
- * \param type The type of transport used
- * \param port Port we are checking to see if it's the standard port.
- * \note port is expected in host byte order
- */
-static int sip_standard_port(enum ast_transport type, int port)
-{
- if (type & AST_TRANSPORT_TLS)
- return port == STANDARD_TLS_PORT;
- else
- return port == STANDARD_SIP_PORT;
-}
-
-static int threadinfo_locate_cb(void *obj, void *arg, int flags)
-{
- struct sip_threadinfo *th = obj;
- struct ast_sockaddr *s = arg;
-
- if (!ast_sockaddr_cmp(s, &th->tcptls_session->remote_address)) {
- return CMP_MATCH | CMP_STOP;
- }
-
- return 0;
-}
-
-/*!
- * \brief Find thread for TCP/TLS session (based on IP/Port
- *
- * \note This function returns an astobj2 reference
- */
-static struct ast_tcptls_session_instance *sip_tcp_locate(struct ast_sockaddr *s)
-{
- struct sip_threadinfo *th;
- struct ast_tcptls_session_instance *tcptls_instance = NULL;
-
- if ((th = ao2_callback(threadt, 0, threadinfo_locate_cb, s))) {
- tcptls_instance = (ao2_ref(th->tcptls_session, +1), th->tcptls_session);
- ao2_t_ref(th, -1, "decrement ref from callback");
- }
-
- return tcptls_instance;
-}
-
-/*!
- * \brief Helper for dns resolution to filter by address family.
- *
- * \note return 0 if addr is [::] else it returns addr's family.
- */
-int get_address_family_filter(unsigned int transport)
-{
- const struct ast_sockaddr *addr = NULL;
-
- if ((transport == AST_TRANSPORT_UDP) || !transport) {
- addr = &bindaddr;
- } else if (transport == AST_TRANSPORT_TCP || transport == AST_TRANSPORT_WS) {
- addr = &sip_tcp_desc.local_address;
- } else if (transport == AST_TRANSPORT_TLS || transport == AST_TRANSPORT_WSS) {
- addr = &sip_tls_desc.local_address;
- }
-
- if (ast_sockaddr_is_ipv6(addr) && ast_sockaddr_is_any(addr)) {
- return 0;
- }
-
- return addr->ss.ss_family;
-}
-
-/*! \todo Get socket for dialog, prepare if needed, and return file handle */
-static int sip_prepare_socket(struct sip_pvt *p)
-{
- struct sip_socket *s = &p->socket;
- static const char name[] = "SIP socket";
- struct sip_threadinfo *th = NULL;
- struct ast_tcptls_session_instance *tcptls_session;
- struct ast_tcptls_session_args *ca;
- struct ast_sockaddr sa_tmp;
- pthread_t launched;
-
- /* check to see if a socket is already active */
- if ((s->fd != -1) && (s->type == AST_TRANSPORT_UDP)) {
- return s->fd;
- }
- if ((s->type & (AST_TRANSPORT_TCP | AST_TRANSPORT_TLS)) &&
- s->tcptls_session && s->tcptls_session->stream) {
- return ast_iostream_get_fd(s->tcptls_session->stream);
- }
- if ((s->type & (AST_TRANSPORT_WS | AST_TRANSPORT_WSS))) {
- return s->ws_session ? ast_websocket_fd(s->ws_session) : -1;
- }
-
- /*! \todo Check this... This might be wrong, depending on the proxy configuration
- If proxy is in "force" mode its correct.
- */
- if (p->outboundproxy && p->outboundproxy->transport) {
- s->type = p->outboundproxy->transport;
- }
-
- if (s->type == AST_TRANSPORT_UDP) {
- s->fd = sipsock;
- return s->fd;
- }
-
- /* At this point we are dealing with a TCP/TLS connection
- * 1. We need to check to see if a connection thread exists
- * for this address, if so use that.
- * 2. If a thread does not exist for this address, but the tcptls_session
- * exists on the socket, the connection was closed.
- * 3. If no tcptls_session thread exists for the address, and no tcptls_session
- * already exists on the socket, create a new one and launch a new thread.
- */
-
- /* 1. check for existing threads */
- ast_sockaddr_copy(&sa_tmp, sip_real_dst(p));
- if ((tcptls_session = sip_tcp_locate(&sa_tmp))) {
- s->fd = ast_iostream_get_fd(tcptls_session->stream);
- if (s->tcptls_session) {
- ao2_ref(s->tcptls_session, -1);
- s->tcptls_session = NULL;
- }
- s->tcptls_session = tcptls_session;
- return s->fd;
- /* 2. Thread not found, if tcptls_session already exists, it once had a thread and is now terminated */
- } else if (s->tcptls_session) {
- return s->fd; /* XXX whether reconnection is ever necessary here needs to be investigated further */
- }
-
- /* 3. Create a new TCP/TLS client connection */
- /* create new session arguments for the client connection */
- if (!(ca = ao2_alloc(sizeof(*ca), sip_tcptls_client_args_destructor)) ||
- !(ca->name = ast_strdup(name))) {
- goto create_tcptls_session_fail;
- }
- ca->accept_fd = -1;
- ast_sockaddr_copy(&ca->remote_address,sip_real_dst(p));
- /* if type is TLS, we need to create a tls cfg for this session arg */
- if (s->type == AST_TRANSPORT_TLS) {
- if (!(ca->tls_cfg = ast_calloc(1, sizeof(*ca->tls_cfg)))) {
- goto create_tcptls_session_fail;
- }
- memcpy(ca->tls_cfg, &default_tls_cfg, sizeof(*ca->tls_cfg));
-
- if (!(ca->tls_cfg->certfile = ast_strdup(default_tls_cfg.certfile)) ||
- !(ca->tls_cfg->pvtfile = ast_strdup(default_tls_cfg.pvtfile)) ||
- !(ca->tls_cfg->cipher = ast_strdup(default_tls_cfg.cipher)) ||
- !(ca->tls_cfg->cafile = ast_strdup(default_tls_cfg.cafile)) ||
- !(ca->tls_cfg->capath = ast_strdup(default_tls_cfg.capath))) {
-
- goto create_tcptls_session_fail;
- }
-
- /* this host is used as the common name in ssl/tls */
- if (!ast_strlen_zero(p->tohost)) {
- ast_copy_string(ca->hostname, p->tohost, sizeof(ca->hostname));
- }
- }
-
- /* If a bind address has been specified, use it */
- if ((s->type == AST_TRANSPORT_TLS) && !ast_sockaddr_isnull(&sip_tls_desc.local_address)) {
- ca->local_address = sip_tls_desc.local_address;
- }
- else if ((s->type == AST_TRANSPORT_TCP) && !ast_sockaddr_isnull(&sip_tcp_desc.local_address)) {
- ca->local_address = sip_tcp_desc.local_address;
- }
- /* Reset tcp source port to zero to let system pick a random one */
- if (!ast_sockaddr_isnull(&ca->local_address)) {
- ast_sockaddr_set_port(&ca->local_address, 0);
- }
- /* Create a client connection for address, this does not start the connection, just sets it up. */
- if (!(s->tcptls_session = ast_tcptls_client_create(ca))) {
- goto create_tcptls_session_fail;
- }
-
- s->fd = ast_iostream_get_fd(s->tcptls_session->stream);
-
- /* client connections need to have the sip_threadinfo object created before
- * the thread is detached. This ensures the alert_pipe is up before it will
- * be used. Note that this function links the new threadinfo object into the
- * threadt container. */
- if (!(th = sip_threadinfo_create(s->tcptls_session, s->type))) {
- goto create_tcptls_session_fail;
- }
-
- /* Give the new thread a reference to the tcptls_session */
- ao2_ref(s->tcptls_session, +1);
-
- if (ast_pthread_create_detached_background(&launched, NULL, sip_tcp_worker_fn, s->tcptls_session)) {
- ast_debug(1, "Unable to launch '%s'.", ca->name);
- ao2_ref(s->tcptls_session, -1); /* take away the thread ref we just gave it */
- goto create_tcptls_session_fail;
- }
-
- ast_set_qos(s->fd, global_tos_sip, global_cos_sip, "SIP");
-
- return s->fd;
-
-create_tcptls_session_fail:
- if (ca) {
- ao2_t_ref(ca, -1, "failed to create client, getting rid of client tcptls_session arguments");
- }
- if (s->tcptls_session) {
- ast_tcptls_close_session_file(s->tcptls_session);
- s->fd = -1;
- ao2_ref(s->tcptls_session, -1);
- s->tcptls_session = NULL;
- }
- if (th) {
- ao2_t_unlink(threadt, th, "Removing tcptls thread info object, thread failed to open");
- }
-
- return -1;
-}
-
-/*!
- * \brief Get cached MWI info
- * \return TRUE if found MWI in cache
- */
-static int get_cached_mwi(struct sip_peer *peer, int *new, int *old)
-{
- struct sip_mailbox *mailbox;
- int in_cache;
-
- in_cache = 0;
- AST_LIST_TRAVERSE(&peer->mailboxes, mailbox, entry) {
- RAII_VAR(struct stasis_message *, msg, NULL, ao2_cleanup);
- struct ast_mwi_state *mwi_state;
-
- msg = stasis_cache_get(ast_mwi_state_cache(), ast_mwi_state_type(), mailbox->id);
- if (!msg) {
- continue;
- }
-
- mwi_state = stasis_message_data(msg);
- *new += mwi_state->new_msgs;
- *old += mwi_state->old_msgs;
- in_cache = 1;
- }
-
- return in_cache;
-}
-
-/*! \brief Send message waiting indication to alert peer that they've got voicemail
- * \note Both peer and associated sip_pvt must be unlocked prior to calling this function.
- * It's possible that this function will get called during peer destruction as final messages
- * are processed. The peer will still be valid however.
- * \retval -1 on failure.
- * \retval 0 on success.
- */
-static int sip_send_mwi_to_peer(struct sip_peer *peer, int cache_only)
-{
- /* Called with peer lock, but releases it */
- struct sip_pvt *p;
- int newmsgs = 0, oldmsgs = 0;
- const char *vmexten = NULL;
-
- ao2_lock(peer);
-
- if (peer->vmexten) {
- vmexten = ast_strdupa(peer->vmexten);
- }
-
- if (ast_test_flag((&peer->flags[1]), SIP_PAGE2_SUBSCRIBEMWIONLY) && !peer->mwipvt) {
- update_peer_lastmsgssent(peer, -1, 1);
- ao2_unlock(peer);
- return -1;
- }
-
- /* Do we have an IP address? If not, skip this peer */
- if (ast_sockaddr_isnull(&peer->addr) && ast_sockaddr_isnull(&peer->defaddr)) {
- update_peer_lastmsgssent(peer, -1, 1);
- ao2_unlock(peer);
- return -1;
- }
-
- /* Attempt to use cached mwi to get message counts. */
- if (!get_cached_mwi(peer, &newmsgs, &oldmsgs) && !cache_only) {
- /* Fall back to manually checking the mailbox if not cache_only and get_cached_mwi failed */
- struct ast_str *mailbox_str = ast_str_alloca(512);
- peer_mailboxes_to_str(&mailbox_str, peer);
- /* if there is no mailbox do nothing */
- if (!ast_str_strlen(mailbox_str)) {
- ao2_unlock(peer);
- return -1;
- }
- ao2_unlock(peer);
- /* If there is no mailbox do nothing */
- if (!ast_str_strlen(mailbox_str)) {
- update_peer_lastmsgssent(peer, -1, 0);
- return 0;
- }
- ast_app_inboxcount(ast_str_buffer(mailbox_str), &newmsgs, &oldmsgs);
- ao2_lock(peer);
- }
-
- if (peer->mwipvt) {
- /* Base message on subscription */
- p = dialog_ref(peer->mwipvt, "sip_send_mwi_to_peer: Setting dialog ptr p from peer->mwipvt");
- ao2_unlock(peer);
- } else {
- ao2_unlock(peer);
- /* Build temporary dialog for this message */
- if (!(p = sip_alloc(NULL, NULL, 0, SIP_NOTIFY, NULL, 0))) {
- update_peer_lastmsgssent(peer, -1, 0);
- return -1;
- }
-
- /* If we don't set the socket type to 0, then create_addr_from_peer will fail immediately if the peer
- * uses any transport other than UDP. We set the type to 0 here and then let create_addr_from_peer copy
- * the peer's socket information to the sip_pvt we just allocated
- */
- set_socket_transport(&p->socket, 0);
- if (create_addr_from_peer(p, peer)) {
- /* Maybe they're not registered, etc. */
- dialog_unlink_all(p);
- dialog_unref(p, "unref dialog p just created via sip_alloc");
- update_peer_lastmsgssent(peer, -1, 0);
- return -1;
- }
- /* Recalculate our side, and recalculate Call ID */
- ast_sip_ouraddrfor(&p->sa, &p->ourip, p);
- build_via(p);
-
- ao2_lock(peer);
- if (!ast_strlen_zero(peer->mwi_from)) {
- ast_string_field_set(p, mwi_from, peer->mwi_from);
- } else if (!ast_strlen_zero(default_mwi_from)) {
- ast_string_field_set(p, mwi_from, default_mwi_from);
- }
- ao2_unlock(peer);
-
- /* Change the dialog callid. */
- change_callid_pvt(p, NULL);
-
- /* Destroy this session after 32 secs */
- sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
- }
-
- /* We have multiple threads (mwi events and monitor retransmits) working with this PVT and as we modify the sip history if that's turned on,
- we really need to have a lock on it */
- sip_pvt_lock(p);
-
- /* Send MWI */
- ast_set_flag(&p->flags[0], SIP_OUTGOING);
- /* the following will decrement the refcount on p as it finishes */
- transmit_notify_with_mwi(p, newmsgs, oldmsgs, vmexten);
- sip_pvt_unlock(p);
- dialog_unref(p, "unref dialog ptr p just before it goes out of scope at the end of sip_send_mwi_to_peer.");
-
- update_peer_lastmsgssent(peer, ((newmsgs > 0x7fff ? 0x7fff0000 : (newmsgs << 16)) | (oldmsgs > 0xffff ? 0xffff : oldmsgs)), 0);
-
- return 0;
-}
-
-static struct ast_manager_event_blob *session_timeout_to_ami(struct stasis_message *msg)
-{
- RAII_VAR(struct ast_str *, channel_string, NULL, ast_free);
- struct ast_channel_blob *obj = stasis_message_data(msg);
- const char *source = ast_json_string_get(ast_json_object_get(obj->blob, "source"));
-
- channel_string = ast_manager_build_channel_state_string(obj->snapshot);
- if (!channel_string) {
- return NULL;
- }
-
- return ast_manager_event_blob_create(EVENT_FLAG_CALL, "SessionTimeout",
- "%s"
- "Source: %s\r\n",
- ast_str_buffer(channel_string), source);
-}
-
-/*! \brief Sends a session timeout channel blob used to produce SessionTimeout AMI messages */
-static void send_session_timeout(struct ast_channel *chan, const char *source)
-{
- RAII_VAR(struct ast_json *, blob, NULL, ast_json_unref);
-
- ast_assert(chan != NULL);
- ast_assert(source != NULL);
-
- blob = ast_json_pack("{s: s}", "source", source);
- if (!blob) {
- return;
- }
-
- ast_channel_publish_blob(chan, session_timeout_type(), blob);
-}
-
-/*!
- * \brief helper function for the monitoring thread -- seems to be called with the assumption that the dialog is locked
- *
- * \return CMP_MATCH for items to be unlinked from dialogs_rtpcheck.
- */
-static int check_rtp_timeout(struct sip_pvt *dialog, time_t t)
-{
- int timeout;
- int hold_timeout;
- int keepalive;
-
- if (!dialog->rtp) {
- /*
- * We have no RTP. Since we don't do much with video RTP for
- * now, stop checking this dialog.
- */
- return CMP_MATCH;
- }
-
- /* If we have no active owner, no need to check timers */
- if (!dialog->owner) {
- return CMP_MATCH;
- }
-
- /* If the call is redirected outside Asterisk, no need to check timers */
- if (!ast_sockaddr_isnull(&dialog->redirip)) {
- return CMP_MATCH;
- }
-
- /* If the call is involved in a T38 fax session do not check RTP timeout */
- if (dialog->t38.state == T38_ENABLED) {
- return CMP_MATCH;
- }
- /* If the call is not in UP state return for later check. */
- if (ast_channel_state(dialog->owner) != AST_STATE_UP) {
- return 0;
- }
-
- /* Store these values locally to avoid multiple function calls */
- timeout = ast_rtp_instance_get_timeout(dialog->rtp);
- hold_timeout = ast_rtp_instance_get_hold_timeout(dialog->rtp);
- keepalive = ast_rtp_instance_get_keepalive(dialog->rtp);
-
- /* If we have no timers set, return now */
- if (!keepalive && !timeout && !hold_timeout) {
- return CMP_MATCH;
- }
-
- /* Check AUDIO RTP keepalives */
- if (dialog->lastrtptx && keepalive && (t > dialog->lastrtptx + keepalive)) {
- /* Need to send an empty RTP packet */
- dialog->lastrtptx = time(NULL);
- ast_rtp_instance_sendcng(dialog->rtp, 0);
- }
-
- /*! \todo Check video RTP keepalives
-
- Do we need to move the lastrtptx to the RTP structure to have one for audio and one
- for video? It really does belong to the RTP structure.
- */
-
- /* Check AUDIO RTP timers */
- if (dialog->lastrtprx && (timeout || hold_timeout) && (t > dialog->lastrtprx + timeout)) {
- if (!ast_test_flag(&dialog->flags[1], SIP_PAGE2_CALL_ONHOLD) || (hold_timeout && (t > dialog->lastrtprx + hold_timeout))) {
- /* Needs a hangup */
- if (timeout) {
- if (!dialog->owner || ast_channel_trylock(dialog->owner)) {
- /*
- * Don't block, just try again later.
- * If there was no owner, the call is dead already.
- */
- return 0;
- }
- ast_log(LOG_NOTICE, "Disconnecting call '%s' for lack of RTP activity in %ld seconds\n",
- ast_channel_name(dialog->owner), (long) (t - dialog->lastrtprx));
- send_session_timeout(dialog->owner, "RTPTimeout");
-
- /* Issue a softhangup - cause 44 (as used by Cisco for RTP timeouts) */
- ast_channel_hangupcause_set(dialog->owner, AST_CAUSE_REQUESTED_CHAN_UNAVAIL);
- ast_softhangup_nolock(dialog->owner, AST_SOFTHANGUP_DEV);
- ast_channel_unlock(dialog->owner);
- /* forget the timeouts for this call, since a hangup
- has already been requested and we don't want to
- repeatedly request hangups
- */
- ast_rtp_instance_set_timeout(dialog->rtp, 0);
- ast_rtp_instance_set_hold_timeout(dialog->rtp, 0);
- if (dialog->vrtp) {
- ast_rtp_instance_set_timeout(dialog->vrtp, 0);
- ast_rtp_instance_set_hold_timeout(dialog->vrtp, 0);
- }
- /* finally unlink the dialog from dialogs_rtpcheck. */
- return CMP_MATCH;
- }
- }
- }
- return 0;
-}
-
-/*! \brief The SIP monitoring thread
-\note This thread monitors all the SIP sessions and peers that needs notification of mwi
- (and thus do not have a separate thread) indefinitely
-*/
-static void *do_monitor(void *data)
-{
- int res;
- time_t t;
- int reloading;
-
- /* Add an I/O event to our SIP UDP socket */
- if (sipsock > -1) {
- sipsock_read_id = ast_io_add(io, sipsock, sipsock_read, AST_IO_IN, NULL);
- }
-
- /* From here on out, we die whenever asked */
- for(;;) {
- /* Check for a reload request */
- ast_mutex_lock(&sip_reload_lock);
- reloading = sip_reloading;
- sip_reloading = FALSE;
- ast_mutex_unlock(&sip_reload_lock);
- if (reloading) {
- ast_verb(1, "Reloading SIP\n");
- sip_do_reload(sip_reloadreason);
-
- /* Change the I/O fd of our UDP socket */
- if (sipsock > -1) {
- if (sipsock_read_id) {
- sipsock_read_id = ast_io_change(io, sipsock_read_id, sipsock, NULL, 0, NULL);
- } else {
- sipsock_read_id = ast_io_add(io, sipsock, sipsock_read, AST_IO_IN, NULL);
- }
- } else if (sipsock_read_id) {
- ast_io_remove(io, sipsock_read_id);
- sipsock_read_id = NULL;
- }
- }
-
- /* Check for dialogs needing to be killed */
- t = time(NULL);
-
- /*
- * Check dialogs with rtp and rtptimeout.
- * All dialogs which have rtp are in dialogs_rtpcheck.
- */
- ao2_t_callback(dialogs_rtpcheck, OBJ_UNLINK | OBJ_NODATA | OBJ_MULTIPLE,
- dialog_checkrtp_cb, &t,
- "callback to check rtptimeout and hangup calls if necessary");
- /*
- * Check dialogs marked to be destroyed.
- * All dialogs with needdestroy set are in dialogs_needdestroy.
- */
- ao2_t_callback(dialogs_needdestroy, OBJ_NODATA | OBJ_MULTIPLE, dialog_needdestroy,
- NULL, "callback to check dialogs which need to be destroyed");
-
- /* XXX TODO The scheduler usage in this module does not have sufficient
- * synchronization being done between running the scheduler and places
- * scheduling tasks. As it is written, any scheduled item may not run
- * any sooner than about 1 second, regardless of whether a sooner time
- * was asked for. */
-
- pthread_testcancel();
- /* Wait for sched or io */
- res = ast_sched_wait(sched);
- if ((res < 0) || (res > 1000)) {
- res = 1000;
- }
- res = ast_io_wait(io, res);
- if (res > 20) {
- ast_debug(1, "chan_sip: ast_io_wait ran %d all at once\n", res);
- }
- ast_mutex_lock(&monlock);
- res = ast_sched_runq(sched);
- if (res >= 20) {
- ast_debug(1, "chan_sip: ast_sched_runq ran %d all at once\n", res);
- }
- ast_mutex_unlock(&monlock);
- }
-
- /* Never reached */
- return NULL;
-}
-
-/*! \brief Start the channel monitor thread */
-static int restart_monitor(void)
-{
- /* If we're supposed to be stopped -- stay stopped */
- if (monitor_thread == AST_PTHREADT_STOP)
- return 0;
- ast_mutex_lock(&monlock);
- if (monitor_thread == pthread_self()) {
- ast_mutex_unlock(&monlock);
- ast_log(LOG_WARNING, "Cannot kill myself\n");
- return -1;
- }
- if (monitor_thread != AST_PTHREADT_NULL && monitor_thread != AST_PTHREADT_STOP) {
- /* Wake up the thread */
- pthread_kill(monitor_thread, SIGURG);
- } else {
- /* Start a new monitor */
- if (ast_pthread_create_background(&monitor_thread, NULL, do_monitor, NULL) < 0) {
- ast_mutex_unlock(&monlock);
- ast_log(LOG_ERROR, "Unable to start monitor thread.\n");
- return -1;
- }
- }
- ast_mutex_unlock(&monlock);
- return 0;
-}
-
-static void acl_change_stasis_cb(void *data, struct stasis_subscription *sub,
- struct stasis_message *message)
-{
- if (stasis_message_type(message) != ast_named_acl_change_type()) {
- return;
- }
-
- ast_log(LOG_NOTICE, "Reloading chan_sip in response to ACL change event.\n");
-
- ast_mutex_lock(&sip_reload_lock);
-
- if (sip_reloading) {
- ast_verbose("Previous SIP reload not yet done\n");
- } else {
- sip_reloading = TRUE;
- sip_reloadreason = CHANNEL_ACL_RELOAD;
- }
-
- ast_mutex_unlock(&sip_reload_lock);
-
- restart_monitor();
-}
-
-/*!
- * \brief Session-Timers: Process session refresh timeout event
- *
- * \note Run by the sched thread.
- */
-static int proc_session_timer(const void *vp)
-{
- struct sip_pvt *p = (struct sip_pvt *) vp;
- struct sip_st_dlg *stimer = p->stimer;
- int res = 0;
-
- ast_assert(stimer != NULL);
-
- ast_debug(2, "Session timer expired: %d - %s\n", stimer->st_schedid, p->callid);
-
- if (!p->owner) {
- goto return_unref;
- }
-
- if ((stimer->st_active != TRUE) || (ast_channel_state(p->owner) != AST_STATE_UP)) {
- goto return_unref;
- }
-
- if (stimer->st_ref == SESSION_TIMER_REFRESHER_US) {
- res = 1;
- if (T38_ENABLED == p->t38.state) {
- transmit_reinvite_with_sdp(p, TRUE, TRUE);
- } else {
- transmit_reinvite_with_sdp(p, FALSE, TRUE);
- }
- } else {
- struct ast_channel *owner;
-
- ast_log(LOG_WARNING, "Session-Timer expired - %s\n", p->callid);
-
- owner = sip_pvt_lock_full(p);
- if (owner) {
- send_session_timeout(owner, "SIPSessionTimer");
- ast_softhangup_nolock(owner, AST_SOFTHANGUP_DEV);
- ast_channel_unlock(owner);
- ast_channel_unref(owner);
- }
- sip_pvt_unlock(p);
- }
-
-return_unref:
- if (!res) {
- /* Session timer processing is no longer needed. */
- ast_debug(2, "Session timer stopped: %d - %s\n",
- stimer->st_schedid, p->callid);
- /* Don't pass go, don't collect $200.. we are the scheduled
- * callback. We can rip ourself out here. */
- stimer->st_schedid = -1;
- stimer->st_active = FALSE;
-
- /* If we are not asking to be rescheduled, then we need to release our
- * reference to the dialog. */
- dialog_unref(p, "Session timer st_schedid complete");
- }
-
- return res;
-}
-
-static void do_stop_session_timer(struct sip_pvt *pvt)
-{
- struct sip_st_dlg *stimer = pvt->stimer;
-
- if (-1 < stimer->st_schedid) {
- ast_debug(2, "Session timer stopped: %d - %s\n",
- stimer->st_schedid, pvt->callid);
- AST_SCHED_DEL_UNREF(sched, stimer->st_schedid,
- dialog_unref(pvt, "Stop scheduled session timer st_schedid"));
- }
-}
-
-/* Run by the sched thread. */
-static int __stop_session_timer(const void *data)
-{
- struct sip_pvt *pvt = (void *) data;
-
- do_stop_session_timer(pvt);
- dialog_unref(pvt, "Stop session timer action");
- return 0;
-}
-
-/*! \brief Session-Timers: Stop session timer */
-static void stop_session_timer(struct sip_pvt *pvt)
-{
- struct sip_st_dlg *stimer = pvt->stimer;
-
- stimer->st_active = FALSE;
- dialog_ref(pvt, "Stop session timer action");
- if (ast_sched_add(sched, 0, __stop_session_timer, pvt) < 0) {
- /* Uh Oh. Expect bad behavior. */
- dialog_unref(pvt, "Failed to schedule stop session timer action");
- }
-}
-
-/* Run by the sched thread. */
-static int __start_session_timer(const void *data)
-{
- struct sip_pvt *pvt = (void *) data;
- struct sip_st_dlg *stimer = pvt->stimer;
- unsigned int timeout_ms;
-
- /*
- * RFC 4028 Section 10
- * If the side not performing refreshes does not receive a
- * session refresh request before the session expiration, it SHOULD send
- * a BYE to terminate the session, slightly before the session
- * expiration. The minimum of 32 seconds and one third of the session
- * interval is RECOMMENDED.
- */
-
- timeout_ms = (1000 * stimer->st_interval);
- if (stimer->st_ref == SESSION_TIMER_REFRESHER_US) {
- timeout_ms /= 2;
- } else {
- timeout_ms -= MIN(timeout_ms / 3, 32000);
- }
-
- /* in the event a timer is already going, stop it */
- do_stop_session_timer(pvt);
-
- dialog_ref(pvt, "Schedule session timer st_schedid");
- stimer->st_schedid = ast_sched_add(sched, timeout_ms, proc_session_timer, pvt);
- if (stimer->st_schedid < 0) {
- dialog_unref(pvt, "Failed to schedule session timer st_schedid");
- } else {
- ast_debug(2, "Session timer started: %d - %s %ums\n",
- stimer->st_schedid, pvt->callid, timeout_ms);
- }
-
- dialog_unref(pvt, "Start session timer action");
- return 0;
-}
-
-/*! \brief Session-Timers: Start session timer */
-static void start_session_timer(struct sip_pvt *pvt)
-{
- struct sip_st_dlg *stimer = pvt->stimer;
-
- stimer->st_active = TRUE;
- dialog_ref(pvt, "Start session timer action");
- if (ast_sched_add(sched, 0, __start_session_timer, pvt) < 0) {
- /* Uh Oh. Expect bad behavior. */
- dialog_unref(pvt, "Failed to schedule start session timer action");
- }
-}
-
-/*! \brief Session-Timers: Restart session timer */
-static void restart_session_timer(struct sip_pvt *p)
-{
- if (p->stimer->st_active == TRUE) {
- start_session_timer(p);
- }
-}
-
-/*! \brief Session-Timers: Function for parsing Min-SE header */
-int parse_minse (const char *p_hdrval, int *const p_interval)
-{
- if (ast_strlen_zero(p_hdrval)) {
- ast_log(LOG_WARNING, "Null Min-SE header\n");
- return -1;
- }
-
- *p_interval = 0;
- p_hdrval = ast_skip_blanks(p_hdrval);
- if (!sscanf(p_hdrval, "%30d", p_interval)) {
- ast_log(LOG_WARNING, "Parsing of Min-SE header failed %s\n", p_hdrval);
- return -1;
- }
-
- ast_debug(2, "Received Min-SE: %d\n", *p_interval);
- return 0;
-}
-
-
-/*! \brief Session-Timers: Function for parsing Session-Expires header */
-int parse_session_expires(const char *p_hdrval, int *const p_interval, enum st_refresher_param *const p_ref)
-{
- char *p_token;
- int ref_idx;
- char *p_se_hdr;
-
- if (ast_strlen_zero(p_hdrval)) {
- ast_log(LOG_WARNING, "Null Session-Expires header\n");
- return -1;
- }
-
- *p_ref = SESSION_TIMER_REFRESHER_PARAM_UNKNOWN;
- *p_interval = 0;
-
- p_se_hdr = ast_strdupa(p_hdrval);
- p_se_hdr = ast_skip_blanks(p_se_hdr);
-
- while ((p_token = strsep(&p_se_hdr, ";"))) {
- p_token = ast_skip_blanks(p_token);
- if (!sscanf(p_token, "%30d", p_interval)) {
- ast_log(LOG_WARNING, "Parsing of Session-Expires failed\n");
- return -1;
- }
-
- ast_debug(2, "Session-Expires: %d\n", *p_interval);
-
- if (!p_se_hdr)
- continue;
-
- p_se_hdr = ast_skip_blanks(p_se_hdr);
- ref_idx = strlen("refresher=");
- if (!strncasecmp(p_se_hdr, "refresher=", ref_idx)) {
- p_se_hdr += ref_idx;
- p_se_hdr = ast_skip_blanks(p_se_hdr);
-
- if (!strncasecmp(p_se_hdr, "uac", strlen("uac"))) {
- *p_ref = SESSION_TIMER_REFRESHER_PARAM_UAC;
- ast_debug(2, "Refresher: UAC\n");
- } else if (!strncasecmp(p_se_hdr, "uas", strlen("uas"))) {
- *p_ref = SESSION_TIMER_REFRESHER_PARAM_UAS;
- ast_debug(2, "Refresher: UAS\n");
- } else {
- ast_log(LOG_WARNING, "Invalid refresher value %s\n", p_se_hdr);
- return -1;
- }
- break;
- }
- }
- return 0;
-}
-
-
-/*! \brief Handle 422 response to INVITE with session-timer requested
-
- Session-Timers: An INVITE originated by Asterisk that asks for session-timers support
- from the UAS can result into a 422 response. This is how a UAS or an intermediary proxy
- server tells Asterisk that the session refresh interval offered by Asterisk is too low
- for them. The proc_422_rsp() function handles a 422 response. It extracts the Min-SE
- header that comes back in 422 and sends a new INVITE accordingly. */
-static void proc_422_rsp(struct sip_pvt *p, struct sip_request *rsp)
-{
- int rtn;
- const char *p_hdrval;
- int minse;
-
- p_hdrval = sip_get_header(rsp, "Min-SE");
- if (ast_strlen_zero(p_hdrval)) {
- ast_log(LOG_WARNING, "422 response without a Min-SE header\n");
- return;
- }
- rtn = parse_minse(p_hdrval, &minse);
- if (rtn != 0) {
- ast_log(LOG_WARNING, "Parsing of Min-SE header failed %s\n", p_hdrval);
- return;
- }
- p->stimer->st_cached_min_se = minse;
- if (p->stimer->st_interval < minse) {
- p->stimer->st_interval = minse;
- }
- transmit_invite(p, SIP_INVITE, 1, 2, NULL);
-}
-
-
-/*! \brief Get Max or Min SE (session timer expiry)
- * \param p pointer to the SIP dialog
- * \param max if true, get max se, otherwise min se
-*/
-int st_get_se(struct sip_pvt *p, int max)
-{
- if (max == TRUE) {
- if (p->stimer->st_cached_max_se) {
- return p->stimer->st_cached_max_se;
- }
- if (p->relatedpeer) {
- p->stimer->st_cached_max_se = p->relatedpeer->stimer.st_max_se;
- return (p->stimer->st_cached_max_se);
- }
- p->stimer->st_cached_max_se = global_max_se;
- return (p->stimer->st_cached_max_se);
- }
- /* Find Min SE timer */
- if (p->stimer->st_cached_min_se) {
- return p->stimer->st_cached_min_se;
- }
- if (p->relatedpeer) {
- p->stimer->st_cached_min_se = p->relatedpeer->stimer.st_min_se;
- return (p->stimer->st_cached_min_se);
- }
- p->stimer->st_cached_min_se = global_min_se;
- return (p->stimer->st_cached_min_se);
-}
-
-
-/*! \brief Get the entity (UAC or UAS) that's acting as the session-timer refresher
- * \note This is only called when processing an INVITE, so in that case Asterisk is
- * always currently the UAS. If this is ever used to process responses, the
- * function will have to be changed.
- * \param p pointer to the SIP dialog
-*/
-enum st_refresher st_get_refresher(struct sip_pvt *p)
-{
- if (p->stimer->st_cached_ref != SESSION_TIMER_REFRESHER_AUTO) {
- return p->stimer->st_cached_ref;
- }
-
- if (p->relatedpeer) {
- p->stimer->st_cached_ref = (p->relatedpeer->stimer.st_ref == SESSION_TIMER_REFRESHER_PARAM_UAC) ? SESSION_TIMER_REFRESHER_THEM : SESSION_TIMER_REFRESHER_US;
- return p->stimer->st_cached_ref;
- }
-
- p->stimer->st_cached_ref = (global_st_refresher == SESSION_TIMER_REFRESHER_PARAM_UAC) ? SESSION_TIMER_REFRESHER_THEM : SESSION_TIMER_REFRESHER_US;
- return p->stimer->st_cached_ref;
-}
-
-
-/*!
- * \brief Get the session-timer mode
- * \param p pointer to the SIP dialog
- * \param no_cached Set this to true in order to force a peername lookup on
- * the session timer mode.
-*/
-enum st_mode st_get_mode(struct sip_pvt *p, int no_cached)
-{
- if (!p->stimer) {
- sip_st_alloc(p);
- if (!p->stimer) {
- return SESSION_TIMER_MODE_INVALID;
- }
- }
-
- if (!no_cached && p->stimer->st_cached_mode != SESSION_TIMER_MODE_INVALID)
- return p->stimer->st_cached_mode;
-
- if (p->relatedpeer) {
- p->stimer->st_cached_mode = p->relatedpeer->stimer.st_mode_oper;
- return p->stimer->st_cached_mode;
- }
-
- p->stimer->st_cached_mode = global_st_mode;
- return global_st_mode;
-}
-
-/*! \brief Send keep alive packet to peer */
-static int sip_send_keepalive(const void *data)
-{
- struct sip_peer *peer = (struct sip_peer*) data;
- int res = 0;
- const char keepalive[] = "\r\n";
- size_t count = sizeof(keepalive) - 1;
-
- peer->keepalivesend = -1;
-
- if (!peer->keepalive || ast_sockaddr_isnull(&peer->addr)) {
- sip_unref_peer(peer, "release keepalive peer ref");
- return 0;
- }
-
- /* Send the packet out using the proper method for this peer */
- if ((peer->socket.fd != -1) && (peer->socket.type == AST_TRANSPORT_UDP)) {
- res = ast_sendto(peer->socket.fd, keepalive, count, 0, &peer->addr);
- } else if ((peer->socket.type & (AST_TRANSPORT_TCP | AST_TRANSPORT_TLS)) &&
- peer->socket.tcptls_session) {
- res = sip_tcptls_write(peer->socket.tcptls_session, keepalive, count);
- if (res < -1) {
- return 0;
- }
- } else if (peer->socket.type == AST_TRANSPORT_UDP) {
- res = ast_sendto(sipsock, keepalive, count, 0, &peer->addr);
- }
-
- if (res == -1) {
- switch (errno) {
- case EBADF: /* Bad file descriptor - seems like this is generated when the host exist, but doesn't accept the UDP packet */
- case EHOSTUNREACH: /* Host can't be reached */
- case ENETDOWN: /* Interface down */
- case ENETUNREACH: /* Network failure */
- case ECONNREFUSED: /* ICMP port unreachable */
- res = XMIT_ERROR; /* Don't bother with trying to transmit again */
- }
- }
-
- if (res != count) {
- ast_log(LOG_WARNING, "sip_send_keepalive to %s returned %d: %s\n", ast_sockaddr_stringify(&peer->addr), res, strerror(errno));
- }
-
- AST_SCHED_REPLACE_UNREF(peer->keepalivesend, sched,
- peer->keepalive * 1000, sip_send_keepalive, peer,
- sip_unref_peer(_data, "removing keepalive peer ref"),
- sip_unref_peer(peer, "removing keepalive peer ref"),
- sip_ref_peer(peer, "adding keepalive peer ref"));
-
- sip_unref_peer(peer, "release keepalive peer ref");
-
- return 0;
-}
-
-/*! \brief React to lack of answer to Qualify poke */
-static int sip_poke_noanswer(const void *data)
-{
- struct sip_peer *peer = (struct sip_peer *)data;
-
- peer->pokeexpire = -1;
-
- if (peer->lastms > -1) {
-
- ast_log(LOG_NOTICE, "Peer '%s' is now UNREACHABLE! Last qualify: %d\n", peer->name, peer->lastms);
- if (sip_cfg.peer_rtupdate) {
- ast_update_realtime(ast_check_realtime("sipregs") ? "sipregs" : "sippeers", "name", peer->name, "lastms", "-1", SENTINEL);
- }
-
- if (peer->endpoint) {
- RAII_VAR(struct ast_json *, blob, NULL, ast_json_unref);
- ast_endpoint_set_state(peer->endpoint, AST_ENDPOINT_OFFLINE);
- blob = ast_json_pack("{s: s, s: s}",
- "peer_status", "Unreachable",
- "time", "-1");
- ast_endpoint_blob_publish(peer->endpoint, ast_endpoint_state_type(), blob);
- }
-
- if (sip_cfg.regextenonqualify) {
- register_peer_exten(peer, FALSE);
- }
- }
-
- if (peer->call) {
- dialog_unlink_all(peer->call);
- peer->call = dialog_unref(peer->call, "unref dialog peer->call");
- /* peer->call = sip_destroy(peer->call);*/
- }
-
- /* Don't send a devstate change if nothing changed. */
- if (peer->lastms > -1) {
- peer->lastms = -1;
- ast_devstate_changed(AST_DEVICE_UNKNOWN, AST_DEVSTATE_CACHABLE, "SIP/%s", peer->name);
- }
-
- /* Try again quickly */
- AST_SCHED_REPLACE_UNREF(peer->pokeexpire, sched,
- DEFAULT_FREQ_NOTOK, sip_poke_peer_s, peer,
- sip_unref_peer(_data, "removing poke peer ref"),
- sip_unref_peer(peer, "removing poke peer ref"),
- sip_ref_peer(peer, "adding poke peer ref"));
-
- /* Release the ref held by the running scheduler entry */
- sip_unref_peer(peer, "release peer poke noanswer ref");
-
- return 0;
-}
-
-/*! \brief Check availability of peer, also keep NAT open
-\note This is done with 60 seconds between each ping,
- unless forced by cli or manager. If peer is unreachable,
- we check every 10th second by default.
-\note Do *not* hold a pvt lock while calling this function.
- This function calls sip_alloc, which can cause a deadlock
- if another sip_pvt is held.
-*/
-static int sip_poke_peer(struct sip_peer *peer, int force)
-{
- struct sip_pvt *p;
- int xmitres = 0;
-
- if ((!peer->maxms && !force) || ast_sockaddr_isnull(&peer->addr)) {
- /* IF we have no IP, or this isn't to be monitored, return
- immediately after clearing things out */
- AST_SCHED_DEL_UNREF(sched, peer->pokeexpire,
- sip_unref_peer(peer, "removing poke peer ref"));
-
- peer->lastms = 0;
- if (peer->call) {
- peer->call = dialog_unref(peer->call, "unref dialog peer->call");
- }
- return 0;
- }
- if (peer->call) {
- if (sipdebug) {
- ast_log(LOG_NOTICE, "Still have a QUALIFY dialog active, deleting\n");
- }
- dialog_unlink_all(peer->call);
- peer->call = dialog_unref(peer->call, "unref dialog peer->call");
- /* peer->call = sip_destroy(peer->call); */
- }
- if (!(p = sip_alloc(NULL, NULL, 0, SIP_OPTIONS, NULL, 0))) {
- return -1;
- }
- peer->call = dialog_ref(p, "copy sip alloc from p to peer->call");
-
- p->sa = peer->addr;
- p->recv = peer->addr;
- copy_socket_data(&p->socket, &peer->socket);
- ast_copy_flags(&p->flags[0], &peer->flags[0], SIP_FLAGS_TO_COPY);
- ast_copy_flags(&p->flags[1], &peer->flags[1], SIP_PAGE2_FLAGS_TO_COPY);
- ast_copy_flags(&p->flags[2], &peer->flags[2], SIP_PAGE3_FLAGS_TO_COPY);
- sip_route_copy(&p->route, &peer->path);
- if (!sip_route_empty(&p->route)) {
- /* Parse SIP URI of first route-set hop and use it as target address */
- __set_address_from_contact(sip_route_first_uri(&p->route), &p->sa, p->socket.type == AST_TRANSPORT_TLS ? 1 : 0);
- }
-
- /* Get the outbound proxy information */
- ref_proxy(p, obproxy_get(p, peer));
-
- /* Send OPTIONs to peer's fullcontact */
- if (!ast_strlen_zero(peer->fullcontact)) {
- ast_string_field_set(p, fullcontact, peer->fullcontact);
- }
-
- if (!ast_strlen_zero(peer->fromuser)) {
- ast_string_field_set(p, fromuser, peer->fromuser);
- }
-
- if (!ast_strlen_zero(peer->tohost)) {
- ast_string_field_set(p, tohost, peer->tohost);
- } else {
- ast_string_field_set(p, tohost, ast_sockaddr_stringify_host_remote(&peer->addr));
- }
-
- /* Recalculate our side, and recalculate Call ID */
- ast_sip_ouraddrfor(&p->sa, &p->ourip, p);
- build_via(p);
-
- /* Change the dialog callid. */
- change_callid_pvt(p, NULL);
-
- AST_SCHED_DEL_UNREF(sched, peer->pokeexpire,
- sip_unref_peer(peer, "removing poke peer ref"));
-
- if (p->relatedpeer)
- p->relatedpeer = sip_unref_peer(p->relatedpeer,"unsetting the relatedpeer field in the dialog, before it is set to something else.");
- p->relatedpeer = sip_ref_peer(peer, "setting the relatedpeer field in the dialog");
- ast_set_flag(&p->flags[0], SIP_OUTGOING);
-#ifdef VOCAL_DATA_HACK
- ast_copy_string(p->username, "__VOCAL_DATA_SHOULD_READ_THE_SIP_SPEC__", sizeof(p->username));
- xmitres = transmit_invite(p, SIP_INVITE, 0, 2, NULL); /* sinks the p refcount */
-#else
- xmitres = transmit_invite(p, SIP_OPTIONS, 0, 2, NULL); /* sinks the p refcount */
-#endif
- peer->ps = ast_tvnow();
- if (xmitres == XMIT_ERROR) {
- /* Immediately unreachable, network problems */
- sip_poke_noanswer(sip_ref_peer(peer, "add ref for peerexpire (fake, for sip_poke_noanswer to remove)"));
- } else if (!force) {
- AST_SCHED_REPLACE_UNREF(peer->pokeexpire, sched, peer->maxms * 2, sip_poke_noanswer, peer,
- sip_unref_peer(_data, "removing poke peer ref"),
- sip_unref_peer(peer, "removing poke peer ref"),
- sip_ref_peer(peer, "adding poke peer ref"));
- }
- dialog_unref(p, "unref dialog at end of sip_poke_peer, obtained from sip_alloc, just before it goes out of scope");
- return 0;
-}
-
-/*! \brief Part of PBX channel interface
-\note
-\par Return values:---
-
- If we have qualify on and the device is not reachable, regardless of registration
- state we return AST_DEVICE_UNAVAILABLE
-
- For peers with call limit:
- - not registered AST_DEVICE_UNAVAILABLE
- - registered, no call AST_DEVICE_NOT_INUSE
- - registered, active calls AST_DEVICE_INUSE
- - registered, call limit reached AST_DEVICE_BUSY
- - registered, onhold AST_DEVICE_ONHOLD
- - registered, ringing AST_DEVICE_RINGING
-
- For peers without call limit:
- - not registered AST_DEVICE_UNAVAILABLE
- - registered AST_DEVICE_NOT_INUSE
- - fixed IP (!dynamic) AST_DEVICE_NOT_INUSE
-
- Peers that does not have a known call and can't be reached by OPTIONS
- - unreachable AST_DEVICE_UNAVAILABLE
-
- If we return AST_DEVICE_UNKNOWN, the device state engine will try to find
- out a state by walking the channel list.
-
- The queue system (\ref app_queue.c) treats a member as "active"
- if devicestate is != AST_DEVICE_UNAVAILBALE && != AST_DEVICE_INVALID
-
- When placing a call to the queue member, queue system sets a member to busy if
- != AST_DEVICE_NOT_INUSE and != AST_DEVICE_UNKNOWN
-
-*/
-static int sip_devicestate(const char *data)
-{
- char *host;
- char *tmp;
- struct sip_peer *p;
-
- int res = AST_DEVICE_INVALID;
-
- /* make sure data is not null. Maybe unnecessary, but better be safe */
- host = ast_strdupa(data ? data : "");
- if ((tmp = strchr(host, '@')))
- host = tmp + 1;
-
- ast_debug(3, "Checking device state for peer %s\n", host);
-
- /* If sip_find_peer asks for a realtime peer, then this breaks rtautoclear. This
- * is because when a peer tries to autoexpire, the last thing it does is to
- * queue up an event telling the system that the devicestate has changed
- * (presumably to unavailable). If we ask for a realtime peer here, this would
- * load it BACK into memory, thus defeating the point of trying to clear dead
- * hosts out of memory.
- */
- if ((p = sip_find_peer(host, NULL, FALSE, FINDALLDEVICES, TRUE, 0))) {
- if (!(ast_sockaddr_isnull(&p->addr) && ast_sockaddr_isnull(&p->defaddr))) {
- /* we have an address for the peer */
-
- /* Check status in this order
- - Hold
- - Ringing
- - Busy (enforced only by call limit)
- - Inuse (we have a call)
- - Unreachable (qualify)
- If we don't find any of these state, report AST_DEVICE_NOT_INUSE
- for registered devices */
-
- if (p->onhold)
- /* First check for hold or ring states */
- res = AST_DEVICE_ONHOLD;
- else if (p->ringing) {
- if (p->ringing == p->inuse)
- res = AST_DEVICE_RINGING;
- else
- res = AST_DEVICE_RINGINUSE;
- } else if (p->call_limit && (p->inuse == p->call_limit))
- /* check call limit */
- res = AST_DEVICE_BUSY;
- else if (p->call_limit && p->busy_level && p->inuse >= p->busy_level)
- /* We're forcing busy before we've reached the call limit */
- res = AST_DEVICE_BUSY;
- else if (p->call_limit && p->inuse)
- /* Not busy, but we do have a call */
- res = AST_DEVICE_INUSE;
- else if (p->maxms && ((p->lastms > p->maxms) || (p->lastms < 0)))
- /* We don't have a call. Are we reachable at all? Requires qualify= */
- res = AST_DEVICE_UNAVAILABLE;
- else /* Default reply if we're registered and have no other data */
- res = AST_DEVICE_NOT_INUSE;
- } else {
- /* there is no address, it's unavailable */
- res = AST_DEVICE_UNAVAILABLE;
- }
- sip_unref_peer(p, "sip_unref_peer, from sip_devicestate, release ref from sip_find_peer");
- }
-
- return res;
-}
-
-/*! \brief PBX interface function -build SIP pvt structure
- * SIP calls initiated by the PBX arrive here.
- *
- * \verbatim
- * SIP Dial string syntax:
- * SIP/devicename
- * or SIP/username@domain (SIP uri)
- * or SIP/username[:password[:md5secret[:authname[:transport]]]]@host[:port]
- * or SIP/devicename/extension
- * or SIP/devicename/extension/IPorHost
- * or SIP/username@domain//IPorHost
- * and there is an optional [!dnid] argument you can append to alter the
- * To: header. And after that, a [![fromuser][@fromdomain]] argument.
- * Leave those blank to use the defaults.
- * \endverbatim
- */
-static struct ast_channel *sip_request_call(const char *type, struct ast_format_cap *cap, const struct ast_assigned_ids *assignedids, const struct ast_channel *requestor, const char *dest, int *cause)
-{
- struct sip_pvt *p;
- struct ast_channel *tmpc = NULL;
- char *ext = NULL, *host;
- char tmp[256];
- struct ast_str *codec_buf = ast_str_alloca(AST_FORMAT_CAP_NAMES_LEN);
- char *dnid;
- char *secret = NULL;
- char *md5secret = NULL;
- char *authname = NULL;
- char *trans = NULL;
- char dialstring[256];
- char *remote_address;
- enum ast_transport transport = 0;
- ast_callid callid;
- AST_DECLARE_APP_ARGS(args,
- AST_APP_ARG(peerorhost);
- AST_APP_ARG(exten);
- AST_APP_ARG(remote_address);
- );
-
- if (ast_format_cap_empty(cap)) {
- ast_log(LOG_NOTICE, "Asked to get a channel without offering any format\n");
- *cause = AST_CAUSE_BEARERCAPABILITY_NOTAVAIL; /* Can't find codec to connect to host */
- return NULL;
- }
- ast_debug(1, "Asked to create a SIP channel with formats: %s\n", ast_format_cap_get_names(cap, &codec_buf));
-
- if (ast_strlen_zero(dest)) {
- ast_log(LOG_ERROR, "Unable to create channel with empty destination.\n");
- *cause = AST_CAUSE_CHANNEL_UNACCEPTABLE;
- return NULL;
- }
-
- callid = ast_read_threadstorage_callid();
- if (!(p = sip_alloc(NULL, NULL, 0, SIP_INVITE, NULL, callid))) {
- ast_log(LOG_ERROR, "Unable to build sip pvt data for '%s' (Out of memory or socket error)\n", dest);
- *cause = AST_CAUSE_SWITCH_CONGESTION;
- return NULL;
- }
-
- p->outgoing_call = TRUE;
-
- snprintf(dialstring, sizeof(dialstring), "%s/%s", type, dest);
- ast_string_field_set(p, dialstring, dialstring);
-
- if (!(p->options = ast_calloc(1, sizeof(*p->options)))) {
- dialog_unlink_all(p);
- dialog_unref(p, "unref dialog p from mem fail");
- /* sip_destroy(p); */
- ast_log(LOG_ERROR, "Unable to build option SIP data structure - Out of memory\n");
- *cause = AST_CAUSE_SWITCH_CONGESTION;
- return NULL;
- }
-
- /* Save the destination, the SIP dial string */
- ast_copy_string(tmp, dest, sizeof(tmp));
-
- /* Find optional DNID (SIP to-uri) and From-CLI (SIP from-uri)
- * and strip it from the dial string:
- * [!touser[@todomain][![fromuser][@fromdomain]]]
- * For historical reasons, the touser@todomain is passed as dnid
- * while fromuser@fromdomain are split immediately. Passing a
- * todomain without touser will create an invalid SIP message. */
- dnid = strchr(tmp, '!');
- if (dnid != NULL) {
- char *fromuser_and_domain;
-
- *dnid++ = '\0';
- if ((fromuser_and_domain = strchr(dnid, '!'))) {
- char *forward_compat;
- char *fromdomain;
-
- *fromuser_and_domain++ = '\0';
-
- /* Cut it at a trailing NUL or trailing '!' for
- * forward compatibility with extra arguments
- * in the future. */
- if ((forward_compat = strchr(fromuser_and_domain, '!'))) {
- /* Ignore the rest.. */
- *forward_compat = '\0';
- }
-
- if ((fromdomain = strchr(fromuser_and_domain, '@'))) {
- *fromdomain++ = '\0';
- /* Set fromdomain. */
- if (!ast_strlen_zero(fromdomain)) {
- ast_string_field_set(p, fromdomain, fromdomain);
- }
- }
-
- /* Set fromuser. */
- if (!ast_strlen_zero(fromuser_and_domain)) {
- ast_string_field_set(p, fromuser, fromuser_and_domain);
- }
- }
-
- /* Set DNID (touser/todomain). */
- if (!ast_strlen_zero(dnid)) {
- ast_string_field_set(p, todnid, dnid);
- }
- }
-
- /* If stripping the DNID left us with nothing, bail out */
- if (ast_strlen_zero(tmp)) {
- dialog_unlink_all(p);
- dialog_unref(p, "unref dialog p from bad destination");
- *cause = AST_CAUSE_DESTINATION_OUT_OF_ORDER;
- return NULL;
- }
-
- /* Divvy up the items separated by slashes */
- AST_NONSTANDARD_APP_ARGS(args, tmp, '/');
-
- /* Find at sign - @ */
- host = strchr(args.peerorhost, '@');
- if (host) {
- *host++ = '\0';
- ext = args.peerorhost;
- secret = strchr(ext, ':');
- }
- if (secret) {
- *secret++ = '\0';
- md5secret = strchr(secret, ':');
- }
- if (md5secret) {
- *md5secret++ = '\0';
- authname = strchr(md5secret, ':');
- }
- if (authname) {
- *authname++ = '\0';
- trans = strchr(authname, ':');
- }
- if (trans) {
- *trans++ = '\0';
- if (!strcasecmp(trans, "tcp"))
- transport = AST_TRANSPORT_TCP;
- else if (!strcasecmp(trans, "tls"))
- transport = AST_TRANSPORT_TLS;
- else {
- if (strcasecmp(trans, "udp"))
- ast_log(LOG_WARNING, "'%s' is not a valid transport option to Dial() for SIP calls, using udp by default.\n", trans);
- transport = AST_TRANSPORT_UDP;
- }
- } else { /* use default */
- transport = AST_TRANSPORT_UDP;
- }
-
- if (!host) {
- ext = args.exten;
- host = args.peerorhost;
- remote_address = args.remote_address;
- } else {
- remote_address = args.remote_address;
- if (!ast_strlen_zero(args.exten)) {
- ast_log(LOG_NOTICE, "Conflicting extension values given. Using '%s' and not '%s'\n", ext, args.exten);
- }
- }
-
- if (!ast_strlen_zero(remote_address)) {
- p->options->outboundproxy = proxy_from_config(remote_address, 0, NULL);
- if (!p->options->outboundproxy) {
- ast_log(LOG_WARNING, "Unable to parse outboundproxy %s. We will not use this remote IP address\n", remote_address);
- }
- }
-
- set_socket_transport(&p->socket, transport);
-
- /* We now have
- host = peer name, DNS host name or DNS domain (for SRV)
- ext = extension (user part of URI)
- dnid = destination of the call (applies to the To: header)
- */
- if (create_addr(p, host, NULL, 1)) {
- *cause = AST_CAUSE_UNREGISTERED;
- ast_debug(3, "Cant create SIP call - target device not registered\n");
- dialog_unlink_all(p);
- dialog_unref(p, "unref dialog p UNREGISTERED");
- /* sip_destroy(p); */
- return NULL;
- }
- if (ast_strlen_zero(p->peername) && ext)
- ast_string_field_set(p, peername, ext);
- /* Recalculate our side, and recalculate Call ID */
- ast_sip_ouraddrfor(&p->sa, &p->ourip, p);
- /* When chan_sip is first loaded or reloaded, we need to check for NAT and set the appropiate flags
- now that we have the auto_* settings. */
- check_for_nat(&p->sa, p);
- /* If there is a peer related to this outgoing call and it hasn't re-registered after
- a reload, we need to set the peer's NAT flags accordingly. */
- if (p->relatedpeer) {
-
- if (!ast_strlen_zero(p->relatedpeer->fullcontact) && !p->natdetected &&
- ((ast_test_flag(&p->flags[2], SIP_PAGE3_NAT_AUTO_RPORT) && !ast_test_flag(&p->flags[0], SIP_NAT_FORCE_RPORT)) ||
- (ast_test_flag(&p->flags[2], SIP_PAGE3_NAT_AUTO_COMEDIA) && !ast_test_flag(&p->flags[1], SIP_PAGE2_SYMMETRICRTP)))) {
- /* We need to make an attempt to determine if a peer is behind NAT
- if the peer has the flags auto_force_rport or auto_comedia set. */
- struct ast_sockaddr tmpaddr;
-
- __set_address_from_contact(p->relatedpeer->fullcontact, &tmpaddr, 0);
-
- check_for_nat(&tmpaddr, p);
- }
-
- set_peer_nat(p, p->relatedpeer);
- }
-
- do_setnat(p);
-
- build_via(p);
-
- /* Change the dialog callid. */
- change_callid_pvt(p, NULL);
-
- /* We have an extension to call, don't use the full contact here */
- /* This to enable dialing registered peers with extension dialling,
- like SIP/peername/extension
- SIP/peername will still use the full contact
- */
- if (ext) {
- ast_string_field_set(p, username, ext);
- ast_string_field_set(p, fullcontact, NULL);
- }
- if (secret && !ast_strlen_zero(secret))
- ast_string_field_set(p, peersecret, secret);
-
- if (md5secret && !ast_strlen_zero(md5secret))
- ast_string_field_set(p, peermd5secret, md5secret);
-
- if (authname && !ast_strlen_zero(authname))
- ast_string_field_set(p, authname, authname);
-#if 0
- printf("Setting up to call extension '%s' at '%s'\n", ext ? ext : "", host);
-#endif
- ast_format_cap_append_from_cap(p->prefcaps, cap, AST_MEDIA_TYPE_UNKNOWN);
- ast_format_cap_get_compatible(cap, p->caps, p->jointcaps);
-
- sip_pvt_lock(p);
-
- tmpc = sip_new(p, AST_STATE_DOWN, host, assignedids, requestor, callid); /* Place the call */
-
- sip_pvt_unlock(p);
- if (!tmpc) {
- dialog_unlink_all(p);
- /* sip_destroy(p); */
- } else {
- ast_channel_unlock(tmpc);
- }
- dialog_unref(p, "toss pvt ptr at end of sip_request_call");
- ast_update_use_count();
- restart_monitor();
- return tmpc;
-}
-
-/*! \brief Parse insecure= setting in sip.conf and set flags according to setting */
-static void set_insecure_flags (struct ast_flags *flags, const char *value, int lineno)
-{
- if (ast_strlen_zero(value))
- return;
-
- if (!ast_false(value)) {
- char buf[64];
- char *word, *next;
-
- ast_copy_string(buf, value, sizeof(buf));
- next = buf;
- while ((word = strsep(&next, ","))) {
- if (!strcasecmp(word, "port"))
- ast_set_flag(&flags[0], SIP_INSECURE_PORT);
- else if (!strcasecmp(word, "invite"))
- ast_set_flag(&flags[0], SIP_INSECURE_INVITE);
- else
- ast_log(LOG_WARNING, "Unknown insecure mode '%s' on line %d\n", value, lineno);
- }
- }
-}
-
-/*!
- \brief Handle T.38 configuration options common to users and peers
- \return non-zero if any config options were handled, zero otherwise
-*/
-static int handle_t38_options(struct ast_flags *flags, struct ast_flags *mask, struct ast_variable *v,
- unsigned int *maxdatagram)
-{
- int res = 1;
-
- if (!strcasecmp(v->name, "t38pt_udptl")) {
- char *buf = ast_strdupa(v->value);
- char *word, *next = buf;
-
- ast_set_flag(&mask[1], SIP_PAGE2_T38SUPPORT);
-
- while ((word = strsep(&next, ","))) {
- if (ast_true(word) || !strcasecmp(word, "fec")) {
- ast_clear_flag(&flags[1], SIP_PAGE2_T38SUPPORT);
- ast_set_flag(&flags[1], SIP_PAGE2_T38SUPPORT_UDPTL_FEC);
- } else if (!strcasecmp(word, "redundancy")) {
- ast_clear_flag(&flags[1], SIP_PAGE2_T38SUPPORT);
- ast_set_flag(&flags[1], SIP_PAGE2_T38SUPPORT_UDPTL_REDUNDANCY);
- } else if (!strcasecmp(word, "none")) {
- ast_clear_flag(&flags[1], SIP_PAGE2_T38SUPPORT);
- ast_set_flag(&flags[1], SIP_PAGE2_T38SUPPORT_UDPTL);
- } else if (!strncasecmp(word, "maxdatagram=", 12)) {
- if (sscanf(&word[12], "%30u", maxdatagram) != 1) {
- ast_log(LOG_WARNING, "Invalid maxdatagram '%s' at line %d of %s\n", v->value, v->lineno, config);
- *maxdatagram = global_t38_maxdatagram;
- }
- }
- }
- } else if (!strcasecmp(v->name, "t38pt_usertpsource")) {
- ast_set_flag(&mask[1], SIP_PAGE2_UDPTL_DESTINATION);
- ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_UDPTL_DESTINATION);
- } else {
- res = 0;
- }
-
- return res;
-}
-
-/*!
- \brief Handle flag-type options common to configuration of devices - peers
- \param flags array of three struct ast_flags
- \param mask array of three struct ast_flags
- \param v linked list of config variables to process
- \return non-zero if any config options were handled, zero otherwise
-*/
-static int handle_common_options(struct ast_flags *flags, struct ast_flags *mask, struct ast_variable *v)
-{
- int res = 1;
-
- if (!strcasecmp(v->name, "trustrpid")) {
- ast_set_flag(&mask[0], SIP_TRUSTRPID);
- ast_set2_flag(&flags[0], ast_true(v->value), SIP_TRUSTRPID);
- } else if (!strcasecmp(v->name, "supportpath")) {
- ast_set_flag(&mask[0], SIP_USEPATH);
- ast_set2_flag(&flags[0], ast_true(v->value), SIP_USEPATH);
- } else if (!strcasecmp(v->name, "sendrpid")) {
- ast_set_flag(&mask[0], SIP_SENDRPID);
- if (!strcasecmp(v->value, "pai")) {
- ast_set_flag(&flags[0], SIP_SENDRPID_PAI);
- } else if (!strcasecmp(v->value, "rpid")) {
- ast_set_flag(&flags[0], SIP_SENDRPID_RPID);
- } else if (ast_true(v->value)) {
- ast_set_flag(&flags[0], SIP_SENDRPID_RPID);
- }
- } else if (!strcasecmp(v->name, "rpid_update")) {
- ast_set_flag(&mask[1], SIP_PAGE2_RPID_UPDATE);
- ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_RPID_UPDATE);
- } else if (!strcasecmp(v->name, "rpid_immediate")) {
- ast_set_flag(&mask[1], SIP_PAGE2_RPID_IMMEDIATE);
- ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_RPID_IMMEDIATE);
- } else if (!strcasecmp(v->name, "trust_id_outbound")) {
- ast_set_flag(&mask[1], SIP_PAGE2_TRUST_ID_OUTBOUND);
- ast_clear_flag(&flags[1], SIP_PAGE2_TRUST_ID_OUTBOUND);
- if (!strcasecmp(v->value, "legacy")) {
- ast_set_flag(&flags[1], SIP_PAGE2_TRUST_ID_OUTBOUND_LEGACY);
- } else if (ast_true(v->value)) {
- ast_set_flag(&flags[1], SIP_PAGE2_TRUST_ID_OUTBOUND_YES);
- } else if (ast_false(v->value)) {
- ast_set_flag(&flags[1], SIP_PAGE2_TRUST_ID_OUTBOUND_NO);
- } else {
- ast_log(LOG_WARNING, "Unknown trust_id_outbound mode '%s' on line %d, using legacy\n", v->value, v->lineno);
- ast_set_flag(&flags[1], SIP_PAGE2_TRUST_ID_OUTBOUND_LEGACY);
- }
- } else if (!strcasecmp(v->name, "g726nonstandard")) {
- ast_set_flag(&mask[0], SIP_G726_NONSTANDARD);
- ast_set2_flag(&flags[0], ast_true(v->value), SIP_G726_NONSTANDARD);
- } else if (!strcasecmp(v->name, "useclientcode")) {
- ast_set_flag(&mask[0], SIP_USECLIENTCODE);
- ast_set2_flag(&flags[0], ast_true(v->value), SIP_USECLIENTCODE);
- } else if (!strcasecmp(v->name, "dtmfmode")) {
- ast_set_flag(&mask[0], SIP_DTMF);
- ast_clear_flag(&flags[0], SIP_DTMF);
- if (!strcasecmp(v->value, "inband"))
- ast_set_flag(&flags[0], SIP_DTMF_INBAND);
- else if (!strcasecmp(v->value, "rfc2833"))
- ast_set_flag(&flags[0], SIP_DTMF_RFC2833);
- else if (!strcasecmp(v->value, "info"))
- ast_set_flag(&flags[0], SIP_DTMF_INFO);
- else if (!strcasecmp(v->value, "shortinfo"))
- ast_set_flag(&flags[0], SIP_DTMF_SHORTINFO);
- else if (!strcasecmp(v->value, "auto"))
- ast_set_flag(&flags[0], SIP_DTMF_AUTO);
- else {
- ast_log(LOG_WARNING, "Unknown dtmf mode '%s' on line %d, using rfc2833\n", v->value, v->lineno);
- ast_set_flag(&flags[0], SIP_DTMF_RFC2833);
- }
- } else if (!strcasecmp(v->name, "nat")) {
- sip_parse_nat_option(v->value, mask, flags);
- } else if (!strcasecmp(v->name, "directmedia") || !strcasecmp(v->name, "canreinvite")) {
- ast_set_flag(&mask[0], SIP_REINVITE);
- ast_clear_flag(&flags[0], SIP_REINVITE);
- if (ast_true(v->value)) {
- ast_set_flag(&flags[0], SIP_DIRECT_MEDIA | SIP_DIRECT_MEDIA_NAT);
- } else if (!ast_false(v->value)) {
- char buf[64];
- char *word, *next = buf;
-
- ast_copy_string(buf, v->value, sizeof(buf));
- while ((word = strsep(&next, ","))) {
- if (!strcasecmp(word, "update")) {
- ast_set_flag(&flags[0], SIP_REINVITE_UPDATE | SIP_DIRECT_MEDIA);
- } else if (!strcasecmp(word, "nonat")) {
- ast_set_flag(&flags[0], SIP_DIRECT_MEDIA);
- ast_clear_flag(&flags[0], SIP_DIRECT_MEDIA_NAT);
- } else if (!strcasecmp(word, "outgoing")) {
- ast_set_flag(&flags[0], SIP_DIRECT_MEDIA);
- ast_set_flag(&mask[2], SIP_PAGE3_DIRECT_MEDIA_OUTGOING);
- ast_set_flag(&flags[2], SIP_PAGE3_DIRECT_MEDIA_OUTGOING);
- } else {
- ast_log(LOG_WARNING, "Unknown directmedia mode '%s' on line %d\n", v->value, v->lineno);
- }
- }
- }
- } else if (!strcasecmp(v->name, "insecure")) {
- ast_set_flag(&mask[0], SIP_INSECURE);
- ast_clear_flag(&flags[0], SIP_INSECURE);
- set_insecure_flags(&flags[0], v->value, v->lineno);
- } else if (!strcasecmp(v->name, "progressinband")) {
- ast_set_flag(&mask[0], SIP_PROG_INBAND);
- ast_clear_flag(&flags[0], SIP_PROG_INBAND);
- if (ast_true(v->value))
- ast_set_flag(&flags[0], SIP_PROG_INBAND_YES);
- else if (!strcasecmp(v->value, "never"))
- ast_set_flag(&flags[0], SIP_PROG_INBAND_NEVER);
- } else if (!strcasecmp(v->name, "promiscredir")) {
- ast_set_flag(&mask[0], SIP_PROMISCREDIR);
- ast_set2_flag(&flags[0], ast_true(v->value), SIP_PROMISCREDIR);
- } else if (!strcasecmp(v->name, "videosupport")) {
- if (!strcasecmp(v->value, "always")) {
- ast_set_flag(&mask[1], SIP_PAGE2_VIDEOSUPPORT_ALWAYS);
- ast_set_flag(&flags[1], SIP_PAGE2_VIDEOSUPPORT_ALWAYS);
- } else {
- ast_set_flag(&mask[1], SIP_PAGE2_VIDEOSUPPORT);
- ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_VIDEOSUPPORT);
- }
- } else if (!strcasecmp(v->name, "textsupport")) {
- ast_set_flag(&mask[1], SIP_PAGE2_TEXTSUPPORT);
- ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_TEXTSUPPORT);
- res = 1;
- } else if (!strcasecmp(v->name, "allowoverlap")) {
- ast_set_flag(&mask[1], SIP_PAGE2_ALLOWOVERLAP);
- ast_clear_flag(&flags[1], SIP_PAGE2_ALLOWOVERLAP);
- if (ast_true(v->value)) {
- ast_set_flag(&flags[1], SIP_PAGE2_ALLOWOVERLAP_YES);
- } else if (!strcasecmp(v->value, "dtmf")){
- ast_set_flag(&flags[1], SIP_PAGE2_ALLOWOVERLAP_DTMF);
- }
- } else if (!strcasecmp(v->name, "allowsubscribe")) {
- ast_set_flag(&mask[1], SIP_PAGE2_ALLOWSUBSCRIBE);
- ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_ALLOWSUBSCRIBE);
- } else if (!strcasecmp(v->name, "ignoresdpversion")) {
- ast_set_flag(&mask[1], SIP_PAGE2_IGNORESDPVERSION);
- ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_IGNORESDPVERSION);
- } else if (!strcasecmp(v->name, "faxdetect")) {
- ast_set_flag(&mask[1], SIP_PAGE2_FAX_DETECT);
- if (ast_true(v->value)) {
- ast_set_flag(&flags[1], SIP_PAGE2_FAX_DETECT_BOTH);
- } else if (ast_false(v->value)) {
- ast_clear_flag(&flags[1], SIP_PAGE2_FAX_DETECT_BOTH);
- } else {
- char *buf = ast_strdupa(v->value);
- char *word, *next = buf;
-
- while ((word = strsep(&next, ","))) {
- if (!strcasecmp(word, "cng")) {
- ast_set_flag(&flags[1], SIP_PAGE2_FAX_DETECT_CNG);
- } else if (!strcasecmp(word, "t38")) {
- ast_set_flag(&flags[1], SIP_PAGE2_FAX_DETECT_T38);
- } else {
- ast_log(LOG_WARNING, "Unknown faxdetect mode '%s' on line %d.\n", word, v->lineno);
- }
- }
- }
- } else if (!strcasecmp(v->name, "rfc2833compensate")) {
- ast_set_flag(&mask[1], SIP_PAGE2_RFC2833_COMPENSATE);
- ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_RFC2833_COMPENSATE);
- } else if (!strcasecmp(v->name, "buggymwi")) {
- ast_set_flag(&mask[1], SIP_PAGE2_BUGGY_MWI);
- ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_BUGGY_MWI);
- } else if (!strcasecmp(v->name, "rtcp_mux")) {
- ast_set_flag(&mask[2], SIP_PAGE3_RTCP_MUX);
- ast_set2_flag(&flags[2], ast_true(v->value), SIP_PAGE3_RTCP_MUX);
- } else
- res = 0;
-
- return res;
-}
-
-/*! \brief Add SIP domain to list of domains we are responsible for */
-static int add_sip_domain(const char *domain, const enum domain_mode mode, const char *context)
-{
- struct domain *d;
-
- if (ast_strlen_zero(domain)) {
- ast_log(LOG_WARNING, "Zero length domain.\n");
- return 1;
- }
-
- if (!(d = ast_calloc(1, sizeof(*d))))
- return 0;
-
- ast_copy_string(d->domain, domain, sizeof(d->domain));
-
- if (!ast_strlen_zero(context))
- ast_copy_string(d->context, context, sizeof(d->context));
-
- d->mode = mode;
-
- AST_LIST_LOCK(&domain_list);
- AST_LIST_INSERT_TAIL(&domain_list, d, list);
- AST_LIST_UNLOCK(&domain_list);
-
- if (sipdebug)
- ast_debug(1, "Added local SIP domain '%s'\n", domain);
-
- return 1;
-}
-
-/*! \brief check_sip_domain: Check if domain part of uri is local to our server */
-static int check_sip_domain(const char *domain, char *context, size_t len)
-{
- struct domain *d;
- int result = 0;
-
- AST_LIST_LOCK(&domain_list);
- AST_LIST_TRAVERSE(&domain_list, d, list) {
- if (strcasecmp(d->domain, domain)) {
- continue;
- }
-
- if (len && !ast_strlen_zero(d->context))
- ast_copy_string(context, d->context, len);
-
- result = 1;
- break;
- }
- AST_LIST_UNLOCK(&domain_list);
-
- return result;
-}
-
-/*! \brief Clear our domain list (at reload) */
-static void clear_sip_domains(void)
-{
- struct domain *d;
-
- AST_LIST_LOCK(&domain_list);
- while ((d = AST_LIST_REMOVE_HEAD(&domain_list, list)))
- ast_free(d);
- AST_LIST_UNLOCK(&domain_list);
-}
-
-/*!
- * \internal
- * \brief Realm authentication container destructor.
- *
- * \param obj Container object to destroy.
- */
-static void destroy_realm_authentication(void *obj)
-{
- struct sip_auth_container *credentials = obj;
- struct sip_auth *auth;
-
- while ((auth = AST_LIST_REMOVE_HEAD(&credentials->list, node))) {
- ast_free(auth);
- }
-}
-
-/*!
- * \internal
- * \brief Add realm authentication to credentials.
- *
- * \param credentials Realm authentication container to create/add authentication credentials.
- * \param configuration Credential configuration value.
- * \param lineno Line number in config file.
- */
-static void add_realm_authentication(struct sip_auth_container **credentials, const char *configuration, int lineno)
-{
- char *authcopy;
- char *username=NULL, *realm=NULL, *secret=NULL, *md5secret=NULL;
- struct sip_auth *auth;
-
- if (ast_strlen_zero(configuration)) {
- /* Nothing to add */
- return;
- }
-
- ast_debug(1, "Auth config :: %s\n", configuration);
-
- authcopy = ast_strdupa(configuration);
- username = authcopy;
-
- /* split user[:secret] and relm */
- realm = strrchr(username, '@');
- if (realm)
- *realm++ = '\0';
- if (ast_strlen_zero(username) || ast_strlen_zero(realm)) {
- ast_log(LOG_WARNING, "Format for authentication entry is user[:secret]@realm at line %d\n", lineno);
- return;
- }
-
- /* parse username at ':' for secret, or '#" for md5secret */
- if ((secret = strchr(username, ':'))) {
- *secret++ = '\0';
- } else if ((md5secret = strchr(username, '#'))) {
- *md5secret++ = '\0';
- }
-
- /* Create the continer if needed. */
- if (!*credentials) {
- *credentials = ao2_t_alloc(sizeof(**credentials), destroy_realm_authentication,
- "Create realm auth container.");
- if (!*credentials) {
- /* Failed to create the credentials container. */
- return;
- }
- }
-
- /* Create the authentication credential entry. */
- auth = ast_calloc(1, sizeof(*auth));
- if (!auth) {
- return;
- }
- ast_copy_string(auth->realm, realm, sizeof(auth->realm));
- ast_copy_string(auth->username, username, sizeof(auth->username));
- if (secret)
- ast_copy_string(auth->secret, secret, sizeof(auth->secret));
- if (md5secret)
- ast_copy_string(auth->md5secret, md5secret, sizeof(auth->md5secret));
-
- /* Add credential to container list. */
- AST_LIST_INSERT_TAIL(&(*credentials)->list, auth, node);
-
- ast_verb(3, "Added authentication for realm %s\n", realm);
-}
-
-/*!
- * \internal
- * \brief Find authentication for a specific realm.
- *
- * \param credentials Realm authentication container to search.
- * \param realm Authentication realm to find.
- *
- * \return Found authentication credential or NULL.
- */
-static struct sip_auth *find_realm_authentication(struct sip_auth_container *credentials, const char *realm)
-{
- struct sip_auth *auth;
-
- if (credentials) {
- AST_LIST_TRAVERSE(&credentials->list, auth, node) {
- if (!strcasecmp(auth->realm, realm)) {
- break;
- }
- }
- } else {
- auth = NULL;
- }
-
- return auth;
-}
-
-/*! \brief
- * implement the setvar config line
- */
-static struct ast_variable *add_var(const char *buf, struct ast_variable *list)
-{
- struct ast_variable *tmpvar = NULL;
- char *varname = ast_strdupa(buf), *varval = NULL;
-
- if ((varval = strchr(varname, '='))) {
- *varval++ = '\0';
- if ((tmpvar = ast_variable_new(varname, varval, ""))) {
- if (ast_variable_list_replace(&list, tmpvar)) {
- tmpvar->next = list;
- list = tmpvar;
- }
- }
- }
- return list;
-}
-
-/*! \brief Set peer defaults before configuring specific configurations */
-static void set_peer_defaults(struct sip_peer *peer)
-{
- if (peer->expire < 0) {
- /* Don't reset expire or port time during reload
- if we have an active registration
- */
- peer_sched_cleanup(peer);
- set_socket_transport(&peer->socket, AST_TRANSPORT_UDP);
- }
- peer->type = SIP_TYPE_PEER;
- ast_copy_flags(&peer->flags[0], &global_flags[0], SIP_FLAGS_TO_COPY);
- ast_copy_flags(&peer->flags[1], &global_flags[1], SIP_PAGE2_FLAGS_TO_COPY);
- ast_copy_flags(&peer->flags[2], &global_flags[2], SIP_PAGE3_FLAGS_TO_COPY);
- ast_string_field_set(peer, context, sip_cfg.default_context);
- ast_string_field_set(peer, record_on_feature, sip_cfg.default_record_on_feature);
- ast_string_field_set(peer, record_off_feature, sip_cfg.default_record_off_feature);
- ast_string_field_set(peer, messagecontext, sip_cfg.messagecontext);
- ast_string_field_set(peer, subscribecontext, sip_cfg.default_subscribecontext);
- ast_string_field_set(peer, language, default_language);
- ast_string_field_set(peer, mohinterpret, default_mohinterpret);
- ast_string_field_set(peer, mohsuggest, default_mohsuggest);
- ast_string_field_set(peer, engine, default_engine);
- ast_sockaddr_setnull(&peer->addr);
- ast_sockaddr_setnull(&peer->defaddr);
- ast_format_cap_append_from_cap(peer->caps, sip_cfg.caps, AST_MEDIA_TYPE_UNKNOWN);
- peer->maxcallbitrate = default_maxcallbitrate;
- peer->rtptimeout = global_rtptimeout;
- peer->rtpholdtimeout = global_rtpholdtimeout;
- peer->rtpkeepalive = global_rtpkeepalive;
- peer->allowtransfer = sip_cfg.allowtransfer;
- peer->autoframing = global_autoframing;
- peer->t38_maxdatagram = global_t38_maxdatagram;
- peer->qualifyfreq = global_qualifyfreq;
- if (global_callcounter)
- peer->call_limit=INT_MAX;
- ast_string_field_set(peer, vmexten, default_vmexten);
- ast_string_field_set(peer, secret, "");
- ast_string_field_set(peer, description, "");
- ast_string_field_set(peer, remotesecret, "");
- ast_string_field_set(peer, md5secret, "");
- ast_string_field_set(peer, cid_num, "");
- ast_string_field_set(peer, cid_name, "");
- ast_string_field_set(peer, cid_tag, "");
- ast_string_field_set(peer, fromdomain, "");
- ast_string_field_set(peer, fromuser, "");
- ast_string_field_set(peer, regexten, "");
- peer->callgroup = 0;
- peer->pickupgroup = 0;
- peer->maxms = default_qualify;
- peer->keepalive = default_keepalive;
- ast_string_field_set(peer, zone, default_zone);
- peer->stimer.st_mode_oper = global_st_mode; /* Session-Timers */
- peer->stimer.st_ref = global_st_refresher;
- peer->stimer.st_min_se = global_min_se;
- peer->stimer.st_max_se = global_max_se;
- peer->timer_t1 = global_t1;
- peer->timer_b = global_timer_b;
- clear_peer_mailboxes(peer);
- peer->disallowed_methods = sip_cfg.disallowed_methods;
- peer->transports = default_transports;
- peer->default_outbound_transport = default_primary_transport;
- if (peer->outboundproxy) {
- ao2_ref(peer->outboundproxy, -1);
- peer->outboundproxy = NULL;
- }
-}
-
-/*! \brief Create temporary peer (used in autocreatepeer mode) */
-static struct sip_peer *temp_peer(const char *name)
-{
- struct sip_peer *peer;
-
- if (!(peer = ao2_t_alloc(sizeof(*peer), sip_destroy_peer_fn, "allocate a peer struct")))
- return NULL;
-
- if (ast_string_field_init(peer, 512)) {
- ao2_t_ref(peer, -1, "failed to string_field_init, drop peer");
- return NULL;
- }
-
- if (!(peer->cc_params = ast_cc_config_params_init())) {
- ao2_t_ref(peer, -1, "failed to allocate cc_params for peer");
- return NULL;
- }
-
- if (!(peer->caps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT))) {
- ao2_t_ref(peer, -1, "failed to allocate format capabilities, drop peer");
- return NULL;
- }
-
- ast_atomic_fetchadd_int(&apeerobjs, 1);
- peer->expire = -1;
- peer->pokeexpire = -1;
- peer->keepalivesend = -1;
-
- set_peer_defaults(peer);
-
- ast_copy_string(peer->name, name, sizeof(peer->name));
-
- peer->selfdestruct = TRUE;
- peer->host_dynamic = TRUE;
- reg_source_db(peer);
-
- return peer;
-}
-
-/*! \todo document this function */
-static void add_peer_mailboxes(struct sip_peer *peer, const char *value)
-{
- char *next;
- char *mbox;
-
- next = ast_strdupa(value);
-
- while ((mbox = strsep(&next, ","))) {
- struct sip_mailbox *mailbox;
- int duplicate = 0;
-
- /* remove leading/trailing whitespace from mailbox string */
- mbox = ast_strip(mbox);
- if (ast_strlen_zero(mbox)) {
- continue;
- }
-
- /* Check whether the mailbox is already in the list */
- AST_LIST_TRAVERSE(&peer->mailboxes, mailbox, entry) {
- if (!strcmp(mailbox->id, mbox)) {
- duplicate = 1;
- mailbox->status = SIP_MAILBOX_STATUS_EXISTING;
- break;
- }
- }
- if (duplicate) {
- continue;
- }
-
- mailbox = ast_calloc(1, sizeof(*mailbox) + strlen(mbox));
- if (!mailbox) {
- continue;
- }
- strcpy(mailbox->id, mbox); /* SAFE */
- mailbox->status = SIP_MAILBOX_STATUS_NEW;
- mailbox->peer = peer;
-
- AST_LIST_INSERT_TAIL(&peer->mailboxes, mailbox, entry);
- }
-}
-
-/*! \brief Build peer from configuration (file or realtime static/dynamic) */
-static struct sip_peer *build_peer(const char *name, struct ast_variable *v_head, struct ast_variable *alt, int realtime, int devstate_only)
-{
- /* We preserve the original value of v_head to make analyzing backtraces easier */
- struct ast_variable *v = v_head;
- struct sip_peer *peer = NULL;
- struct ast_acl_list *oldacl = NULL;
- struct ast_acl_list *oldcontactacl = NULL;
- struct ast_acl_list *olddirectmediaacl = NULL;
- int found = 0;
- int firstpass = 1;
- uint16_t port = 0;
- int format = 0; /* Ama flags */
- int timerb_set = 0, timert1_set = 0;
- time_t regseconds = 0;
- struct ast_flags peerflags[3] = {{(0)}};
- struct ast_flags mask[3] = {{(0)}};
- struct sip_peer tmp_peer;
- const char *srvlookup = NULL;
- static int deprecation_warning = 1;
- int alt_fullcontact = alt ? 1 : 0, headercount = 0;
- struct ast_str *fullcontact = ast_str_alloca(512);
- int acl_change_subscription_needed = 0;
-
- if (!realtime || ast_test_flag(&global_flags[1], SIP_PAGE2_RTCACHEFRIENDS)) {
- /* Note we do NOT use sip_find_peer here, to avoid realtime recursion */
- /* We also use a case-sensitive comparison (unlike sip_find_peer) so
- that case changes made to the peer name will be properly handled
- during reload
- */
- ast_copy_string(tmp_peer.name, name, sizeof(tmp_peer.name));
- peer = ao2_t_find(peers, &tmp_peer, OBJ_POINTER | OBJ_UNLINK, "find and unlink peer from peers table");
- }
-
- if (peer) {
- /* Already in the list, remove it and it will be added back (or FREE'd) */
- found++;
- /* we've unlinked the peer from the peers container but not unlinked from the peers_by_ip container yet
- this leads to a wrong refcounter and the peer object is never destroyed */
- if (!ast_sockaddr_isnull(&peer->addr)) {
- ao2_t_unlink(peers_by_ip, peer, "ao2_unlink peer from peers_by_ip table");
- }
- if (!(peer->the_mark)) {
- firstpass = 0;
- } else {
- ast_format_cap_remove_by_type(peer->caps, AST_MEDIA_TYPE_UNKNOWN);
- }
- } else {
- if (!(peer = ao2_t_alloc(sizeof(*peer), sip_destroy_peer_fn, "allocate a peer struct"))) {
- return NULL;
- }
- if (!(peer->endpoint = ast_endpoint_create("SIP", name))) {
- ao2_t_ref(peer, -1, "failed to allocate endpoint, drop peer");
- return NULL;
- }
- if (!(peer->caps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT))) {
- ao2_t_ref(peer, -1, "failed to allocate format capabilities, drop peer");
- return NULL;
- }
- if (ast_string_field_init(peer, 512)) {
- ao2_t_ref(peer, -1, "failed to string_field_init, drop peer");
- return NULL;
- }
-
- if (!(peer->cc_params = ast_cc_config_params_init())) {
- ao2_t_ref(peer, -1, "failed to allocate cc_params for peer");
- return NULL;
- }
-
-
- if (realtime && !ast_test_flag(&global_flags[1], SIP_PAGE2_RTCACHEFRIENDS)) {
- ast_atomic_fetchadd_int(&rpeerobjs, 1);
- ast_debug(3, "-REALTIME- peer built. Name: %s. Peer objects: %d\n", name, rpeerobjs);
- } else
- ast_atomic_fetchadd_int(&speerobjs, 1);
-
- peer->expire = -1;
- peer->pokeexpire = -1;
- peer->keepalivesend = -1;
- }
-
- /* Note that our peer HAS had its reference count increased */
- if (firstpass) {
- oldacl = peer->acl;
- peer->acl = NULL;
- oldcontactacl = peer->contactacl;
- peer->contactacl = NULL;
- olddirectmediaacl = peer->directmediaacl;
- peer->directmediaacl = NULL;
- set_peer_defaults(peer); /* Set peer defaults */
- peer->type = 0;
- }
-
- /* in case the case of the peer name has changed, update the name */
- ast_copy_string(peer->name, name, sizeof(peer->name));
-
- /* If we have channel variables, remove them (reload) */
- if (peer->chanvars) {
- ast_variables_destroy(peer->chanvars);
- peer->chanvars = NULL;
- /* XXX should unregister ? */
- }
-
- if (found)
- peer->portinuri = 0;
-
- /* If we have realm authentication information, remove them (reload) */
- ao2_lock(peer);
- if (peer->auth) {
- ao2_t_ref(peer->auth, -1, "Removing old peer authentication");
- peer->auth = NULL;
- }
- ao2_unlock(peer);
-
- /* clear the transport information. We will detect if a default value is required after parsing the config */
- peer->default_outbound_transport = 0;
- peer->transports = 0;
-
- if (!devstate_only) {
- struct sip_mailbox *mailbox;
- AST_LIST_TRAVERSE(&peer->mailboxes, mailbox, entry) {
- mailbox->status = SIP_MAILBOX_STATUS_UNKNOWN;
- }
- }
-
- /* clear named callgroup and named pickup group container */
- peer->named_callgroups = ast_unref_namedgroups(peer->named_callgroups);
- peer->named_pickupgroups = ast_unref_namedgroups(peer->named_pickupgroups);
-
- /* Set the default DTLS settings from default_tls_cfg */
- ast_rtp_dtls_cfg_free(&peer->dtls_cfg);
- ast_rtp_dtls_cfg_copy(&default_dtls_cfg, &peer->dtls_cfg);
- peer->dtls_cfg.enabled = FALSE;
-
- for (; v || ((v = alt) && !(alt=NULL)); v = v->next) {
- if (!devstate_only) {
- if (handle_common_options(&peerflags[0], &mask[0], v)) {
- continue;
- }
- if (handle_t38_options(&peerflags[0], &mask[0], v, &peer->t38_maxdatagram)) {
- continue;
- }
- if (!strcasecmp(v->name, "transport")) {
- char *val = ast_strdupa(v->value);
- char *trans;
-
- peer->transports = peer->default_outbound_transport = 0;
- while ((trans = strsep(&val, ","))) {
- trans = ast_skip_blanks(trans);
-
- if (!strncasecmp(trans, "udp", 3)) {
- peer->transports |= AST_TRANSPORT_UDP;
- } else if (!strncasecmp(trans, "wss", 3)) {
- peer->transports |= AST_TRANSPORT_WSS;
- } else if (!strncasecmp(trans, "ws", 2)) {
- peer->transports |= AST_TRANSPORT_WS;
- } else if (sip_cfg.tcp_enabled && !strncasecmp(trans, "tcp", 3)) {
- peer->transports |= AST_TRANSPORT_TCP;
- } else if (default_tls_cfg.enabled && !strncasecmp(trans, "tls", 3)) {
- peer->transports |= AST_TRANSPORT_TLS;
- } else if (!strncasecmp(trans, "tcp", 3) || !strncasecmp(trans, "tls", 3)) {
- ast_log(LOG_WARNING, "'%.3s' is not a valid transport type when %.3senable=no. If no other is specified, the defaults from general will be used.\n", trans, trans);
- } else {
- ast_log(LOG_NOTICE, "'%s' is not a valid transport type. if no other is specified, the defaults from general will be used.\n", trans);
- }
-
- if (!peer->default_outbound_transport) { /*!< The first transport listed should be default outbound */
- peer->default_outbound_transport = peer->transports;
- }
- }
- } else if (realtime && !strcasecmp(v->name, "regseconds")) {
- ast_get_time_t(v->value, ®seconds, 0, NULL);
- } else if (realtime && !strcasecmp(v->name, "name")) {
- ast_copy_string(peer->name, v->value, sizeof(peer->name));
- } else if (realtime && !strcasecmp(v->name, "useragent")) {
- ast_string_field_set(peer, useragent, v->value);
- } else if (!strcasecmp(v->name, "type")) {
- if (!strcasecmp(v->value, "peer")) {
- peer->type |= SIP_TYPE_PEER;
- } else if (!strcasecmp(v->value, "user")) {
- peer->type |= SIP_TYPE_USER;
- } else if (!strcasecmp(v->value, "friend")) {
- peer->type = SIP_TYPE_USER | SIP_TYPE_PEER;
- }
- } else if (!strcasecmp(v->name, "remotesecret")) {
- ast_string_field_set(peer, remotesecret, v->value);
- } else if (!strcasecmp(v->name, "secret")) {
- ast_string_field_set(peer, secret, v->value);
- } else if (!strcasecmp(v->name, "description")) {
- ast_string_field_set(peer, description, v->value);
- } else if (!strcasecmp(v->name, "md5secret")) {
- ast_string_field_set(peer, md5secret, v->value);
- } else if (!strcasecmp(v->name, "auth")) {
- add_realm_authentication(&peer->auth, v->value, v->lineno);
- } else if (!strcasecmp(v->name, "callerid")) {
- char cid_name[80] = { '\0' }, cid_num[80] = { '\0' };
-
- ast_callerid_split(v->value, cid_name, sizeof(cid_name), cid_num, sizeof(cid_num));
- ast_string_field_set(peer, cid_name, cid_name);
- ast_string_field_set(peer, cid_num, cid_num);
- } else if (!strcasecmp(v->name, "mwi_from")) {
- ast_string_field_set(peer, mwi_from, v->value);
- } else if (!strcasecmp(v->name, "fullname")) {
- ast_string_field_set(peer, cid_name, v->value);
- } else if (!strcasecmp(v->name, "trunkname")) {
- /* This is actually for a trunk, so we don't want to override callerid */
- ast_string_field_set(peer, cid_name, "");
- } else if (!strcasecmp(v->name, "cid_number")) {
- ast_string_field_set(peer, cid_num, v->value);
- } else if (!strcasecmp(v->name, "cid_tag")) {
- ast_string_field_set(peer, cid_tag, v->value);
- } else if (!strcasecmp(v->name, "context")) {
- ast_string_field_set(peer, context, v->value);
- ast_set_flag(&peer->flags[1], SIP_PAGE2_HAVEPEERCONTEXT);
- } else if (!strcasecmp(v->name, "recordonfeature")) {
- ast_string_field_set(peer, record_on_feature, v->value);
- } else if (!strcasecmp(v->name, "recordofffeature")) {
- ast_string_field_set(peer, record_off_feature, v->value);
- } else if (!strcasecmp(v->name, "outofcall_message_context")) {
- ast_string_field_set(peer, messagecontext, v->value);
- } else if (!strcasecmp(v->name, "subscribecontext")) {
- ast_string_field_set(peer, subscribecontext, v->value);
- } else if (!strcasecmp(v->name, "fromdomain")) {
- char *fromdomainport;
- ast_string_field_set(peer, fromdomain, v->value);
- if ((fromdomainport = strchr(peer->fromdomain, ':'))) {
- *fromdomainport++ = '\0';
- if (!(peer->fromdomainport = port_str2int(fromdomainport, 0))) {
- ast_log(LOG_NOTICE, "'%s' is not a valid port number for fromdomain.\n",fromdomainport);
- }
- } else {
- peer->fromdomainport = STANDARD_SIP_PORT;
- }
- } else if (!strcasecmp(v->name, "usereqphone")) {
- ast_set2_flag(&peer->flags[0], ast_true(v->value), SIP_USEREQPHONE);
- } else if (!strcasecmp(v->name, "fromuser")) {
- ast_string_field_set(peer, fromuser, v->value);
- } else if (!strcasecmp(v->name, "outboundproxy")) {
- struct sip_proxy *proxy;
- if (ast_strlen_zero(v->value)) {
- ast_log(LOG_WARNING, "no value given for outbound proxy on line %d of sip.conf\n", v->lineno);
- continue;
- }
- proxy = proxy_from_config(v->value, v->lineno, peer->outboundproxy);
- if (!proxy) {
- ast_log(LOG_WARNING, "failure parsing the outbound proxy on line %d of sip.conf.\n", v->lineno);
- continue;
- }
- peer->outboundproxy = proxy;
- } else if (!strcasecmp(v->name, "host")) {
- if (!strcasecmp(v->value, "dynamic")) {
- /* They'll register with us */
- if ((!found && !ast_test_flag(&global_flags[1], SIP_PAGE2_RTCACHEFRIENDS)) || !peer->host_dynamic) {
- /* Initialize stuff if this is a new peer, or if it used to
- * not be dynamic before the reload. */
- ast_string_field_set(peer, tohost, NULL);
- ast_sockaddr_setnull(&peer->addr);
- }
- peer->host_dynamic = TRUE;
- } else {
- /* Non-dynamic. Make sure we become that way if we're not */
- AST_SCHED_DEL_UNREF(sched, peer->expire,
- sip_unref_peer(peer, "removing register expire ref"));
- peer->host_dynamic = FALSE;
- srvlookup = v->value;
- }
- } else if (!strcasecmp(v->name, "defaultip")) {
- peer->defaddr.ss.ss_family = AST_AF_UNSPEC;
- if (!ast_strlen_zero(v->value) && ast_get_ip(&peer->defaddr, v->value)) {
- sip_unref_peer(peer, "sip_unref_peer: from build_peer defaultip");
- return NULL;
- }
- } else if (!strcasecmp(v->name, "permit") || !strcasecmp(v->name, "deny") || !strcasecmp(v->name, "acl")) {
- int ha_error = 0;
- if (!ast_strlen_zero(v->value)) {
- ast_append_acl(v->name, v->value, &peer->acl, &ha_error, &acl_change_subscription_needed);
- }
- if (ha_error) {
- ast_log(LOG_ERROR, "Bad ACL entry in configuration line %d : %s. Deleting peer\n", v->lineno, v->value);
- sip_unref_peer(peer, "Removing peer due to bad ACL configuration");
- return NULL;
- }
- } else if (!strcasecmp(v->name, "contactpermit") || !strcasecmp(v->name, "contactdeny") || !strcasecmp(v->name, "contactacl")) {
- int ha_error = 0;
- if (!ast_strlen_zero(v->value)) {
- ast_append_acl(v->name + 7, v->value, &peer->contactacl, &ha_error, &acl_change_subscription_needed);
- }
- if (ha_error) {
- ast_log(LOG_ERROR, "Bad ACL entry in configuration line %d : %s. Deleting peer\n", v->lineno, v->value);
- sip_unref_peer(peer, "Removing peer due to bad contact ACL configuration");
- return NULL;
- }
- } else if (!strcasecmp(v->name, "directmediapermit") || !strcasecmp(v->name, "directmediadeny") || !strcasecmp(v->name, "directmediaacl")) {
- int ha_error = 0;
- ast_append_acl(v->name + 11, v->value, &peer->directmediaacl, &ha_error, &acl_change_subscription_needed);
- if (ha_error) {
- ast_log(LOG_ERROR, "Bad directmedia ACL entry in configuration line %d : %s. Deleting peer\n", v->lineno, v->value);
- sip_unref_peer(peer, "Removing peer due to bad direct media ACL configuration");
- return NULL;
- }
- } else if (!strcasecmp(v->name, "port")) {
- peer->portinuri = 1;
- if (!(port = port_str2int(v->value, 0))) {
- if (realtime) {
- /* If stored as integer, could be 0 for some DBs (notably MySQL) */
- peer->portinuri = 0;
- } else {
- ast_log(LOG_WARNING, "Invalid peer port configuration at line %d : %s\n", v->lineno, v->value);
- }
- }
- } else if (!strcasecmp(v->name, "callingpres")) {
- peer->callingpres = ast_parse_caller_presentation(v->value);
- if (peer->callingpres == -1) {
- peer->callingpres = atoi(v->value);
- }
- } else if (!strcasecmp(v->name, "username") || !strcasecmp(v->name, "defaultuser")) { /* "username" is deprecated */
- ast_string_field_set(peer, username, v->value);
- if (!strcasecmp(v->name, "username")) {
- if (deprecation_warning) {
- ast_log(LOG_NOTICE, "The 'username' field for sip peers has been deprecated in favor of the term 'defaultuser'\n");
- deprecation_warning = 0;
- }
- peer->deprecated_username = 1;
- }
- } else if (!strcasecmp(v->name, "tonezone")) {
- struct ast_tone_zone *new_zone;
- if (!(new_zone = ast_get_indication_zone(v->value))) {
- ast_log(LOG_ERROR, "Unknown country code '%s' for tonezone in device [%s] at line %d. Check indications.conf for available country codes.\n", v->value, peer->name, v->lineno);
- } else {
- ast_tone_zone_unref(new_zone);
- ast_string_field_set(peer, zone, v->value);
- }
- } else if (!strcasecmp(v->name, "language")) {
- ast_string_field_set(peer, language, v->value);
- } else if (!strcasecmp(v->name, "regexten")) {
- ast_string_field_set(peer, regexten, v->value);
- } else if (!strcasecmp(v->name, "callbackextension")) {
- ast_string_field_set(peer, callback, v->value);
- } else if (!strcasecmp(v->name, "amaflags")) {
- format = ast_channel_string2amaflag(v->value);
- if (format < 0) {
- ast_log(LOG_WARNING, "Invalid AMA Flags for peer: %s at line %d\n", v->value, v->lineno);
- } else {
- peer->amaflags = format;
- }
- } else if (!strcasecmp(v->name, "maxforwards")) {
- if (sscanf(v->value, "%30d", &peer->maxforwards) != 1
- || peer->maxforwards < 1 || 255 < peer->maxforwards) {
- ast_log(LOG_WARNING, "'%s' is not a valid maxforwards value at line %d. Using default.\n", v->value, v->lineno);
- peer->maxforwards = sip_cfg.default_max_forwards;
- }
- } else if (!strcasecmp(v->name, "accountcode")) {
- ast_string_field_set(peer, accountcode, v->value);
- } else if (!strcasecmp(v->name, "mohinterpret")) {
- ast_string_field_set(peer, mohinterpret, v->value);
- } else if (!strcasecmp(v->name, "mohsuggest")) {
- ast_string_field_set(peer, mohsuggest, v->value);
- } else if (!strcasecmp(v->name, "parkinglot")) {
- ast_string_field_set(peer, parkinglot, v->value);
- } else if (!strcasecmp(v->name, "rtp_engine")) {
- ast_string_field_set(peer, engine, v->value);
- } else if (!strcasecmp(v->name, "mailbox")) {
- add_peer_mailboxes(peer, v->value);
- } else if (!strcasecmp(v->name, "hasvoicemail")) {
- /* People expect that if 'hasvoicemail' is set, that the mailbox will
- * be also set, even if not explicitly specified. */
- if (ast_true(v->value) && AST_LIST_EMPTY(&peer->mailboxes)) {
- /*
- * hasvoicemail is a users.conf legacy voicemail enable method.
- * hasvoicemail is only going to work for app_voicemail mailboxes.
- */
- if (strchr(name, '@')) {
- add_peer_mailboxes(peer, name);
- } else {
- char mailbox[AST_MAX_MAILBOX_UNIQUEID];
-
- snprintf(mailbox, sizeof(mailbox), "%s@default", name);
- add_peer_mailboxes(peer, mailbox);
- }
- }
- } else if (!strcasecmp(v->name, "subscribemwi")) {
- ast_set2_flag(&peer->flags[1], ast_true(v->value), SIP_PAGE2_SUBSCRIBEMWIONLY);
- } else if (!strcasecmp(v->name, "vmexten")) {
- ast_string_field_set(peer, vmexten, v->value);
- } else if (!strcasecmp(v->name, "callgroup")) {
- peer->callgroup = ast_get_group(v->value);
- } else if (!strcasecmp(v->name, "allowtransfer")) {
- peer->allowtransfer = ast_true(v->value) ? TRANSFER_OPENFORALL : TRANSFER_CLOSED;
- } else if (!strcasecmp(v->name, "pickupgroup")) {
- peer->pickupgroup = ast_get_group(v->value);
- } else if (!strcasecmp(v->name, "namedcallgroup")) {
- peer->named_callgroups = ast_get_namedgroups(v->value);
- } else if (!strcasecmp(v->name, "namedpickupgroup")) {
- peer->named_pickupgroups = ast_get_namedgroups(v->value);
- } else if (!strcasecmp(v->name, "allow")) {
- int error = ast_format_cap_update_by_allow_disallow(peer->caps, v->value, TRUE);
- if (error) {
- ast_log(LOG_WARNING, "Codec configuration errors found in line %d : %s = %s\n", v->lineno, v->name, v->value);
- }
- } else if (!strcasecmp(v->name, "disallow")) {
- int error = ast_format_cap_update_by_allow_disallow(peer->caps, v->value, FALSE);
- if (error) {
- ast_log(LOG_WARNING, "Codec configuration errors found in line %d : %s = %s\n", v->lineno, v->name, v->value);
- }
- } else if (!strcasecmp(v->name, "preferred_codec_only")) {
- ast_set2_flag(&peer->flags[1], ast_true(v->value), SIP_PAGE2_PREFERRED_CODEC);
- } else if (!strcasecmp(v->name, "autoframing")) {
- peer->autoframing = ast_true(v->value);
- } else if (!strcasecmp(v->name, "rtptimeout")) {
- if ((sscanf(v->value, "%30d", &peer->rtptimeout) != 1) || (peer->rtptimeout < 0)) {
- ast_log(LOG_WARNING, "'%s' is not a valid RTP hold time at line %d. Using default.\n", v->value, v->lineno);
- peer->rtptimeout = global_rtptimeout;
- }
- } else if (!strcasecmp(v->name, "rtpholdtimeout")) {
- if ((sscanf(v->value, "%30d", &peer->rtpholdtimeout) != 1) || (peer->rtpholdtimeout < 0)) {
- ast_log(LOG_WARNING, "'%s' is not a valid RTP hold time at line %d. Using default.\n", v->value, v->lineno);
- peer->rtpholdtimeout = global_rtpholdtimeout;
- }
- } else if (!strcasecmp(v->name, "rtpkeepalive")) {
- if ((sscanf(v->value, "%30d", &peer->rtpkeepalive) != 1) || (peer->rtpkeepalive < 0)) {
- ast_log(LOG_WARNING, "'%s' is not a valid RTP keepalive time at line %d. Using default.\n", v->value, v->lineno);
- peer->rtpkeepalive = global_rtpkeepalive;
- }
- } else if (!strcasecmp(v->name, "timert1")) {
- if ((sscanf(v->value, "%30d", &peer->timer_t1) != 1) || (peer->timer_t1 < 200) || (peer->timer_t1 < global_t1min)) {
- ast_log(LOG_WARNING, "'%s' is not a valid T1 time at line %d. Using default.\n", v->value, v->lineno);
- peer->timer_t1 = global_t1min;
- }
- timert1_set = 1;
- } else if (!strcasecmp(v->name, "timerb")) {
- if ((sscanf(v->value, "%30d", &peer->timer_b) != 1) || (peer->timer_b < 200)) {
- ast_log(LOG_WARNING, "'%s' is not a valid Timer B time at line %d. Using default.\n", v->value, v->lineno);
- peer->timer_b = global_timer_b;
- }
- timerb_set = 1;
- } else if (!strcasecmp(v->name, "setvar")) {
- peer->chanvars = add_var(v->value, peer->chanvars);
- } else if (!strcasecmp(v->name, "header")) {
- char tmp[4096];
- snprintf(tmp, sizeof(tmp), "__SIPADDHEADERpre%2d=%s", ++headercount, v->value);
- peer->chanvars = add_var(tmp, peer->chanvars);
- } else if (!strcasecmp(v->name, "qualifyfreq")) {
- int i;
- if (sscanf(v->value, "%30d", &i) == 1) {
- peer->qualifyfreq = i * 1000;
- } else {
- ast_log(LOG_WARNING, "Invalid qualifyfreq number '%s' at line %d of %s\n", v->value, v->lineno, config);
- peer->qualifyfreq = global_qualifyfreq;
- }
- } else if (!strcasecmp(v->name, "maxcallbitrate")) {
- peer->maxcallbitrate = atoi(v->value);
- if (peer->maxcallbitrate < 0) {
- peer->maxcallbitrate = default_maxcallbitrate;
- }
- } else if (!strcasecmp(v->name, "session-timers")) {
- int i = (int) str2stmode(v->value);
- if (i < 0) {
- ast_log(LOG_WARNING, "Invalid session-timers '%s' at line %d of %s\n", v->value, v->lineno, config);
- peer->stimer.st_mode_oper = global_st_mode;
- } else {
- peer->stimer.st_mode_oper = i;
- }
- } else if (!strcasecmp(v->name, "session-expires")) {
- if (sscanf(v->value, "%30d", &peer->stimer.st_max_se) != 1) {
- ast_log(LOG_WARNING, "Invalid session-expires '%s' at line %d of %s\n", v->value, v->lineno, config);
- peer->stimer.st_max_se = global_max_se;
- }
- } else if (!strcasecmp(v->name, "session-minse")) {
- if (sscanf(v->value, "%30d", &peer->stimer.st_min_se) != 1) {
- ast_log(LOG_WARNING, "Invalid session-minse '%s' at line %d of %s\n", v->value, v->lineno, config);
- peer->stimer.st_min_se = global_min_se;
- }
- if (peer->stimer.st_min_se < DEFAULT_MIN_SE) {
- ast_log(LOG_WARNING, "session-minse '%s' at line %d of %s is not allowed to be < %d secs\n", v->value, v->lineno, config, DEFAULT_MIN_SE);
- peer->stimer.st_min_se = global_min_se;
- }
- } else if (!strcasecmp(v->name, "session-refresher")) {
- int i = (int) str2strefresherparam(v->value);
- if (i < 0) {
- ast_log(LOG_WARNING, "Invalid session-refresher '%s' at line %d of %s\n", v->value, v->lineno, config);
- peer->stimer.st_ref = global_st_refresher;
- } else {
- peer->stimer.st_ref = i;
- }
- } else if (!strcasecmp(v->name, "disallowed_methods")) {
- char *disallow = ast_strdupa(v->value);
- mark_parsed_methods(&peer->disallowed_methods, disallow);
- } else if (!strcasecmp(v->name, "unsolicited_mailbox")) {
- ast_string_field_set(peer, unsolicited_mailbox, v->value);
- } else if (!strcasecmp(v->name, "use_q850_reason")) {
- ast_set2_flag(&peer->flags[1], ast_true(v->value), SIP_PAGE2_Q850_REASON);
- } else if (!strcasecmp(v->name, "encryption")) {
- ast_set2_flag(&peer->flags[1], ast_true(v->value), SIP_PAGE2_USE_SRTP);
- } else if (!strcasecmp(v->name, "encryption_taglen")) {
- ast_set2_flag(&peer->flags[2], !strcasecmp(v->value, "32"), SIP_PAGE3_SRTP_TAG_32);
- } else if (!strcasecmp(v->name, "snom_aoc_enabled")) {
- ast_set2_flag(&peer->flags[2], ast_true(v->value), SIP_PAGE3_SNOM_AOC);
- } else if (!strcasecmp(v->name, "avpf")) {
- ast_set2_flag(&peer->flags[2], ast_true(v->value), SIP_PAGE3_USE_AVPF);
- } else if (!strcasecmp(v->name, "icesupport")) {
- ast_set2_flag(&peer->flags[2], ast_true(v->value), SIP_PAGE3_ICE_SUPPORT);
- } else if (!strcasecmp(v->name, "ignore_requested_pref")) {
- ast_set2_flag(&peer->flags[2], ast_true(v->value), SIP_PAGE3_IGNORE_PREFCAPS);
- } else if (!strcasecmp(v->name, "discard_remote_hold_retrieval")) {
- ast_set2_flag(&peer->flags[2], ast_true(v->value), SIP_PAGE3_DISCARD_REMOTE_HOLD_RETRIEVAL);
- } else if (!strcasecmp(v->name, "force_avp")) {
- ast_set2_flag(&peer->flags[2], ast_true(v->value), SIP_PAGE3_FORCE_AVP);
- } else {
- ast_rtp_dtls_cfg_parse(&peer->dtls_cfg, v->name, v->value);
- }
- }
-
- /* Validate DTLS configuration */
- if (ast_rtp_dtls_cfg_validate(&peer->dtls_cfg)) {
- sip_unref_peer(peer, "Removing peer due to bad DTLS configuration");
- return NULL;
- }
-
- /* SRB */
-
- /* Apply the encryption tag length to the DTLS configuration, in case DTLS is in use */
- peer->dtls_cfg.suite = (ast_test_flag(&peer->flags[2], SIP_PAGE3_SRTP_TAG_32) ? AST_AES_CM_128_HMAC_SHA1_32 : AST_AES_CM_128_HMAC_SHA1_80);
-
- /* These apply to devstate lookups */
- if (realtime && !strcasecmp(v->name, "lastms")) {
- sscanf(v->value, "%30d", &peer->lastms);
- } else if (realtime && !strcasecmp(v->name, "ipaddr") && !ast_strlen_zero(v->value) ) {
- ast_sockaddr_parse(&peer->addr, v->value, PARSE_PORT_FORBID);
- } else if (realtime && !strcasecmp(v->name, "fullcontact")) {
- if (alt_fullcontact && !alt) {
- /* Reset, because the alternate also has a fullcontact and we
- * do NOT want the field value to be doubled. It might be
- * tempting to skip this, but the first table might not have
- * fullcontact and since we're here, we know that the alternate
- * absolutely does. */
- alt_fullcontact = 0;
- ast_str_reset(fullcontact);
- }
- /* Reconstruct field, because realtime separates our value at the ';' */
- if (ast_str_strlen(fullcontact) > 0) {
- ast_str_append(&fullcontact, 0, ";%s", v->value);
- } else {
- ast_str_set(&fullcontact, 0, "%s", v->value);
- }
- } else if (!strcasecmp(v->name, "qualify")) {
- if (!strcasecmp(v->value, "no")) {
- peer->maxms = 0;
- } else if (!strcasecmp(v->value, "yes")) {
- peer->maxms = default_qualify ? default_qualify : DEFAULT_MAXMS;
- } else if (sscanf(v->value, "%30d", &peer->maxms) != 1) {
- ast_log(LOG_WARNING, "Qualification of peer '%s' should be 'yes', 'no', or a number of milliseconds at line %d of sip.conf\n", peer->name, v->lineno);
- peer->maxms = 0;
- }
- if (realtime && !ast_test_flag(&global_flags[1], SIP_PAGE2_RTCACHEFRIENDS) && peer->maxms > 0) {
- /* This would otherwise cause a network storm, where the
- * qualify response refreshes the peer from the database,
- * which in turn causes another qualify to be sent, ad
- * infinitum. */
- ast_log(LOG_WARNING, "Qualify is incompatible with dynamic uncached realtime. Please either turn rtcachefriends on or turn qualify off on peer '%s'\n", peer->name);
- peer->maxms = 0;
- }
- } else if (!strcasecmp(v->name, "keepalive")) {
- if (!strcasecmp(v->value, "no")) {
- peer->keepalive = 0;
- } else if (!strcasecmp(v->value, "yes")) {
- peer->keepalive = DEFAULT_KEEPALIVE_INTERVAL;
- } else if (sscanf(v->value, "%30d", &peer->keepalive) != 1) {
- ast_log(LOG_WARNING, "Keep alive of peer '%s' should be 'yes', 'no', or a number of milliseconds at line %d of sip.conf\n", peer->name, v->lineno);
- peer->keepalive = 0;
- }
- } else if (!strcasecmp(v->name, "callcounter")) {
- peer->call_limit = ast_true(v->value) ? INT_MAX : 0;
- } else if (!strcasecmp(v->name, "call-limit")) {
- peer->call_limit = atoi(v->value);
- if (peer->call_limit < 0) {
- peer->call_limit = 0;
- }
- } else if (!strcasecmp(v->name, "busylevel")) {
- peer->busy_level = atoi(v->value);
- if (peer->busy_level < 0) {
- peer->busy_level = 0;
- }
- } else if (ast_cc_is_config_param(v->name)) {
- ast_cc_set_param(peer->cc_params, v->name, v->value);
- }
- }
-
- if (!devstate_only) {
- struct sip_mailbox *mailbox;
- AST_LIST_TRAVERSE_SAFE_BEGIN(&peer->mailboxes, mailbox, entry) {
- if (mailbox->status == SIP_MAILBOX_STATUS_UNKNOWN) {
- AST_LIST_REMOVE_CURRENT(entry);
- destroy_mailbox(mailbox);
- }
- }
- AST_LIST_TRAVERSE_SAFE_END;
- }
-
- if (!can_parse_xml && (ast_get_cc_agent_policy(peer->cc_params) == AST_CC_AGENT_NATIVE)) {
- ast_log(LOG_WARNING, "Peer %s has a cc_agent_policy of 'native' but required libxml2 dependency is not installed. Changing policy to 'never'\n", peer->name);
- ast_set_cc_agent_policy(peer->cc_params, AST_CC_AGENT_NEVER);
- }
-
- /* Note that Timer B is dependent upon T1 and MUST NOT be lower
- * than T1 * 64, according to RFC 3261, Section 17.1.1.2 */
- if (peer->timer_b < peer->timer_t1 * 64) {
- if (timerb_set && timert1_set) {
- ast_log(LOG_WARNING, "Timer B has been set lower than recommended for peer %s (%d < 64 * Timer-T1=%d)\n", peer->name, peer->timer_b, peer->timer_t1);
- } else if (timerb_set) {
- if ((peer->timer_t1 = peer->timer_b / 64) < global_t1min) {
- ast_log(LOG_WARNING, "Timer B has been set lower than recommended (%d < 64 * timert1=%d). (RFC 3261, 17.1.1.2)\n", peer->timer_b, peer->timer_t1);
- peer->timer_t1 = global_t1min;
- peer->timer_b = peer->timer_t1 * 64;
- }
- peer->timer_t1 = peer->timer_b / 64;
- } else {
- peer->timer_b = peer->timer_t1 * 64;
- }
- }
-
- if (!peer->default_outbound_transport) {
- /* Set default set of transports */
- peer->transports = default_transports;
- /* Set default primary transport */
- peer->default_outbound_transport = default_primary_transport;
- }
-
- /* The default transport type set during build_peer should only replace the socket.type when...
- * 1. Registration is not present and the socket.type and default transport types are different.
- * 2. The socket.type is not an acceptable transport type after rebuilding peer.
- * 3. The socket.type is not set yet. */
- if (((peer->socket.type != peer->default_outbound_transport) && (peer->expire == -1)) ||
- !(peer->socket.type & peer->transports) || !(peer->socket.type)) {
- set_socket_transport(&peer->socket, peer->default_outbound_transport);
- }
-
- ast_copy_flags(&peer->flags[0], &peerflags[0], mask[0].flags);
- ast_copy_flags(&peer->flags[1], &peerflags[1], mask[1].flags);
- ast_copy_flags(&peer->flags[2], &peerflags[2], mask[2].flags);
-
- if (ast_str_strlen(fullcontact)) {
- ast_string_field_set(peer, fullcontact, ast_str_buffer(fullcontact));
- peer->rt_fromcontact = TRUE;
- /* We have a hostname in the fullcontact, but if we don't have an
- * address listed on the entry (or if it's 'dynamic'), then we need to
- * parse the entry to obtain the IP address, so a dynamic host can be
- * contacted immediately after reload (as opposed to waiting for it to
- * register once again). But if we have an address for this peer and NAT was
- * specified, use that address instead. */
- /* XXX May need to revisit the final argument; does the realtime DB store whether
- * the original contact was over TLS or not? XXX */
- if ((!ast_test_flag(&peer->flags[2], SIP_PAGE3_NAT_AUTO_RPORT) && !ast_test_flag(&peer->flags[0], SIP_NAT_FORCE_RPORT))
- || ast_sockaddr_isnull(&peer->addr)) {
- __set_address_from_contact(ast_str_buffer(fullcontact), &peer->addr, 0);
- }
- }
-
- if (srvlookup && peer->dnsmgr == NULL) {
- char transport[MAXHOSTNAMELEN];
- char _srvlookup[MAXHOSTNAMELEN];
- char *params;
-
- ast_copy_string(_srvlookup, srvlookup, sizeof(_srvlookup));
- if ((params = strchr(_srvlookup, ';'))) {
- *params++ = '\0';
- }
-
- snprintf(transport, sizeof(transport), "_%s._%s", get_srv_service(peer->socket.type), get_srv_protocol(peer->socket.type));
-
- peer->addr.ss.ss_family = get_address_family_filter(peer->socket.type); /* Filter address family */
- if (ast_dnsmgr_lookup_cb(_srvlookup, &peer->addr, &peer->dnsmgr, sip_cfg.srvlookup && !peer->portinuri ? transport : NULL,
- on_dns_update_peer, sip_ref_peer(peer, "Store peer on dnsmgr"))) {
- ast_log(LOG_ERROR, "srvlookup failed for host: %s, on peer %s, removing peer\n", _srvlookup, peer->name);
- sip_unref_peer(peer, "dnsmgr lookup failed, getting rid of peer dnsmgr ref");
- sip_unref_peer(peer, "getting rid of a peer pointer");
- return NULL;
- }
- if (!peer->dnsmgr) {
- /* dnsmgr refresh disabeld, release reference */
- sip_unref_peer(peer, "dnsmgr disabled, unref peer");
- }
-
- ast_string_field_set(peer, tohost, srvlookup);
-
- if (global_dynamic_exclude_static && !ast_sockaddr_isnull(&peer->addr)) {
- int ha_error = 0;
-
- ast_append_acl("deny", ast_sockaddr_stringify_addr(&peer->addr), &sip_cfg.contact_acl, &ha_error, NULL);
- if (ha_error) {
- ast_log(LOG_ERROR, "Bad or unresolved host/IP entry in configuration for peer %s, cannot add to contact ACL\n", peer->name);
- }
- }
- } else if (peer->dnsmgr && !peer->host_dynamic) {
- /* force a refresh here on reload if dnsmgr already exists and host is set. */
- ast_dnsmgr_refresh(peer->dnsmgr);
- }
-
- if (port && !realtime && peer->host_dynamic) {
- ast_sockaddr_set_port(&peer->defaddr, port);
- } else if (port) {
- ast_sockaddr_set_port(&peer->addr, port);
- }
-
- if (ast_sockaddr_port(&peer->addr) == 0) {
- ast_sockaddr_set_port(&peer->addr,
- (peer->socket.type & AST_TRANSPORT_TLS) ?
- STANDARD_TLS_PORT : STANDARD_SIP_PORT);
- }
- if (ast_sockaddr_port(&peer->defaddr) == 0) {
- ast_sockaddr_set_port(&peer->defaddr,
- (peer->socket.type & AST_TRANSPORT_TLS) ?
- STANDARD_TLS_PORT : STANDARD_SIP_PORT);
- }
-
- if (realtime) {
- int enablepoke = 1;
-
- if (!sip_cfg.ignore_regexpire && peer->host_dynamic) {
- time_t nowtime = time(NULL);
-
- if ((nowtime - regseconds) > 0) {
- destroy_association(peer);
- memset(&peer->addr, 0, sizeof(peer->addr));
- peer->lastms = -1;
- enablepoke = 0;
- ast_debug(1, "Bah, we're expired (%d/%d/%d)!\n", (int)(nowtime - regseconds), (int)regseconds, (int)nowtime);
- }
- }
-
- /* Startup regular pokes */
- if (!devstate_only && enablepoke) {
- /*
- * We cannot poke the peer now in this thread without
- * a lock inversion so pass it off to the scheduler
- * thread.
- */
- AST_SCHED_REPLACE_UNREF(peer->pokeexpire, sched,
- 0, /* Poke the peer ASAP */
- sip_poke_peer_now, peer,
- sip_unref_peer(_data, "removing poke peer ref"),
- sip_unref_peer(peer, "removing poke peer ref"),
- sip_ref_peer(peer, "adding poke peer ref"));
- }
- }
-
- if (ast_test_flag(&peer->flags[1], SIP_PAGE2_ALLOWSUBSCRIBE)) {
- sip_cfg.allowsubscribe = TRUE; /* No global ban any more */
- }
- /* If read-only RT backend, then refresh from local DB cache */
- if (peer->host_dynamic && (!peer->is_realtime || !sip_cfg.peer_rtupdate)) {
- reg_source_db(peer);
- }
-
- /* If they didn't request that MWI is sent *only* on subscribe, go ahead and
- * subscribe to it now. */
- if (!devstate_only && !ast_test_flag(&peer->flags[1], SIP_PAGE2_SUBSCRIBEMWIONLY) &&
- !AST_LIST_EMPTY(&peer->mailboxes)) {
- add_peer_mwi_subs(peer);
- /* Send MWI from the event cache only. This is so we can send initial
- * MWI if app_voicemail got loaded before chan_sip. If it is the other
- * way, then we will get events when app_voicemail gets loaded. */
- sip_send_mwi_to_peer(peer, 1);
- }
-
- peer->the_mark = 0;
-
- oldacl = ast_free_acl_list(oldacl);
- oldcontactacl = ast_free_acl_list(oldcontactacl);
- olddirectmediaacl = ast_free_acl_list(olddirectmediaacl);
- if (!ast_strlen_zero(peer->callback)) { /* build string from peer info */
- char *reg_string;
- if (ast_asprintf(®_string, "%s?%s:%s:%s@%s/%s", peer->name, S_OR(peer->fromuser, peer->username), S_OR(peer->remotesecret, peer->secret), peer->username, peer->tohost, peer->callback) >= 0) {
- sip_register(reg_string, 0); /* XXX TODO: count in registry_count */
- ast_free(reg_string);
- }
- }
-
- /* If an ACL change subscription is needed and doesn't exist, we need one. */
- if (acl_change_subscription_needed) {
- acl_change_stasis_subscribe();
- }
-
- return peer;
-}
-
-static int peer_markall_func(void *device, void *arg, int flags)
-{
- struct sip_peer *peer = device;
- if (!peer->selfdestruct) {
- peer->the_mark = 1;
- }
- return 0;
-}
-
-static int peer_markall_autopeers_func(void *device, void *arg, int flags)
-{
- struct sip_peer *peer = device;
- if (peer->selfdestruct) {
- peer->the_mark = 1;
- }
- return 0;
-}
-
-/*!
- * \internal
- * \brief If no default formats are set in config, these are used
- */
-static void sip_set_default_format_capabilities(struct ast_format_cap *cap)
-{
- ast_format_cap_remove_by_type(cap, AST_MEDIA_TYPE_UNKNOWN);
- ast_format_cap_append(cap, ast_format_ulaw, 0);
- ast_format_cap_append(cap, ast_format_alaw, 0);
- ast_format_cap_append(cap, ast_format_gsm, 0);
- ast_format_cap_append(cap, ast_format_h263, 0);
-}
-
-static void display_nat_warning(const char *cat, int reason, struct ast_flags *flags) {
- int global_nat, specific_nat;
-
- if (reason == CHANNEL_MODULE_LOAD && (specific_nat = ast_test_flag(&flags[0], SIP_NAT_FORCE_RPORT)) != (global_nat = ast_test_flag(&global_flags[0], SIP_NAT_FORCE_RPORT))) {
- ast_log(LOG_WARNING, "!!! PLEASE NOTE: Setting 'nat' for a peer/user that differs from the global setting can make\n");
- ast_log(LOG_WARNING, "!!! the name of that peer/user discoverable by an attacker. Replies for non-existent peers/users\n");
- ast_log(LOG_WARNING, "!!! will be sent to a different port than replies for an existing peer/user. If at all possible,\n");
- ast_log(LOG_WARNING, "!!! use the global 'nat' setting and do not set 'nat' per peer/user.\n");
- ast_log(LOG_WARNING, "!!! (config category='%s' global force_rport='%s' peer/user force_rport='%s')\n", cat, AST_CLI_YESNO(global_nat), AST_CLI_YESNO(specific_nat));
- }
-}
-
-/* Run by the sched thread. */
-static int __cleanup_registration(const void *data)
-{
- struct sip_registry *reg = (struct sip_registry *) data;
-
- ao2_lock(reg);
-
- if (reg->call) {
- ast_debug(3, "Destroying active SIP dialog for registry %s@%s\n", reg->username, reg->hostname);
- /* This will also remove references to the registry */
- dialog_unlink_all(reg->call);
- reg->call = dialog_unref(reg->call, "remove iterator->call from registry traversal");
- }
-
- AST_SCHED_DEL_UNREF(sched, reg->expire,
- ao2_t_ref(reg, -1, "Stop scheduled reregister timeout"));
- AST_SCHED_DEL_UNREF(sched, reg->timeout,
- ao2_t_ref(reg, -1, "Stop scheduled register timeout"));
-
- if (reg->dnsmgr) {
- ast_dnsmgr_release(reg->dnsmgr);
- reg->dnsmgr = NULL;
- ao2_t_ref(reg, -1, "reg ptr unref from dnsmgr");
- }
-
- ao2_unlock(reg);
-
- ao2_t_ref(reg, -1, "cleanup_registration action");
- return 0;
-}
-
-static int cleanup_registration(void *obj, void *arg, int flags)
-{
- struct sip_registry *reg = obj;
-
- ao2_t_ref(reg, +1, "cleanup_registration action");
- if (ast_sched_add(sched, 0, __cleanup_registration, reg) < 0) {
- /* Uh Oh. Expect bad behavior. */
- ao2_t_ref(reg, -1, "Failed to schedule cleanup_registration action");
- }
-
- return CMP_MATCH;
-}
-
-static void cleanup_all_regs(void)
-{
- ao2_t_callback(registry_list, OBJ_NODATA | OBJ_UNLINK | OBJ_MULTIPLE,
- cleanup_registration, NULL, "remove all SIP registry items");
-}
-
-/*! \brief Re-read SIP.conf config file
-\note This function reloads all config data, except for
- active peers (with registrations). They will only
- change configuration data at restart, not at reload.
- SIP debug and recordhistory state will not change
- */
-static int reload_config(enum channelreloadreason reason)
-{
- struct ast_config *cfg, *ucfg;
- struct ast_variable *v;
- struct sip_peer *peer;
- char *cat, *stringp, *context, *oldregcontext;
- char newcontexts[AST_MAX_CONTEXT], oldcontexts[AST_MAX_CONTEXT];
- struct ast_flags mask[3] = {{0}};
- struct ast_flags setflags[3] = {{0}};
- struct ast_flags config_flags = { (reason == CHANNEL_MODULE_LOAD || reason == CHANNEL_ACL_RELOAD) ? 0 : ast_test_flag(&global_flags[1], SIP_PAGE2_RTCACHEFRIENDS) ? 0 : CONFIG_FLAG_FILEUNCHANGED };
- int auto_sip_domains = FALSE;
- struct ast_sockaddr old_bindaddr = bindaddr;
- int registry_count = 0, peer_count = 0, timerb_set = 0, timert1_set = 0;
- int subscribe_network_change = 1;
- time_t run_start, run_end;
- int bindport = 0;
- int acl_change_subscription_needed = 0;
- int min_subexpiry_set = 0, max_subexpiry_set = 0;
- int websocket_was_enabled = sip_cfg.websocket_enabled;
-
- run_start = time(0);
- ast_unload_realtime("sipregs");
- ast_unload_realtime("sippeers");
- cfg = ast_config_load(config, config_flags);
-
- /* We *must* have a config file otherwise stop immediately */
- if (!cfg) {
- ast_log(LOG_NOTICE, "Unable to load config %s\n", config);
- return -1;
- } else if (cfg == CONFIG_STATUS_FILEUNCHANGED) {
- ucfg = ast_config_load("users.conf", config_flags);
- if (ucfg == CONFIG_STATUS_FILEUNCHANGED) {
- return 1;
- } else if (ucfg == CONFIG_STATUS_FILEINVALID) {
- ast_log(LOG_ERROR, "Contents of users.conf are invalid and cannot be parsed\n");
- return 1;
- }
- /* Must reread both files, because one changed */
- ast_clear_flag(&config_flags, CONFIG_FLAG_FILEUNCHANGED);
- if ((cfg = ast_config_load(config, config_flags)) == CONFIG_STATUS_FILEINVALID) {
- ast_log(LOG_ERROR, "Contents of %s are invalid and cannot be parsed\n", config);
- ast_config_destroy(ucfg);
- return 1;
- }
- if (!cfg) {
- /* should have been able to reload here */
- ast_log(LOG_NOTICE, "Unable to load config %s\n", config);
- return -1;
- }
- } else if (cfg == CONFIG_STATUS_FILEINVALID) {
- ast_log(LOG_ERROR, "Contents of %s are invalid and cannot be parsed\n", config);
- return 1;
- } else {
- ast_clear_flag(&config_flags, CONFIG_FLAG_FILEUNCHANGED);
- if ((ucfg = ast_config_load("users.conf", config_flags)) == CONFIG_STATUS_FILEINVALID) {
- ast_log(LOG_ERROR, "Contents of users.conf are invalid and cannot be parsed\n");
- ast_config_destroy(cfg);
- return 1;
- }
- }
-
- sip_cfg.contact_acl = ast_free_acl_list(sip_cfg.contact_acl);
-
- default_tls_cfg.enabled = FALSE; /* Default: Disable TLS */
- default_dtls_cfg.enabled = FALSE; /* Default: Disable DTLS too */
-
- if (reason != CHANNEL_MODULE_LOAD) {
- ast_debug(4, "--------------- SIP reload started\n");
-
- clear_sip_domains();
- ast_mutex_lock(&authl_lock);
- if (authl) {
- ao2_t_ref(authl, -1, "Removing old global authentication");
- authl = NULL;
- }
- ast_mutex_unlock(&authl_lock);
-
- /* Then, actually destroy users and registry */
- cleanup_all_regs();
- ast_debug(4, "--------------- Done destroying registry list\n");
- ao2_t_callback(peers, OBJ_NODATA, peer_markall_func, NULL, "callback to mark all peers");
- }
-
- /* Reset certificate handling for TLS and DTLS sessions */
- if (reason != CHANNEL_MODULE_LOAD) {
- ast_free(default_tls_cfg.certfile);
- ast_free(default_tls_cfg.pvtfile);
- ast_free(default_tls_cfg.cipher);
- ast_free(default_tls_cfg.cafile);
- ast_free(default_tls_cfg.capath);
- ast_rtp_dtls_cfg_free(&default_dtls_cfg);
- }
- default_tls_cfg.certfile = ast_strdup(AST_CERTFILE); /*XXX Not sure if this is useful */
- default_tls_cfg.pvtfile = ast_strdup("");
- default_tls_cfg.cipher = ast_strdup("");
- default_tls_cfg.cafile = ast_strdup("");
- default_tls_cfg.capath = ast_strdup("");
- /* Using the same idea fro DTLS as the code block above for TLS */
- default_dtls_cfg.certfile = ast_strdup("");
- default_dtls_cfg.pvtfile = ast_strdup("");
- default_dtls_cfg.cipher = ast_strdup("");
- default_dtls_cfg.cafile = ast_strdup("");
- default_dtls_cfg.capath = ast_strdup("");
-
- /* Initialize copy of current sip_cfg.regcontext for later use in removing stale contexts */
- ast_copy_string(oldcontexts, sip_cfg.regcontext, sizeof(oldcontexts));
- oldregcontext = oldcontexts;
-
- /* Clear all flags before setting default values */
- /* Preserve debugging settings for console */
- sipdebug &= sip_debug_console;
- ast_clear_flag(&global_flags[0], AST_FLAGS_ALL);
- ast_clear_flag(&global_flags[1], AST_FLAGS_ALL);
- ast_clear_flag(&global_flags[2], AST_FLAGS_ALL);
-
- /* Reset IP addresses */
- ast_sockaddr_parse(&bindaddr, "0.0.0.0:0", 0);
- memset(&internip, 0, sizeof(internip));
-
- /* Free memory for local network address mask */
- ast_free_ha(localaddr);
- memset(&localaddr, 0, sizeof(localaddr));
- memset(&externaddr, 0, sizeof(externaddr));
- memset(&media_address, 0, sizeof(media_address));
- memset(&rtpbindaddr, 0, sizeof(rtpbindaddr));
- memset(&sip_cfg.outboundproxy, 0, sizeof(struct sip_proxy));
- sip_cfg.outboundproxy.force = FALSE; /*!< Don't force proxy usage, use route: headers */
- default_transports = AST_TRANSPORT_UDP;
- default_primary_transport = AST_TRANSPORT_UDP;
- ourport_tcp = STANDARD_SIP_PORT;
- ourport_tls = STANDARD_TLS_PORT;
- externtcpport = 0;
- externtlsport = 0;
- sip_cfg.srvlookup = DEFAULT_SRVLOOKUP;
- global_tos_sip = DEFAULT_TOS_SIP;
- global_tos_audio = DEFAULT_TOS_AUDIO;
- global_tos_video = DEFAULT_TOS_VIDEO;
- global_tos_text = DEFAULT_TOS_TEXT;
- global_cos_sip = DEFAULT_COS_SIP;
- global_cos_audio = DEFAULT_COS_AUDIO;
- global_cos_video = DEFAULT_COS_VIDEO;
- global_cos_text = DEFAULT_COS_TEXT;
-
- externhost[0] = '\0'; /* External host name (for behind NAT DynDNS support) */
- externexpire = 0; /* Expiration for DNS re-issuing */
- externrefresh = 10;
-
- /* Reset channel settings to default before re-configuring */
- sip_cfg.allow_external_domains = DEFAULT_ALLOW_EXT_DOM; /* Allow external invites */
- sip_cfg.regcontext[0] = '\0';
- sip_set_default_format_capabilities(sip_cfg.caps);
- sip_cfg.regextenonqualify = DEFAULT_REGEXTENONQUALIFY;
- sip_cfg.legacy_useroption_parsing = DEFAULT_LEGACY_USEROPTION_PARSING;
- sip_cfg.send_diversion = DEFAULT_SEND_DIVERSION;
- sip_cfg.notifyringing = DEFAULT_NOTIFYRINGING;
- sip_cfg.notifycid = DEFAULT_NOTIFYCID;
- sip_cfg.notifyhold = FALSE; /*!< Keep track of hold status for a peer */
- sip_cfg.directrtpsetup = FALSE; /* Experimental feature, disabled by default */
- sip_cfg.alwaysauthreject = DEFAULT_ALWAYSAUTHREJECT;
- sip_cfg.auth_options_requests = DEFAULT_AUTH_OPTIONS;
- sip_cfg.auth_message_requests = DEFAULT_AUTH_MESSAGE;
- sip_cfg.messagecontext[0] = '\0';
- sip_cfg.accept_outofcall_message = DEFAULT_ACCEPT_OUTOFCALL_MESSAGE;
- sip_cfg.allowsubscribe = FALSE;
- sip_cfg.disallowed_methods = SIP_UNKNOWN;
- sip_cfg.contact_acl = NULL; /* Reset the contact ACL */
- snprintf(global_useragent, sizeof(global_useragent), "%s %s", DEFAULT_USERAGENT, ast_get_version());
- snprintf(global_sdpsession, sizeof(global_sdpsession), "%s %s", DEFAULT_SDPSESSION, ast_get_version());
- snprintf(global_sdpowner, sizeof(global_sdpowner), "%s", DEFAULT_SDPOWNER);
- global_prematuremediafilter = TRUE;
- ast_copy_string(default_notifymime, DEFAULT_NOTIFYMIME, sizeof(default_notifymime));
- ast_copy_string(sip_cfg.realm, S_OR(ast_config_AST_SYSTEM_NAME, DEFAULT_REALM), sizeof(sip_cfg.realm));
- sip_cfg.domainsasrealm = DEFAULT_DOMAINSASREALM;
- ast_copy_string(default_callerid, DEFAULT_CALLERID, sizeof(default_callerid));
- ast_copy_string(default_mwi_from, DEFAULT_MWI_FROM, sizeof(default_mwi_from));
- sip_cfg.compactheaders = DEFAULT_COMPACTHEADERS;
- global_reg_timeout = DEFAULT_REGISTRATION_TIMEOUT;
- global_regattempts_max = 0;
- global_reg_retry_403 = 0;
- sip_cfg.pedanticsipchecking = DEFAULT_PEDANTIC;
- sip_cfg.autocreatepeer = DEFAULT_AUTOCREATEPEER;
- global_autoframing = 0;
- sip_cfg.allowguest = DEFAULT_ALLOWGUEST;
- global_callcounter = DEFAULT_CALLCOUNTER;
- global_match_auth_username = FALSE; /*!< Match auth username if available instead of From: Default off. */
- global_rtptimeout = 0;
- global_rtpholdtimeout = 0;
- global_rtpkeepalive = DEFAULT_RTPKEEPALIVE;
- sip_cfg.allowtransfer = TRANSFER_OPENFORALL; /* Merrily accept all transfers by default */
- sip_cfg.rtautoclear = 120;
- ast_set_flag(&global_flags[1], SIP_PAGE2_ALLOWSUBSCRIBE); /* Default for all devices: TRUE */
- ast_set_flag(&global_flags[1], SIP_PAGE2_ALLOWOVERLAP_YES); /* Default for all devices: Yes */
- sip_cfg.peer_rtupdate = TRUE;
- global_dynamic_exclude_static = 0; /* Exclude static peers */
- sip_cfg.tcp_enabled = FALSE;
- sip_cfg.websocket_enabled = TRUE;
- sip_cfg.websocket_write_timeout = AST_DEFAULT_WEBSOCKET_WRITE_TIMEOUT;
-
- /* Session-Timers */
- global_st_mode = SESSION_TIMER_MODE_ACCEPT;
- global_st_refresher = SESSION_TIMER_REFRESHER_PARAM_UAS;
- global_min_se = DEFAULT_MIN_SE;
- global_max_se = DEFAULT_MAX_SE;
-
- /* Peer poking settings */
- global_qualify_gap = DEFAULT_QUALIFY_GAP;
- global_qualify_peers = DEFAULT_QUALIFY_PEERS;
-
- /* Initialize some reasonable defaults at SIP reload (used both for channel and as default for devices */
- ast_copy_string(sip_cfg.default_context, DEFAULT_CONTEXT, sizeof(sip_cfg.default_context));
- ast_copy_string(sip_cfg.default_record_on_feature, DEFAULT_RECORD_FEATURE, sizeof(sip_cfg.default_record_on_feature));
- ast_copy_string(sip_cfg.default_record_off_feature, DEFAULT_RECORD_FEATURE, sizeof(sip_cfg.default_record_off_feature));
- sip_cfg.default_subscribecontext[0] = '\0';
- sip_cfg.default_max_forwards = DEFAULT_MAX_FORWARDS;
- default_language[0] = '\0';
- default_fromdomain[0] = '\0';
- default_fromdomainport = 0;
- default_qualify = DEFAULT_QUALIFY;
- default_keepalive = DEFAULT_KEEPALIVE;
- default_zone[0] = '\0';
- default_maxcallbitrate = DEFAULT_MAX_CALL_BITRATE;
- ast_copy_string(default_mohinterpret, DEFAULT_MOHINTERPRET, sizeof(default_mohinterpret));
- ast_copy_string(default_mohsuggest, DEFAULT_MOHSUGGEST, sizeof(default_mohsuggest));
- ast_copy_string(default_vmexten, DEFAULT_VMEXTEN, sizeof(default_vmexten));
- ast_set_flag(&global_flags[0], SIP_DTMF_RFC2833); /*!< Default DTMF setting: RFC2833 */
- ast_set_flag(&global_flags[0], SIP_DIRECT_MEDIA); /*!< Allow re-invites */
- ast_set_flag(&global_flags[2], SIP_PAGE3_NAT_AUTO_RPORT); /*!< Default to nat=auto_force_rport */
- ast_copy_string(default_engine, DEFAULT_ENGINE, sizeof(default_engine));
- ast_copy_string(default_parkinglot, DEFAULT_PARKINGLOT, sizeof(default_parkinglot));
-
- /* Debugging settings, always default to off */
- dumphistory = FALSE;
- recordhistory = FALSE;
- sipdebug &= ~sip_debug_config;
-
- /* Misc settings for the channel */
- global_relaxdtmf = FALSE;
- global_authfailureevents = FALSE;
- global_t1 = DEFAULT_TIMER_T1;
- global_timer_b = 64 * DEFAULT_TIMER_T1;
- global_t1min = DEFAULT_T1MIN;
- global_qualifyfreq = DEFAULT_QUALIFYFREQ;
- global_t38_maxdatagram = -1;
- global_shrinkcallerid = 1;
- global_refer_addheaders = TRUE;
- authlimit = DEFAULT_AUTHLIMIT;
- authtimeout = DEFAULT_AUTHTIMEOUT;
- global_store_sip_cause = DEFAULT_STORE_SIP_CAUSE;
- min_expiry = DEFAULT_MIN_EXPIRY;
- max_expiry = DEFAULT_MAX_EXPIRY;
- default_expiry = DEFAULT_DEFAULT_EXPIRY;
-
- sip_cfg.matchexternaddrlocally = DEFAULT_MATCHEXTERNADDRLOCALLY;
-
- /* Copy the default jb config over global_jbconf */
- memcpy(&global_jbconf, &default_jbconf, sizeof(struct ast_jb_conf));
-
- ast_clear_flag(&global_flags[1], SIP_PAGE2_FAX_DETECT);
- ast_clear_flag(&global_flags[1], SIP_PAGE2_VIDEOSUPPORT | SIP_PAGE2_VIDEOSUPPORT_ALWAYS);
- ast_clear_flag(&global_flags[1], SIP_PAGE2_TEXTSUPPORT);
- ast_clear_flag(&global_flags[1], SIP_PAGE2_IGNORESDPVERSION);
-
- /* Read the [general] config section of sip.conf (or from realtime config) */
- for (v = ast_variable_browse(cfg, "general"); v; v = v->next) {
- if (handle_common_options(&setflags[0], &mask[0], v)) {
- continue;
- }
- if (handle_t38_options(&setflags[0], &mask[0], v, &global_t38_maxdatagram)) {
- continue;
- }
- /* handle jb conf */
- if (!ast_jb_read_conf(&global_jbconf, v->name, v->value)) {
- continue;
- }
-
- /* Load default dtls configuration */
- ast_rtp_dtls_cfg_parse(&default_dtls_cfg, v->name, v->value);
-
- /* handle tls conf, don't allow setting of tlsverifyclient as it isn't supported by chan_sip */
- if (!strcasecmp(v->name, "tlsverifyclient")) {
- ast_log(LOG_WARNING, "Ignoring unsupported option 'tlsverifyclient'\n");
- continue;
- } else if (!ast_tls_read_conf(&default_tls_cfg, &sip_tls_desc, v->name, v->value)) {
- continue;
- }
-
- if (!strcasecmp(v->name, "context")) {
- ast_copy_string(sip_cfg.default_context, v->value, sizeof(sip_cfg.default_context));
- } else if (!strcasecmp(v->name, "recordonfeature")) {
- ast_copy_string(sip_cfg.default_record_on_feature, v->value, sizeof(sip_cfg.default_record_on_feature));
- } else if (!strcasecmp(v->name, "recordofffeature")) {
- ast_copy_string(sip_cfg.default_record_off_feature, v->value, sizeof(sip_cfg.default_record_off_feature));
- } else if (!strcasecmp(v->name, "subscribecontext")) {
- ast_copy_string(sip_cfg.default_subscribecontext, v->value, sizeof(sip_cfg.default_subscribecontext));
- } else if (!strcasecmp(v->name, "callcounter")) {
- global_callcounter = ast_true(v->value) ? 1 : 0;
- } else if (!strcasecmp(v->name, "allowguest")) {
- sip_cfg.allowguest = ast_true(v->value) ? 1 : 0;
- } else if (!strcasecmp(v->name, "realm")) {
- ast_copy_string(sip_cfg.realm, v->value, sizeof(sip_cfg.realm));
- } else if (!strcasecmp(v->name, "domainsasrealm")) {
- sip_cfg.domainsasrealm = ast_true(v->value);
- } else if (!strcasecmp(v->name, "useragent")) {
- ast_copy_string(global_useragent, v->value, sizeof(global_useragent));
- ast_debug(1, "Setting SIP channel User-Agent Name to %s\n", global_useragent);
- } else if (!strcasecmp(v->name, "sdpsession")) {
- ast_copy_string(global_sdpsession, v->value, sizeof(global_sdpsession));
- } else if (!strcasecmp(v->name, "sdpowner")) {
- /* Field cannot contain spaces */
- if (!strstr(v->value, " ")) {
- ast_copy_string(global_sdpowner, v->value, sizeof(global_sdpowner));
- } else {
- ast_log(LOG_WARNING, "'%s' must not contain spaces at line %d. Using default.\n", v->value, v->lineno);
- }
- } else if (!strcasecmp(v->name, "allowtransfer")) {
- sip_cfg.allowtransfer = ast_true(v->value) ? TRANSFER_OPENFORALL : TRANSFER_CLOSED;
- } else if (!strcasecmp(v->name, "rtcachefriends")) {
- ast_set2_flag(&global_flags[1], ast_true(v->value), SIP_PAGE2_RTCACHEFRIENDS);
- } else if (!strcasecmp(v->name, "rtsavesysname")) {
- sip_cfg.rtsave_sysname = ast_true(v->value);
- } else if (!strcasecmp(v->name, "rtsavepath")) {
- sip_cfg.rtsave_path = ast_true(v->value);
- } else if (!strcasecmp(v->name, "rtupdate")) {
- sip_cfg.peer_rtupdate = ast_true(v->value);
- } else if (!strcasecmp(v->name, "ignoreregexpire")) {
- sip_cfg.ignore_regexpire = ast_true(v->value);
- } else if (!strcasecmp(v->name, "timert1")) {
- /* Defaults to 500ms, but RFC 3261 states that it is recommended
- * for the value to be set higher, though a lower value is only
- * allowed on private networks unconnected to the Internet. */
- global_t1 = atoi(v->value);
- } else if (!strcasecmp(v->name, "timerb")) {
- int tmp = atoi(v->value);
- if (tmp < 500) {
- global_timer_b = global_t1 * 64;
- ast_log(LOG_WARNING, "Invalid value for timerb ('%s'). Setting to default ('%d').\n", v->value, global_timer_b);
- }
- timerb_set = 1;
- } else if (!strcasecmp(v->name, "t1min")) {
- global_t1min = atoi(v->value);
- } else if (!strcasecmp(v->name, "transport")) {
- char *val = ast_strdupa(v->value);
- char *trans;
-
- default_transports = default_primary_transport = 0;
- while ((trans = strsep(&val, ","))) {
- trans = ast_skip_blanks(trans);
-
- if (!strncasecmp(trans, "udp", 3)) {
- default_transports |= AST_TRANSPORT_UDP;
- } else if (!strncasecmp(trans, "tcp", 3)) {
- default_transports |= AST_TRANSPORT_TCP;
- } else if (!strncasecmp(trans, "tls", 3)) {
- default_transports |= AST_TRANSPORT_TLS;
- } else if (!strncasecmp(trans, "wss", 3)) {
- default_transports |= AST_TRANSPORT_WSS;
- } else if (!strncasecmp(trans, "ws", 2)) {
- default_transports |= AST_TRANSPORT_WS;
- } else {
- ast_log(LOG_NOTICE, "'%s' is not a valid transport type. if no other is specified, udp will be used.\n", trans);
- }
- if (default_primary_transport == 0) {
- default_primary_transport = default_transports;
- }
- }
- } else if (!strcasecmp(v->name, "tcpenable")) {
- if (!ast_false(v->value)) {
- ast_debug(2, "Enabling TCP socket for listening\n");
- sip_cfg.tcp_enabled = TRUE;
- }
- } else if (!strcasecmp(v->name, "tcpbindaddr")) {
- if (ast_parse_arg(v->value, PARSE_ADDR,
- &sip_tcp_desc.local_address)) {
- ast_log(LOG_WARNING, "Invalid %s '%s' at line %d of %s\n",
- v->name, v->value, v->lineno, config);
- }
- ast_debug(2, "Setting TCP socket address to %s\n",
- ast_sockaddr_stringify(&sip_tcp_desc.local_address));
- } else if (!strcasecmp(v->name, "dynamic_exclude_static") || !strcasecmp(v->name, "dynamic_excludes_static")) {
- global_dynamic_exclude_static = ast_true(v->value);
- } else if (!strcasecmp(v->name, "contactpermit") || !strcasecmp(v->name, "contactdeny") || !strcasecmp(v->name, "contactacl")) {
- int ha_error = 0;
- ast_append_acl(v->name + 7, v->value, &sip_cfg.contact_acl, &ha_error, &acl_change_subscription_needed);
- if (ha_error) {
- ast_log(LOG_ERROR, "Bad ACL entry in configuration line %d : %s. Failing to load chan_sip.so\n", v->lineno, v->value);
- return -1;
- }
- } else if (!strcasecmp(v->name, "rtautoclear")) {
- int i = atoi(v->value);
- if (i > 0) {
- sip_cfg.rtautoclear = i;
- } else {
- i = 0;
- }
- ast_set2_flag(&global_flags[1], i || ast_true(v->value), SIP_PAGE2_RTAUTOCLEAR);
- } else if (!strcasecmp(v->name, "usereqphone")) {
- ast_set2_flag(&global_flags[0], ast_true(v->value), SIP_USEREQPHONE);
- } else if (!strcasecmp(v->name, "prematuremedia")) {
- global_prematuremediafilter = ast_true(v->value);
- } else if (!strcasecmp(v->name, "relaxdtmf")) {
- global_relaxdtmf = ast_true(v->value);
- } else if (!strcasecmp(v->name, "vmexten")) {
- ast_copy_string(default_vmexten, v->value, sizeof(default_vmexten));
- } else if (!strcasecmp(v->name, "rtptimeout")) {
- if ((sscanf(v->value, "%30d", &global_rtptimeout) != 1) || (global_rtptimeout < 0)) {
- ast_log(LOG_WARNING, "'%s' is not a valid RTP hold time at line %d. Using default.\n", v->value, v->lineno);
- global_rtptimeout = 0;
- }
- } else if (!strcasecmp(v->name, "rtpholdtimeout")) {
- if ((sscanf(v->value, "%30d", &global_rtpholdtimeout) != 1) || (global_rtpholdtimeout < 0)) {
- ast_log(LOG_WARNING, "'%s' is not a valid RTP hold time at line %d. Using default.\n", v->value, v->lineno);
- global_rtpholdtimeout = 0;
- }
- } else if (!strcasecmp(v->name, "rtpkeepalive")) {
- if ((sscanf(v->value, "%30d", &global_rtpkeepalive) != 1) || (global_rtpkeepalive < 0)) {
- ast_log(LOG_WARNING, "'%s' is not a valid RTP keepalive time at line %d. Using default.\n", v->value, v->lineno);
- global_rtpkeepalive = DEFAULT_RTPKEEPALIVE;
- }
- } else if (!strcasecmp(v->name, "compactheaders")) {
- sip_cfg.compactheaders = ast_true(v->value);
- } else if (!strcasecmp(v->name, "notifymimetype")) {
- ast_copy_string(default_notifymime, v->value, sizeof(default_notifymime));
- } else if (!strcasecmp(v->name, "directrtpsetup")) {
- sip_cfg.directrtpsetup = ast_true(v->value);
- } else if (!strcasecmp(v->name, "notifyringing")) {
- if (!strcasecmp(v->value, "notinuse")) {
- sip_cfg.notifyringing = NOTIFYRINGING_NOTINUSE;
- } else {
- sip_cfg.notifyringing = ast_true(v->value) ? NOTIFYRINGING_ENABLED : NOTIFYRINGING_DISABLED;
- }
- } else if (!strcasecmp(v->name, "notifyhold")) {
- sip_cfg.notifyhold = ast_true(v->value);
- } else if (!strcasecmp(v->name, "notifycid")) {
- if (!strcasecmp(v->value, "ignore-context")) {
- sip_cfg.notifycid = IGNORE_CONTEXT;
- } else {
- sip_cfg.notifycid = ast_true(v->value) ? ENABLED : DISABLED;
- }
- } else if (!strcasecmp(v->name, "alwaysauthreject")) {
- sip_cfg.alwaysauthreject = ast_true(v->value);
- } else if (!strcasecmp(v->name, "auth_options_requests")) {
- if (ast_true(v->value)) {
- sip_cfg.auth_options_requests = 1;
- }
- } else if (!strcasecmp(v->name, "auth_message_requests")) {
- sip_cfg.auth_message_requests = ast_true(v->value) ? 1 : 0;
- } else if (!strcasecmp(v->name, "accept_outofcall_message")) {
- sip_cfg.accept_outofcall_message = ast_true(v->value) ? 1 : 0;
- } else if (!strcasecmp(v->name, "outofcall_message_context")) {
- ast_copy_string(sip_cfg.messagecontext, v->value, sizeof(sip_cfg.messagecontext));
- } else if (!strcasecmp(v->name, "mohinterpret")) {
- ast_copy_string(default_mohinterpret, v->value, sizeof(default_mohinterpret));
- } else if (!strcasecmp(v->name, "mohsuggest")) {
- ast_copy_string(default_mohsuggest, v->value, sizeof(default_mohsuggest));
- } else if (!strcasecmp(v->name, "tonezone")) {
- struct ast_tone_zone *new_zone;
- if (!(new_zone = ast_get_indication_zone(v->value))) {
- ast_log(LOG_ERROR, "Unknown country code '%s' for tonezone in [general] at line %d. Check indications.conf for available country codes.\n", v->value, v->lineno);
- } else {
- ast_tone_zone_unref(new_zone);
- ast_copy_string(default_zone, v->value, sizeof(default_zone));
- }
- } else if (!strcasecmp(v->name, "language")) {
- ast_copy_string(default_language, v->value, sizeof(default_language));
- } else if (!strcasecmp(v->name, "regcontext")) {
- ast_copy_string(newcontexts, v->value, sizeof(newcontexts));
- stringp = newcontexts;
- /* Let's remove any contexts that are no longer defined in regcontext */
- cleanup_stale_contexts(stringp, oldregcontext);
- /* Create contexts if they don't exist already */
- while ((context = strsep(&stringp, "&"))) {
- ast_copy_string(used_context, context, sizeof(used_context));
- ast_context_find_or_create(NULL, NULL, context, "SIP");
- }
- ast_copy_string(sip_cfg.regcontext, v->value, sizeof(sip_cfg.regcontext));
- } else if (!strcasecmp(v->name, "regextenonqualify")) {
- sip_cfg.regextenonqualify = ast_true(v->value);
- } else if (!strcasecmp(v->name, "legacy_useroption_parsing")) {
- sip_cfg.legacy_useroption_parsing = ast_true(v->value);
- } else if (!strcasecmp(v->name, "send_diversion")) {
- sip_cfg.send_diversion = ast_true(v->value);
- } else if (!strcasecmp(v->name, "callerid")) {
- ast_copy_string(default_callerid, v->value, sizeof(default_callerid));
- } else if (!strcasecmp(v->name, "mwi_from")) {
- ast_copy_string(default_mwi_from, v->value, sizeof(default_mwi_from));
- } else if (!strcasecmp(v->name, "fromdomain")) {
- char *fromdomainport;
- ast_copy_string(default_fromdomain, v->value, sizeof(default_fromdomain));
- if ((fromdomainport = strchr(default_fromdomain, ':'))) {
- *fromdomainport++ = '\0';
- if (!(default_fromdomainport = port_str2int(fromdomainport, 0))) {
- ast_log(LOG_NOTICE, "'%s' is not a valid port number for fromdomain.\n",fromdomainport);
- }
- } else {
- default_fromdomainport = STANDARD_SIP_PORT;
- }
- } else if (!strcasecmp(v->name, "outboundproxy")) {
- struct sip_proxy *proxy;
- if (ast_strlen_zero(v->value)) {
- ast_log(LOG_WARNING, "no value given for outbound proxy on line %d of sip.conf\n", v->lineno);
- continue;
- }
- proxy = proxy_from_config(v->value, v->lineno, &sip_cfg.outboundproxy);
- if (!proxy) {
- ast_log(LOG_WARNING, "failure parsing the outbound proxy on line %d of sip.conf.\n", v->lineno);
- continue;
- }
- } else if (!strcasecmp(v->name, "autocreatepeer")) {
- if (!strcasecmp(v->value, "persist")) {
- sip_cfg.autocreatepeer = AUTOPEERS_PERSIST;
- } else {
- sip_cfg.autocreatepeer = ast_true(v->value) ? AUTOPEERS_VOLATILE : AUTOPEERS_DISABLED;
- }
- } else if (!strcasecmp(v->name, "match_auth_username")) {
- global_match_auth_username = ast_true(v->value);
- } else if (!strcasecmp(v->name, "srvlookup")) {
- sip_cfg.srvlookup = ast_true(v->value);
- } else if (!strcasecmp(v->name, "pedantic")) {
- sip_cfg.pedanticsipchecking = ast_true(v->value);
- } else if (!strcasecmp(v->name, "maxexpirey") || !strcasecmp(v->name, "maxexpiry")) {
- max_expiry = atoi(v->value);
- if (max_expiry < 1) {
- max_expiry = DEFAULT_MAX_EXPIRY;
- }
- } else if (!strcasecmp(v->name, "minexpirey") || !strcasecmp(v->name, "minexpiry")) {
- min_expiry = atoi(v->value);
- if (min_expiry < 1) {
- min_expiry = DEFAULT_MIN_EXPIRY;
- }
- } else if (!strcasecmp(v->name, "defaultexpiry") || !strcasecmp(v->name, "defaultexpirey")) {
- default_expiry = atoi(v->value);
- if (default_expiry < 1) {
- default_expiry = DEFAULT_DEFAULT_EXPIRY;
- }
- } else if (!strcasecmp(v->name, "submaxexpirey") || !strcasecmp(v->name, "submaxexpiry")) {
- max_subexpiry = atoi(v->value);
- if (max_subexpiry < 1) {
- max_subexpiry = DEFAULT_MAX_EXPIRY;
- }
- max_subexpiry_set = 1;
- } else if (!strcasecmp(v->name, "subminexpirey") || !strcasecmp(v->name, "subminexpiry")) {
- min_subexpiry = atoi(v->value);
- if (min_subexpiry < 1) {
- min_subexpiry = DEFAULT_MIN_EXPIRY;
- }
- min_subexpiry_set = 1;
- } else if (!strcasecmp(v->name, "mwiexpiry") || !strcasecmp(v->name, "mwiexpirey")) {
- mwi_expiry = atoi(v->value);
- if (mwi_expiry < 1) {
- mwi_expiry = DEFAULT_MWI_EXPIRY;
- }
- } else if (!strcasecmp(v->name, "tcpauthtimeout")) {
- if (ast_parse_arg(v->value, PARSE_INT32|PARSE_DEFAULT|PARSE_IN_RANGE,
- &authtimeout, DEFAULT_AUTHTIMEOUT, 1, INT_MAX)) {
- ast_log(LOG_WARNING, "Invalid %s '%s' at line %d of %s\n",
- v->name, v->value, v->lineno, config);
- }
- } else if (!strcasecmp(v->name, "tcpauthlimit")) {
- if (ast_parse_arg(v->value, PARSE_INT32|PARSE_DEFAULT|PARSE_IN_RANGE,
- &authlimit, DEFAULT_AUTHLIMIT, 1, INT_MAX)) {
- ast_log(LOG_WARNING, "Invalid %s '%s' at line %d of %s\n",
- v->name, v->value, v->lineno, config);
- }
- } else if (!strcasecmp(v->name, "sipdebug")) {
- if (ast_true(v->value))
- sipdebug |= sip_debug_config;
- } else if (!strcasecmp(v->name, "dumphistory")) {
- dumphistory = ast_true(v->value);
- } else if (!strcasecmp(v->name, "recordhistory")) {
- recordhistory = ast_true(v->value);
- } else if (!strcasecmp(v->name, "registertimeout")) {
- global_reg_timeout = atoi(v->value);
- if (global_reg_timeout < 1) {
- global_reg_timeout = DEFAULT_REGISTRATION_TIMEOUT;
- }
- } else if (!strcasecmp(v->name, "registerattempts")) {
- global_regattempts_max = atoi(v->value);
- } else if (!strcasecmp(v->name, "register_retry_403")) {
- global_reg_retry_403 = ast_true(v->value);
- } else if (!strcasecmp(v->name, "bindaddr") || !strcasecmp(v->name, "udpbindaddr")) {
- if (ast_parse_arg(v->value, PARSE_ADDR, &bindaddr)) {
- ast_log(LOG_WARNING, "Invalid address: %s\n", v->value);
- }
- } else if (!strcasecmp(v->name, "localnet")) {
- struct ast_ha *na;
- int ha_error = 0;
-
- if (!(na = ast_append_ha("d", v->value, localaddr, &ha_error))) {
- ast_log(LOG_WARNING, "Invalid localnet value: %s\n", v->value);
- } else {
- localaddr = na;
- }
- if (ha_error) {
- ast_log(LOG_ERROR, "Bad localnet configuration value line %d : %s\n", v->lineno, v->value);
- }
- } else if (!strcasecmp(v->name, "media_address")) {
- if (ast_parse_arg(v->value, PARSE_ADDR, &media_address))
- ast_log(LOG_WARNING, "Invalid address for media_address keyword: %s\n", v->value);
- } else if (!strcasecmp(v->name, "rtpbindaddr")) {
- if (ast_parse_arg(v->value, PARSE_ADDR, &rtpbindaddr)) {
- ast_log(LOG_WARNING, "Invalid address for rtpbindaddr keyword: %s\n", v->value);
- }
- } else if (!strcasecmp(v->name, "externaddr") || !strcasecmp(v->name, "externip")) {
- if (ast_parse_arg(v->value, PARSE_ADDR, &externaddr)) {
- ast_log(LOG_WARNING,
- "Invalid address for externaddr keyword: %s\n",
- v->value);
- }
- externexpire = 0;
- } else if (!strcasecmp(v->name, "externhost")) {
- ast_copy_string(externhost, v->value, sizeof(externhost));
- if (ast_sockaddr_resolve_first_af(&externaddr, externhost, 0, AST_AF_INET)) {
- ast_log(LOG_WARNING, "Invalid address for externhost keyword: %s\n", externhost);
- }
- externexpire = time(NULL);
- } else if (!strcasecmp(v->name, "externrefresh")) {
- if (sscanf(v->value, "%30d", &externrefresh) != 1) {
- ast_log(LOG_WARNING, "Invalid externrefresh value '%s', must be an integer >0 at line %d\n", v->value, v->lineno);
- externrefresh = 10;
- }
- } else if (!strcasecmp(v->name, "externtcpport")) {
- if (!(externtcpport = port_str2int(v->value, 0))) {
- ast_log(LOG_WARNING, "Invalid externtcpport value, must be a positive integer between 1 and 65535 at line %d\n", v->lineno);
- }
- } else if (!strcasecmp(v->name, "externtlsport")) {
- if (!(externtlsport = port_str2int(v->value, 0))) {
- ast_log(LOG_WARNING, "Invalid externtlsport value, must be a positive integer between 1 and 65535 at line %d\n", v->lineno);
- }
- } else if (!strcasecmp(v->name, "allow")) {
- int error = ast_format_cap_update_by_allow_disallow(sip_cfg.caps, v->value, TRUE);
- if (error) {
- ast_log(LOG_WARNING, "Codec configuration errors found in line %d : %s = %s\n", v->lineno, v->name, v->value);
- }
- } else if (!strcasecmp(v->name, "disallow")) {
- int error = ast_format_cap_update_by_allow_disallow(sip_cfg.caps, v->value, FALSE);
- if (error) {
- ast_log(LOG_WARNING, "Codec configuration errors found in line %d : %s = %s\n", v->lineno, v->name, v->value);
- }
- } else if (!strcasecmp(v->name, "preferred_codec_only")) {
- ast_set2_flag(&global_flags[1], ast_true(v->value), SIP_PAGE2_PREFERRED_CODEC);
- } else if (!strcasecmp(v->name, "autoframing")) {
- global_autoframing = ast_true(v->value);
- } else if (!strcasecmp(v->name, "allowexternaldomains")) {
- sip_cfg.allow_external_domains = ast_true(v->value);
- } else if (!strcasecmp(v->name, "autodomain")) {
- auto_sip_domains = ast_true(v->value);
- } else if (!strcasecmp(v->name, "domain")) {
- char *domain = ast_strdupa(v->value);
- char *cntx = strchr(domain, ',');
-
- if (cntx) {
- *cntx++ = '\0';
- }
-
- if (ast_strlen_zero(cntx)) {
- ast_debug(1, "No context specified at line %d for domain '%s'\n", v->lineno, domain);
- }
- if (ast_strlen_zero(domain)) {
- ast_log(LOG_WARNING, "Empty domain specified at line %d\n", v->lineno);
- } else {
- add_sip_domain(ast_strip(domain), SIP_DOMAIN_CONFIG, cntx ? ast_strip(cntx) : "");
- }
- } else if (!strcasecmp(v->name, "register")) {
- if (sip_register(v->value, v->lineno) == 0) {
- registry_count++;
- }
- } else if (!strcasecmp(v->name, "mwi")) {
- sip_subscribe_mwi(v->value, v->lineno);
- } else if (!strcasecmp(v->name, "tos_sip")) {
- if (ast_str2tos(v->value, &global_tos_sip)) {
- ast_log(LOG_WARNING, "Invalid tos_sip value at line %d, refer to QoS documentation\n", v->lineno);
- }
- } else if (!strcasecmp(v->name, "tos_audio")) {
- if (ast_str2tos(v->value, &global_tos_audio)) {
- ast_log(LOG_WARNING, "Invalid tos_audio value at line %d, refer to QoS documentation\n", v->lineno);
- }
- } else if (!strcasecmp(v->name, "tos_video")) {
- if (ast_str2tos(v->value, &global_tos_video)) {
- ast_log(LOG_WARNING, "Invalid tos_video value at line %d, refer to QoS documentation\n", v->lineno);
- }
- } else if (!strcasecmp(v->name, "tos_text")) {
- if (ast_str2tos(v->value, &global_tos_text)) {
- ast_log(LOG_WARNING, "Invalid tos_text value at line %d, refer to QoS documentation\n", v->lineno);
- }
- } else if (!strcasecmp(v->name, "cos_sip")) {
- if (ast_str2cos(v->value, &global_cos_sip)) {
- ast_log(LOG_WARNING, "Invalid cos_sip value at line %d, refer to QoS documentation\n", v->lineno);
- }
- } else if (!strcasecmp(v->name, "cos_audio")) {
- if (ast_str2cos(v->value, &global_cos_audio)) {
- ast_log(LOG_WARNING, "Invalid cos_audio value at line %d, refer to QoS documentation\n", v->lineno);
- }
- } else if (!strcasecmp(v->name, "cos_video")) {
- if (ast_str2cos(v->value, &global_cos_video)) {
- ast_log(LOG_WARNING, "Invalid cos_video value at line %d, refer to QoS documentation\n", v->lineno);
- }
- } else if (!strcasecmp(v->name, "cos_text")) {
- if (ast_str2cos(v->value, &global_cos_text)) {
- ast_log(LOG_WARNING, "Invalid cos_text value at line %d, refer to QoS documentation\n", v->lineno);
- }
- } else if (!strcasecmp(v->name, "bindport")) {
- if (sscanf(v->value, "%5d", &bindport) != 1) {
- ast_log(LOG_WARNING, "Invalid port number '%s' at line %d of %s\n", v->value, v->lineno, config);
- }
- } else if (!strcasecmp(v->name, "qualify")) {
- if (!strcasecmp(v->value, "no")) {
- default_qualify = 0;
- } else if (!strcasecmp(v->value, "yes")) {
- default_qualify = DEFAULT_MAXMS;
- } else if (sscanf(v->value, "%30d", &default_qualify) != 1) {
- ast_log(LOG_WARNING, "Qualification default should be 'yes', 'no', or a number of milliseconds at line %d of sip.conf\n", v->lineno);
- default_qualify = 0;
- }
- } else if (!strcasecmp(v->name, "keepalive")) {
- if (!strcasecmp(v->value, "no")) {
- default_keepalive = 0;
- } else if (!strcasecmp(v->value, "yes")) {
- default_keepalive = DEFAULT_KEEPALIVE_INTERVAL;
- } else if (sscanf(v->value, "%30d", &default_keepalive) != 1) {
- ast_log(LOG_WARNING, "Keep alive default should be 'yes', 'no', or a number of milliseconds at line %d of sip.conf\n", v->lineno);
- default_keepalive = 0;
- }
- } else if (!strcasecmp(v->name, "qualifyfreq")) {
- int i;
- if (sscanf(v->value, "%30d", &i) == 1) {
- global_qualifyfreq = i * 1000;
- } else {
- ast_log(LOG_WARNING, "Invalid qualifyfreq number '%s' at line %d of %s\n", v->value, v->lineno, config);
- global_qualifyfreq = DEFAULT_QUALIFYFREQ;
- }
- } else if (!strcasecmp(v->name, "authfailureevents")) {
- global_authfailureevents = ast_true(v->value);
- } else if (!strcasecmp(v->name, "maxcallbitrate")) {
- default_maxcallbitrate = atoi(v->value);
- if (default_maxcallbitrate < 0) {
- default_maxcallbitrate = DEFAULT_MAX_CALL_BITRATE;
- }
- } else if (!strcasecmp(v->name, "matchexternaddrlocally") || !strcasecmp(v->name, "matchexterniplocally")) {
- sip_cfg.matchexternaddrlocally = ast_true(v->value);
- } else if (!strcasecmp(v->name, "session-timers")) {
- int i = (int) str2stmode(v->value);
- if (i < 0) {
- ast_log(LOG_WARNING, "Invalid session-timers '%s' at line %d of %s\n", v->value, v->lineno, config);
- global_st_mode = SESSION_TIMER_MODE_ACCEPT;
- } else {
- global_st_mode = i;
- }
- } else if (!strcasecmp(v->name, "session-expires")) {
- if (sscanf(v->value, "%30d", &global_max_se) != 1) {
- ast_log(LOG_WARNING, "Invalid session-expires '%s' at line %d of %s\n", v->value, v->lineno, config);
- global_max_se = DEFAULT_MAX_SE;
- }
- } else if (!strcasecmp(v->name, "session-minse")) {
- if (sscanf(v->value, "%30d", &global_min_se) != 1) {
- ast_log(LOG_WARNING, "Invalid session-minse '%s' at line %d of %s\n", v->value, v->lineno, config);
- global_min_se = DEFAULT_MIN_SE;
- }
- if (global_min_se < DEFAULT_MIN_SE) {
- ast_log(LOG_WARNING, "session-minse '%s' at line %d of %s is not allowed to be < %d secs\n", v->value, v->lineno, config, DEFAULT_MIN_SE);
- global_min_se = DEFAULT_MIN_SE;
- }
- } else if (!strcasecmp(v->name, "session-refresher")) {
- int i = (int) str2strefresherparam(v->value);
- if (i < 0) {
- ast_log(LOG_WARNING, "Invalid session-refresher '%s' at line %d of %s\n", v->value, v->lineno, config);
- global_st_refresher = SESSION_TIMER_REFRESHER_PARAM_UAS;
- } else {
- global_st_refresher = i;
- }
- } else if (!strcasecmp(v->name, "storesipcause")) {
- global_store_sip_cause = ast_true(v->value);
- if (global_store_sip_cause) {
- ast_log(LOG_WARNING, "Usage of SIP_CAUSE is deprecated. Please use HANGUPCAUSE instead.\n");
- }
- } else if (!strcasecmp(v->name, "qualifygap")) {
- if (sscanf(v->value, "%30d", &global_qualify_gap) != 1
- || global_qualify_gap < 0) {
- ast_log(LOG_WARNING, "Invalid qualifygap '%s' at line %d of %s\n", v->value, v->lineno, config);
- global_qualify_gap = DEFAULT_QUALIFY_GAP;
- }
- } else if (!strcasecmp(v->name, "qualifypeers")) {
- if (sscanf(v->value, "%30d", &global_qualify_peers) != 1) {
- ast_log(LOG_WARNING, "Invalid pokepeers '%s' at line %d of %s\n", v->value, v->lineno, config);
- global_qualify_peers = DEFAULT_QUALIFY_PEERS;
- }
- } else if (!strcasecmp(v->name, "disallowed_methods")) {
- char *disallow = ast_strdupa(v->value);
- mark_parsed_methods(&sip_cfg.disallowed_methods, disallow);
- } else if (!strcasecmp(v->name, "shrinkcallerid")) {
- if (ast_true(v->value)) {
- global_shrinkcallerid = 1;
- } else if (ast_false(v->value)) {
- global_shrinkcallerid = 0;
- } else {
- ast_log(LOG_WARNING, "shrinkcallerid value %s is not valid at line %d.\n", v->value, v->lineno);
- }
- } else if (!strcasecmp(v->name, "use_q850_reason")) {
- ast_set2_flag(&global_flags[1], ast_true(v->value), SIP_PAGE2_Q850_REASON);
- } else if (!strcasecmp(v->name, "maxforwards")) {
- if (sscanf(v->value, "%30d", &sip_cfg.default_max_forwards) != 1
- || sip_cfg.default_max_forwards < 1 || 255 < sip_cfg.default_max_forwards) {
- ast_log(LOG_WARNING, "'%s' is not a valid maxforwards value at line %d. Using default.\n", v->value, v->lineno);
- sip_cfg.default_max_forwards = DEFAULT_MAX_FORWARDS;
- }
- } else if (!strcasecmp(v->name, "subscribe_network_change_event")) {
- if (ast_true(v->value)) {
- subscribe_network_change = 1;
- } else if (ast_false(v->value)) {
- subscribe_network_change = 0;
- } else {
- ast_log(LOG_WARNING, "subscribe_network_change_event value %s is not valid at line %d.\n", v->value, v->lineno);
- }
- } else if (!strcasecmp(v->name, "snom_aoc_enabled")) {
- ast_set2_flag(&global_flags[2], ast_true(v->value), SIP_PAGE3_SNOM_AOC);
- } else if (!strcasecmp(v->name, "icesupport")) {
- ast_set2_flag(&global_flags[2], ast_true(v->value), SIP_PAGE3_ICE_SUPPORT);
- } else if (!strcasecmp(v->name, "discard_remote_hold_retrieval")) {
- ast_set2_flag(&global_flags[2], ast_true(v->value), SIP_PAGE3_DISCARD_REMOTE_HOLD_RETRIEVAL);
- } else if (!strcasecmp(v->name, "parkinglot")) {
- ast_copy_string(default_parkinglot, v->value, sizeof(default_parkinglot));
- } else if (!strcasecmp(v->name, "refer_addheaders")) {
- global_refer_addheaders = ast_true(v->value);
- } else if (!strcasecmp(v->name, "websocket_write_timeout")) {
- if (sscanf(v->value, "%30d", &sip_cfg.websocket_write_timeout) != 1
- || sip_cfg.websocket_write_timeout < 0) {
- ast_log(LOG_WARNING, "'%s' is not a valid websocket_write_timeout value at line %d. Using default '%d'.\n", v->value, v->lineno, AST_DEFAULT_WEBSOCKET_WRITE_TIMEOUT);
- sip_cfg.websocket_write_timeout = AST_DEFAULT_WEBSOCKET_WRITE_TIMEOUT;
- }
- } else if (!strcasecmp(v->name, "websocket_enabled")) {
- sip_cfg.websocket_enabled = ast_true(v->value);
- }
- }
-
- /* Validate DTLS configuration */
- if (ast_rtp_dtls_cfg_validate(&default_dtls_cfg)) {
- return -1;
- }
-
- /* Override global defaults if setting found in general section */
- ast_copy_flags(&global_flags[0], &setflags[0], mask[0].flags);
- ast_copy_flags(&global_flags[1], &setflags[1], mask[1].flags);
- ast_copy_flags(&global_flags[2], &setflags[2], mask[2].flags);
-
- /* For backwards compatibility the corresponding registration timer value is used if subscription timer value isn't set by configuration */
- if (!min_subexpiry_set) {
- min_subexpiry = min_expiry;
- }
- if (!max_subexpiry_set) {
- max_subexpiry = max_expiry;
- }
-
- if (reason != CHANNEL_MODULE_LOAD && sip_cfg.autocreatepeer != AUTOPEERS_PERSIST) {
- ao2_t_callback(peers, OBJ_NODATA, peer_markall_autopeers_func, NULL, "callback to mark autopeers for destruction");
- }
-
- if (subscribe_network_change) {
- network_change_stasis_subscribe();
- } else {
- network_change_stasis_unsubscribe();
- }
-
- if (global_t1 < global_t1min) {
- ast_log(LOG_WARNING, "'t1min' (%d) cannot be greater than 't1timer' (%d). Resetting 't1timer' to the value of 't1min'\n", global_t1min, global_t1);
- global_t1 = global_t1min;
- }
-
- if (global_timer_b < global_t1 * 64) {
- if (timerb_set && timert1_set) {
- ast_log(LOG_WARNING, "Timer B has been set lower than recommended (%d < 64 * timert1=%d). (RFC 3261, 17.1.1.2)\n", global_timer_b, global_t1);
- } else if (timerb_set) {
- if ((global_t1 = global_timer_b / 64) < global_t1min) {
- ast_log(LOG_WARNING, "Timer B has been set lower than recommended (%d < 64 * timert1=%d). (RFC 3261, 17.1.1.2)\n", global_timer_b, global_t1);
- global_t1 = global_t1min;
- global_timer_b = global_t1 * 64;
- }
- } else {
- global_timer_b = global_t1 * 64;
- }
- }
- if (!sip_cfg.allow_external_domains && AST_LIST_EMPTY(&domain_list)) {
- ast_log(LOG_WARNING, "To disallow external domains, you need to configure local SIP domains.\n");
- sip_cfg.allow_external_domains = 1;
- }
- /* If not or badly configured, set default transports */
- if (!sip_cfg.tcp_enabled && (default_transports & AST_TRANSPORT_TCP)) {
- ast_log(LOG_WARNING, "Cannot use 'tcp' transport with tcpenable=no. Removing from available transports.\n");
- default_primary_transport &= ~AST_TRANSPORT_TCP;
- default_transports &= ~AST_TRANSPORT_TCP;
- }
- if (!default_tls_cfg.enabled && (default_transports & AST_TRANSPORT_TLS)) {
- ast_log(LOG_WARNING, "Cannot use 'tls' transport with tlsenable=no. Removing from available transports.\n");
- default_primary_transport &= ~AST_TRANSPORT_TLS;
- default_transports &= ~AST_TRANSPORT_TLS;
- }
- if (!default_transports) {
- ast_log(LOG_WARNING, "No valid transports available, falling back to 'udp'.\n");
- default_transports = default_primary_transport = AST_TRANSPORT_UDP;
- } else if (!default_primary_transport) {
- ast_log(LOG_WARNING, "No valid default transport. Selecting 'udp' as default.\n");
- default_primary_transport = AST_TRANSPORT_UDP;
- }
-
- /* Build list of authentication to various SIP realms, i.e. service providers */
- for (v = ast_variable_browse(cfg, "authentication"); v ; v = v->next) {
- /* Format for authentication is auth = username:password@realm */
- if (!strcasecmp(v->name, "auth")) {
- add_realm_authentication(&authl, v->value, v->lineno);
- }
- }
-
- if (bindport) {
- if (ast_sockaddr_port(&bindaddr)) {
- ast_log(LOG_WARNING, "bindport is also specified in bindaddr. "
- "Using %d.\n", bindport);
- }
- ast_sockaddr_set_port(&bindaddr, bindport);
- }
-
- if (!ast_sockaddr_port(&bindaddr)) {
- ast_sockaddr_set_port(&bindaddr, STANDARD_SIP_PORT);
- }
-
- /* Set UDP address and open socket */
- ast_sockaddr_copy(&internip, &bindaddr);
- if (ast_find_ourip(&internip, &bindaddr, 0)) {
- ast_log(LOG_WARNING, "Unable to get own IP address, SIP disabled\n");
- ast_config_destroy(cfg);
- return 0;
- }
-
- ast_mutex_lock(&netlock);
- if ((sipsock > -1) && (ast_sockaddr_cmp(&old_bindaddr, &bindaddr))) {
- close(sipsock);
- sipsock = -1;
- }
- if (sipsock < 0) {
- sipsock = socket(ast_sockaddr_is_ipv6(&bindaddr) ?
- AF_INET6 : AF_INET, SOCK_DGRAM, 0);
- if (sipsock < 0) {
- ast_log(LOG_WARNING, "Unable to create SIP socket: %s\n", strerror(errno));
- ast_config_destroy(cfg);
- ast_mutex_unlock(&netlock);
- return -1;
- } else {
- /* Allow SIP clients on the same host to access us: */
- const int reuseFlag = 1;
-
- setsockopt(sipsock, SOL_SOCKET, SO_REUSEADDR,
- (const char*)&reuseFlag,
- sizeof reuseFlag);
-
- ast_enable_packet_fragmentation(sipsock);
-
- if (ast_bind(sipsock, &bindaddr) < 0) {
- ast_log(LOG_WARNING, "Failed to bind to %s: %s\n",
- ast_sockaddr_stringify(&bindaddr), strerror(errno));
- close(sipsock);
- sipsock = -1;
- } else {
- ast_verb(2, "SIP Listening on %s\n", ast_sockaddr_stringify(&bindaddr));
- ast_set_qos(sipsock, global_tos_sip, global_cos_sip, "SIP");
- }
- }
- } else {
- ast_set_qos(sipsock, global_tos_sip, global_cos_sip, "SIP");
- }
- ast_mutex_unlock(&netlock);
-
- /* Start TCP server */
- if (sip_cfg.tcp_enabled) {
- if (ast_sockaddr_isnull(&sip_tcp_desc.local_address)) {
- ast_sockaddr_copy(&sip_tcp_desc.local_address, &bindaddr);
- }
- if (!ast_sockaddr_port(&sip_tcp_desc.local_address)) {
- ast_sockaddr_set_port(&sip_tcp_desc.local_address, STANDARD_SIP_PORT);
- }
- } else {
- ast_sockaddr_setnull(&sip_tcp_desc.local_address);
- }
- ast_tcptls_server_start(&sip_tcp_desc);
- if (sip_cfg.tcp_enabled && sip_tcp_desc.accept_fd == -1) {
- /* TCP server start failed. Tell the admin */
- ast_log(LOG_ERROR, "SIP TCP Server start failed. Not listening on TCP socket.\n");
- } else {
- ast_debug(2, "SIP TCP server started\n");
- if (sip_tcp_desc.accept_fd >= 0) {
- int flags = 1;
- if (setsockopt(sip_tcp_desc.accept_fd, SOL_SOCKET, SO_KEEPALIVE, &flags, sizeof(flags))) {
- ast_log(LOG_ERROR, "Error enabling TCP keep-alive on sip socket: %s\n", strerror(errno));
- }
- ast_set_qos(sip_tcp_desc.accept_fd, global_tos_sip, global_cos_sip, "SIP");
- }
- }
-
- /* Start TLS server if needed */
- memcpy(sip_tls_desc.tls_cfg, &default_tls_cfg, sizeof(default_tls_cfg));
-
- if (ast_ssl_setup(sip_tls_desc.tls_cfg)) {
- if (ast_sockaddr_isnull(&sip_tls_desc.local_address)) {
- ast_sockaddr_copy(&sip_tls_desc.local_address, &bindaddr);
- ast_sockaddr_set_port(&sip_tls_desc.local_address,
- STANDARD_TLS_PORT);
- }
- if (!ast_sockaddr_port(&sip_tls_desc.local_address)) {
- ast_sockaddr_set_port(&sip_tls_desc.local_address,
- STANDARD_TLS_PORT);
- }
- ast_tcptls_server_start(&sip_tls_desc);
- if (default_tls_cfg.enabled && sip_tls_desc.accept_fd == -1) {
- ast_log(LOG_ERROR, "TLS Server start failed. Not listening on TLS socket.\n");
- sip_tls_desc.tls_cfg = NULL;
- }
- if (sip_tls_desc.accept_fd >= 0) {
- int flags = 1;
- if (setsockopt(sip_tls_desc.accept_fd, SOL_SOCKET, SO_KEEPALIVE, &flags, sizeof(flags))) {
- ast_log(LOG_ERROR, "Error enabling TCP keep-alive on sip socket: %s\n", strerror(errno));
- sip_tls_desc.tls_cfg = NULL;
- }
- ast_set_qos(sip_tls_desc.accept_fd, global_tos_sip, global_cos_sip, "SIP");
- }
- } else if (sip_tls_desc.tls_cfg->enabled) {
- sip_tls_desc.tls_cfg = NULL;
- ast_log(LOG_WARNING, "SIP TLS server did not load because of errors.\n");
- }
-
- if (ucfg) {
- struct ast_variable *gen;
- int genhassip, genregistersip;
- const char *hassip, *registersip;
-
- genhassip = ast_true(ast_variable_retrieve(ucfg, "general", "hassip"));
- genregistersip = ast_true(ast_variable_retrieve(ucfg, "general", "registersip"));
- gen = ast_variable_browse(ucfg, "general");
- cat = ast_category_browse(ucfg, NULL);
- while (cat) {
- if (strcasecmp(cat, "general")) {
- hassip = ast_variable_retrieve(ucfg, cat, "hassip");
- registersip = ast_variable_retrieve(ucfg, cat, "registersip");
- if (ast_true(hassip) || (!hassip && genhassip)) {
- peer = build_peer(cat, gen, ast_variable_browse(ucfg, cat), 0, 0);
- if (peer) {
- /* user.conf entries are always of type friend */
- peer->type = SIP_TYPE_USER | SIP_TYPE_PEER;
- ao2_t_link(peers, peer, "link peer into peer table");
- if ((peer->type & SIP_TYPE_PEER) && !ast_sockaddr_isnull(&peer->addr)) {
- ao2_t_link(peers_by_ip, peer, "link peer into peers_by_ip table");
- }
-
- sip_unref_peer(peer, "sip_unref_peer: from reload_config");
- peer_count++;
- }
- }
- if (ast_true(registersip) || (!registersip && genregistersip)) {
- char tmp[256];
- const char *host = ast_variable_retrieve(ucfg, cat, "host");
- const char *username = ast_variable_retrieve(ucfg, cat, "username");
- const char *secret = ast_variable_retrieve(ucfg, cat, "secret");
- const char *contact = ast_variable_retrieve(ucfg, cat, "contact");
- const char *authuser = ast_variable_retrieve(ucfg, cat, "authuser");
- if (!host) {
- host = ast_variable_retrieve(ucfg, "general", "host");
- }
- if (!username) {
- username = ast_variable_retrieve(ucfg, "general", "username");
- }
- if (!secret) {
- secret = ast_variable_retrieve(ucfg, "general", "secret");
- }
- if (!contact) {
- contact = "s";
- }
- if (!ast_strlen_zero(username) && !ast_strlen_zero(host)) {
- if (!ast_strlen_zero(secret)) {
- if (!ast_strlen_zero(authuser)) {
- snprintf(tmp, sizeof(tmp), "%s?%s:%s:%s@%s/%s", cat, username, secret, authuser, host, contact);
- } else {
- snprintf(tmp, sizeof(tmp), "%s?%s:%s@%s/%s", cat, username, secret, host, contact);
- }
- } else if (!ast_strlen_zero(authuser)) {
- snprintf(tmp, sizeof(tmp), "%s?%s::%s@%s/%s", cat, username, authuser, host, contact);
- } else {
- snprintf(tmp, sizeof(tmp), "%s?%s@%s/%s", cat, username, host, contact);
- }
- if (sip_register(tmp, 0) == 0) {
- registry_count++;
- }
- }
- }
- }
- cat = ast_category_browse(ucfg, cat);
- }
- ast_config_destroy(ucfg);
- }
-
- /* Load peers, users and friends */
- cat = NULL;
- while ( (cat = ast_category_browse(cfg, cat)) ) {
- const char *utype;
- if (!strcasecmp(cat, "general") || !strcasecmp(cat, "authentication"))
- continue;
- utype = ast_variable_retrieve(cfg, cat, "type");
- if (!utype) {
- ast_log(LOG_WARNING, "Section '%s' lacks type\n", cat);
- continue;
- } else {
- if (!strcasecmp(utype, "user")) {
- ;
- } else if (!strcasecmp(utype, "friend")) {
- ;
- } else if (!strcasecmp(utype, "peer")) {
- ;
- } else {
- ast_log(LOG_WARNING, "Unknown type '%s' for '%s' in %s\n", utype, cat, "sip.conf");
- continue;
- }
- peer = build_peer(cat, ast_variable_browse(cfg, cat), NULL, 0, 0);
- if (peer) {
- display_nat_warning(cat, reason, &peer->flags[0]);
- ao2_t_link(peers, peer, "link peer into peers table");
- if ((peer->type & SIP_TYPE_PEER) && !ast_sockaddr_isnull(&peer->addr)) {
- ao2_t_link(peers_by_ip, peer, "link peer into peers_by_ip table");
- }
- sip_unref_peer(peer, "unref the result of the build_peer call. Now, the links from the tables are the only ones left.");
- peer_count++;
- }
- }
- }
-
- /* Add default domains - host name, IP address and IP:port
- * Only do this if user added any sip domain with "localdomains"
- * In order to *not* break backwards compatibility
- * Some phones address us at IP only, some with additional port number
- */
- if (auto_sip_domains) {
- char temp[MAXHOSTNAMELEN];
-
- /* First our default IP address */
- if (!ast_sockaddr_isnull(&bindaddr) && !ast_sockaddr_is_any(&bindaddr)) {
- add_sip_domain(ast_sockaddr_stringify_addr(&bindaddr),
- SIP_DOMAIN_AUTO, NULL);
- } else if (!ast_sockaddr_isnull(&internip) && !ast_sockaddr_is_any(&internip)) {
- /* Our internal IP address, if configured */
- add_sip_domain(ast_sockaddr_stringify_addr(&internip),
- SIP_DOMAIN_AUTO, NULL);
- } else {
- ast_log(LOG_NOTICE, "Can't add wildcard IP address to domain list, please add IP address to domain manually.\n");
- }
-
- /* If TCP is running on a different IP than UDP, then add it too */
- if (!ast_sockaddr_isnull(&sip_tcp_desc.local_address) &&
- ast_sockaddr_cmp_addr(&bindaddr, &sip_tcp_desc.local_address)) {
- add_sip_domain(ast_sockaddr_stringify_addr(&sip_tcp_desc.local_address),
- SIP_DOMAIN_AUTO, NULL);
- }
-
- /* If TLS is running on a different IP than UDP and TCP, then add that too */
- if (!ast_sockaddr_isnull(&sip_tls_desc.local_address) &&
- ast_sockaddr_cmp_addr(&bindaddr, &sip_tls_desc.local_address) &&
- ast_sockaddr_cmp_addr(&sip_tcp_desc.local_address,
- &sip_tls_desc.local_address)) {
- add_sip_domain(ast_sockaddr_stringify_addr(&sip_tls_desc.local_address),
- SIP_DOMAIN_AUTO, NULL);
- }
-
- /* Our extern IP address, if configured */
- if (!ast_sockaddr_isnull(&externaddr)) {
- add_sip_domain(ast_sockaddr_stringify_addr(&externaddr),
- SIP_DOMAIN_AUTO, NULL);
- }
-
- /* Extern host name (NAT traversal support) */
- if (!ast_strlen_zero(externhost)) {
- add_sip_domain(externhost, SIP_DOMAIN_AUTO, NULL);
- }
-
- /* Our host name */
- if (!gethostname(temp, sizeof(temp))) {
- add_sip_domain(temp, SIP_DOMAIN_AUTO, NULL);
- }
- }
-
- /* Release configuration from memory */
- ast_config_destroy(cfg);
-
- register_realtime_peers_with_callbackextens();
-
- /* Load the list of manual NOTIFY types to support */
- if (notify_types) {
- ast_config_destroy(notify_types);
- }
- if ((notify_types = ast_config_load(notify_config, config_flags)) == CONFIG_STATUS_FILEINVALID) {
- ast_log(LOG_ERROR, "Contents of %s are invalid and cannot be parsed.\n", notify_config);
- notify_types = NULL;
- }
-
- /* If the module is loading it's not time to enable websockets yet. */
- if (reason != CHANNEL_MODULE_LOAD && websocket_was_enabled != sip_cfg.websocket_enabled) {
- if (sip_cfg.websocket_enabled) {
- ast_websocket_add_protocol("sip", sip_websocket_callback);
- } else {
- ast_websocket_remove_protocol("sip", sip_websocket_callback);
- }
- }
-
- run_end = time(0);
- ast_debug(4, "SIP reload_config done...Runtime= %d sec\n", (int)(run_end-run_start));
-
- /* If an ACL change subscription is needed and doesn't exist, we need one. */
- if (acl_change_subscription_needed) {
- acl_change_stasis_subscribe();
- }
-
- return 0;
-}
-
-static int sip_allow_anyrtp_remote(struct ast_channel *chan1, struct ast_rtp_instance *instance, const char *rtptype)
-{
- struct sip_pvt *p;
- struct ast_acl_list *acl = NULL;
- int res = 1;
-
- if (!(p = ast_channel_tech_pvt(chan1))) {
- return 0;
- }
-
- sip_pvt_lock(p);
- if (p->relatedpeer && p->relatedpeer->directmediaacl) {
- acl = ast_duplicate_acl_list(p->relatedpeer->directmediaacl);
- }
- sip_pvt_unlock(p);
-
- if (!acl) {
- return res;
- }
-
- if (ast_test_flag(&p->flags[0], SIP_DIRECT_MEDIA)) {
- struct ast_sockaddr us = { { 0, }, }, them = { { 0, }, };
-
- ast_rtp_instance_get_requested_target_address(instance, &them);
- ast_rtp_instance_get_local_address(instance, &us);
-
- if (ast_apply_acl(acl, &them, "SIP Direct Media ACL: ") == AST_SENSE_DENY) {
- const char *us_addr = ast_strdupa(ast_sockaddr_stringify(&us));
- const char *them_addr = ast_strdupa(ast_sockaddr_stringify(&them));
-
- ast_debug(3, "Reinvite %s to %s denied by directmedia ACL on %s\n",
- rtptype, them_addr, us_addr);
-
- res = 0;
- }
- }
-
- ast_free_acl_list(acl);
-
- return res;
-}
-
-static int sip_allow_rtp_remote(struct ast_channel *chan1, struct ast_rtp_instance *instance)
-{
- return sip_allow_anyrtp_remote(chan1, instance, "audio");
-}
-
-static int sip_allow_vrtp_remote(struct ast_channel *chan1, struct ast_rtp_instance *instance)
-{
- return sip_allow_anyrtp_remote(chan1, instance, "video");
-}
-
-static enum ast_rtp_glue_result sip_get_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance **instance)
-{
- struct sip_pvt *p = NULL;
- enum ast_rtp_glue_result res = AST_RTP_GLUE_RESULT_LOCAL;
-
- if (!(p = ast_channel_tech_pvt(chan))) {
- return AST_RTP_GLUE_RESULT_FORBID;
- }
-
- sip_pvt_lock(p);
- if (!(p->rtp)) {
- sip_pvt_unlock(p);
- return AST_RTP_GLUE_RESULT_FORBID;
- }
-
- ao2_ref(p->rtp, +1);
- *instance = p->rtp;
-
- if (ast_test_flag(&p->flags[0], SIP_DIRECT_MEDIA)) {
- res = AST_RTP_GLUE_RESULT_REMOTE;
- } else if (ast_test_flag(&p->flags[0], SIP_DIRECT_MEDIA_NAT)) {
- res = AST_RTP_GLUE_RESULT_REMOTE;
- } else if (ast_test_flag(&global_jbconf, AST_JB_FORCED)) {
- res = AST_RTP_GLUE_RESULT_FORBID;
- }
-
- if (ast_test_flag(&p->flags[1], SIP_PAGE2_T38SUPPORT)) {
- switch (p->t38.state) {
- case T38_LOCAL_REINVITE:
- case T38_PEER_REINVITE:
- case T38_ENABLED:
- res = AST_RTP_GLUE_RESULT_LOCAL;
- break;
- case T38_REJECTED:
- default:
- break;
- }
- }
-
- if (p->srtp) {
- res = AST_RTP_GLUE_RESULT_FORBID;
- }
-
- sip_pvt_unlock(p);
-
- return res;
-}
-
-static enum ast_rtp_glue_result sip_get_vrtp_peer(struct ast_channel *chan, struct ast_rtp_instance **instance)
-{
- struct sip_pvt *p = NULL;
- enum ast_rtp_glue_result res = AST_RTP_GLUE_RESULT_FORBID;
-
- if (!(p = ast_channel_tech_pvt(chan))) {
- return AST_RTP_GLUE_RESULT_FORBID;
- }
-
- sip_pvt_lock(p);
- if (!(p->vrtp)) {
- sip_pvt_unlock(p);
- return AST_RTP_GLUE_RESULT_FORBID;
- }
-
- ao2_ref(p->vrtp, +1);
- *instance = p->vrtp;
-
- if (ast_test_flag(&p->flags[0], SIP_DIRECT_MEDIA)) {
- res = AST_RTP_GLUE_RESULT_REMOTE;
- }
-
- sip_pvt_unlock(p);
-
- return res;
-}
-
-static enum ast_rtp_glue_result sip_get_trtp_peer(struct ast_channel *chan, struct ast_rtp_instance **instance)
-{
- struct sip_pvt *p = NULL;
- enum ast_rtp_glue_result res = AST_RTP_GLUE_RESULT_FORBID;
-
- if (!(p = ast_channel_tech_pvt(chan))) {
- return AST_RTP_GLUE_RESULT_FORBID;
- }
-
- sip_pvt_lock(p);
- if (!(p->trtp)) {
- sip_pvt_unlock(p);
- return AST_RTP_GLUE_RESULT_FORBID;
- }
-
- ao2_ref(p->trtp, +1);
- *instance = p->trtp;
-
- if (ast_test_flag(&p->flags[0], SIP_DIRECT_MEDIA)) {
- res = AST_RTP_GLUE_RESULT_REMOTE;
- }
-
- sip_pvt_unlock(p);
-
- return res;
-}
-
-static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance *instance, struct ast_rtp_instance *vinstance, struct ast_rtp_instance *tinstance, const struct ast_format_cap *cap, int nat_active)
-{
- struct sip_pvt *p;
- int changed = 0;
-
- p = ast_channel_tech_pvt(chan);
- if (!p) {
- return -1;
- }
- sip_pvt_lock(p);
- if (p->owner != chan) {
- /* I suppose it could be argued that if this happens it is a bug. */
- ast_debug(1, "The private is not owned by channel %s anymore.\n", ast_channel_name(chan));
- sip_pvt_unlock(p);
- return 0;
- }
-
- /* Disable early RTP bridge */
- if ((instance || vinstance || tinstance) &&
- !ast_channel_is_bridged(chan) &&
- !sip_cfg.directrtpsetup) {
- sip_pvt_unlock(p);
- return 0;
- }
-
- if (p->alreadygone) {
- /* If we're destroyed, don't bother */
- sip_pvt_unlock(p);
- return 0;
- }
-
- /* if this peer cannot handle reinvites of the media stream to devices
- that are known to be behind a NAT, then stop the process now
- */
- if (nat_active && !ast_test_flag(&p->flags[0], SIP_DIRECT_MEDIA_NAT)) {
- sip_pvt_unlock(p);
- return 0;
- }
-
- if (instance) {
- changed |= ast_rtp_instance_get_and_cmp_remote_address(instance, &p->redirip);
-
- if (p->rtp) {
- /* Prevent audio RTCP reads */
- ast_channel_set_fd(chan, SIP_AUDIO_RTCP_FD, -1);
- /* Silence RTCP while audio RTP is inactive */
- ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_RTCP, AST_RTP_INSTANCE_RTCP_DISABLED);
- }
- } else if (!ast_sockaddr_isnull(&p->redirip)) {
- memset(&p->redirip, 0, sizeof(p->redirip));
- changed = 1;
- }
-
- if (vinstance) {
- changed |= ast_rtp_instance_get_and_cmp_remote_address(vinstance, &p->vredirip);
-
- if (p->vrtp) {
- /* Prevent video RTCP reads */
- ast_channel_set_fd(chan, SIP_VIDEO_RTCP_FD, -1);
- /* Silence RTCP while video RTP is inactive */
- ast_rtp_instance_set_prop(p->vrtp, AST_RTP_PROPERTY_RTCP, AST_RTP_INSTANCE_RTCP_DISABLED);
- }
- } else if (!ast_sockaddr_isnull(&p->vredirip)) {
- memset(&p->vredirip, 0, sizeof(p->vredirip));
- changed = 1;
-
- if (p->vrtp) {
- /* Enable RTCP since it will be inactive if we're coming back
- * from a reinvite */
- ast_rtp_instance_set_prop(p->vrtp, AST_RTP_PROPERTY_RTCP, AST_RTP_INSTANCE_RTCP_STANDARD);
- /* Enable video RTCP reads */
- ast_channel_set_fd(chan, SIP_VIDEO_RTCP_FD, ast_rtp_instance_fd(p->vrtp, 1));
- }
- }
-
- if (tinstance) {
- changed |= ast_rtp_instance_get_and_cmp_remote_address(tinstance, &p->tredirip);
- } else if (!ast_sockaddr_isnull(&p->tredirip)) {
- memset(&p->tredirip, 0, sizeof(p->tredirip));
- changed = 1;
- }
- if (cap && ast_format_cap_count(cap) && !ast_format_cap_identical(cap, p->redircaps)) {
- ast_format_cap_remove_by_type(p->redircaps, AST_MEDIA_TYPE_UNKNOWN);
- ast_format_cap_append_from_cap(p->redircaps, cap, AST_MEDIA_TYPE_UNKNOWN);
- changed = 1;
- }
-
- if (ast_test_flag(&p->flags[2], SIP_PAGE3_DIRECT_MEDIA_OUTGOING) && !p->outgoing_call) {
- /* We only wish to withhold sending the initial direct media reinvite on the incoming dialog.
- * Further direct media reinvites beyond the initial should be sent. In order to allow further
- * direct media reinvites to be sent, we clear this flag.
- */
- ast_clear_flag(&p->flags[2], SIP_PAGE3_DIRECT_MEDIA_OUTGOING);
- sip_pvt_unlock(p);
- return 0;
- }
-
- if (changed && !ast_test_flag(&p->flags[0], SIP_GOTREFER) && !ast_test_flag(&p->flags[0], SIP_DEFER_BYE_ON_TRANSFER)) {
- if (ast_channel_state(chan) != AST_STATE_UP) { /* We are in early state */
- if (p->do_history)
- append_history(p, "ExtInv", "Initial invite sent with remote bridge proposal.");
- ast_debug(1, "Early remote bridge setting SIP '%s' - Sending media to %s\n", p->callid, ast_sockaddr_stringify(instance ? &p->redirip : &p->ourip));
- } else if (!p->pendinginvite) { /* We are up, and have no outstanding invite */
- ast_debug(3, "Sending reinvite on SIP '%s' - It's audio soon redirected to IP %s\n", p->callid, ast_sockaddr_stringify(instance ? &p->redirip : &p->ourip));
- transmit_reinvite_with_sdp(p, FALSE, FALSE);
- } else if (!ast_test_flag(&p->flags[0], SIP_PENDINGBYE)) {
- ast_debug(3, "Deferring reinvite on SIP '%s' - It's audio will be redirected to IP %s\n", p->callid, ast_sockaddr_stringify(instance ? &p->redirip : &p->ourip));
- /* We have a pending Invite. Send re-invite when we're done with the invite */
- ast_set_flag(&p->flags[0], SIP_NEEDREINVITE);
- }
- }
- /* Reset lastrtprx timer */
- p->lastrtprx = p->lastrtptx = time(NULL);
- sip_pvt_unlock(p);
- return 0;
-}
-
-static void sip_get_codec(struct ast_channel *chan, struct ast_format_cap *result)
-{
- ast_format_cap_append_from_cap(result, ast_channel_nativeformats(chan), AST_MEDIA_TYPE_UNKNOWN);
-}
-
-static struct ast_rtp_glue sip_rtp_glue = {
- .type = "SIP",
- .get_rtp_info = sip_get_rtp_peer,
- .allow_rtp_remote = sip_allow_rtp_remote,
- .get_vrtp_info = sip_get_vrtp_peer,
- .allow_vrtp_remote = sip_allow_vrtp_remote,
- .get_trtp_info = sip_get_trtp_peer,
- .update_peer = sip_set_rtp_peer,
- .get_codec = sip_get_codec,
-};
-
-static char *app_dtmfmode = "SIPDtmfMode";
-static char *app_sipaddheader = "SIPAddHeader";
-static char *app_sipremoveheader = "SIPRemoveHeader";
-#ifdef TEST_FRAMEWORK
-static char *app_sipsendcustominfo = "SIPSendCustomINFO";
-#endif
-
-/*! \brief Set the DTMFmode for an outbound SIP call (application) */
-static int sip_dtmfmode(struct ast_channel *chan, const char *data)
-{
- struct sip_pvt *p;
- const char *mode = data;
-
- if (!data) {
- ast_log(LOG_WARNING, "This application requires the argument: info, inband, rfc2833\n");
- return 0;
- }
- ast_channel_lock(chan);
- if (!IS_SIP_TECH(ast_channel_tech(chan))) {
- ast_log(LOG_WARNING, "Call this application only on SIP incoming calls\n");
- ast_channel_unlock(chan);
- return 0;
- }
- p = ast_channel_tech_pvt(chan);
- if (!p) {
- ast_channel_unlock(chan);
- return 0;
- }
- sip_pvt_lock(p);
- if (!strcasecmp(mode, "info")) {
- ast_clear_flag(&p->flags[0], SIP_DTMF);
- ast_set_flag(&p->flags[0], SIP_DTMF_INFO);
- p->jointnoncodeccapability &= ~AST_RTP_DTMF;
- } else if (!strcasecmp(mode, "shortinfo")) {
- ast_clear_flag(&p->flags[0], SIP_DTMF);
- ast_set_flag(&p->flags[0], SIP_DTMF_SHORTINFO);
- p->jointnoncodeccapability &= ~AST_RTP_DTMF;
- } else if (!strcasecmp(mode, "rfc2833")) {
- ast_clear_flag(&p->flags[0], SIP_DTMF);
- ast_set_flag(&p->flags[0], SIP_DTMF_RFC2833);
- p->jointnoncodeccapability |= AST_RTP_DTMF;
- } else if (!strcasecmp(mode, "inband")) {
- ast_clear_flag(&p->flags[0], SIP_DTMF);
- ast_set_flag(&p->flags[0], SIP_DTMF_INBAND);
- p->jointnoncodeccapability &= ~AST_RTP_DTMF;
- } else {
- ast_log(LOG_WARNING, "I don't know about this dtmf mode: %s\n", mode);
- }
- if (p->rtp)
- ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_DTMF, ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833);
- if ((ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_INBAND) ||
- (ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_AUTO)) {
- enable_dsp_detect(p);
- } else {
- disable_dsp_detect(p);
- }
- sip_pvt_unlock(p);
- ast_channel_unlock(chan);
- return 0;
-}
-
-/*! \brief Add a SIP header to an outbound INVITE */
-static int sip_addheader(struct ast_channel *chan, const char *data)
-{
- int no = 0;
- int ok = FALSE;
- char varbuf[30];
- const char *inbuf = data;
- char *subbuf;
-
- if (ast_strlen_zero(inbuf)) {
- ast_log(LOG_WARNING, "This application requires the argument: Header\n");
- return 0;
- }
- ast_channel_lock(chan);
-
- /* Check for headers */
- while (!ok && no <= 50) {
- no++;
- snprintf(varbuf, sizeof(varbuf), "__SIPADDHEADER%.2d", no);
-
- /* Compare without the leading underscores */
- if ((pbx_builtin_getvar_helper(chan, (const char *) varbuf + 2) == (const char *) NULL)) {
- ok = TRUE;
- }
- }
- if (ok) {
- size_t len = strlen(inbuf);
- subbuf = ast_alloca(len + 1);
- ast_get_encoded_str(inbuf, subbuf, len + 1);
- pbx_builtin_setvar_helper(chan, varbuf, subbuf);
- if (sipdebug) {
- ast_debug(1, "SIP Header added \"%s\" as %s\n", inbuf, varbuf);
- }
- } else {
- ast_log(LOG_WARNING, "Too many SIP headers added, max 50\n");
- }
- ast_channel_unlock(chan);
- return 0;
-}
-
-/*! \brief Remove SIP headers added previously with SipAddHeader application */
-static int sip_removeheader(struct ast_channel *chan, const char *data)
-{
- struct ast_var_t *newvariable;
- struct varshead *headp;
- int removeall = 0;
- char *inbuf = (char *) data;
-
- if (ast_strlen_zero(inbuf)) {
- removeall = 1;
- }
- ast_channel_lock(chan);
-
- headp=ast_channel_varshead(chan);
- AST_LIST_TRAVERSE_SAFE_BEGIN (headp, newvariable, entries) {
- if (strncmp(ast_var_name(newvariable), "SIPADDHEADER", strlen("SIPADDHEADER")) == 0) {
- if (removeall || (!strncasecmp(ast_var_value(newvariable),inbuf,strlen(inbuf)))) {
- if (sipdebug) {
- ast_debug(1,"removing SIP Header \"%s\" as %s\n",
- ast_var_value(newvariable),
- ast_var_name(newvariable));
- }
- AST_LIST_REMOVE_CURRENT(entries);
- ast_var_delete(newvariable);
- }
- }
- }
- AST_LIST_TRAVERSE_SAFE_END;
-
- ast_channel_unlock(chan);
- return 0;
-}
-
-#ifdef TEST_FRAMEWORK
-/*! \brief Send a custom INFO message via AST_CONTROL_CUSTOM indication */
-static int sip_sendcustominfo(struct ast_channel *chan, const char *data)
-{
- char *info_data, *useragent;
-
- if (ast_strlen_zero(data)) {
- ast_log(LOG_WARNING, "You must provide data to be sent\n");
- return 0;
- }
-
- useragent = ast_strdupa(data);
- info_data = strsep(&useragent, ",");
-
- if (ast_sipinfo_send(chan, NULL, "text/plain", info_data, useragent)) {
- ast_log(LOG_WARNING, "Failed to create payload for custom SIP INFO\n");
- return 0;
- }
- return 0;
-}
-#endif
-
-/*! \brief Transfer call before connect with a 302 redirect
-\note Called by the transfer() dialplan application through the sip_transfer()
- pbx interface function if the call is in ringing state
-\todo Fix this function so that we wait for reply to the REFER and
- react to errors, denials or other issues the other end might have.
- */
-static int sip_sipredirect(struct sip_pvt *p, const char *dest)
-{
- char *cdest;
- char *extension, *domain;
-
- cdest = ast_strdupa(dest);
-
- extension = strsep(&cdest, "@");
- domain = cdest;
- if (ast_strlen_zero(extension)) {
- ast_log(LOG_ERROR, "Missing mandatory argument: extension\n");
- return 0;
- }
-
- /* we'll issue the redirect message here */
- if (!domain) {
- char *local_to_header;
- char to_header[256];
-
- ast_copy_string(to_header, sip_get_header(&p->initreq, "To"), sizeof(to_header));
- if (ast_strlen_zero(to_header)) {
- ast_log(LOG_ERROR, "Cannot retrieve the 'To' header from the original SIP request!\n");
- return 0;
- }
- if (((local_to_header = strcasestr(to_header, "sip:")) || (local_to_header = strcasestr(to_header, "sips:")))
- && (local_to_header = strchr(local_to_header, '@'))) {
- char ldomain[256];
-
- memset(ldomain, 0, sizeof(ldomain));
- local_to_header++;
- /* Will copy no more than 255 chars plus null terminator. */
- sscanf(local_to_header, "%255[^<>; ]", ldomain);
- if (ast_strlen_zero(ldomain)) {
- ast_log(LOG_ERROR, "Can't find the host address\n");
- return 0;
- }
- domain = ast_strdupa(ldomain);
- }
- }
-
- ast_string_field_build(p, our_contact, "Transfer ", extension, domain);
- transmit_response_reliable(p, "302 Moved Temporarily", &p->initreq);
-
- sip_scheddestroy(p, SIP_TRANS_TIMEOUT); /* Make sure we stop send this reply. */
- sip_alreadygone(p);
-
- if (p->owner) {
- enum ast_control_transfer message = AST_TRANSFER_SUCCESS;
- ast_queue_control_data(p->owner, AST_CONTROL_TRANSFER, &message, sizeof(message));
- }
- /* hangup here */
- return 0;
-}
-
-static int sip_is_xml_parsable(void)
-{
-#ifdef HAVE_LIBXML2
- return TRUE;
-#else
- return FALSE;
-#endif
-}
-
-/*! \brief Send a poke to all known peers */
-static void sip_poke_all_peers(void)
-{
- int ms = 0, num = 0;
- struct ao2_iterator i;
- struct sip_peer *peer;
-
- if (!speerobjs) { /* No peers, just give up */
- return;
- }
-
- i = ao2_iterator_init(peers, 0);
- while ((peer = ao2_t_iterator_next(&i, "iterate thru peers table"))) {
- ao2_lock(peer);
- /* Don't schedule poking on a peer without qualify */
- if (peer->maxms) {
- if (num == global_qualify_peers) {
- ms += global_qualify_gap;
- num = 0;
- } else {
- num++;
- }
- AST_SCHED_REPLACE_UNREF(peer->pokeexpire, sched, ms, sip_poke_peer_s, peer,
- sip_unref_peer(_data, "removing poke peer ref"),
- sip_unref_peer(peer, "removing poke peer ref"),
- sip_ref_peer(peer, "adding poke peer ref"));
- }
- ao2_unlock(peer);
- sip_unref_peer(peer, "toss iterator peer ptr");
- }
- ao2_iterator_destroy(&i);
-}
-
-/*! \brief Send a keepalive to all known peers */
-static void sip_keepalive_all_peers(void)
-{
- struct ao2_iterator i;
- struct sip_peer *peer;
-
- if (!speerobjs) { /* No peers, just give up */
- return;
- }
-
- i = ao2_iterator_init(peers, 0);
- while ((peer = ao2_t_iterator_next(&i, "iterate thru peers table"))) {
- ao2_lock(peer);
- AST_SCHED_REPLACE_UNREF(peer->keepalivesend, sched, 0, sip_send_keepalive, peer,
- sip_unref_peer(_data, "removing poke peer ref"),
- sip_unref_peer(peer, "removing poke peer ref"),
- sip_ref_peer(peer, "adding poke peer ref"));
- ao2_unlock(peer);
- sip_unref_peer(peer, "toss iterator peer ptr");
- }
- ao2_iterator_destroy(&i);
-}
-
-/*! \brief Send all known registrations */
-static void sip_send_all_registers(void)
-{
- int ms;
- int regspacing;
- struct ao2_iterator iter;
- struct sip_registry *iterator;
-
- if (!ao2_container_count(registry_list)) {
- return;
- }
- regspacing = default_expiry * 1000 / ao2_container_count(registry_list);
- if (regspacing > 100) {
- regspacing = 100;
- }
- ms = regspacing;
-
- iter = ao2_iterator_init(registry_list, 0);
- while ((iterator = ao2_t_iterator_next(&iter, "sip_send_all_registers iter"))) {
- ao2_lock(iterator);
- ms += regspacing;
- start_reregister_timeout(iterator, ms);
- ao2_unlock(iterator);
- ao2_t_ref(iterator, -1, "sip_send_all_registers iter");
- }
- ao2_iterator_destroy(&iter);
-}
-
-/*! \brief Send all MWI subscriptions */
-static void sip_send_all_mwi_subscriptions(void)
-{
- struct ao2_iterator iter;
- struct sip_subscription_mwi *mwi;
-
- iter = ao2_iterator_init(subscription_mwi_list, 0);
- while ((mwi = ao2_t_iterator_next(&iter, "sip_send_all_mwi_subscriptions iter"))) {
- start_mwi_subscription(mwi, 1);
- ao2_t_ref(mwi, -1, "sip_send_all_mwi_subscriptions iter");
- }
- ao2_iterator_destroy(&iter);
-}
-
-static int process_crypto(struct sip_pvt *p, struct ast_rtp_instance *rtp, struct ast_sdp_srtp **srtp,
- const char *a)
-{
- struct ast_rtp_engine_dtls *dtls;
-
- /* If no RTP instance exists for this media stream don't bother processing the crypto line */
- if (!rtp) {
- ast_debug(3, "Received offer with crypto line for media stream that is not enabled\n");
- return FALSE;
- }
-
- if (strncasecmp(a, "crypto:", 7)) {
- return FALSE;
- }
- /* skip "crypto:" */
- a += strlen("crypto:");
-
- if (!*srtp) {
- if (ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
- ast_log(LOG_WARNING, "Ignoring unexpected crypto attribute in SDP answer\n");
- return FALSE;
- }
-
- if (!(*srtp = ast_sdp_srtp_alloc())) {
- return FALSE;
- }
- }
-
- if (!(*srtp)->crypto && !((*srtp)->crypto = ast_sdp_crypto_alloc())) {
- return FALSE;
- }
-
- if (ast_sdp_crypto_process(rtp, *srtp, a) < 0) {
- return FALSE;
- }
-
- if ((dtls = ast_rtp_instance_get_dtls(rtp))) {
- dtls->stop(rtp);
- p->dtls_cfg.enabled = 0;
- }
-
- return TRUE;
-}
-
-/*! \brief Reload module */
-static int sip_do_reload(enum channelreloadreason reason)
-{
- time_t start_poke, end_poke;
-
- reload_config(reason);
- ast_sched_dump(sched);
-
- start_poke = time(0);
- /* Prune peers who still are supposed to be deleted */
- unlink_marked_peers_from_tables();
-
- ast_debug(4, "--------------- Done destroying pruned peers\n");
-
- /* Send qualify (OPTIONS) to all peers */
- sip_poke_all_peers();
-
- /* Send keepalive to all peers */
- sip_keepalive_all_peers();
-
- /* Register with all services */
- sip_send_all_registers();
-
- sip_send_all_mwi_subscriptions();
-
- end_poke = time(0);
-
- ast_debug(4, "do_reload finished. peer poke/prune reg contact time = %d sec.\n", (int)(end_poke-start_poke));
-
- ast_debug(4, "--------------- SIP reload done\n");
-
- return 0;
-}
-
-/*! \brief Force reload of module from cli */
-static char *sip_reload(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
-{
- static struct sip_peer *new_peer;
-
- switch (cmd) {
- case CLI_INIT:
- e->command = "sip reload";
- e->usage =
- "Usage: sip reload\n"
- " Reloads SIP configuration from sip.conf\n";
- return NULL;
- case CLI_GENERATE:
- return NULL;
- }
-
- ast_mutex_lock(&sip_reload_lock);
- if (sip_reloading) {
- ast_verbose("Previous SIP reload not yet done\n");
- } else {
- sip_reloading = TRUE;
- sip_reloadreason = (a && a->fd) ? CHANNEL_CLI_RELOAD : CHANNEL_MODULE_RELOAD;
- }
- ast_mutex_unlock(&sip_reload_lock);
- restart_monitor();
-
- /* Create new bogus peer possibly with new global settings. */
- if ((new_peer = temp_peer("(bogus_peer)"))) {
- ast_string_field_set(new_peer, md5secret, BOGUS_PEER_MD5SECRET);
- ast_clear_flag(&new_peer->flags[0], SIP_INSECURE);
- ao2_t_global_obj_replace_unref(g_bogus_peer, new_peer,
- "Replacing the old bogus peer during reload.");
- ao2_t_ref(new_peer, -1, "done with new_peer");
- } else {
- ast_log(LOG_ERROR, "Could not update the fake authentication peer.\n");
- /* You probably have bigger (memory?) issues to worry about though.. */
- }
-
- return CLI_SUCCESS;
-}
-
-/*! \brief Part of Asterisk module interface */
-static int reload(void)
-{
- sip_reload(0, 0, NULL);
- return AST_MODULE_LOAD_SUCCESS;
-}
-
-/*! \brief Return the first entry from ast_sockaddr_resolve filtered by family of binddaddr
- *
- * \warning Using this function probably means you have a faulty design.
- */
-static int ast_sockaddr_resolve_first(struct ast_sockaddr *addr,
- const char* name, int flag)
-{
- return ast_sockaddr_resolve_first_af(addr, name, flag, get_address_family_filter(AST_TRANSPORT_UDP));
-}
-
-/*! \brief Return the first entry from ast_sockaddr_resolve filtered by family of binddaddr
- *
- * \warning Using this function probably means you have a faulty design.
- */
-static int ast_sockaddr_resolve_first_transport(struct ast_sockaddr *addr,
- const char* name, int flag, unsigned int transport)
-{
- return ast_sockaddr_resolve_first_af(addr, name, flag, get_address_family_filter(transport));
-}
-
-/*! \brief
- * \note The only member of the peer used here is the name field
- */
-static int peer_hash_cb(const void *obj, const int flags)
-{
- const struct sip_peer *peer = obj;
-
- return ast_str_case_hash(peer->name);
-}
-
-/*!
- * \note The only member of the peer used here is the name field
- */
-static int peer_cmp_cb(void *obj, void *arg, int flags)
-{
- struct sip_peer *peer = obj, *peer2 = arg;
-
- return !strcasecmp(peer->name, peer2->name) ? CMP_MATCH | CMP_STOP : 0;
-}
-
-/*!
- * Hash function based on the peer's ip address. For IPv6, we use the end
- * of the address.
- * \todo Find a better hashing function
- */
-static int peer_iphash_cb(const void *obj, const int flags)
-{
- const struct sip_peer *peer = obj;
- int ret = 0;
-
- if (ast_sockaddr_isnull(&peer->addr)) {
- ast_log(LOG_ERROR, "Empty address\n");
- }
-
- ret = ast_sockaddr_hash(&peer->addr);
-
- if (ret < 0) {
- ret = -ret;
- }
-
- return ret;
-}
-
-/*!
- * Match Peers by IP and Port number.
- *
- * This function has two modes.
- * - If the peer arg does not have INSECURE_PORT set, then we will only return
- * a match for a peer that matches both the IP and port.
- * - If the peer arg does have the INSECURE_PORT flag set, then we will return
- * a match for UDP peers with insecure=port set, or a peer that does NOT have
- * host=dynamic for other protocols (or have a valid Contact: header in REGISTER).
- * This callback will be used twice when doing peer matching, as per the two modes
- * described above.
- *
- * \note the peer's addr struct provides to fields combined to make a key: the
- * sin_addr.s_addr and sin_port fields (transport is compared separately).
- */
-static int peer_ipcmp_cb_full(void *obj, void *arg, void *data, int flags)
-{
- struct sip_peer *peer = obj, *peer2 = arg;
- char *callback = data;
-
- if (!ast_strlen_zero(callback) && strcasecmp(peer->callback, callback)) {
- /* We require a callback extension match, but don't have one */
- return 0;
- }
-
- /* At this point, we match the callback extension if we need to. Carry on. */
-
- if (ast_sockaddr_cmp_addr(&peer->addr, &peer2->addr)) {
- /* IP doesn't match */
- return 0;
- }
-
- if ((peer->transports & peer2->transports) == 0) {
- /* transport setting doesn't match */
- return 0;
- }
-
- if (!ast_test_flag(&peer2->flags[0], SIP_INSECURE_PORT)) {
- /* On the first pass only match if ports match. */
- return ast_sockaddr_port(&peer->addr) == ast_sockaddr_port(&peer2->addr) ?
- (CMP_MATCH | CMP_STOP) : 0;
- }
-
- /* We can reach here only if peer2 is for SIP_INSECURE_PORT, in
- * other words, the second pass where we only try to match against IP.
- *
- * Some special handling for UDP vs non-UDP (TCP, TLS, WS and WSS), since
- * for non-UDP the source port won't typically be controlled, we only want
- * to check the source IP, but only if the host isn't dynamic. This isn't
- * done in the first pass so that if a peer registers from the same IP as
- * a static IP peer that registration (port match) will take prescedence).
- */
- if (peer2->transports == AST_TRANSPORT_UDP) {
- /* We are allowing match without port for peers configured that
- * way in this pass through the peers. */
- return ast_test_flag(&peer->flags[0], SIP_INSECURE_PORT) ?
- (CMP_MATCH | CMP_STOP) : 0;
- }
-
- if (!peer->host_dynamic) {
- return CMP_MATCH | CMP_STOP;
- }
-
- /* Conditions taken from parse_register_contact() */
- if (peer2->transports & (AST_TRANSPORT_WS | AST_TRANSPORT_WSS)) {
- /* The contact address of websockets is always the transport source address and port */
- return 0;
- }
-
- if (ast_test_flag(&peer->flags[0], SIP_NAT_FORCE_RPORT)) {
- /* The contact address of NATed peers is always the transport source address and port */
- return 0;
- }
-
- /* Have to assume that we used the registered contact header (non-NAT) */
- return CMP_MATCH | CMP_STOP;
-}
-
-static int threadt_hash_cb(const void *obj, const int flags)
-{
- const struct sip_threadinfo *th = obj;
-
- return ast_sockaddr_hash(&th->tcptls_session->remote_address);
-}
-
-static int threadt_cmp_cb(void *obj, void *arg, int flags)
-{
- struct sip_threadinfo *th = obj, *th2 = arg;
-
- return (th->tcptls_session == th2->tcptls_session) ? CMP_MATCH | CMP_STOP : 0;
-}
-
-/*!
- * \note The only member of the dialog used here callid string
- */
-static int dialog_hash_cb(const void *obj, const int flags)
-{
- const struct sip_pvt *pvt = obj;
-
- return ast_str_case_hash(pvt->callid);
-}
-
-/*!
- * \note Same as dialog_cmp_cb, except without the CMP_STOP on match
- */
-static int dialog_find_multiple(void *obj, void *arg, int flags)
-{
- struct sip_pvt *pvt = obj, *pvt2 = arg;
-
- return !strcasecmp(pvt->callid, pvt2->callid) ? CMP_MATCH : 0;
-}
-
-/*!
- * \note The only member of the dialog used here callid string
- */
-static int dialog_cmp_cb(void *obj, void *arg, int flags)
-{
- struct sip_pvt *pvt = obj, *pvt2 = arg;
-
- return !strcasecmp(pvt->callid, pvt2->callid) ? CMP_MATCH | CMP_STOP : 0;
-}
-
-
-static int registry_hash_cb(const void *obj, const int flags)
-{
- const struct sip_registry *object;
- const char *key;
-
- switch (flags & OBJ_SEARCH_MASK) {
- case OBJ_SEARCH_KEY:
- key = obj;
- break;
- case OBJ_SEARCH_OBJECT:
- object = obj;
- key = object->configvalue;
- break;
- default:
- /* Hash can only work on something with a full key. */
- ast_assert(0);
- return 0;
- }
- return ast_str_hash(key);
-}
-
-static int registry_cmp_cb(void *obj, void *arg, int flags)
-{
- const struct sip_registry *object_left = obj;
- const struct sip_registry *object_right = arg;
- const char *right_key = arg;
- int cmp;
-
- switch (flags & OBJ_SEARCH_MASK) {
- case OBJ_SEARCH_OBJECT:
- right_key = object_right->configvalue;
- /* Fall through */
- case OBJ_SEARCH_KEY:
- cmp = strcmp(object_left->configvalue, right_key);
- break;
- default:
- cmp = 0;
- break;
- }
- if (cmp) {
- return 0;
- }
- return CMP_MATCH;
-}
-
-
-/*! \brief SIP Cli commands definition */
-static struct ast_cli_entry cli_sip[] = {
- AST_CLI_DEFINE(sip_show_channels, "List active SIP channels or subscriptions"),
- AST_CLI_DEFINE(sip_show_channelstats, "List statistics for active SIP channels"),
- AST_CLI_DEFINE(sip_show_domains, "List our local SIP domains"),
- AST_CLI_DEFINE(sip_show_inuse, "List all inuse/limits"),
- AST_CLI_DEFINE(sip_show_objects, "List all SIP object allocations"),
- AST_CLI_DEFINE(sip_show_peers, "List defined SIP peers"),
- AST_CLI_DEFINE(sip_show_registry, "List SIP registration status"),
- AST_CLI_DEFINE(sip_unregister, "Unregister (force expiration) a SIP peer from the registry"),
- AST_CLI_DEFINE(sip_show_settings, "Show SIP global settings"),
- AST_CLI_DEFINE(sip_show_mwi, "Show MWI subscriptions"),
- AST_CLI_DEFINE(sip_cli_notify, "Send a notify packet to a SIP peer"),
- AST_CLI_DEFINE(sip_show_channel, "Show detailed SIP channel info"),
- AST_CLI_DEFINE(sip_show_history, "Show SIP dialog history"),
- AST_CLI_DEFINE(sip_show_peer, "Show details on specific SIP peer"),
- AST_CLI_DEFINE(sip_show_users, "List defined SIP users"),
- AST_CLI_DEFINE(sip_show_user, "Show details on specific SIP user"),
- AST_CLI_DEFINE(sip_qualify_peer, "Send an OPTIONS packet to a peer"),
- AST_CLI_DEFINE(sip_show_sched, "Present a report on the status of the scheduler queue"),
- AST_CLI_DEFINE(sip_prune_realtime, "Prune cached Realtime users/peers"),
- AST_CLI_DEFINE(sip_do_debug, "Enable/Disable SIP debugging"),
- AST_CLI_DEFINE(sip_set_history, "Enable/Disable SIP history"),
- AST_CLI_DEFINE(sip_reload, "Reload SIP configuration"),
- AST_CLI_DEFINE(sip_show_tcp, "List TCP Connections")
-};
-
-/*! \brief SIP test registration */
-static void sip_register_tests(void)
-{
- sip_config_parser_register_tests();
- sip_request_parser_register_tests();
- sip_dialplan_function_register_tests();
-}
-
-/*! \brief SIP test registration */
-static void sip_unregister_tests(void)
-{
- sip_config_parser_unregister_tests();
- sip_request_parser_unregister_tests();
- sip_dialplan_function_unregister_tests();
-}
-
-#ifdef TEST_FRAMEWORK
-AST_TEST_DEFINE(test_sip_mwi_subscribe_parse)
-{
- struct ao2_iterator iter;
- struct sip_subscription_mwi *iterator;
- int found = 0;
- enum ast_test_result_state res = AST_TEST_PASS;
- const char *mwi1 = "1234@mysipprovider.com/1234";
- const char *mwi2 = "1234:password@mysipprovider.com/1234";
- const char *mwi3 = "1234:password@mysipprovider.com:5061/1234";
- const char *mwi4 = "1234:password:authuser@mysipprovider.com/1234";
- const char *mwi5 = "1234:password:authuser@mysipprovider.com:5061/1234";
- const char *mwi6 = "1234:password";
-
- switch (cmd) {
- case TEST_INIT:
- info->name = "sip_mwi_subscribe_parse_test";
- info->category = "/channels/chan_sip/";
- info->summary = "SIP MWI subscribe line parse unit test";
- info->description =
- "Tests the parsing of mwi subscription lines (e.g., mwi => from sip.conf)";
- return AST_TEST_NOT_RUN;
- case TEST_EXECUTE:
- break;
- }
-
- if (sip_subscribe_mwi(mwi1, 1)) {
- res = AST_TEST_FAIL;
- } else {
- found = 0;
- res = AST_TEST_FAIL;
-
- iter = ao2_iterator_init(subscription_mwi_list, 0);
- while ((iterator = ao2_t_iterator_next(&iter, "test_sip_mwi_subscribe_parse mwi1"))) {
- ao2_lock(iterator);
- if (
- !strcmp(iterator->hostname, "mysipprovider.com") &&
- !strcmp(iterator->username, "1234") &&
- !strcmp(iterator->secret, "") &&
- !strcmp(iterator->authuser, "") &&
- !strcmp(iterator->mailbox, "1234") &&
- iterator->portno == 0) {
- found = 1;
- res = AST_TEST_PASS;
- }
- ao2_unlock(iterator);
- ao2_t_ref(iterator, -1, "test_sip_mwi_subscribe_parse mwi1");
- }
- ao2_iterator_destroy(&iter);
- if (!found) {
- ast_test_status_update(test, "sip_subscribe_mwi test 1 failed\n");
- }
- }
-
- if (sip_subscribe_mwi(mwi2, 1)) {
- res = AST_TEST_FAIL;
- } else {
- found = 0;
- res = AST_TEST_FAIL;
-
- iter = ao2_iterator_init(subscription_mwi_list, 0);
- while ((iterator = ao2_t_iterator_next(&iter, "test_sip_mwi_subscribe_parse mwi2"))) {
- ao2_lock(iterator);
- if (
- !strcmp(iterator->hostname, "mysipprovider.com") &&
- !strcmp(iterator->username, "1234") &&
- !strcmp(iterator->secret, "password") &&
- !strcmp(iterator->authuser, "") &&
- !strcmp(iterator->mailbox, "1234") &&
- iterator->portno == 0) {
- found = 1;
- res = AST_TEST_PASS;
- }
- ao2_unlock(iterator);
- ao2_t_ref(iterator, -1, "test_sip_mwi_subscribe_parse mwi2");
- }
- ao2_iterator_destroy(&iter);
- if (!found) {
- ast_test_status_update(test, "sip_subscribe_mwi test 2 failed\n");
- }
- }
-
- if (sip_subscribe_mwi(mwi3, 1)) {
- res = AST_TEST_FAIL;
- } else {
- found = 0;
- res = AST_TEST_FAIL;
-
- iter = ao2_iterator_init(subscription_mwi_list, 0);
- while ((iterator = ao2_t_iterator_next(&iter, "test_sip_mwi_subscribe_parse mwi3"))) {
- ao2_lock(iterator);
- if (
- !strcmp(iterator->hostname, "mysipprovider.com") &&
- !strcmp(iterator->username, "1234") &&
- !strcmp(iterator->secret, "password") &&
- !strcmp(iterator->authuser, "") &&
- !strcmp(iterator->mailbox, "1234") &&
- iterator->portno == 5061) {
- found = 1;
- res = AST_TEST_PASS;
- }
- ao2_unlock(iterator);
- ao2_t_ref(iterator, -1, "test_sip_mwi_subscribe_parse mwi3");
- }
- ao2_iterator_destroy(&iter);
- if (!found) {
- ast_test_status_update(test, "sip_subscribe_mwi test 3 failed\n");
- }
- }
-
- if (sip_subscribe_mwi(mwi4, 1)) {
- res = AST_TEST_FAIL;
- } else {
- found = 0;
- res = AST_TEST_FAIL;
-
- iter = ao2_iterator_init(subscription_mwi_list, 0);
- while ((iterator = ao2_t_iterator_next(&iter, "test_sip_mwi_subscribe_parse mwi4"))) {
- ao2_lock(iterator);
- if (
- !strcmp(iterator->hostname, "mysipprovider.com") &&
- !strcmp(iterator->username, "1234") &&
- !strcmp(iterator->secret, "password") &&
- !strcmp(iterator->authuser, "authuser") &&
- !strcmp(iterator->mailbox, "1234") &&
- iterator->portno == 0) {
- found = 1;
- res = AST_TEST_PASS;
- }
- ao2_unlock(iterator);
- ao2_t_ref(iterator, -1, "test_sip_mwi_subscribe_parse mwi4");
- }
- ao2_iterator_destroy(&iter);
- if (!found) {
- ast_test_status_update(test, "sip_subscribe_mwi test 4 failed\n");
- }
- }
-
- if (sip_subscribe_mwi(mwi5, 1)) {
- res = AST_TEST_FAIL;
- } else {
- found = 0;
- res = AST_TEST_FAIL;
-
- iter = ao2_iterator_init(subscription_mwi_list, 0);
- while ((iterator = ao2_t_iterator_next(&iter, "test_sip_mwi_subscribe_parse mwi4"))) {
- ao2_lock(iterator);
- if (
- !strcmp(iterator->hostname, "mysipprovider.com") &&
- !strcmp(iterator->username, "1234") &&
- !strcmp(iterator->secret, "password") &&
- !strcmp(iterator->authuser, "authuser") &&
- !strcmp(iterator->mailbox, "1234") &&
- iterator->portno == 5061) {
- found = 1;
- res = AST_TEST_PASS;
- }
- ao2_unlock(iterator);
- ao2_t_ref(iterator, -1, "test_sip_mwi_subscribe_parse mwi4");
- }
- ao2_iterator_destroy(&iter);
- if (!found) {
- ast_test_status_update(test, "sip_subscribe_mwi test 5 failed\n");
- }
- }
-
- if (sip_subscribe_mwi(mwi6, 1)) {
- res = AST_TEST_PASS;
- } else {
- res = AST_TEST_FAIL;
- }
- return res;
-}
-
-/*!
- * \brief Imitation TCP reception loop
- *
- * This imitates the logic used by SIP's TCP code. Its purpose
- * is to either
- * 1) Combine fragments into a single message
- * 2) Break up combined messages into single messages
- *
- * \param fragments The message fragments. This simulates the data received on a TCP socket.
- * \param num_fragments This indicates the number of fragments to receive
- * \param overflow This is a place to stash extra data if more than one message is received
- * in a single fragment
- * \param[out] messages The parsed messages are placed in this array
- * \param[out] num_messages The number of messages that were parsed
- * \param test Used for printing messages
- * \retval 0 Success
- * \retval -1 Failure
- */
-static int mock_tcp_loop(char *fragments[], size_t num_fragments,
- struct ast_str **overflow, char **messages, int *num_messages, struct ast_test* test)
-{
- struct ast_str *req_data;
- int i = 0;
- int res = 0;
-
- req_data = ast_str_create(128);
- ast_str_reset(*overflow);
-
- while (i < num_fragments || ast_str_strlen(*overflow) > 0) {
- enum message_integrity message_integrity = MESSAGE_FRAGMENT;
- ast_str_reset(req_data);
- while (message_integrity == MESSAGE_FRAGMENT) {
- if (ast_str_strlen(*overflow) > 0) {
- ast_str_append(&req_data, 0, "%s", ast_str_buffer(*overflow));
- ast_str_reset(*overflow);
- } else {
- ast_str_append(&req_data, 0, "%s", fragments[i++]);
- }
- message_integrity = check_message_integrity(&req_data, overflow);
- }
- if (strcmp(ast_str_buffer(req_data), messages[*num_messages])) {
- ast_test_status_update(test, "Mismatch in SIP messages.\n");
- ast_test_status_update(test, "Expected message:\n%s", messages[*num_messages]);
- ast_test_status_update(test, "Parsed message:\n%s", ast_str_buffer(req_data));
- res = -1;
- goto end;
- } else {
- ast_test_status_update(test, "Successfully read message:\n%s", ast_str_buffer(req_data));
- }
- (*num_messages)++;
- }
-
-end:
- ast_free(req_data);
- return res;
-};
-
-AST_TEST_DEFINE(test_tcp_message_fragmentation)
-{
- /* Normal single message in one fragment */
- char *normal[] = {
- "INVITE sip:bob@example.org SIP/2.0\r\n"
- "Via: SIP/2.0/TCP 127.0.0.1:5060;branch=[branch]\r\n"
- "From: sipp ;tag=12345\r\n"
- "To: \r\n"
- "Call-ID: 12345\r\n"
- "CSeq: 1 INVITE\r\n"
- "Contact: sip:127.0.0.1:5061\r\n"
- "Max-Forwards: 70\r\n"
- "Content-Type: application/sdp\r\n"
- "Content-Length: 130\r\n"
- "\r\n"
- "v=0\r\n"
- "o=user1 53655765 2353687637 IN IP4 127.0.0.1\r\n"
- "s=-\r\n"
- "c=IN IP4 127.0.0.1\r\n"
- "t=0 0\r\n"
- "m=audio 10000 RTP/AVP 0\r\n"
- "a=rtpmap:0 PCMU/8000\r\n"
- };
-
- /* Single message in two fragments.
- * Fragments combine to make "normal"
- */
- char *fragmented[] = {
- "INVITE sip:bob@example.org SIP/2.0\r\n"
- "Via: SIP/2.0/TCP 127.0.0.1:5060;branch=[branch]\r\n"
- "From: sipp ;tag=12345\r\n"
- "To: \r\n"
- "Call-ID: 12345\r\n"
- "CSeq: 1 INVITE\r\n"
- "Contact: sip:127.0.0.1:5061\r\n"
- "Max-Forwards: ",
-
- "70\r\n"
- "Content-Type: application/sdp\r\n"
- "Content-Length: 130\r\n"
- "\r\n"
- "v=0\r\n"
- "o=user1 53655765 2353687637 IN IP4 127.0.0.1\r\n"
- "s=-\r\n"
- "c=IN IP4 127.0.0.1\r\n"
- "t=0 0\r\n"
- "m=audio 10000 RTP/AVP 0\r\n"
- "a=rtpmap:0 PCMU/8000\r\n"
- };
- /* Single message in two fragments, divided precisely at the body
- * Fragments combine to make "normal"
- */
- char *fragmented_body[] = {
- "INVITE sip:bob@example.org SIP/2.0\r\n"
- "Via: SIP/2.0/TCP 127.0.0.1:5060;branch=[branch]\r\n"
- "From: sipp ;tag=12345\r\n"
- "To: \r\n"
- "Call-ID: 12345\r\n"
- "CSeq: 1 INVITE\r\n"
- "Contact: sip:127.0.0.1:5061\r\n"
- "Max-Forwards: 70\r\n"
- "Content-Type: application/sdp\r\n"
- "Content-Length: 130\r\n"
- "\r\n",
-
- "v=0\r\n"
- "o=user1 53655765 2353687637 IN IP4 127.0.0.1\r\n"
- "s=-\r\n"
- "c=IN IP4 127.0.0.1\r\n"
- "t=0 0\r\n"
- "m=audio 10000 RTP/AVP 0\r\n"
- "a=rtpmap:0 PCMU/8000\r\n"
- };
-
- /* Single message in three fragments
- * Fragments combine to make "normal"
- */
- char *multi_fragment[] = {
- "INVITE sip:bob@example.org SIP/2.0\r\n"
- "Via: SIP/2.0/TCP 127.0.0.1:5060;branch=[branch]\r\n"
- "From: sipp ;tag=12345\r\n"
- "To: \r\n"
- "Call-ID: 12345\r\n"
- "CSeq: 1 INVITE\r\n",
-
- "Contact: sip:127.0.0.1:5061\r\n"
- "Max-Forwards: 70\r\n"
- "Content-Type: application/sdp\r\n"
- "Content-Length: 130\r\n"
- "\r\n"
- "v=0\r\n"
- "o=user1 53655765 2353687637 IN IP4 127.0.0.1\r\n"
- "s=-\r\n"
- "c=IN IP4",
-
- " 127.0.0.1\r\n"
- "t=0 0\r\n"
- "m=audio 10000 RTP/AVP 0\r\n"
- "a=rtpmap:0 PCMU/8000\r\n"
- };
-
- /* Two messages in a single fragment
- * Fragments split into "multi_message_divided"
- */
- char *multi_message[] = {
- "SIP/2.0 100 Trying\r\n"
- "Via: SIP/2.0/TCP 127.0.0.1:5060;branch=[branch]\r\n"
- "From: sipp ;tag=12345\r\n"
- "To: \r\n"
- "Call-ID: 12345\r\n"
- "CSeq: 1 INVITE\r\n"
- "Contact: \r\n"
- "Content-Length: 0\r\n"
- "\r\n"
- "SIP/2.0 180 Ringing\r\n"
- "Via: SIP/2.0/TCP 127.0.0.1:5060;branch=[branch]\r\n"
- "From: sipp ;tag=12345\r\n"
- "To: \r\n"
- "Call-ID: 12345\r\n"
- "CSeq: 1 INVITE\r\n"
- "Contact: \r\n"
- "Content-Length: 0\r\n"
- "\r\n"
- };
- char *multi_message_divided[] = {
- "SIP/2.0 100 Trying\r\n"
- "Via: SIP/2.0/TCP 127.0.0.1:5060;branch=[branch]\r\n"
- "From: sipp ;tag=12345\r\n"
- "To: \r\n"
- "Call-ID: 12345\r\n"
- "CSeq: 1 INVITE\r\n"
- "Contact: \r\n"
- "Content-Length: 0\r\n"
- "\r\n",
-
- "SIP/2.0 180 Ringing\r\n"
- "Via: SIP/2.0/TCP 127.0.0.1:5060;branch=[branch]\r\n"
- "From: sipp ;tag=12345\r\n"
- "To: \r\n"
- "Call-ID: 12345\r\n"
- "CSeq: 1 INVITE\r\n"
- "Contact: \r\n"
- "Content-Length: 0\r\n"
- "\r\n"
- };
- /* Two messages with bodies combined into one fragment
- * Fragments split into "multi_message_body_divided"
- */
- char *multi_message_body[] = {
- "INVITE sip:bob@example.org SIP/2.0\r\n"
- "Via: SIP/2.0/TCP 127.0.0.1:5060;branch=[branch]\r\n"
- "From: sipp ;tag=12345\r\n"
- "To: \r\n"
- "Call-ID: 12345\r\n"
- "CSeq: 1 INVITE\r\n"
- "Contact: sip:127.0.0.1:5061\r\n"
- "Max-Forwards: 70\r\n"
- "Content-Type: application/sdp\r\n"
- "Content-Length: 130\r\n"
- "\r\n"
- "v=0\r\n"
- "o=user1 53655765 2353687637 IN IP4 127.0.0.1\r\n"
- "s=-\r\n"
- "c=IN IP4 127.0.0.1\r\n"
- "t=0 0\r\n"
- "m=audio 10000 RTP/AVP 0\r\n"
- "a=rtpmap:0 PCMU/8000\r\n"
- "INVITE sip:bob@example.org SIP/2.0\r\n"
- "Via: SIP/2.0/TCP 127.0.0.1:5060;branch=[branch]\r\n"
- "From: sipp ;tag=12345\r\n"
- "To: \r\n"
- "Call-ID: 12345\r\n"
- "CSeq: 2 INVITE\r\n"
- "Contact: sip:127.0.0.1:5061\r\n"
- "Max-Forwards: 70\r\n"
- "Content-Type: application/sdp\r\n"
- "Content-Length: 130\r\n"
- "\r\n"
- "v=0\r\n"
- "o=user1 53655765 2353687637 IN IP4 127.0.0.1\r\n"
- "s=-\r\n"
- "c=IN IP4 127.0.0.1\r\n"
- "t=0 0\r\n"
- "m=audio 10000 RTP/AVP 0\r\n"
- "a=rtpmap:0 PCMU/8000\r\n"
- };
- char *multi_message_body_divided[] = {
- "INVITE sip:bob@example.org SIP/2.0\r\n"
- "Via: SIP/2.0/TCP 127.0.0.1:5060;branch=[branch]\r\n"
- "From: sipp ;tag=12345\r\n"
- "To: \r\n"
- "Call-ID: 12345\r\n"
- "CSeq: 1 INVITE\r\n"
- "Contact: sip:127.0.0.1:5061\r\n"
- "Max-Forwards: 70\r\n"
- "Content-Type: application/sdp\r\n"
- "Content-Length: 130\r\n"
- "\r\n"
- "v=0\r\n"
- "o=user1 53655765 2353687637 IN IP4 127.0.0.1\r\n"
- "s=-\r\n"
- "c=IN IP4 127.0.0.1\r\n"
- "t=0 0\r\n"
- "m=audio 10000 RTP/AVP 0\r\n"
- "a=rtpmap:0 PCMU/8000\r\n",
-
- "INVITE sip:bob@example.org SIP/2.0\r\n"
- "Via: SIP/2.0/TCP 127.0.0.1:5060;branch=[branch]\r\n"
- "From: sipp ;tag=12345\r\n"
- "To: \r\n"
- "Call-ID: 12345\r\n"
- "CSeq: 2 INVITE\r\n"
- "Contact: sip:127.0.0.1:5061\r\n"
- "Max-Forwards: 70\r\n"
- "Content-Type: application/sdp\r\n"
- "Content-Length: 130\r\n"
- "\r\n"
- "v=0\r\n"
- "o=user1 53655765 2353687637 IN IP4 127.0.0.1\r\n"
- "s=-\r\n"
- "c=IN IP4 127.0.0.1\r\n"
- "t=0 0\r\n"
- "m=audio 10000 RTP/AVP 0\r\n"
- "a=rtpmap:0 PCMU/8000\r\n"
- };
-
- /* Two messages that appear in two fragments. Fragment
- * boundaries do not align with message boundaries.
- * Fragments combine to make "multi_message_divided"
- */
- char *multi_message_in_fragments[] = {
- "SIP/2.0 100 Trying\r\n"
- "Via: SIP/2.0/TCP 127.0.0.1:5060;branch=[branch]\r\n"
- "From: sipp ;tag=12345\r\n"
- "To: \r\n"
- "Call-ID: 12345\r\n"
- "CSeq: 1 INVI",
-
- "TE\r\n"
- "Contact: \r\n"
- "Content-Length: 0\r\n"
- "\r\n"
- "SIP/2.0 180 Ringing\r\n"
- "Via: SIP/2.0/TCP 127.0.0.1:5060;branch=[branch]\r\n"
- "From: sipp ;tag=12345\r\n"
- "To: \r\n"
- "Call-ID: 12345\r\n"
- "CSeq: 1 INVITE\r\n"
- "Contact: \r\n"
- "Content-Length: 0\r\n"
- "\r\n"
- };
-
- /* Message with compact content-length header
- * Same as "normal" but with compact content-length header
- */
- char *compact[] = {
- "INVITE sip:bob@example.org SIP/2.0\r\n"
- "Via: SIP/2.0/TCP 127.0.0.1:5060;branch=[branch]\r\n"
- "From: sipp ;tag=12345\r\n"
- "To: \r\n"
- "Call-ID: 12345\r\n"
- "CSeq: 1 INVITE\r\n"
- "Contact: sip:127.0.0.1:5061\r\n"
- "Max-Forwards: 70\r\n"
- "Content-Type: application/sdp\r\n"
- "l:130\r\n" /* intentionally no space */
- "\r\n"
- "v=0\r\n"
- "o=user1 53655765 2353687637 IN IP4 127.0.0.1\r\n"
- "s=-\r\n"
- "c=IN IP4 127.0.0.1\r\n"
- "t=0 0\r\n"
- "m=audio 10000 RTP/AVP 0\r\n"
- "a=rtpmap:0 PCMU/8000\r\n"
- };
-
- /* Message with faux content-length headers
- * Same as "normal" but with extra fake content-length headers
- */
- char *faux[] = {
- "INVITE sip:bob@example.org SIP/2.0\r\n"
- "Via: SIP/2.0/TCP 127.0.0.1:5060;branch=[branch]\r\n"
- "From: sipp ;tag=12345\r\n"
- "To: \r\n"
- "Call-ID: 12345\r\n"
- "CSeq: 1 INVITE\r\n"
- "Contact: sip:127.0.0.1:5061\r\n"
- "Max-Forwards: 70\r\n"
- "Content-Type: application/sdp\r\n"
- "DisContent-Length: 0\r\n"
- "MalContent-Length: 60\r\n"
- "Content-Length:130\r\n" /* intentionally no space */
- "\r\n"
- "v=0\r\n"
- "o=user1 53655765 2353687637 IN IP4 127.0.0.1\r\n"
- "s=-\r\n"
- "c=IN IP4 127.0.0.1\r\n"
- "t=0 0\r\n"
- "m=audio 10000 RTP/AVP 0\r\n"
- "a=rtpmap:0 PCMU/8000\r\n"
- };
-
- /* Message with folded Content-Length header
- * Message is "normal" with Content-Length spread across three lines
- *
- * This is the test that requires pedantic=yes in order to pass
- */
- char *folded[] = {
- "INVITE sip:bob@example.org SIP/2.0\r\n"
- "Via: SIP/2.0/TCP 127.0.0.1:5060;branch=[branch]\r\n"
- "From: sipp ;tag=12345\r\n"
- "To: \r\n"
- "Call-ID: 12345\r\n"
- "CSeq: 1 INVITE\r\n"
- "Contact: sip:127.0.0.1:5061\r\n"
- "Max-Forwards: 70\r\n"
- "Content-Type: application/sdp\r\n"
- "Content-Length: \t\r\n"
- "\t \r\n"
- " 130\t \r\n"
- "\r\n"
- "v=0\r\n"
- "o=user1 53655765 2353687637 IN IP4 127.0.0.1\r\n"
- "s=-\r\n"
- "c=IN IP4 127.0.0.1\r\n"
- "t=0 0\r\n"
- "m=audio 10000 RTP/AVP 0\r\n"
- "a=rtpmap:0 PCMU/8000\r\n"
- };
-
- /* Message with compact Content-length header in message and
- * full Content-Length header in the body. Ensure that the header
- * in the message is read and that the one in the body is ignored
- */
- char *cl_in_body[] = {
- "INVITE sip:bob@example.org SIP/2.0\r\n"
- "Via: SIP/2.0/TCP 127.0.0.1:5060;branch=[branch]\r\n"
- "From: sipp ;tag=12345\r\n"
- "To: \r\n"
- "Call-ID: 12345\r\n"
- "CSeq: 1 INVITE\r\n"
- "Contact: sip:127.0.0.1:5061\r\n"
- "Max-Forwards: 70\r\n"
- "Content-Type: application/sdp\r\n"
- "l: 149\r\n"
- "\r\n"
- "v=0\r\n"
- "Content-Length: 0\r\n"
- "o=user1 53655765 2353687637 IN IP4 127.0.0.1\r\n"
- "s=-\r\n"
- "c=IN IP4 127.0.0.1\r\n"
- "t=0 0\r\n"
- "m=audio 10000 RTP/AVP 0\r\n"
- "a=rtpmap:0 PCMU/8000\r\n"
- };
-
- struct ast_str *overflow;
- struct {
- char **fragments;
- size_t fragment_count;
- char **expected;
- int num_expected;
- const char *description;
- } tests[] = {
- { normal, ARRAY_LEN(normal), normal, 1, "normal" },
- { fragmented, ARRAY_LEN(fragmented), normal, 1, "fragmented" },
- { fragmented_body, ARRAY_LEN(fragmented_body), normal, 1, "fragmented_body" },
- { multi_fragment, ARRAY_LEN(multi_fragment), normal, 1, "multi_fragment" },
- { multi_message, ARRAY_LEN(multi_message), multi_message_divided, 2, "multi_message" },
- { multi_message_body, ARRAY_LEN(multi_message_body), multi_message_body_divided, 2, "multi_message_body" },
- { multi_message_in_fragments, ARRAY_LEN(multi_message_in_fragments), multi_message_divided, 2, "multi_message_in_fragments" },
- { compact, ARRAY_LEN(compact), compact, 1, "compact" },
- { faux, ARRAY_LEN(faux), faux, 1, "faux" },
- { folded, ARRAY_LEN(folded), folded, 1, "folded" },
- { cl_in_body, ARRAY_LEN(cl_in_body), cl_in_body, 1, "cl_in_body" },
- };
- int i;
- enum ast_test_result_state res = AST_TEST_PASS;
-
- switch (cmd) {
- case TEST_INIT:
- info->name = "sip_tcp_message_fragmentation";
- info->category = "/main/sip/transport/";
- info->summary = "SIP TCP message fragmentation test";
- info->description =
- "Tests reception of different TCP messages that have been fragmented or"
- "run together. This test mimics the code that TCP reception uses.";
- return AST_TEST_NOT_RUN;
- case TEST_EXECUTE:
- break;
- }
- if (!sip_cfg.pedanticsipchecking) {
- ast_log(LOG_WARNING, "Not running test. Pedantic SIP checking is not enabled, so it is guaranteed to fail\n");
- return AST_TEST_NOT_RUN;
- }
-
- overflow = ast_str_create(128);
- if (!overflow) {
- return AST_TEST_FAIL;
- }
- for (i = 0; i < ARRAY_LEN(tests); ++i) {
- int num_messages = 0;
- if (mock_tcp_loop(tests[i].fragments, tests[i].fragment_count,
- &overflow, tests[i].expected, &num_messages, test)) {
- ast_test_status_update(test, "Failed to parse message '%s'\n", tests[i].description);
- res = AST_TEST_FAIL;
- break;
- }
- if (num_messages != tests[i].num_expected) {
- ast_test_status_update(test, "Did not receive the expected number of messages. "
- "Expected %d but received %d\n", tests[i].num_expected, num_messages);
- res = AST_TEST_FAIL;
- break;
- }
- }
- ast_free(overflow);
- return res;
-}
-
-AST_TEST_DEFINE(get_in_brackets_const_test)
-{
- const char *input;
- const char *start = NULL;
- int len = 0;
- int res;
-
-#define CHECK_RESULTS(in, expected_res, expected_start, expected_len) do { \
- input = (in); \
- res = get_in_brackets_const(input, &start, &len); \
- if ((expected_res) != res) { \
- ast_test_status_update(test, "Unexpected result: %d != %d\n", expected_res, res); \
- return AST_TEST_FAIL; \
- } \
- if ((void *)(expected_start) != (void *)start) { \
- const char *e = ((void *)expected_start != (void *)NULL) ? expected_start : "(null)"; \
- const char *a = start ? start : "(null)"; \
- ast_test_status_update(test, "Unexpected start: %s != %s\n", e, a); \
- return AST_TEST_FAIL; \
- } \
- if ((expected_len) != len) { \
- ast_test_status_update(test, "Unexpected len: %d != %d\n", expected_len, len); \
- return AST_TEST_FAIL; \
- } \
- } while(0)
-
- switch (cmd) {
- case TEST_INIT:
- info->name = __func__;
- info->category = "/channels/chan_sip/";
- info->summary = "get_in_brackets_const test";
- info->description =
- "Tests the get_in_brackets_const function";
- return AST_TEST_NOT_RUN;
- case TEST_EXECUTE:
- break;
- }
-
- CHECK_RESULTS("", 1, NULL, -1);
- CHECK_RESULTS("normal ", 0, input + 8, 4);
- CHECK_RESULTS("\"normal\" ", 0, input + 10, 4);
- CHECK_RESULTS("not normal ", 0, input + 16, 4);
- CHECK_RESULTS("\"even > this\" ", 0, input + 15, 4);
- CHECK_RESULTS("", 0, input + 1, 22);
- CHECK_RESULTS(", ", 0, input + 1, 22);
- CHECK_RESULTS("", 0, input + 1, 26);
- CHECK_RESULTS("", 0, input + 1, 36);
- CHECK_RESULTS("\"quoted text\" ", 0, input + 15, 23);
-
- return AST_TEST_PASS;
-}
-
-#endif
-
-static const struct ast_sip_api_tech chan_sip_api_provider = {
- .version = AST_SIP_API_VERSION,
- .name = "chan_sip",
- .sipinfo_send = sipinfo_send,
-};
-
-static void deprecation_notice(void)
-{
- ast_log(LOG_WARNING, "chan_sip has no official maintainer and is deprecated. Migration to\n");
- ast_log(LOG_WARNING, "chan_pjsip is recommended. See guides at the Asterisk Wiki:\n");
- ast_log(LOG_WARNING, "https://wiki.asterisk.org/wiki/display/AST/Migrating+from+chan_sip+to+res_pjsip\n");
- ast_log(LOG_WARNING, "https://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip\n");
-}
-
-/*! \brief Event callback which indicates we're fully booted */
-static void startup_event_cb(void *data, struct stasis_subscription *sub, struct stasis_message *message)
-{
- struct ast_json_payload *payload;
- const char *type;
-
- if (stasis_message_type(message) != ast_manager_get_generic_type()) {
- return;
- }
-
- payload = stasis_message_data(message);
- type = ast_json_string_get(ast_json_object_get(payload->json, "type"));
-
- if (strcmp(type, "FullyBooted")) {
- return;
- }
-
- deprecation_notice();
-
- stasis_unsubscribe(sub);
-}
-
-
-static int unload_module(void);
-
-/*!
- * \brief Load the module
- *
- * Module loading including tests for configuration or dependencies.
- * This function can return AST_MODULE_LOAD_FAILURE, AST_MODULE_LOAD_DECLINE,
- * or AST_MODULE_LOAD_SUCCESS. If a dependency or environment variable fails
- * tests return AST_MODULE_LOAD_FAILURE. If the module can not load the
- * configuration file or other non-critical problem return
- * AST_MODULE_LOAD_DECLINE. On success return AST_MODULE_LOAD_SUCCESS.
- */
-static int load_module(void)
-{
- struct sip_peer *bogus_peer;
-
- ast_verbose("SIP channel loading...\n");
- log_level = ast_logger_register_level("SIP_HISTORY");
- if (log_level < 0) {
- ast_log(LOG_WARNING, "Unable to register history log level\n");
- }
-
- if (STASIS_MESSAGE_TYPE_INIT(session_timeout_type)) {
- unload_module();
- return AST_MODULE_LOAD_DECLINE;
- }
-
- if (!(sip_tech.capabilities = ast_format_cap_alloc(0))) {
- unload_module();
- return AST_MODULE_LOAD_DECLINE;
- }
-
- if (ast_sip_api_provider_register(&chan_sip_api_provider)) {
- unload_module();
- return AST_MODULE_LOAD_DECLINE;
- }
-
- /* the fact that ao2_containers can't resize automatically is a major worry! */
- /* if the number of objects gets above MAX_XXX_BUCKETS, things will slow down */
- peers = ao2_t_container_alloc_hash(AO2_ALLOC_OPT_LOCK_MUTEX, 0, HASH_PEER_SIZE,
- peer_hash_cb, NULL, peer_cmp_cb, "allocate peers");
- peers_by_ip = ao2_t_container_alloc_hash(AO2_ALLOC_OPT_LOCK_MUTEX, 0, HASH_PEER_SIZE,
- peer_iphash_cb, NULL, NULL, "allocate peers_by_ip");
- dialogs = ao2_t_container_alloc_hash(AO2_ALLOC_OPT_LOCK_MUTEX, 0, HASH_DIALOG_SIZE,
- dialog_hash_cb, NULL, dialog_cmp_cb, "allocate dialogs");
- dialogs_needdestroy = ao2_t_container_alloc_hash(AO2_ALLOC_OPT_LOCK_MUTEX, 0, 1,
- NULL, NULL, NULL, "allocate dialogs_needdestroy");
- dialogs_rtpcheck = ao2_t_container_alloc_hash(AO2_ALLOC_OPT_LOCK_MUTEX, 0, HASH_DIALOG_SIZE,
- dialog_hash_cb, NULL, dialog_cmp_cb, "allocate dialogs for rtpchecks");
- threadt = ao2_t_container_alloc_hash(AO2_ALLOC_OPT_LOCK_MUTEX, 0, HASH_DIALOG_SIZE,
- threadt_hash_cb, NULL, threadt_cmp_cb, "allocate threadt table");
- if (!peers || !peers_by_ip || !dialogs || !dialogs_needdestroy || !dialogs_rtpcheck
- || !threadt) {
- ast_log(LOG_ERROR, "Unable to create primary SIP container(s)\n");
- unload_module();
- return AST_MODULE_LOAD_DECLINE;
- }
-
- if (!(sip_cfg.caps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT))) {
- unload_module();
- return AST_MODULE_LOAD_DECLINE;
- }
- ast_format_cap_append_by_type(sip_tech.capabilities, AST_MEDIA_TYPE_AUDIO);
-
- registry_list = ao2_t_container_alloc_hash(AO2_ALLOC_OPT_LOCK_MUTEX, 0, HASH_REGISTRY_SIZE,
- registry_hash_cb, NULL, registry_cmp_cb, "allocate registry_list");
- subscription_mwi_list = ao2_t_container_alloc_list(AO2_ALLOC_OPT_LOCK_MUTEX,
- AO2_CONTAINER_ALLOC_OPT_INSERT_BEGIN, NULL, NULL, "allocate subscription_mwi_list");
-
- if (!(sched = ast_sched_context_create())) {
- ast_log(LOG_ERROR, "Unable to create scheduler context\n");
- unload_module();
- return AST_MODULE_LOAD_DECLINE;
- }
-
- if (!(io = io_context_create())) {
- ast_log(LOG_ERROR, "Unable to create I/O context\n");
- unload_module();
- return AST_MODULE_LOAD_DECLINE;
- }
-
- sip_reloadreason = CHANNEL_MODULE_LOAD;
-
- can_parse_xml = sip_is_xml_parsable();
- if (reload_config(sip_reloadreason)) { /* Load the configuration from sip.conf */
- unload_module();
- return AST_MODULE_LOAD_DECLINE;
- }
-
- /* Initialize bogus peer. Can be done first after reload_config() */
- if (!(bogus_peer = temp_peer("(bogus_peer)"))) {
- ast_log(LOG_ERROR, "Unable to create bogus_peer for authentication\n");
- unload_module();
- return AST_MODULE_LOAD_DECLINE;
- }
- /* Make sure the auth will always fail. */
- ast_string_field_set(bogus_peer, md5secret, BOGUS_PEER_MD5SECRET);
- ast_clear_flag(&bogus_peer->flags[0], SIP_INSECURE);
- ao2_t_global_obj_replace_unref(g_bogus_peer, bogus_peer, "Set the initial bogus peer.");
- ao2_t_ref(bogus_peer, -1, "Module load is done with the bogus peer.");
-
- /* Prepare the version that does not require DTMF BEGIN frames.
- * We need to use tricks such as memcpy and casts because the variable
- * has const fields.
- */
- memcpy(&sip_tech_info, &sip_tech, sizeof(sip_tech));
- memset((void *) &sip_tech_info.send_digit_begin, 0, sizeof(sip_tech_info.send_digit_begin));
-
- if (ast_msg_tech_register(&sip_msg_tech)) {
- unload_module();
- return AST_MODULE_LOAD_DECLINE;
- }
-
- /* Make sure we can register our sip channel type */
- if (ast_channel_register(&sip_tech)) {
- ast_log(LOG_ERROR, "Unable to register channel type 'SIP'\n");
- unload_module();
- return AST_MODULE_LOAD_DECLINE;
- }
-
-#ifdef TEST_FRAMEWORK
- AST_TEST_REGISTER(test_sip_mwi_subscribe_parse);
- AST_TEST_REGISTER(test_tcp_message_fragmentation);
- AST_TEST_REGISTER(get_in_brackets_const_test);
-#endif
-
- /* Register all CLI functions for SIP */
- ast_cli_register_multiple(cli_sip, ARRAY_LEN(cli_sip));
-
- /* Tell the RTP engine about our RTP glue */
- ast_rtp_glue_register(&sip_rtp_glue);
-
- /* Register dialplan applications */
- ast_register_application_xml(app_dtmfmode, sip_dtmfmode);
- ast_register_application_xml(app_sipaddheader, sip_addheader);
- ast_register_application_xml(app_sipremoveheader, sip_removeheader);
-#ifdef TEST_FRAMEWORK
- ast_register_application_xml(app_sipsendcustominfo, sip_sendcustominfo);
-#endif
-
- /* Register dialplan functions */
- ast_custom_function_register(&sip_header_function);
- ast_custom_function_register(&sip_headers_function);
- ast_custom_function_register(&sippeer_function);
- ast_custom_function_register(&checksipdomain_function);
-
- /* Register manager commands */
- ast_manager_register_xml("SIPpeers", EVENT_FLAG_SYSTEM | EVENT_FLAG_REPORTING, manager_sip_show_peers);
- ast_manager_register_xml("SIPshowpeer", EVENT_FLAG_SYSTEM | EVENT_FLAG_REPORTING, manager_sip_show_peer);
- ast_manager_register_xml("SIPqualifypeer", EVENT_FLAG_SYSTEM | EVENT_FLAG_REPORTING, manager_sip_qualify_peer);
- ast_manager_register_xml("SIPshowregistry", EVENT_FLAG_SYSTEM | EVENT_FLAG_REPORTING, manager_show_registry);
- ast_manager_register_xml("SIPnotify", EVENT_FLAG_SYSTEM, manager_sipnotify);
- ast_manager_register_xml("SIPpeerstatus", EVENT_FLAG_SYSTEM, manager_sip_peer_status);
- sip_poke_all_peers();
- sip_keepalive_all_peers();
- sip_send_all_registers();
- sip_send_all_mwi_subscriptions();
- initialize_escs();
-
- if (sip_epa_register(&cc_epa_static_data)) {
- unload_module();
- return AST_MODULE_LOAD_DECLINE;
- }
-
- if (sip_reqresp_parser_init() == -1) {
- ast_log(LOG_ERROR, "Unable to initialize the SIP request and response parser\n");
- unload_module();
- return AST_MODULE_LOAD_DECLINE;
- }
-
- if (can_parse_xml) {
- /* SIP CC agents require the ability to parse XML PIDF bodies
- * in incoming PUBLISH requests
- */
- if (ast_cc_agent_register(&sip_cc_agent_callbacks)) {
- unload_module();
- return AST_MODULE_LOAD_DECLINE;
- }
- }
- if (ast_cc_monitor_register(&sip_cc_monitor_callbacks)) {
- unload_module();
- return AST_MODULE_LOAD_DECLINE;
- }
- sip_monitor_instances = ao2_container_alloc_hash(AO2_ALLOC_OPT_LOCK_MUTEX, 0, 37,
- sip_monitor_instance_hash_fn, NULL, sip_monitor_instance_cmp_fn);
- if (!sip_monitor_instances) {
- unload_module();
- return AST_MODULE_LOAD_DECLINE;
- }
-
- /* And start the monitor for the first time */
- restart_monitor();
-
- if (sip_cfg.peer_rtupdate) {
- ast_realtime_require_field(ast_check_realtime("sipregs") ? "sipregs" : "sippeers",
- "name", RQ_CHAR, 10,
- "ipaddr", RQ_CHAR, INET6_ADDRSTRLEN - 1,
- "port", RQ_UINTEGER2, 5,
- "regseconds", RQ_INTEGER4, 11,
- "defaultuser", RQ_CHAR, 10,
- "fullcontact", RQ_CHAR, 35,
- "regserver", RQ_CHAR, 20,
- "useragent", RQ_CHAR, 20,
- "lastms", RQ_INTEGER4, 11,
- SENTINEL);
- }
-
-
- sip_register_tests();
- network_change_stasis_subscribe();
-
- if (sip_cfg.websocket_enabled) {
- ast_websocket_add_protocol("sip", sip_websocket_callback);
- }
-
- if (ast_fully_booted) {
- deprecation_notice();
- } else {
- stasis_subscribe_pool(ast_manager_get_topic(), startup_event_cb, NULL);
- }
-
- return AST_MODULE_LOAD_SUCCESS;
-}
-
-/*! \brief PBX unload module API */
-static int unload_module(void)
-{
- struct sip_pvt *p;
- struct sip_threadinfo *th;
- struct ao2_iterator i;
- struct timeval start;
-
- ast_sched_dump(sched);
-
- ast_sip_api_provider_unregister();
-
- if (sip_cfg.websocket_enabled) {
- ast_websocket_remove_protocol("sip", sip_websocket_callback);
- }
-
- network_change_stasis_unsubscribe();
- acl_change_event_stasis_unsubscribe();
-
- /* First, take us out of the channel type list */
- ast_channel_unregister(&sip_tech);
- ast_msg_tech_unregister(&sip_msg_tech);
- ast_cc_monitor_unregister(&sip_cc_monitor_callbacks);
- ast_cc_agent_unregister(&sip_cc_agent_callbacks);
-
- /* Unregister dial plan functions */
- ast_custom_function_unregister(&sippeer_function);
- ast_custom_function_unregister(&sip_headers_function);
- ast_custom_function_unregister(&sip_header_function);
- ast_custom_function_unregister(&checksipdomain_function);
-
- /* Unregister dial plan applications */
- ast_unregister_application(app_dtmfmode);
- ast_unregister_application(app_sipaddheader);
- ast_unregister_application(app_sipremoveheader);
-#ifdef TEST_FRAMEWORK
- ast_unregister_application(app_sipsendcustominfo);
-
- AST_TEST_UNREGISTER(test_sip_mwi_subscribe_parse);
- AST_TEST_UNREGISTER(test_tcp_message_fragmentation);
- AST_TEST_UNREGISTER(get_in_brackets_const_test);
-#endif
- /* Unregister CLI commands */
- ast_cli_unregister_multiple(cli_sip, ARRAY_LEN(cli_sip));
-
- /* Disconnect from RTP engine */
- ast_rtp_glue_unregister(&sip_rtp_glue);
-
- /* Unregister AMI actions */
- ast_manager_unregister("SIPpeers");
- ast_manager_unregister("SIPshowpeer");
- ast_manager_unregister("SIPqualifypeer");
- ast_manager_unregister("SIPshowregistry");
- ast_manager_unregister("SIPnotify");
- ast_manager_unregister("SIPpeerstatus");
-
- /* Kill TCP/TLS server threads */
- if (sip_tcp_desc.master) {
- ast_tcptls_server_stop(&sip_tcp_desc);
- }
- if (sip_tls_desc.master) {
- ast_tcptls_server_stop(&sip_tls_desc);
- }
- ast_ssl_teardown(sip_tls_desc.tls_cfg);
-
- /* Kill all existing TCP/TLS threads */
- i = ao2_iterator_init(threadt, 0);
- while ((th = ao2_t_iterator_next(&i, "iterate through tcp threads for 'sip show tcp'"))) {
- pthread_t thread = th->threadid;
- th->stop = 1;
- pthread_kill(thread, SIGURG);
- ao2_t_ref(th, -1, "decrement ref from iterator");
- }
- ao2_iterator_destroy(&i);
-
- /* Hangup all dialogs if they have an owner */
- i = ao2_iterator_init(dialogs, 0);
- while ((p = ao2_t_iterator_next(&i, "iterate thru dialogs"))) {
- if (p->owner)
- ast_softhangup(p->owner, AST_SOFTHANGUP_APPUNLOAD);
- ao2_t_ref(p, -1, "toss dialog ptr from iterator_next");
- }
- ao2_iterator_destroy(&i);
-
- ast_mutex_lock(&monlock);
- if (monitor_thread && (monitor_thread != AST_PTHREADT_STOP) && (monitor_thread != AST_PTHREADT_NULL)) {
- pthread_t th = monitor_thread;
- monitor_thread = AST_PTHREADT_STOP;
- pthread_cancel(th);
- pthread_kill(th, SIGURG);
- ast_mutex_unlock(&monlock);
- pthread_join(th, NULL);
- } else {
- monitor_thread = AST_PTHREADT_STOP;
- ast_mutex_unlock(&monlock);
- }
-
- /* Clear containers */
- unlink_all_peers_from_tables();
- cleanup_all_regs();
- sip_epa_unregister_all();
- destroy_escs();
- clear_sip_domains();
-
- {
- struct ao2_iterator iter;
- struct sip_subscription_mwi *mwi;
-
- iter = ao2_iterator_init(subscription_mwi_list, 0);
- while ((mwi = ao2_t_iterator_next(&iter, "unload_module iter"))) {
- shutdown_mwi_subscription(mwi);
- ao2_t_ref(mwi, -1, "unload_module iter");
- }
- ao2_iterator_destroy(&iter);
- }
-
- /* Destroy all the dialogs and free their memory */
- i = ao2_iterator_init(dialogs, 0);
- while ((p = ao2_t_iterator_next(&i, "iterate thru dialogs"))) {
- dialog_unlink_all(p);
- ao2_t_ref(p, -1, "throw away iterator result");
- }
- ao2_iterator_destroy(&i);
-
- /*
- * Since the monitor thread runs the scheduled events and we
- * just stopped the monitor thread above, we have to run any
- * pending scheduled immediate events in this thread.
- */
- ast_sched_runq(sched);
-
- /*
- * Wait awhile for the TCP/TLS thread container to become empty.
- *
- * XXX This is a hack, but the worker threads cannot be created
- * joinable. They can die on their own and remove themselves
- * from the container thus resulting in a huge memory leak.
- */
- start = ast_tvnow();
- while (ao2_container_count(threadt) && (ast_tvdiff_sec(ast_tvnow(), start) < 5)) {
- sched_yield();
- }
- if (ao2_container_count(threadt)) {
- ast_debug(2, "TCP/TLS thread container did not become empty :(\n");
-
- return -1;
- }
-
- /* Free memory for local network address mask */
- ast_free_ha(localaddr);
-
- ast_mutex_lock(&authl_lock);
- if (authl) {
- ao2_t_cleanup(authl, "Removing global authentication");
- authl = NULL;
- }
- ast_mutex_unlock(&authl_lock);
-
- ast_free(default_tls_cfg.certfile);
- ast_free(default_tls_cfg.pvtfile);
- ast_free(default_tls_cfg.cipher);
- ast_free(default_tls_cfg.cafile);
- ast_free(default_tls_cfg.capath);
-
- ast_rtp_dtls_cfg_free(&default_dtls_cfg);
-
- ao2_cleanup(registry_list);
- ao2_cleanup(subscription_mwi_list);
-
- ao2_t_global_obj_release(g_bogus_peer, "Release the bogus peer.");
-
- ao2_t_cleanup(peers, "unref the peers table");
- ao2_t_cleanup(peers_by_ip, "unref the peers_by_ip table");
- ao2_t_cleanup(dialogs, "unref the dialogs table");
- ao2_t_cleanup(dialogs_needdestroy, "unref dialogs_needdestroy");
- ao2_t_cleanup(dialogs_rtpcheck, "unref dialogs_rtpcheck");
- ao2_t_cleanup(threadt, "unref the thread table");
- ao2_t_cleanup(sip_monitor_instances, "unref the sip_monitor_instances table");
-
- sip_cfg.contact_acl = ast_free_acl_list(sip_cfg.contact_acl);
- if (sipsock_read_id) {
- ast_io_remove(io, sipsock_read_id);
- sipsock_read_id = NULL;
- }
- close(sipsock);
- io_context_destroy(io);
- ast_sched_context_destroy(sched);
- sched = NULL;
- ast_context_destroy_by_name(used_context, "SIP");
- ast_unload_realtime("sipregs");
- ast_unload_realtime("sippeers");
-
- sip_reqresp_parser_exit();
- sip_unregister_tests();
-
- if (notify_types) {
- ast_config_destroy(notify_types);
- notify_types = NULL;
- }
-
- ao2_cleanup(sip_tech.capabilities);
- sip_tech.capabilities = NULL;
- ao2_cleanup(sip_cfg.caps);
- sip_cfg.caps = NULL;
-
- STASIS_MESSAGE_TYPE_CLEANUP(session_timeout_type);
- if (log_level != -1) {
- ast_logger_unregister_level("SIP_HISTORY");
- }
-
- return 0;
-}
-
-AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "Session Initiation Protocol (SIP)",
- .support_level = AST_MODULE_SUPPORT_DEPRECATED,
- .load = load_module,
- .unload = unload_module,
- .reload = reload,
- .load_pri = AST_MODPRI_CHANNEL_DRIVER,
- .requires = "ccss,dnsmgr,udptl",
- .optional_modules = "res_crypto,res_http_websocket",
-);
diff --git a/channels/console_video.c b/channels/console_video.c
index 1975f06d60..f2c296b9b3 100644
--- a/channels/console_video.c
+++ b/channels/console_video.c
@@ -99,14 +99,20 @@ if the formats are equivalent. This will save some unnecessary format
conversion.
-In order to handle video you need to add to sip.conf (and presumably
-iax.conf too) the following:
+In order to handle video you need to add the following to the endpoint in
+pjsip.conf
- [general](+)
- videosupport=yes
allow=h263 ; this or other video formats
allow=h263p ; this or other video formats
+(Presumably, iax.conf would require):
+
+ [general](+)
+ videosupport=yes
+ allow=h263 ; this or other video formats
+ allow=h263p ; this or other video formats
+
+
*/
/*
diff --git a/channels/sip/config_parser.c b/channels/sip/config_parser.c
deleted file mode 100644
index b4d4fe5aed..0000000000
--- a/channels/sip/config_parser.c
+++ /dev/null
@@ -1,927 +0,0 @@
-/*
- * Asterisk -- An open source telephony toolkit.
- *
- * Copyright (C) 2010, Digium, Inc.
- *
- * See http://www.asterisk.org for more information about
- * the Asterisk project. Please do not directly contact
- * any of the maintainers of this project for assistance;
- * the project provides a web site, mailing lists and IRC
- * channels for your use.
- *
- * This program is free software, distributed under the terms of
- * the GNU General Public License Version 2. See the LICENSE file
- * at the top of the source tree.
- */
-
-/*!
- * \file
- * \brief sip config parsing functions and unit tests
- */
-
-/*** MODULEINFO
- deprecated
- ***/
-
-#include "asterisk.h"
-
-#include "include/sip.h"
-#include "include/config_parser.h"
-#include "include/sip_utils.h"
-
-/*! \brief Parse register=> line in sip.conf
- *
- * \retval 0 on success
- * \retval -1 on failure
- */
-int sip_parse_register_line(struct sip_registry *reg, int default_expiry, const char *value, int lineno)
-{
- int portnum = 0;
- int domainport = 0;
- enum ast_transport transport = AST_TRANSPORT_UDP;
- char buf[256] = "";
- char *userpart = NULL, *hostpart = NULL;
- /* register => [peer?][transport://]user[@domain][:secret[:authuser]]@host[:port][/extension][~expiry] */
- AST_DECLARE_APP_ARGS(pre1,
- AST_APP_ARG(peer);
- AST_APP_ARG(userpart);
- );
- AST_DECLARE_APP_ARGS(pre2,
- AST_APP_ARG(transport);
- AST_APP_ARG(blank);
- AST_APP_ARG(userpart);
- );
- AST_DECLARE_APP_ARGS(user1,
- AST_APP_ARG(userpart);
- AST_APP_ARG(secret);
- AST_APP_ARG(authuser);
- );
- AST_DECLARE_APP_ARGS(user2,
- AST_APP_ARG(user);
- AST_APP_ARG(domain);
- );
- AST_DECLARE_APP_ARGS(user3,
- AST_APP_ARG(authuser);
- AST_APP_ARG(domainport);
- );
- AST_DECLARE_APP_ARGS(host1,
- AST_APP_ARG(hostpart);
- AST_APP_ARG(expiry);
- );
- AST_DECLARE_APP_ARGS(host2,
- AST_APP_ARG(hostpart);
- AST_APP_ARG(extension);
- );
- AST_DECLARE_APP_ARGS(host3,
- AST_APP_ARG(host);
- AST_APP_ARG(port);
- );
-
- if (!reg) {
- return -1;
- }
-
- reg->expire = -1;
- reg->timeout = -1;
-
- if (!value) {
- return -1;
- }
-
- ast_copy_string(buf, value, sizeof(buf));
-
- /*
- * register => [peer?][transport://]user[@domain][:secret[:authuser]]@host[:port][/extension][~expiry]
- * becomes
- * userpart => [peer?][transport://]user[@domain][:secret[:authuser]]
- * hostpart => host[:port][/extension][~expiry]
- */
- if ((hostpart = strrchr(buf, '@'))) {
- *hostpart++ = '\0';
- userpart = buf;
- }
-
- if (ast_strlen_zero(userpart) || ast_strlen_zero(hostpart)) {
- ast_log(LOG_WARNING, "Format for registration is [peer?][transport://]user[@domain][:secret[:authuser]]@host[:port][/extension][~expiry] at line %d\n", lineno);
- return -1;
- }
-
- /*
- * pre1.peer => peer
- * pre1.userpart => [transport://]user[@domain][:secret[:authuser]]
- * hostpart => host[:port][/extension][~expiry]
- */
- AST_NONSTANDARD_RAW_ARGS(pre1, userpart, '?');
- if (ast_strlen_zero(pre1.userpart)) {
- pre1.userpart = pre1.peer;
- pre1.peer = NULL;
- }
-
- /*
- * pre1.peer => peer
- * pre2.transport = transport
- * pre2.userpart => user[@domain][:secret[:authuser]]
- * hostpart => host[:port][/extension][~expiry]
- */
- AST_NONSTANDARD_RAW_ARGS(pre2, pre1.userpart, '/');
- if (ast_strlen_zero(pre2.userpart)) {
- pre2.userpart = pre2.transport;
- pre2.transport = NULL;
- } else {
- pre2.transport[strlen(pre2.transport) - 1] = '\0'; /* Remove trailing : */
- }
-
- if (!ast_strlen_zero(pre2.blank)) {
- ast_log(LOG_WARNING, "Format for registration is [peer?][transport://]user[@domain][:secret[:authuser]]@host[:port][/extension][~expiry] at line %d\n", lineno);
- return -1;
- }
-
- /*
- * pre1.peer => peer
- * pre2.transport = transport
- * user1.userpart => user[@domain]
- * user1.secret => secret
- * user1.authuser => authuser
- * hostpart => host[:port][/extension][~expiry]
- */
- AST_NONSTANDARD_RAW_ARGS(user1, pre2.userpart, ':');
-
- /*
- * pre1.peer => peer
- * pre2.transport = transport
- * user1.userpart => user[@domain]
- * user1.secret => secret
- * user1.authuser => authuser
- * host1.hostpart => host[:port][/extension]
- * host1.expiry => [expiry]
- */
- AST_NONSTANDARD_RAW_ARGS(host1, hostpart, '~');
-
- /*
- * pre1.peer => peer
- * pre2.transport = transport
- * user1.userpart => user[@domain]
- * user1.secret => secret
- * user1.authuser => authuser
- * host2.hostpart => host[:port]
- * host2.extension => [extension]
- * host1.expiry => [expiry]
- */
- AST_NONSTANDARD_RAW_ARGS(host2, host1.hostpart, '/');
-
- /*
- * pre1.peer => peer
- * pre2.transport = transport
- * user1.userpart => user[@domain]
- * user1.secret => secret
- * user1.authuser => authuser
- * host3.host => host
- * host3.port => port
- * host2.extension => extension
- * host1.expiry => expiry
- */
- AST_NONSTANDARD_RAW_ARGS(host3, host2.hostpart, ':');
-
- /*
- * pre1.peer => peer
- * pre2.transport = transport
- * user2.user => user
- * user2.domain => domain
- * user1.secret => secret
- * user1.authuser => authuser
- * host3.host => host
- * host3.port => port
- * host2.extension => extension
- * host1.expiry => expiry
- */
- AST_NONSTANDARD_RAW_ARGS(user2, user1.userpart, '@');
-
- /*
- * pre1.peer => peer
- * pre2.transport = transport
- * user2.user => user
- * user2.domain => domain
- * user1.secret => secret
- * user3.authuser => authuser
- * user3.domainport => domainport
- * host3.host => host
- * host3.port => port
- * host2.extension => extension
- * host1.expiry => expiry
- */
- AST_NONSTANDARD_RAW_ARGS(user3, user1.authuser, ':');
-
- /* Reordering needed due to fields being [(:secret[:username])|(:regdomainport:secret:username)]
- but parsing being [secret[:username[:regdomainport]]] */
- if (user3.argc == 2) {
- char *reorder = user3.domainport;
- user3.domainport = user1.secret;
- user1.secret = user3.authuser;
- user3.authuser = reorder;
- }
-
- if (host3.port) {
- if (!(portnum = port_str2int(host3.port, 0))) {
- ast_log(LOG_NOTICE, "'%s' is not a valid port number on line %d of sip.conf. using default.\n", host3.port, lineno);
- }
- }
- if (user3.domainport) {
- if (!(domainport = port_str2int(user3.domainport, 0))) {
- ast_log(LOG_NOTICE, "'%s' is not a valid domain port number on line %d of sip.conf. using default.\n", user3.domainport, lineno);
- }
- }
-
- /* set transport type */
- if (!pre2.transport) {
- transport = AST_TRANSPORT_UDP;
- } else if (!strncasecmp(pre2.transport, "tcp", 3)) {
- transport = AST_TRANSPORT_TCP;
- } else if (!strncasecmp(pre2.transport, "tls", 3)) {
- transport = AST_TRANSPORT_TLS;
- } else if (!strncasecmp(pre2.transport, "udp", 3)) {
- transport = AST_TRANSPORT_UDP;
- } else {
- transport = AST_TRANSPORT_UDP;
- ast_log(LOG_NOTICE, "'%.3s' is not a valid transport type on line %d of sip.conf. defaulting to udp.\n", pre2.transport, lineno);
- }
-
- /* if no portnum specified, set default for transport */
- if (!portnum) {
- if (transport == AST_TRANSPORT_TLS) {
- portnum = STANDARD_TLS_PORT;
- } else {
- portnum = STANDARD_SIP_PORT;
- }
- }
-
- /* copy into sip_registry object */
- ast_string_field_set(reg, callback, ast_strip_quoted(S_OR(host2.extension, "s"), "\"", "\""));
- ast_string_field_set(reg, username, ast_strip_quoted(S_OR(user2.user, ""), "\"", "\""));
- ast_string_field_set(reg, hostname, ast_strip_quoted(S_OR(host3.host, ""), "\"", "\""));
- ast_string_field_set(reg, authuser, ast_strip_quoted(S_OR(user3.authuser, ""), "\"", "\""));
- ast_string_field_set(reg, secret, ast_strip_quoted(S_OR(user1.secret, ""), "\"", "\""));
- ast_string_field_set(reg, peername, ast_strip_quoted(S_OR(pre1.peer, ""), "\"", "\""));
- ast_string_field_set(reg, regdomain, ast_strip_quoted(S_OR(user2.domain, ""), "\"", "\""));
-
- reg->transport = transport;
- reg->portno = portnum;
- reg->regdomainport = domainport;
- reg->callid_valid = FALSE;
- reg->ocseq = INITIAL_CSEQ;
- reg->refresh = reg->expiry = reg->configured_expiry = (host1.expiry ? atoi(ast_strip_quoted(host1.expiry, "\"", "\"")) : default_expiry);
-
- return 0;
-}
-
-#ifdef TEST_FRAMEWORK
-AST_TEST_DEFINE(sip_parse_register_line_test)
-{
- int res = AST_TEST_PASS;
- struct sip_registry *reg;
- int default_expiry = 120;
- const char *reg1 = "name@domain";
- const char *reg2 = "name:pass@domain";
- const char *reg3 = "name@namedomain:pass:authuser@domain";
- const char *reg4 = "name@namedomain:pass:authuser@domain/extension";
- const char *reg5 = "tcp://name@namedomain:pass:authuser@domain/extension";
- const char *reg6 = "tls://name@namedomain:pass:authuser@domain/extension~111";
- const char *reg7 = "peer?tcp://name@namedomain:pass:authuser@domain:1234/extension~111";
- const char *reg8 = "peer?name@namedomain:pass:authuser@domain:1234/extension~111";
- const char *reg9 = "peer?name:pass:authuser:1234/extension~111";
- const char *reg10 = "@domin:1234";
- const char *reg12 = "name@namedomain:4321:pass:authuser@domain";
- const char *reg13 = "name@namedomain:4321::@domain";
-
- switch (cmd) {
- case TEST_INIT:
- info->name = "sip_parse_register_line_test";
- info->category = "/channels/chan_sip/";
- info->summary = "tests sip register line parsing";
- info->description =
- "Tests parsing of various register line configurations. "
- "Verifies output matches expected behavior.";
- return AST_TEST_NOT_RUN;
- case TEST_EXECUTE:
- break;
- }
-
- /* ---Test reg 1, simple config --- */
- if (!(reg = ast_calloc_with_stringfields(1, struct sip_registry, 256))) {
- goto alloc_fail;
- } else if (
- sip_parse_register_line(reg, default_expiry, reg1, 1) ||
- strcmp(reg->callback, "s") ||
- strcmp(reg->username, "name") ||
- strcmp(reg->regdomain, "") ||
- strcmp(reg->hostname, "domain") ||
- strcmp(reg->authuser, "") ||
- strcmp(reg->secret, "") ||
- strcmp(reg->peername, "") ||
- reg->transport != AST_TRANSPORT_UDP ||
- reg->timeout != -1 ||
- reg->expire != -1 ||
- reg->refresh != default_expiry ||
- reg->expiry != default_expiry ||
- reg->configured_expiry != default_expiry ||
- reg->portno != STANDARD_SIP_PORT ||
- (reg->regdomainport) ||
- reg->callid_valid != FALSE ||
- reg->ocseq != INITIAL_CSEQ) {
-
- ast_test_status_update(test, "Test 1: simple config failed\n");
- res = AST_TEST_FAIL;
- }
- ast_string_field_free_memory(reg);
- ast_free(reg);
-
- /* ---Test reg 2, add secret --- */
- if (!(reg = ast_calloc_with_stringfields(1, struct sip_registry, 256))) {
- goto alloc_fail;
- } else if (
- sip_parse_register_line(reg, default_expiry, reg2, 1) ||
- strcmp(reg->callback, "s") ||
- strcmp(reg->username, "name") ||
- strcmp(reg->regdomain, "") ||
- strcmp(reg->hostname, "domain") ||
- strcmp(reg->authuser, "") ||
- strcmp(reg->secret, "pass") ||
- strcmp(reg->peername, "") ||
- reg->transport != AST_TRANSPORT_UDP ||
- reg->timeout != -1 ||
- reg->expire != -1 ||
- reg->refresh != default_expiry ||
- reg->expiry != default_expiry ||
- reg->configured_expiry != default_expiry ||
- reg->portno != STANDARD_SIP_PORT ||
- (reg->regdomainport) ||
- reg->callid_valid != FALSE ||
- reg->ocseq != INITIAL_CSEQ) {
-
- ast_test_status_update(test, "Test 2: add secret failed\n");
- res = AST_TEST_FAIL;
- }
- ast_string_field_free_memory(reg);
- ast_free(reg);
-
- /* ---Test reg 3, add userdomain and authuser --- */
- if (!(reg = ast_calloc_with_stringfields(1, struct sip_registry, 256))) {
- goto alloc_fail;
- } else if (
- sip_parse_register_line(reg, default_expiry, reg3, 1) ||
- strcmp(reg->callback, "s") ||
- strcmp(reg->username, "name") ||
- strcmp(reg->regdomain, "namedomain") ||
- strcmp(reg->hostname, "domain") ||
- strcmp(reg->authuser, "authuser") ||
- strcmp(reg->secret, "pass") ||
- strcmp(reg->peername, "") ||
- reg->transport != AST_TRANSPORT_UDP ||
- reg->timeout != -1 ||
- reg->expire != -1 ||
- reg->refresh != default_expiry ||
- reg->expiry != default_expiry ||
- reg->configured_expiry != default_expiry ||
- reg->portno != STANDARD_SIP_PORT ||
- (reg->regdomainport) ||
- reg->callid_valid != FALSE ||
- reg->ocseq != INITIAL_CSEQ) {
-
- ast_test_status_update(test, "Test 3: add userdomain and authuser failed\n");
- res = AST_TEST_FAIL;
- }
- ast_string_field_free_memory(reg);
- ast_free(reg);
-
- /* ---Test reg 4, add callback extension --- */
- if (!(reg = ast_calloc_with_stringfields(1, struct sip_registry, 256))) {
- goto alloc_fail;
- } else if (
- sip_parse_register_line(reg, default_expiry, reg4, 1) ||
- strcmp(reg->callback, "extension") ||
- strcmp(reg->username, "name") ||
- strcmp(reg->regdomain, "namedomain") ||
- strcmp(reg->hostname, "domain") ||
- strcmp(reg->authuser, "authuser") ||
- strcmp(reg->secret, "pass") ||
- strcmp(reg->peername, "") ||
- reg->transport != AST_TRANSPORT_UDP ||
- reg->timeout != -1 ||
- reg->expire != -1 ||
- reg->refresh != default_expiry ||
- reg->expiry != default_expiry ||
- reg->configured_expiry != default_expiry ||
- reg->portno != STANDARD_SIP_PORT ||
- (reg->regdomainport) ||
- reg->callid_valid != FALSE ||
- reg->ocseq != INITIAL_CSEQ) {
-
- ast_test_status_update(test, "Test 4: add callback extension failed\n");
- res = AST_TEST_FAIL;
- }
- ast_string_field_free_memory(reg);
- ast_free(reg);
-
- /* ---Test reg 5, add transport --- */
- if (!(reg = ast_calloc_with_stringfields(1, struct sip_registry, 256))) {
- goto alloc_fail;
- } else if (
- sip_parse_register_line(reg, default_expiry, reg5, 1) ||
- strcmp(reg->callback, "extension") ||
- strcmp(reg->username, "name") ||
- strcmp(reg->regdomain, "namedomain") ||
- strcmp(reg->hostname, "domain") ||
- strcmp(reg->authuser, "authuser") ||
- strcmp(reg->secret, "pass") ||
- strcmp(reg->peername, "") ||
- reg->transport != AST_TRANSPORT_TCP ||
- reg->timeout != -1 ||
- reg->expire != -1 ||
- reg->refresh != default_expiry ||
- reg->expiry != default_expiry ||
- reg->configured_expiry != default_expiry ||
- reg->portno != STANDARD_SIP_PORT ||
- (reg->regdomainport) ||
- reg->callid_valid != FALSE ||
- reg->ocseq != INITIAL_CSEQ) {
-
- ast_test_status_update(test, "Test 5: add transport failed\n");
- res = AST_TEST_FAIL;
- }
- ast_string_field_free_memory(reg);
- ast_free(reg);
-
- /* ---Test reg 6, change to tls transport, add expiry --- */
- if (!(reg = ast_calloc_with_stringfields(1, struct sip_registry, 256))) {
- goto alloc_fail;
- } else if (
- sip_parse_register_line(reg, default_expiry, reg6, 1) ||
- strcmp(reg->callback, "extension") ||
- strcmp(reg->username, "name") ||
- strcmp(reg->regdomain, "namedomain") ||
- strcmp(reg->hostname, "domain") ||
- strcmp(reg->authuser, "authuser") ||
- strcmp(reg->secret, "pass") ||
- strcmp(reg->peername, "") ||
- reg->transport != AST_TRANSPORT_TLS ||
- reg->timeout != -1 ||
- reg->expire != -1 ||
- reg->refresh != 111 ||
- reg->expiry != 111 ||
- reg->configured_expiry != 111 ||
- reg->portno != STANDARD_TLS_PORT ||
- (reg->regdomainport) ||
- reg->callid_valid != FALSE ||
- reg->ocseq != INITIAL_CSEQ) {
-
- ast_test_status_update(test, "Test 6: change to tls transport and add expiry failed\n");
- res = AST_TEST_FAIL;
- }
- ast_string_field_free_memory(reg);
- ast_free(reg);
-
- /* ---Test reg 7, change transport to tcp, add custom port, and add peer --- */
- if (!(reg = ast_calloc_with_stringfields(1, struct sip_registry, 256))) {
- goto alloc_fail;
- } else if (
- sip_parse_register_line(reg, default_expiry, reg7, 1) ||
- strcmp(reg->callback, "extension") ||
- strcmp(reg->username, "name") ||
- strcmp(reg->regdomain, "namedomain") ||
- strcmp(reg->hostname, "domain") ||
- strcmp(reg->authuser, "authuser") ||
- strcmp(reg->secret, "pass") ||
- strcmp(reg->peername, "peer") ||
- reg->transport != AST_TRANSPORT_TCP ||
- reg->timeout != -1 ||
- reg->expire != -1 ||
- reg->refresh != 111 ||
- reg->expiry != 111 ||
- reg->configured_expiry != 111 ||
- reg->portno != 1234 ||
- (reg->regdomainport) ||
- reg->callid_valid != FALSE ||
- reg->ocseq != INITIAL_CSEQ) {
-
- ast_test_status_update(test, "Test 7, change transport to tcp, add custom port, and add peer failed.\n");
- res = AST_TEST_FAIL;
- }
- ast_string_field_free_memory(reg);
- ast_free(reg);
-
- /* ---Test reg 8, remove transport --- */
- if (!(reg = ast_calloc_with_stringfields(1, struct sip_registry, 256))) {
- goto alloc_fail;
- } else if (
- sip_parse_register_line(reg, default_expiry, reg8, 1) ||
- strcmp(reg->callback, "extension") ||
- strcmp(reg->username, "name") ||
- strcmp(reg->regdomain, "namedomain") ||
- strcmp(reg->hostname, "domain") ||
- strcmp(reg->authuser, "authuser") ||
- strcmp(reg->secret, "pass") ||
- strcmp(reg->peername, "peer") ||
- reg->transport != AST_TRANSPORT_UDP ||
- reg->timeout != -1 ||
- reg->expire != -1 ||
- reg->refresh != 111 ||
- reg->expiry != 111 ||
- reg->configured_expiry != 111 ||
- reg->portno != 1234 ||
- (reg->regdomainport) ||
- reg->callid_valid != FALSE ||
- reg->ocseq != INITIAL_CSEQ) {
-
- ast_test_status_update(test, "Test 8, remove transport failed.\n");
- res = AST_TEST_FAIL;
- }
- ast_string_field_free_memory(reg);
- ast_free(reg);
-
- /* ---Test reg 9, missing domain, expected to fail --- */
- if (!(reg = ast_calloc_with_stringfields(1, struct sip_registry, 256))) {
- goto alloc_fail;
- } else if (!sip_parse_register_line(reg, default_expiry, reg9, 1)) {
- ast_test_status_update(test,
- "Test 9, missing domain, expected to fail but did not.\n");
- res = AST_TEST_FAIL;
- }
- ast_string_field_free_memory(reg);
- ast_free(reg);
-
- /* ---Test reg 10, missing user, expected to fail --- */
- if (!(reg = ast_calloc_with_stringfields(1, struct sip_registry, 256))) {
- goto alloc_fail;
- } else if (!sip_parse_register_line(reg, default_expiry, reg10, 1)) {
- ast_test_status_update(test,
- "Test 10, missing user expected to fail but did not\n");
- res = AST_TEST_FAIL;
- }
- ast_string_field_free_memory(reg);
- ast_free(reg);
-
- /* ---Test reg 11, no registry object, expected to fail--- */
- if (!sip_parse_register_line(NULL, default_expiry, reg1, 1)) {
- ast_test_status_update(test,
- "Test 11, no registry object, expected to fail but did not.\n");
- res = AST_TEST_FAIL;
- }
-
- /* ---Test reg 12, no registry line, expected to fail --- */
- if (!(reg = ast_calloc_with_stringfields(1, struct sip_registry, 256))) {
- goto alloc_fail;
- } else if (!sip_parse_register_line(reg, default_expiry, NULL, 1)) {
-
- ast_test_status_update(test,
- "Test 12, NULL register line expected to fail but did not.\n");
- res = AST_TEST_FAIL;
- }
- ast_string_field_free_memory(reg);
- ast_free(reg);
-
- /* ---Test reg13, add domain port --- */
- if (!(reg = ast_calloc_with_stringfields(1, struct sip_registry, 256))) {
- goto alloc_fail;
- } else if (
- sip_parse_register_line(reg, default_expiry, reg12, 1) ||
- strcmp(reg->callback, "s") ||
- strcmp(reg->username, "name") ||
- strcmp(reg->regdomain, "namedomain") ||
- strcmp(reg->hostname, "domain") ||
- strcmp(reg->authuser, "authuser") ||
- strcmp(reg->secret, "pass") ||
- strcmp(reg->peername, "") ||
- reg->transport != AST_TRANSPORT_UDP ||
- reg->timeout != -1 ||
- reg->expire != -1 ||
- reg->refresh != default_expiry ||
- reg->expiry != default_expiry ||
- reg->configured_expiry != default_expiry ||
- reg->portno != STANDARD_SIP_PORT ||
- reg->regdomainport != 4321 ||
- reg->callid_valid != FALSE ||
- reg->ocseq != INITIAL_CSEQ) {
-
- ast_test_status_update(test, "Test 13, add domain port failed.\n");
- res = AST_TEST_FAIL;
- }
- ast_string_field_free_memory(reg);
- ast_free(reg);
-
- /* ---Test reg14, domain port without secret --- */
- if (!(reg = ast_calloc_with_stringfields(1, struct sip_registry, 256))) {
- goto alloc_fail;
- } else if (
- sip_parse_register_line(reg, default_expiry, reg13, 1) ||
- strcmp(reg->callback, "s") ||
- strcmp(reg->username, "name") ||
- strcmp(reg->regdomain, "namedomain") ||
- strcmp(reg->hostname, "domain") ||
- strcmp(reg->authuser, "") ||
- strcmp(reg->secret, "") ||
- strcmp(reg->peername, "") ||
- reg->transport != AST_TRANSPORT_UDP ||
- reg->timeout != -1 ||
- reg->expire != -1 ||
- reg->refresh != default_expiry ||
- reg->expiry != default_expiry ||
- reg->configured_expiry != default_expiry ||
- reg->portno != STANDARD_SIP_PORT ||
- reg->regdomainport != 4321 ||
- reg->callid_valid != FALSE ||
- reg->ocseq != INITIAL_CSEQ) {
-
- ast_test_status_update(test, "Test 14, domain port without secret failed.\n");
- res = AST_TEST_FAIL;
- }
- ast_string_field_free_memory(reg);
- ast_free(reg);
-
-
- return res;
-
-alloc_fail:
- ast_test_status_update(test, "Out of memory. \n");
- return res;
-}
-#endif
-
-int sip_parse_host(char *line, int lineno, char **hostname, int *portnum, enum ast_transport *transport)
-{
- char *port;
-
- if (ast_strlen_zero(line)) {
- *hostname = NULL;
- return -1;
- }
- if ((*hostname = strstr(line, "://"))) {
- *hostname += 3;
-
- if (!strncasecmp(line, "tcp", 3)) {
- *transport = AST_TRANSPORT_TCP;
- } else if (!strncasecmp(line, "tls", 3)) {
- *transport = AST_TRANSPORT_TLS;
- } else if (!strncasecmp(line, "udp", 3)) {
- *transport = AST_TRANSPORT_UDP;
- } else if (lineno) {
- ast_log(LOG_NOTICE, "'%.3s' is not a valid transport type on line %d of sip.conf. defaulting to udp.\n", line, lineno);
- } else {
- ast_log(LOG_NOTICE, "'%.3s' is not a valid transport type in sip config. defaulting to udp.\n", line);
- }
- } else {
- *hostname = line;
- *transport = AST_TRANSPORT_UDP;
- }
-
- if ((line = strrchr(*hostname, '@')))
- line++;
- else
- line = *hostname;
-
- if (ast_sockaddr_split_hostport(line, hostname, &port, 0) == 0) {
- if (lineno) {
- ast_log(LOG_WARNING, "Cannot parse host '%s' on line %d of sip.conf.\n",
- line, lineno);
- } else {
- ast_log(LOG_WARNING, "Cannot parse host '%s' in sip config.\n", line);
- }
- return -1;
- }
-
- if (port) {
- if (!sscanf(port, "%5d", portnum)) {
- if (lineno) {
- ast_log(LOG_NOTICE, "'%s' is not a valid port number on line %d of sip.conf. using default.\n", port, lineno);
- } else {
- ast_log(LOG_NOTICE, "'%s' is not a valid port number in sip config. using default.\n", port);
- }
- port = NULL;
- }
- }
-
- if (!port) {
- if (*transport & AST_TRANSPORT_TLS) {
- *portnum = STANDARD_TLS_PORT;
- } else {
- *portnum = STANDARD_SIP_PORT;
- }
- }
-
- return 0;
-}
-
-#ifdef TEST_FRAMEWORK
-AST_TEST_DEFINE(sip_parse_host_line_test)
-{
- int res = AST_TEST_PASS;
- char *host;
- int port;
- enum ast_transport transport;
- char host1[] = "www.blah.com";
- char host2[] = "tcp://www.blah.com";
- char host3[] = "tls://10.10.10.10";
- char host4[] = "tls://10.10.10.10:1234";
- char host5[] = "10.10.10.10:1234";
-
- switch (cmd) {
- case TEST_INIT:
- info->name = "sip_parse_host_line_test";
- info->category = "/channels/chan_sip/";
- info->summary = "tests sip.conf host line parsing";
- info->description =
- "Tests parsing of various host line configurations. "
- "Verifies output matches expected behavior.";
- return AST_TEST_NOT_RUN;
- case TEST_EXECUTE:
- break;
- }
-
- /* test 1, simple host */
- sip_parse_host(host1, 1, &host, &port, &transport);
- if (port != STANDARD_SIP_PORT ||
- ast_strlen_zero(host) || strcmp(host, "www.blah.com") ||
- transport != AST_TRANSPORT_UDP) {
- ast_test_status_update(test, "Test 1: simple host failed.\n");
- res = AST_TEST_FAIL;
- }
-
- /* test 2, add tcp transport */
- sip_parse_host(host2, 1, &host, &port, &transport);
- if (port != STANDARD_SIP_PORT ||
- ast_strlen_zero(host) || strcmp(host, "www.blah.com") ||
- transport != AST_TRANSPORT_TCP) {
- ast_test_status_update(test, "Test 2: tcp host failed.\n");
- res = AST_TEST_FAIL;
- }
-
- /* test 3, add tls transport */
- sip_parse_host(host3, 1, &host, &port, &transport);
- if (port != STANDARD_TLS_PORT ||
- ast_strlen_zero(host) || strcmp(host, "10.10.10.10") ||
- transport != AST_TRANSPORT_TLS) {
- ast_test_status_update(test, "Test 3: tls host failed. \n");
- res = AST_TEST_FAIL;
- }
-
- /* test 4, add custom port with tls */
- sip_parse_host(host4, 1, &host, &port, &transport);
- if (port != 1234 || ast_strlen_zero(host) ||
- strcmp(host, "10.10.10.10") ||
- transport != AST_TRANSPORT_TLS) {
- ast_test_status_update(test, "Test 4: tls host with custom port failed.\n");
- res = AST_TEST_FAIL;
- }
-
- /* test 5, simple host with custom port */
- sip_parse_host(host5, 1, &host, &port, &transport);
- if (port != 1234 || ast_strlen_zero(host) ||
- strcmp(host, "10.10.10.10") ||
- transport != AST_TRANSPORT_UDP) {
- ast_test_status_update(test, "Test 5: simple host with custom port failed.\n");
- res = AST_TEST_FAIL;
- }
-
- /* test 6, expected failure with NULL input */
- if (!sip_parse_host(NULL, 1, &host, &port, &transport)) {
- ast_test_status_update(test, "Test 6: expected error on NULL input did not occur.\n");
- res = AST_TEST_FAIL;
- }
-
- return res;
-
-}
-#endif
-
-/*! \brief Parse the comma-separated nat= option values */
-void sip_parse_nat_option(const char *value, struct ast_flags *mask, struct ast_flags *flags)
-{
- char *parse, *this;
-
- if (!(parse = ast_strdupa(value))) {
- return;
- }
-
- /* Since we need to completely override the general settings if we are being called
- * later for a peer, always set the flags for all options on the mask */
- ast_set_flag(&mask[0], SIP_NAT_FORCE_RPORT);
- ast_set_flag(&mask[1], SIP_PAGE2_SYMMETRICRTP);
- ast_set_flag(&mask[2], SIP_PAGE3_NAT_AUTO_RPORT);
- ast_set_flag(&mask[2], SIP_PAGE3_NAT_AUTO_COMEDIA);
-
- while ((this = strsep(&parse, ","))) {
- if (ast_false(this)) {
- ast_clear_flag(&flags[0], SIP_NAT_FORCE_RPORT);
- ast_clear_flag(&flags[1], SIP_PAGE2_SYMMETRICRTP);
- ast_clear_flag(&flags[2], SIP_PAGE3_NAT_AUTO_RPORT);
- ast_clear_flag(&flags[2], SIP_PAGE3_NAT_AUTO_COMEDIA);
- break; /* It doesn't make sense to have no + something else */
- } else if (!strcasecmp(this, "yes")) {
- ast_log(LOG_WARNING, "nat=yes is deprecated, use nat=force_rport,comedia instead\n");
- ast_set_flag(&flags[0], SIP_NAT_FORCE_RPORT);
- ast_set_flag(&flags[1], SIP_PAGE2_SYMMETRICRTP);
- ast_clear_flag(&flags[2], SIP_PAGE3_NAT_AUTO_RPORT);
- ast_clear_flag(&flags[2], SIP_PAGE3_NAT_AUTO_COMEDIA);
- break; /* It doesn't make sense to have yes + something else */
- } else if (!strcasecmp(this, "force_rport") && !ast_test_flag(&flags[2], SIP_PAGE3_NAT_AUTO_RPORT)) {
- ast_set_flag(&flags[0], SIP_NAT_FORCE_RPORT);
- } else if (!strcasecmp(this, "comedia") && !ast_test_flag(&flags[2], SIP_PAGE3_NAT_AUTO_COMEDIA)) {
- ast_set_flag(&flags[1], SIP_PAGE2_SYMMETRICRTP);
- } else if (!strcasecmp(this, "auto_force_rport")) {
- ast_set_flag(&flags[2], SIP_PAGE3_NAT_AUTO_RPORT);
- /* In case someone did something dumb like nat=force_rport,auto_force_rport */
- ast_clear_flag(&flags[0], SIP_NAT_FORCE_RPORT);
- } else if (!strcasecmp(this, "auto_comedia")) {
- ast_set_flag(&flags[2], SIP_PAGE3_NAT_AUTO_COMEDIA);
- /* In case someone did something dumb like nat=comedia,auto_comedia*/
- ast_clear_flag(&flags[1], SIP_PAGE2_SYMMETRICRTP);
- }
- }
-}
-
-#ifdef TEST_FRAMEWORK
-#define TEST_FORCE_RPORT 1 << 0
-#define TEST_COMEDIA 1 << 1
-#define TEST_AUTO_FORCE_RPORT 1 << 2
-#define TEST_AUTO_COMEDIA 1 << 3
-static int match_nat_options(int val, struct ast_flags *flags)
-{
- if ((!ast_test_flag(&flags[0], SIP_NAT_FORCE_RPORT)) != !(val & TEST_FORCE_RPORT)) {
- return 0;
- }
- if (!ast_test_flag(&flags[1], SIP_PAGE2_SYMMETRICRTP) != !(val & TEST_COMEDIA)) {
- return 0;
- }
- if (!ast_test_flag(&flags[2], SIP_PAGE3_NAT_AUTO_RPORT) != !(val & TEST_AUTO_FORCE_RPORT)) {
- return 0;
- }
- if (!ast_test_flag(&flags[2], SIP_PAGE3_NAT_AUTO_COMEDIA) != !(val & TEST_AUTO_COMEDIA)) {
- return 0;
- }
- return 1;
-}
-
-AST_TEST_DEFINE(sip_parse_nat_test)
-{
- int i, res = AST_TEST_PASS;
- struct ast_flags mask[3] = {{0}}, flags[3] = {{0}};
- struct {
- const char *str;
- int i;
- } options[] = {
- { "yes", TEST_FORCE_RPORT | TEST_COMEDIA },
- { "no", 0 },
- { "force_rport", TEST_FORCE_RPORT },
- { "comedia", TEST_COMEDIA },
- { "auto_force_rport", TEST_AUTO_FORCE_RPORT },
- { "auto_comedia", TEST_AUTO_COMEDIA },
- { "force_rport,auto_force_rport", TEST_AUTO_FORCE_RPORT },
- { "auto_force_rport,force_rport", TEST_AUTO_FORCE_RPORT },
- { "comedia,auto_comedia", TEST_AUTO_COMEDIA },
- { "auto_comedia,comedia", TEST_AUTO_COMEDIA },
- { "force_rport,comedia", TEST_FORCE_RPORT | TEST_COMEDIA },
- { "force_rport,auto_comedia", TEST_FORCE_RPORT | TEST_AUTO_COMEDIA },
- { "force_rport,yes,no", TEST_FORCE_RPORT | TEST_COMEDIA },
- { "auto_comedia,no,yes", 0 },
- };
-
- switch (cmd) {
- case TEST_INIT:
- info->name = "sip_parse_nat_test";
- info->category = "/channels/chan_sip/";
- info->summary = "tests sip.conf nat line parsing";
- info->description =
- "Tests parsing of various nat line configurations. "
- "Verifies output matches expected behavior.";
- return AST_TEST_NOT_RUN;
- case TEST_EXECUTE:
- break;
- }
-
- for (i = 0; i < ARRAY_LEN(options); i++) {
- sip_parse_nat_option(options[i].str, mask, flags);
- if (!match_nat_options(options[i].i, flags)) {
- ast_test_status_update(test, "Failed nat=%s\n", options[i].str);
- res = AST_TEST_FAIL;
- }
- memset(flags, 0, sizeof(flags));
- memset(mask, 0, sizeof(mask));
- }
-
- return res;
-}
-#endif
-
-/*! \brief SIP test registration */
-void sip_config_parser_register_tests(void)
-{
- AST_TEST_REGISTER(sip_parse_register_line_test);
- AST_TEST_REGISTER(sip_parse_host_line_test);
- AST_TEST_REGISTER(sip_parse_nat_test);
-}
-
-/*! \brief SIP test registration */
-void sip_config_parser_unregister_tests(void)
-{
- AST_TEST_UNREGISTER(sip_parse_register_line_test);
- AST_TEST_UNREGISTER(sip_parse_host_line_test);
- AST_TEST_UNREGISTER(sip_parse_nat_test);
-}
diff --git a/channels/sip/dialplan_functions.c b/channels/sip/dialplan_functions.c
deleted file mode 100644
index 74106d1be7..0000000000
--- a/channels/sip/dialplan_functions.c
+++ /dev/null
@@ -1,515 +0,0 @@
-/*
- * Asterisk -- An open source telephony toolkit.
- *
- * Copyright (C) 2010, Digium, Inc.
- *
- * See http://www.asterisk.org for more information about
- * the Asterisk project. Please do not directly contact
- * any of the maintainers of this project for assistance;
- * the project provides a web site, mailing lists and IRC
- * channels for your use.
- *
- * This program is free software, distributed under the terms of
- * the GNU General Public License Version 2. See the LICENSE file
- * at the top of the source tree.
- */
-
-/*!
- * \file
- * \brief sip channel dialplan functions and unit tests
- */
-
-/*** MODULEINFO
- deprecated
- ***/
-
-/*** DOCUMENTATION
-
-
-
- R/O Get the IP address of the peer.
-
-
- R/O Get the source IP address of the peer.
-
-
- R/O Get the source port of the peer.
-
-
- R/O Get the URI from the From: header.
-
-
- R/O Get the URI from the Contact: header.
-
-
- R/O Get the Request-URI from the INVITE header.
-
-
- R/O Get the useragent.
-
-
- R/O Get the name of the peer.
-
-
- R/O 1 if T38 is offered or enabled in this channel,
- otherwise 0
-
-
- R/O Get QOS information about the RTP stream
- This option takes two additional arguments:
- Argument 1:
- audio Get data about the audio stream
- video Get data about the video stream
- text Get data about the text stream
- Argument 2:
- local_ssrc Local SSRC (stream ID)
- local_lostpackets Local lost packets
- local_jitter Local calculated jitter
- local_maxjitter Local calculated jitter (maximum)
- local_minjitter Local calculated jitter (minimum)
- local_normdevjitter Local calculated jitter (normal deviation)
- local_stdevjitter Local calculated jitter (standard deviation)
- local_count Number of received packets
- remote_ssrc Remote SSRC (stream ID)
- remote_lostpackets Remote lost packets
- remote_jitter Remote reported jitter
- remote_maxjitter Remote calculated jitter (maximum)
- remote_minjitter Remote calculated jitter (minimum)
- remote_normdevjitter Remote calculated jitter (normal deviation)
- remote_stdevjitter Remote calculated jitter (standard deviation)
- remote_count Number of transmitted packets
- rtt Round trip time
- maxrtt Round trip time (maximum)
- minrtt Round trip time (minimum)
- normdevrtt Round trip time (normal deviation)
- stdevrtt Round trip time (standard deviation)
- all All statistics (in a form suited to logging,
- but not for parsing)
-
-
- R/O Get remote RTP destination information.
- This option takes one additional argument:
- Argument 1:
- audio Get audio destination
- video Get video destination
- text Get text destination
- Defaults to audio if unspecified.
-
-
- R/O Get source RTP destination information.
- This option takes one additional argument:
- Argument 1:
- audio Get audio destination
- video Get video destination
- text Get text destination
- Defaults to audio if unspecified.
-
-
-
- ***/
-
-#include "asterisk.h"
-
-#include
-
-#include "asterisk/channel.h"
-#include "asterisk/rtp_engine.h"
-#include "asterisk/pbx.h"
-#include "asterisk/acl.h"
-
-#include "include/sip.h"
-#include "include/globals.h"
-#include "include/dialog.h"
-#include "include/dialplan_functions.h"
-#include "include/sip_utils.h"
-
-
-int sip_acf_channel_read(struct ast_channel *chan, const char *funcname, char *preparse, char *buf, size_t buflen)
-{
- struct sip_pvt *p = ast_channel_tech_pvt(chan);
- char *parse = ast_strdupa(preparse);
- int res = 0;
- AST_DECLARE_APP_ARGS(args,
- AST_APP_ARG(param);
- AST_APP_ARG(type);
- AST_APP_ARG(field);
- );
-
- /* Check for zero arguments */
- if (ast_strlen_zero(parse)) {
- ast_log(LOG_ERROR, "Cannot call %s without arguments\n", funcname);
- return -1;
- }
-
- AST_STANDARD_APP_ARGS(args, parse);
-
- /* Sanity check */
- if (!IS_SIP_TECH(ast_channel_tech(chan))) {
- ast_log(LOG_ERROR, "Cannot call %s on a non-SIP channel\n", funcname);
- return 0;
- }
-
- memset(buf, 0, buflen);
-
- if (p == NULL) {
- return -1;
- }
-
- if (!strcasecmp(args.param, "peerip")) {
- ast_copy_string(buf, ast_sockaddr_isnull(&p->sa) ? "" : ast_sockaddr_stringify_addr(&p->sa), buflen);
- } else if (!strcasecmp(args.param, "recvip")) {
- ast_copy_string(buf, ast_sockaddr_isnull(&p->recv) ? "" : ast_sockaddr_stringify_addr(&p->recv), buflen);
- } else if (!strcasecmp(args.param, "recvport")) {
- ast_copy_string(buf, ast_sockaddr_isnull(&p->recv) ? "" : ast_sockaddr_stringify_port(&p->recv), buflen);
- } else if (!strcasecmp(args.param, "from")) {
- ast_copy_string(buf, p->from, buflen);
- } else if (!strcasecmp(args.param, "uri")) {
- ast_copy_string(buf, p->uri, buflen);
- } else if (!strcasecmp(args.param, "ruri")) {
- if (p->initreq.data) {
- char *tmpruri = REQ_OFFSET_TO_STR(&p->initreq, rlpart2);
- ast_copy_string(buf, tmpruri, buflen);
- } else {
- return -1;
- }
- } else if (!strcasecmp(args.param, "useragent")) {
- ast_copy_string(buf, p->useragent, buflen);
- } else if (!strcasecmp(args.param, "peername")) {
- ast_copy_string(buf, p->peername, buflen);
- } else if (!strcasecmp(args.param, "t38passthrough")) {
- ast_copy_string(buf, (p->t38.state == T38_DISABLED) ? "0" : "1", buflen);
- } else if (!strcasecmp(args.param, "rtpdest")) {
- struct ast_sockaddr addr;
- struct ast_rtp_instance *stream;
-
- if (ast_strlen_zero(args.type))
- args.type = "audio";
-
- if (!strcasecmp(args.type, "audio"))
- stream = p->rtp;
- else if (!strcasecmp(args.type, "video"))
- stream = p->vrtp;
- else if (!strcasecmp(args.type, "text"))
- stream = p->trtp;
- else
- return -1;
-
- /* Return 0 to suppress a console warning message */
- if (!stream) {
- return 0;
- }
-
- ast_rtp_instance_get_remote_address(stream, &addr);
- snprintf(buf, buflen, "%s", ast_sockaddr_stringify(&addr));
- } else if (!strcasecmp(args.param, "rtpsource")) {
- struct ast_sockaddr sa;
- struct ast_rtp_instance *stream;
-
- if (ast_strlen_zero(args.type))
- args.type = "audio";
-
- if (!strcasecmp(args.type, "audio"))
- stream = p->rtp;
- else if (!strcasecmp(args.type, "video"))
- stream = p->vrtp;
- else if (!strcasecmp(args.type, "text"))
- stream = p->trtp;
- else
- return -1;
-
- /* Return 0 to suppress a console warning message */
- if (!stream) {
- return 0;
- }
-
- ast_rtp_instance_get_local_address(stream, &sa);
-
- if (ast_sockaddr_isnull(&sa)) {
- struct ast_sockaddr dest_sa;
- ast_rtp_instance_get_remote_address(stream, &dest_sa);
- ast_ouraddrfor(&dest_sa, &sa);
- }
-
- snprintf(buf, buflen, "%s", ast_sockaddr_stringify(&sa));
- } else if (!strcasecmp(args.param, "rtpqos")) {
- struct ast_rtp_instance *rtp = NULL;
-
- if (ast_strlen_zero(args.type)) {
- args.type = "audio";
- }
-
- if (!strcasecmp(args.type, "audio")) {
- rtp = p->rtp;
- } else if (!strcasecmp(args.type, "video")) {
- rtp = p->vrtp;
- } else if (!strcasecmp(args.type, "text")) {
- rtp = p->trtp;
- } else {
- return -1;
- }
-
- if (ast_strlen_zero(args.field) || !strcasecmp(args.field, "all")) {
- char quality_buf[AST_MAX_USER_FIELD];
-
- if (!ast_rtp_instance_get_quality(rtp, AST_RTP_INSTANCE_STAT_FIELD_QUALITY, quality_buf, sizeof(quality_buf))) {
- return -1;
- }
-
- ast_copy_string(buf, quality_buf, buflen);
- return res;
- } else {
- struct ast_rtp_instance_stats stats;
- int i;
- struct {
- const char *name;
- enum { INT, DBL } type;
- union {
- unsigned int *i4;
- double *d8;
- };
- } lookup[] = {
- { "txcount", INT, { .i4 = &stats.txcount, }, },
- { "rxcount", INT, { .i4 = &stats.rxcount, }, },
- { "txjitter", DBL, { .d8 = &stats.txjitter, }, },
- { "rxjitter", DBL, { .d8 = &stats.rxjitter, }, },
- { "remote_maxjitter", DBL, { .d8 = &stats.remote_maxjitter, }, },
- { "remote_minjitter", DBL, { .d8 = &stats.remote_minjitter, }, },
- { "remote_normdevjitter", DBL, { .d8 = &stats.remote_normdevjitter, }, },
- { "remote_stdevjitter", DBL, { .d8 = &stats.remote_stdevjitter, }, },
- { "local_maxjitter", DBL, { .d8 = &stats.local_maxjitter, }, },
- { "local_minjitter", DBL, { .d8 = &stats.local_minjitter, }, },
- { "local_normdevjitter", DBL, { .d8 = &stats.local_normdevjitter, }, },
- { "local_stdevjitter", DBL, { .d8 = &stats.local_stdevjitter, }, },
- { "txploss", INT, { .i4 = &stats.txploss, }, },
- { "rxploss", INT, { .i4 = &stats.rxploss, }, },
- { "remote_maxrxploss", DBL, { .d8 = &stats.remote_maxrxploss, }, },
- { "remote_minrxploss", DBL, { .d8 = &stats.remote_minrxploss, }, },
- { "remote_normdevrxploss", DBL, { .d8 = &stats.remote_normdevrxploss, }, },
- { "remote_stdevrxploss", DBL, { .d8 = &stats.remote_stdevrxploss, }, },
- { "local_maxrxploss", DBL, { .d8 = &stats.local_maxrxploss, }, },
- { "local_minrxploss", DBL, { .d8 = &stats.local_minrxploss, }, },
- { "local_normdevrxploss", DBL, { .d8 = &stats.local_normdevrxploss, }, },
- { "local_stdevrxploss", DBL, { .d8 = &stats.local_stdevrxploss, }, },
- { "rtt", DBL, { .d8 = &stats.rtt, }, },
- { "maxrtt", DBL, { .d8 = &stats.maxrtt, }, },
- { "minrtt", DBL, { .d8 = &stats.minrtt, }, },
- { "normdevrtt", DBL, { .d8 = &stats.normdevrtt, }, },
- { "stdevrtt", DBL, { .d8 = &stats.stdevrtt, }, },
- { "local_ssrc", INT, { .i4 = &stats.local_ssrc, }, },
- { "remote_ssrc", INT, { .i4 = &stats.remote_ssrc, }, },
- { NULL, },
- };
-
- if (ast_rtp_instance_get_stats(rtp, &stats, AST_RTP_INSTANCE_STAT_ALL)) {
- return -1;
- }
-
- for (i = 0; !ast_strlen_zero(lookup[i].name); i++) {
- if (!strcasecmp(args.field, lookup[i].name)) {
- if (lookup[i].type == INT) {
- snprintf(buf, buflen, "%u", *lookup[i].i4);
- } else {
- snprintf(buf, buflen, "%f", *lookup[i].d8);
- }
- return 0;
- }
- }
- ast_log(LOG_WARNING, "Unrecognized argument '%s' to %s\n", preparse, funcname);
- return -1;
- }
- } else if (!strcasecmp(args.param, "secure_signaling")) {
- snprintf(buf, buflen, "%s", p->socket.type == AST_TRANSPORT_TLS ? "1" : "");
- } else if (!strcasecmp(args.param, "secure_media")) {
- snprintf(buf, buflen, "%s", p->srtp ? "1" : "");
- } else {
- res = -1;
- }
- return res;
-}
-
-#ifdef TEST_FRAMEWORK
-static int test_sip_rtpqos_1_new(struct ast_rtp_instance *instance, struct ast_sched_context *sched, struct ast_sockaddr *addr, void *data)
-{
- /* Needed to pass sanity checks */
- ast_rtp_instance_set_data(instance, data);
- return 0;
-}
-
-static int test_sip_rtpqos_1_destroy(struct ast_rtp_instance *instance)
-{
- /* Needed to pass sanity checks */
- return 0;
-}
-
-static struct ast_frame *test_sip_rtpqos_1_read(struct ast_rtp_instance *instance, int rtcp)
-{
- /* Needed to pass sanity checks */
- return &ast_null_frame;
-}
-
-static int test_sip_rtpqos_1_write(struct ast_rtp_instance *instance, struct ast_frame *frame)
-{
- /* Needed to pass sanity checks */
- return 0;
-}
-
-static int test_sip_rtpqos_1_get_stat(struct ast_rtp_instance *instance, struct ast_rtp_instance_stats *stats, enum ast_rtp_instance_stat stat)
-{
- struct ast_rtp_instance_stats *s = ast_rtp_instance_get_data(instance);
- memcpy(stats, s, sizeof(*stats));
- return 0;
-}
-
-AST_TEST_DEFINE(test_sip_rtpqos_1)
-{
- int i, res = AST_TEST_PASS;
- static struct ast_rtp_engine test_engine = {
- .name = "test",
- .new = test_sip_rtpqos_1_new,
- .destroy = test_sip_rtpqos_1_destroy,
- .read = test_sip_rtpqos_1_read,
- .write = test_sip_rtpqos_1_write,
- .get_stat = test_sip_rtpqos_1_get_stat,
- };
- struct ast_sockaddr sa = { {0, } };
- struct ast_rtp_instance_stats mine = { 0, };
- struct sip_pvt *p = NULL;
- struct ast_channel *chan = NULL;
- struct ast_str *varstr = NULL, *buffer = NULL;
- struct {
- const char *name;
- enum { INT, DBL } type;
- union {
- unsigned int *i4;
- double *d8;
- };
- } lookup[] = {
- { "txcount", INT, { .i4 = &mine.txcount, }, },
- { "rxcount", INT, { .i4 = &mine.rxcount, }, },
- { "txjitter", DBL, { .d8 = &mine.txjitter, }, },
- { "rxjitter", DBL, { .d8 = &mine.rxjitter, }, },
- { "remote_maxjitter", DBL, { .d8 = &mine.remote_maxjitter, }, },
- { "remote_minjitter", DBL, { .d8 = &mine.remote_minjitter, }, },
- { "remote_normdevjitter", DBL, { .d8 = &mine.remote_normdevjitter, }, },
- { "remote_stdevjitter", DBL, { .d8 = &mine.remote_stdevjitter, }, },
- { "local_maxjitter", DBL, { .d8 = &mine.local_maxjitter, }, },
- { "local_minjitter", DBL, { .d8 = &mine.local_minjitter, }, },
- { "local_normdevjitter", DBL, { .d8 = &mine.local_normdevjitter, }, },
- { "local_stdevjitter", DBL, { .d8 = &mine.local_stdevjitter, }, },
- { "txploss", INT, { .i4 = &mine.txploss, }, },
- { "rxploss", INT, { .i4 = &mine.rxploss, }, },
- { "remote_maxrxploss", DBL, { .d8 = &mine.remote_maxrxploss, }, },
- { "remote_minrxploss", DBL, { .d8 = &mine.remote_minrxploss, }, },
- { "remote_normdevrxploss", DBL, { .d8 = &mine.remote_normdevrxploss, }, },
- { "remote_stdevrxploss", DBL, { .d8 = &mine.remote_stdevrxploss, }, },
- { "local_maxrxploss", DBL, { .d8 = &mine.local_maxrxploss, }, },
- { "local_minrxploss", DBL, { .d8 = &mine.local_minrxploss, }, },
- { "local_normdevrxploss", DBL, { .d8 = &mine.local_normdevrxploss, }, },
- { "local_stdevrxploss", DBL, { .d8 = &mine.local_stdevrxploss, }, },
- { "rtt", DBL, { .d8 = &mine.rtt, }, },
- { "maxrtt", DBL, { .d8 = &mine.maxrtt, }, },
- { "minrtt", DBL, { .d8 = &mine.minrtt, }, },
- { "normdevrtt", DBL, { .d8 = &mine.normdevrtt, }, },
- { "stdevrtt", DBL, { .d8 = &mine.stdevrtt, }, },
- { "local_ssrc", INT, { .i4 = &mine.local_ssrc, }, },
- { "remote_ssrc", INT, { .i4 = &mine.remote_ssrc, }, },
- { NULL, },
- };
-
- switch (cmd) {
- case TEST_INIT:
- info->name = "test_sip_rtpqos";
- info->category = "/channels/chan_sip/";
- info->summary = "Test retrieval of SIP RTP QOS stats";
- info->description =
- "Verify values in the RTP instance structure can be accessed through the dialplan.";
- return AST_TEST_NOT_RUN;
- case TEST_EXECUTE:
- break;
- }
-
- ast_rtp_engine_register(&test_engine);
- /* Have to associate this with a SIP pvt and an ast_channel */
- if (!(p = sip_alloc(0, NULL, 0, SIP_NOTIFY, NULL, 0))) {
- res = AST_TEST_NOT_RUN;
- goto done;
- }
-
- if (!(p->rtp = ast_rtp_instance_new("test", sched, &bindaddr, &mine))) {
- res = AST_TEST_NOT_RUN;
- goto done;
- }
- ast_rtp_instance_set_remote_address(p->rtp, &sa);
- if (!(chan = ast_dummy_channel_alloc())) {
- res = AST_TEST_NOT_RUN;
- goto done;
- }
- ast_channel_tech_set(chan, &sip_tech);
- ast_channel_tech_pvt_set(chan, dialog_ref(p, "Give the owner channel a reference to the dialog"));
- p->owner = chan;
-
- varstr = ast_str_create(16);
- buffer = ast_str_create(16);
- if (!varstr || !buffer) {
- res = AST_TEST_NOT_RUN;
- goto done;
- }
-
- /* Populate "mine" with values, then retrieve them with the CHANNEL dialplan function */
- for (i = 0; !ast_strlen_zero(lookup[i].name); i++) {
- ast_str_set(&varstr, 0, "${CHANNEL(rtpqos,audio,%s)}", lookup[i].name);
- if (lookup[i].type == INT) {
- int j;
- char cmpstr[256];
- for (j = 1; j < 25; j++) {
- *lookup[i].i4 = j;
- ast_str_substitute_variables(&buffer, 0, chan, ast_str_buffer(varstr));
- snprintf(cmpstr, sizeof(cmpstr), "%d", j);
- if (strcmp(cmpstr, ast_str_buffer(buffer))) {
- res = AST_TEST_FAIL;
- ast_test_status_update(test, "%s != %s != %s\n", ast_str_buffer(varstr), cmpstr, ast_str_buffer(buffer));
- break;
- }
- }
- } else {
- double j, cmpdbl = 0.0;
- for (j = 1.0; j < 10.0; j += 0.3) {
- *lookup[i].d8 = j;
- ast_str_substitute_variables(&buffer, 0, chan, ast_str_buffer(varstr));
- if (sscanf(ast_str_buffer(buffer), "%lf", &cmpdbl) != 1 || fabs(j - cmpdbl) > .05) {
- res = AST_TEST_FAIL;
- ast_test_status_update(test, "%s != %f != %s\n", ast_str_buffer(varstr), j, ast_str_buffer(buffer));
- break;
- }
- }
- }
- }
-
-done:
- ast_free(varstr);
- ast_free(buffer);
-
- /* This unlink and unref will take care of destroying the channel, RTP instance, and SIP pvt */
- if (p) {
- dialog_unlink_all(p);
- dialog_unref(p, "Destroy test object");
- }
- if (chan) {
- ast_channel_unref(chan);
- }
- ast_rtp_engine_unregister(&test_engine);
- return res;
-}
-#endif
-
-/*! \brief SIP test registration */
-void sip_dialplan_function_register_tests(void)
-{
- AST_TEST_REGISTER(test_sip_rtpqos_1);
-}
-
-/*! \brief SIP test registration */
-void sip_dialplan_function_unregister_tests(void)
-{
- AST_TEST_UNREGISTER(test_sip_rtpqos_1);
-}
diff --git a/channels/sip/include/config_parser.h b/channels/sip/include/config_parser.h
deleted file mode 100644
index 41d1cc6fba..0000000000
--- a/channels/sip/include/config_parser.h
+++ /dev/null
@@ -1,68 +0,0 @@
-/*
- * Asterisk -- An open source telephony toolkit.
- *
- * Copyright (C) 2010, Digium, Inc.
- *
- * See http://www.asterisk.org for more information about
- * the Asterisk project. Please do not directly contact
- * any of the maintainers of this project for assistance;
- * the project provides a web site, mailing lists and IRC
- * channels for your use.
- *
- * This program is free software, distributed under the terms of
- * the GNU General Public License Version 2. See the LICENSE file
- * at the top of the source tree.
- */
-
-/*!
- * \file
- * \brief sip.conf parser header file
- */
-
-#include "sip.h"
-
-#ifndef _SIP_CONF_PARSE_H
-#define _SIP_CONF_PARSE_H
-
-/*!
- * \brief Parse register=> line in sip.conf
- *
- * \retval 0 on success
- * \retval -1 on failure
- */
-int sip_parse_register_line(struct sip_registry *reg, int default_expiry, const char *value, int lineno);
-
-/*!
- * \brief parses a config line for a host with a transport
- *
- * An example input would be:
- * tls://www.google.com:8056
- *
- * \retval 0 on success
- * \retval -1 on failure
- */
-int sip_parse_host(char *line, int lineno, char **hostname, int *portnum, enum ast_transport *transport);
-
-/*! \brief Parse the comma-separated nat= option values
- * \param value The comma-separated value
- * \param mask An array of ast_flags that will be set by this function
- * and used as a mask for copying the flags later
- * \param flags An array of ast_flags that will be set by this function
- *
- * \note The nat-related values in both mask and flags are assumed to empty. This function
- * will treat the first "yes" or "no" value in a list of values as overriding all other values
- * and will stop parsing. Auto values will override their non-auto counterparts.
- */
-void sip_parse_nat_option(const char *value, struct ast_flags *mask, struct ast_flags *flags);
-
-/*!
- * \brief register config parsing tests
- */
-void sip_config_parser_register_tests(void);
-
-/*!
- * \brief unregister config parsing tests
- */
-void sip_config_parser_unregister_tests(void);
-
-#endif
diff --git a/channels/sip/include/dialog.h b/channels/sip/include/dialog.h
deleted file mode 100644
index 0694170978..0000000000
--- a/channels/sip/include/dialog.h
+++ /dev/null
@@ -1,82 +0,0 @@
-/*
- * Asterisk -- An open source telephony toolkit.
- *
- * Copyright (C) 2010, Digium, Inc.
- *
- * See http://www.asterisk.org for more information about
- * the Asterisk project. Please do not directly contact
- * any of the maintainers of this project for assistance;
- * the project provides a web site, mailing lists and IRC
- * channels for your use.
- *
- * This program is free software, distributed under the terms of
- * the GNU General Public License Version 2. See the LICENSE file
- * at the top of the source tree.
- */
-
-/*!
- * \file
- * \brief sip dialog management header file
- */
-
-#include "sip.h"
-
-#ifndef _SIP_DIALOG_H
-#define _SIP_DIALOG_H
-
-/*! \brief
- * when we create or delete references, make sure to use these
- * functions so we keep track of the refcounts.
- * To simplify the code, we allow a NULL to be passed to dialog_unref().
- */
-#define dialog_ref(dialog, tag) ao2_t_bump(dialog, tag)
-#define dialog_unref(dialog, tag) ({ ao2_t_cleanup(dialog, tag); (NULL); })
-
-struct sip_pvt *__sip_alloc(ast_string_field callid, struct ast_sockaddr *sin,
- int useglobal_nat, const int intended_method, struct sip_request *req, ast_callid logger_callid,
- const char *file, int line, const char *func);
-
-#define sip_alloc(callid, addr, useglobal_nat, intended_method, req, logger_callid) \
- __sip_alloc(callid, addr, useglobal_nat, intended_method, req, logger_callid, __FILE__, __LINE__, __PRETTY_FUNCTION__)
-
-/*!
- * \brief Schedule final destruction of SIP dialog.
- *
- * \note This cannot be canceled.
- *
- * \details
- * This function is used to keep a dialog around for a period of time in order
- * to properly respond to any retransmits.
- */
-void sip_scheddestroy_final(struct sip_pvt *p, int ms);
-
-/*! \brief Schedule destruction of SIP dialog */
-void sip_scheddestroy(struct sip_pvt *p, int ms);
-
-/*! \brief Cancel destruction of SIP dialog. */
-void sip_cancel_destroy(struct sip_pvt *pvt);
-
-/*!
- * \brief Unlink a dialog from the dialogs container, as well as any other places
- * that it may be currently stored.
- *
- * \note A reference to the dialog must be held before calling
- * this function, and this function does not release that
- * reference.
- *
- * \note The dialog must not be locked when called.
- */
-void dialog_unlink_all(struct sip_pvt *dialog);
-
-/*! \brief Acknowledges receipt of a packet and stops retransmission
- * called with p locked*/
-int __sip_ack(struct sip_pvt *p, uint32_t seqno, int resp, int sipmethod);
-
-/*! \brief Pretend to ack all packets
- * called with p locked */
-void __sip_pretend_ack(struct sip_pvt *p);
-
-/*! \brief Acks receipt of packet, keep it around (used for provisional responses) */
-int __sip_semi_ack(struct sip_pvt *p, uint32_t seqno, int resp, int sipmethod);
-
-#endif /* defined(_SIP_DIALOG_H) */
diff --git a/channels/sip/include/dialplan_functions.h b/channels/sip/include/dialplan_functions.h
deleted file mode 100644
index 1600d43142..0000000000
--- a/channels/sip/include/dialplan_functions.h
+++ /dev/null
@@ -1,41 +0,0 @@
-/*
- * Asterisk -- An open source telephony toolkit.
- *
- * Copyright (C) 2010, Digium, Inc.
- *
- * See http://www.asterisk.org for more information about
- * the Asterisk project. Please do not directly contact
- * any of the maintainers of this project for assistance;
- * the project provides a web site, mailing lists and IRC
- * channels for your use.
- *
- * This program is free software, distributed under the terms of
- * the GNU General Public License Version 2. See the LICENSE file
- * at the top of the source tree.
- */
-
-/*!
- * \file
- * \brief SIP dialplan functions header file
- */
-
-#include "sip.h"
-
-#ifndef _SIP_DIALPLAN_FUNCTIONS_H
-#define _SIP_DIALPLAN_FUNCTIONS_H
-
-/*!
- * \brief Channel read dialplan function for SIP
- */
-int sip_acf_channel_read(struct ast_channel *chan, const char *funcname, char *preparse, char *buf, size_t buflen);
-
-/*!
- * \brief register dialplan function tests
- */
-void sip_dialplan_function_register_tests(void);
-/*!
- * \brief unregister dialplan function tests
- */
-void sip_dialplan_function_unregister_tests(void);
-
-#endif /* !defined(_SIP_DIALPLAN_FUNCTIONS_H) */
diff --git a/channels/sip/include/globals.h b/channels/sip/include/globals.h
deleted file mode 100644
index 3c3ba47bd3..0000000000
--- a/channels/sip/include/globals.h
+++ /dev/null
@@ -1,41 +0,0 @@
-/*
- * Asterisk -- An open source telephony toolkit.
- *
- * Copyright (C) 2010, Digium, Inc.
- *
- * See http://www.asterisk.org for more information about
- * the Asterisk project. Please do not directly contact
- * any of the maintainers of this project for assistance;
- * the project provides a web site, mailing lists and IRC
- * channels for your use.
- *
- * This program is free software, distributed under the terms of
- * the GNU General Public License Version 2. See the LICENSE file
- * at the top of the source tree.
- */
-
-/*!
- * \file
- * \brief sip global declaration header file
- */
-
-#include "sip.h"
-
-#ifndef _SIP_GLOBALS_H
-#define _SIP_GLOBALS_H
-
-extern struct ast_sockaddr bindaddr; /*!< UDP: The address we bind to */
-extern struct ast_sched_context *sched; /*!< The scheduling context */
-
-/*! \brief Definition of this channel for PBX channel registration */
-extern struct ast_channel_tech sip_tech;
-
-/*! \brief This version of the sip channel tech has no send_digit_begin
- * callback so that the core knows that the channel does not want
- * DTMF BEGIN frames.
- * The struct is initialized just before registering the channel driver,
- * and is for use with channels using SIP INFO DTMF.
- */
-extern struct ast_channel_tech sip_tech_info;
-
-#endif /* !defined(SIP_GLOBALS_H) */
diff --git a/channels/sip/include/reqresp_parser.h b/channels/sip/include/reqresp_parser.h
deleted file mode 100644
index 67d9da43e7..0000000000
--- a/channels/sip/include/reqresp_parser.h
+++ /dev/null
@@ -1,250 +0,0 @@
-/*
- * Asterisk -- An open source telephony toolkit.
- *
- * Copyright (C) 2010, Digium, Inc.
- *
- * See http://www.asterisk.org for more information about
- * the Asterisk project. Please do not directly contact
- * any of the maintainers of this project for assistance;
- * the project provides a web site, mailing lists and IRC
- * channels for your use.
- *
- * This program is free software, distributed under the terms of
- * the GNU General Public License Version 2. See the LICENSE file
- * at the top of the source tree.
- */
-
-/*!
- * \file
- * \brief sip request response parser header file
- */
-
-#ifndef _SIP_REQRESP_H
-#define _SIP_REQRESP_H
-
-/*! \brief uri parameters */
-struct uriparams {
- char *transport;
- char *user;
- char *method;
- char *ttl;
- char *maddr;
- int lr;
-};
-
-struct contact {
- AST_LIST_ENTRY(contact) list;
- char *name;
- char *user;
- char *pass;
- char *hostport;
- struct uriparams params;
- char *headers;
- char *expires;
- char *q;
-};
-
-AST_LIST_HEAD_NOLOCK(contactliststruct, contact);
-
-/*!
- * \brief parses a URI in its components.
- *
- * \note
- * - Multiple scheme's can be specified ',' delimited. ex: "sip:,sips:"
- * - If a component is not requested, do not split around it. This means
- * that if we don't have domain, we cannot split name:pass.
- * - It is safe to call with ret_name, pass, hostport pointing all to
- * the same place.
- * - If no secret parameter is provided, ret_name will return with both
- * parts, user:secret.
- * - If the URI contains a port number, hostport will return with both
- * parts, host:port.
- * - This function overwrites the URI string.
- *
- * \retval 0 on success
- * \retval -1 on error.
- *
- * \verbatim
- * general form we are expecting is sip:user:password;user-parameters@host:port;uri-parameters?headers
- * \endverbatim
- */
-int parse_uri(char *uri, const char *scheme, char **ret_name, char **pass,
- char **hostport, char **transport);
-
-/*!
- * \brief parses a URI in to all of its components and any trailing residue
- *
- * \retval 0 on success
- * \retval -1 on error.
- *
- */
-int parse_uri_full(char *uri, const char *scheme, char **user, char **pass,
- char **hostport, struct uriparams *params, char **headers,
- char **residue);
-
-/*!
- * \brief Get caller id name from SIP headers, copy into output buffer
- *
- * \return input string pointer placed after display-name field if possible
- */
-const char *get_calleridname(const char *input, char *output, size_t outputsize);
-
-/*!
- * \brief Get name and number from sip header
- *
- * \note name and number point to malloced memory on return and must be
- * freed. If name or number is not found, they will be returned as NULL.
- *
- * \retval 0 success
- * \retval -1 failure
- */
-int get_name_and_number(const char *hdr, char **name, char **number);
-
-/*! \brief Pick out text in brackets from character string
- * \return pointer to terminated stripped string
- * \param tmp input string that will be modified
- *
- * Examples:
- * \verbatim
- * "foo" valid input, returns bar
- * foo returns the whole string
- * < "foo ... > returns the string between brackets
- * < "foo... bogus (missing closing bracket), returns the whole string
- * \endverbatim
- */
-char *get_in_brackets(char *tmp);
-
-/*! \brief Get text in brackets on a const without copy
- *
- * \param src String to search
- * \param[out] start Set to first character inside left bracket.
- * \param[out] length Set to lenght of string inside brackets
- * \retval 0 success
- * \retval -1 failure
- * \retval 1 no brackets so got all
- */
-int get_in_brackets_const(const char *src,const char **start,int *length);
-
-/*! \brief Get text in brackets and any trailing residue
- *
- * \retval 0 success
- * \retval -1 failure
- * \retval 1 no brackets so got all
- */
-int get_in_brackets_full(char *tmp, char **out, char **residue);
-
-/*! \brief Parse the ABNF structure
- * name-andor-addr = name-addr / addr-spec
- * into its components and return any trailing message-header parameters
- *
- * \retval 0 success
- * \retval -1 failure
- */
-int parse_name_andor_addr(char *uri, const char *scheme, char **name,
- char **user, char **pass, char **domain,
- struct uriparams *params, char **headers,
- char **residue);
-
-/*! \brief Parse all contact header contacts
- * \retval 0 success
- * \retval -1 failure
- * \retval 1 all contacts (star)
- */
-
-int get_comma(char *parse, char **out);
-
-int parse_contact_header(char *contactheader, struct contactliststruct *contactlist);
-
-/*!
- * \brief register request parsing tests
- */
-void sip_request_parser_register_tests(void);
-
-/*!
- * \brief unregister request parsing tests
- */
-void sip_request_parser_unregister_tests(void);
-
-/*!
- * \brief Parse supported header in incoming packet
- *
- * \details This function parses through the options parameters and
- * builds a bit field representing all the SIP options in that field. When an
- * item is found that is not supported, it is copied to the unsupported
- * out buffer.
- *
- * \param options list
- * \param[in,out] unsupported buffer (optional)
- * \param[in,out] unsupported_len buffer length
- *
- * \note Because this function can be called multiple times, it will append
- * whatever options are specified in \c options to \c unsupported. Callers
- * of this function should make sure the unsupported buffer is clear before
- * calling this function.
- */
-unsigned int parse_sip_options(const char *options, char *unsupported, size_t unsupported_len);
-
-/*!
- * \brief Compare two URIs as described in RFC 3261 Section 19.1.4
- *
- * \param input1 First URI
- * \param input2 Second URI
- * \retval 0 URIs match
- * \retval nonzero URIs do not match or one or both is malformed
- */
-int sip_uri_cmp(const char *input1, const char *input2);
-
-/*!
- * \brief initialize request and response parser data
- *
- * \retval 0 Success
- * \retval -1 Failure
- */
-int sip_reqresp_parser_init(void);
-
-/*!
- * \brief Free resources used by request and response parser
- */
-void sip_reqresp_parser_exit(void);
-
-/*!
- * \brief Parse a Via header
- *
- * This function parses the Via header and processes it according to section
- * 18.2 of RFC 3261 and RFC 3581. Since we don't have a transport layer, we
- * only care about the maddr and ttl parms. The received and rport params are
- * not parsed.
- *
- * \note This function fails to parse some odd combinations of SWS in parameter
- * lists.
- *
- * \code
- * VIA syntax. RFC 3261 section 25.1
- * Via = ( "Via" / "v" ) HCOLON via-parm *(COMMA via-parm)
- * via-parm = sent-protocol LWS sent-by *( SEMI via-params )
- * via-params = via-ttl / via-maddr
- * / via-received / via-branch
- * / via-extension
- * via-ttl = "ttl" EQUAL ttl
- * via-maddr = "maddr" EQUAL host
- * via-received = "received" EQUAL (IPv4address / IPv6address)
- * via-branch = "branch" EQUAL token
- * via-extension = generic-param
- * sent-protocol = protocol-name SLASH protocol-version
- * SLASH transport
- * protocol-name = "SIP" / token
- * protocol-version = token
- * transport = "UDP" / "TCP" / "TLS" / "SCTP"
- * / other-transport
- * sent-by = host [ COLON port ]
- * ttl = 1*3DIGIT ; 0 to 255
- * \endcode
- */
-struct sip_via *parse_via(const char *header);
-
-/*!
- * \brief Free parsed Via data.
- */
-void free_via(struct sip_via *v);
-
-#endif
diff --git a/channels/sip/include/route.h b/channels/sip/include/route.h
deleted file mode 100644
index 13acd2cddd..0000000000
--- a/channels/sip/include/route.h
+++ /dev/null
@@ -1,117 +0,0 @@
-/*
- * Asterisk -- An open source telephony toolkit.
- *
- * Copyright (C) 2013, Digium, Inc.
- *
- * See http://www.asterisk.org for more information about
- * the Asterisk project. Please do not directly contact
- * any of the maintainers of this project for assistance;
- * the project provides a web site, mailing lists and IRC
- * channels for your use.
- *
- * This program is free software, distributed under the terms of
- * the GNU General Public License Version 2. See the LICENSE file
- * at the top of the source tree.
- */
-
-/*!
- * \file
- * \brief sip_route header file
- */
-
-#ifndef _SIP_ROUTE_H
-#define _SIP_ROUTE_H
-
-#include "asterisk/linkedlists.h"
-#include "asterisk/strings.h"
-
-/*!
- * \brief Opaque storage of a sip route hop
- */
-struct sip_route_hop;
-
-/*!
- * \internal \brief Internal enum to remember last calculated
- */
-enum sip_route_type {
- route_loose = 0, /*!< The first hop contains ;lr or does not exist */
- route_strict, /*!< The first hop exists and does not contain ;lr */
- route_invalidated, /*!< strict/loose routing needs to be rechecked */
-};
-
-/*!
- * \brief Structure to store route information
- *
- * \note This must be zero-filled on allocation
- */
-struct sip_route {
- AST_LIST_HEAD_NOLOCK(, sip_route_hop) list;
- enum sip_route_type type;
-};
-
-/*!
- * \brief Add a new hop to the route
- *
- * \param route Route
- * \param uri Address of this hop
- * \param len Length of hop not including null terminator
- * \param inserthead If true then inserted the new route to the top of the list
- *
- * \return Pointer to null terminated copy of URI on success
- * \retval NULL on error
- */
-const char *sip_route_add(struct sip_route *route, const char *uri, size_t len, int inserthead);
-
-/*!
- * \brief Add routes from header
- *
- * \note This procedure is for headers that require use of \.
- */
-void sip_route_process_header(struct sip_route *route, const char *header, int inserthead);
-
-/*!
- * \brief copy route-set
- */
-void sip_route_copy(struct sip_route *dst, const struct sip_route *src);
-
-/*!
- * \brief Free all routes in the list
- */
-void sip_route_clear(struct sip_route *route);
-
-/*!
- * \brief Verbose dump of all hops for debugging
- */
-void sip_route_dump(const struct sip_route *route);
-
-/*!
- * \brief Make the comma separated list of route hops
- *
- * \param route Source of route list
- * \param formatcli Add's space after comma's, print's N/A if list is empty.
- * \param skip Number of hops to skip
- *
- * \return an allocated struct ast_str on success
- * \retval NULL on failure
- */
-struct ast_str *sip_route_list(const struct sip_route *route, int formatcli, int skip)
- __attribute__((__malloc__)) __attribute__((__warn_unused_result__));
-
-/*!
- * \brief Check if the route is strict
- *
- * \note The result is cached in route->type
- */
-int sip_route_is_strict(struct sip_route *route);
-
-/*!
- * \brief Get the URI of the route's first hop
- */
-const char *sip_route_first_uri(const struct sip_route *route);
-
-/*!
- * \brief Check if route has no URI's
- */
-#define sip_route_empty(route) AST_LIST_EMPTY(&(route)->list)
-
-#endif
diff --git a/channels/sip/include/security_events.h b/channels/sip/include/security_events.h
deleted file mode 100644
index 9f4bb2ec9b..0000000000
--- a/channels/sip/include/security_events.h
+++ /dev/null
@@ -1,44 +0,0 @@
-/*
- * Asterisk -- An open source telephony toolkit.
- *
- * Copyright (C) 2011, Digium, Inc.
- *
- * Michael L. Young
- *
- * See http://www.asterisk.org for more information about
- * the Asterisk project. Please do not directly contact
- * any of the maintainers of this project for assistance;
- * the project provides a web site, mailing lists and IRC
- * channels for your use.
- *
- * This program is free software, distributed under the terms of
- * the GNU General Public License Version 2. See the LICENSE file
- * at the top of the source tree.
- */
-
-/*!
- * \file
- *
- * \brief Generate security events in the SIP channel
- *
- * \author Michael L. Young
- */
-
-#include "sip.h"
-
-#ifndef _SIP_SECURITY_EVENTS_H
-#define _SIP_SECURITY_EVENTS_H
-
-void sip_report_invalid_peer(const struct sip_pvt *p);
-void sip_report_failed_acl(const struct sip_pvt *p, const char *aclname);
-void sip_report_inval_password(const struct sip_pvt *p, const char *responsechallenge, const char *responsehash);
-void sip_report_auth_success(const struct sip_pvt *p, uint32_t using_password);
-void sip_report_session_limit(const struct sip_pvt *p);
-void sip_report_failed_challenge_response(const struct sip_pvt *p, const char *response, const char *expected_response);
-void sip_report_chal_sent(const struct sip_pvt *p);
-void sip_report_inval_transport(const struct sip_pvt *p, const char *transport);
-void sip_digest_parser(char *c, struct digestkeys *keys);
-int sip_report_security_event(const char *peer, struct ast_sockaddr *addr, const struct sip_pvt *p,
- const struct sip_request *req, const int res);
-
-#endif
diff --git a/channels/sip/include/sip.h b/channels/sip/include/sip.h
deleted file mode 100644
index 1396d81d2e..0000000000
--- a/channels/sip/include/sip.h
+++ /dev/null
@@ -1,1896 +0,0 @@
-/*
- * Asterisk -- An open source telephony toolkit.
- *
- * Copyright (C) 2010, Digium, Inc.
- *
- * See http://www.asterisk.org for more information about
- * the Asterisk project. Please do not directly contact
- * any of the maintainers of this project for assistance;
- * the project provides a web site, mailing lists and IRC
- * channels for your use.
- *
- * This program is free software, distributed under the terms of
- * the GNU General Public License Version 2. See the LICENSE file
- * at the top of the source tree.
- */
-
-/*!
- * \file
- * \brief chan_sip header file
- */
-
-#ifndef _SIP_H
-#define _SIP_H
-
-#include "asterisk.h"
-
-#include "asterisk/stringfields.h"
-#include "asterisk/linkedlists.h"
-#include "asterisk/strings.h"
-#include "asterisk/tcptls.h"
-#include "asterisk/test.h"
-#include "asterisk/channel.h"
-#include "asterisk/app.h"
-#include "asterisk/indications.h"
-#include "asterisk/security_events.h"
-#include "asterisk/features.h"
-#include "asterisk/rtp_engine.h"
-#include "asterisk/netsock2.h"
-#include "asterisk/features_config.h"
-
-#include "route.h"
-
-#ifndef FALSE
-#define FALSE 0
-#endif
-
-#ifndef TRUE
-#define TRUE 1
-#endif
-
-/* Arguments for sip_find_peer */
-#define FINDUSERS (1 << 0)
-#define FINDPEERS (1 << 1)
-#define FINDALLDEVICES (FINDUSERS | FINDPEERS)
-
-#define SIPBUFSIZE 512 /*!< Buffer size for many operations */
-
-#define XMIT_ERROR -2
-
-#define SIP_RESERVED ";/?:@&=+$,# " /*!< Reserved characters in the username part of the URI */
-
-#define DEFAULT_DEFAULT_EXPIRY 120
-#define DEFAULT_MIN_EXPIRY 60
-#define DEFAULT_MAX_EXPIRY 3600
-#define DEFAULT_MWI_EXPIRY 3600
-#define DEFAULT_REGISTRATION_TIMEOUT 20
-#define DEFAULT_MAX_FORWARDS 70
-
-#define DEFAULT_AUTHLIMIT 100
-#define DEFAULT_AUTHTIMEOUT 30
-
-/* guard limit must be larger than guard secs */
-/* guard min must be < 1000, and should be >= 250 */
-#define EXPIRY_GUARD_SECS 15 /*!< How long before expiry do we reregister */
-#define EXPIRY_GUARD_LIMIT 30 /*!< Below here, we use EXPIRY_GUARD_PCT instead of EXPIRY_GUARD_SECS */
-#define EXPIRY_GUARD_MIN 500 /*!< This is the minimum guard time applied. If
- * GUARD_PCT turns out to be lower than this, it
- * will use this time instead.
- * This is in milliseconds.
- */
-#define EXPIRY_GUARD_PCT 0.20 /*!< Percentage of expires timeout to use when
- * below EXPIRY_GUARD_LIMIT */
-#define DEFAULT_EXPIRY 900 /*!< Expire slowly */
-
-#define DEFAULT_QUALIFY_GAP 100
-#define DEFAULT_QUALIFY_PEERS 1
-
-#define CALLERID_UNKNOWN "Anonymous"
-#define FROMDOMAIN_INVALID "anonymous.invalid"
-
-#define DEFAULT_MAXMS 2000 /*!< Qualification: Must be faster than 2 seconds by default */
-#define DEFAULT_QUALIFYFREQ 60 * 1000 /*!< Qualification: How often to check for the host to be up */
-#define DEFAULT_FREQ_NOTOK 10 * 1000 /*!< Qualification: How often to check, if the host is down... */
-
-#define DEFAULT_RETRANS 1000 /*!< How frequently to retransmit Default: 2 * 500 ms in RFC 3261 */
-#define DEFAULT_TIMER_T1 500 /*!< SIP timer T1 (according to RFC 3261) */
-#define SIP_TRANS_TIMEOUT 64 * DEFAULT_TIMER_T1 /*!< SIP request timeout (rfc 3261) 64*T1
- * \todo Use known T1 for timeout (peerpoke)
- */
-#define DEFAULT_TRANS_TIMEOUT -1 /*!< Use default SIP transaction timeout */
-#define PROVIS_KEEPALIVE_TIMEOUT 60000 /*!< How long to wait before retransmitting a provisional response (rfc 3261 13.3.1.1) */
-#define MAX_AUTHTRIES 3 /*!< Try authentication three times, then fail */
-
-#define SIP_MAX_HEADERS 64 /*!< Max amount of SIP headers to read */
-#define SIP_MAX_LINES 256 /*!< Max amount of lines in SIP attachment (like SDP) */
-#define SIP_MAX_PACKET_SIZE 20480 /*!< Max SIP packet size */
-#define SIP_MIN_PACKET 4096 /*!< Initialize size of memory to allocate for packets */
-#define MAX_HISTORY_ENTRIES 50 /*!< Max entires in the history list for a sip_pvt */
-
-#define INITIAL_CSEQ 101 /*!< Our initial sip sequence number */
-
-#define DEFAULT_MAX_SE 1800 /*!< Session-Timer Default Session-Expires period (RFC 4028) */
-#define DEFAULT_MIN_SE 90 /*!< Session-Timer Default Min-SE period (RFC 4028) */
-
-#define SDP_MAX_RTPMAP_CODECS 32 /*!< Maximum number of codecs allowed in received SDP */
-
-#define RTP 1
-#define NO_RTP 0
-
-#define DEC_CALL_LIMIT 0
-#define INC_CALL_LIMIT 1
-#define DEC_CALL_RINGING 2
-#define INC_CALL_RINGING 3
-
-/*! Define SIP option tags, used in Require: and Supported: headers
- * We need to be aware of these properties in the phones to use
- * the replace: header. We should not do that without knowing
- * that the other end supports it...
- * This is nothing we can configure, we learn by the dialog
- * Supported: header on the REGISTER (peer) or the INVITE
- * (other devices)
- * We are not using many of these today, but will in the future.
- * This is documented in RFC 3261
- */
-#define SUPPORTED 1
-#define NOT_SUPPORTED 0
-
-/* SIP options */
-#define SIP_OPT_REPLACES (1 << 0)
-#define SIP_OPT_100REL (1 << 1)
-#define SIP_OPT_TIMER (1 << 2)
-#define SIP_OPT_EARLY_SESSION (1 << 3)
-#define SIP_OPT_JOIN (1 << 4)
-#define SIP_OPT_PATH (1 << 5)
-#define SIP_OPT_PREF (1 << 6)
-#define SIP_OPT_PRECONDITION (1 << 7)
-#define SIP_OPT_PRIVACY (1 << 8)
-#define SIP_OPT_SDP_ANAT (1 << 9)
-#define SIP_OPT_SEC_AGREE (1 << 10)
-#define SIP_OPT_EVENTLIST (1 << 11)
-#define SIP_OPT_GRUU (1 << 12)
-#define SIP_OPT_TARGET_DIALOG (1 << 13)
-#define SIP_OPT_NOREFERSUB (1 << 14)
-#define SIP_OPT_HISTINFO (1 << 15)
-#define SIP_OPT_RESPRIORITY (1 << 16)
-#define SIP_OPT_FROMCHANGE (1 << 17)
-#define SIP_OPT_RECLISTINV (1 << 18)
-#define SIP_OPT_RECLISTSUB (1 << 19)
-#define SIP_OPT_OUTBOUND (1 << 20)
-#define SIP_OPT_UNKNOWN (1 << 21)
-
-/*! \brief SIP Methods we support
- * \todo This string should be set dynamically. We only support REFER and SUBSCRIBE if we have
- * allowsubscribe and allowrefer on in sip.conf.
- */
-#define ALLOWED_METHODS "INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE"
-
-/*! \brief Standard SIP unsecure port for UDP and TCP from RFC 3261. DO NOT CHANGE THIS */
-#define STANDARD_SIP_PORT 5060
-/*! \brief Standard SIP TLS port from RFC 3261. DO NOT CHANGE THIS */
-#define STANDARD_TLS_PORT 5061
-
-/*! \note in many SIP headers, absence of a port number implies port 5060,
- * and this is why we cannot change the above constant.
- * There is a limited number of places in asterisk where we could,
- * in principle, use a different "default" port number, but
- * we do not support this feature at the moment.
- * You can run Asterisk with SIP on a different port with a configuration
- * option. If you change this value in the source code, the signalling will be incorrect.
- *
- */
-
-/*! \name DefaultValues Default values, set and reset in reload_config before reading configuration
-
- These are default values in the source. There are other recommended values in the
- sip.conf.sample for new installations. These may differ to keep backwards compatibility,
- yet encouraging new behaviour on new installations
-
- @{
- */
-#define DEFAULT_CONTEXT "default" /*!< The default context for [general] section as well as devices */
-#define DEFAULT_RECORD_FEATURE "automon" /*!< The default feature specified for use with INFO */
-#define DEFAULT_MOHINTERPRET "default" /*!< The default music class */
-#define DEFAULT_MOHSUGGEST ""
-#define DEFAULT_VMEXTEN "asterisk" /*!< Default voicemail extension */
-#define DEFAULT_CALLERID "asterisk" /*!< Default caller ID */
-#define DEFAULT_MWI_FROM ""
-#define DEFAULT_NOTIFYMIME "application/simple-message-summary"
-#define DEFAULT_ALLOWGUEST TRUE
-#define DEFAULT_RTPKEEPALIVE 0 /*!< Default RTPkeepalive setting */
-#define DEFAULT_CALLCOUNTER FALSE /*!< Do not enable call counters by default */
-#define DEFAULT_SRVLOOKUP TRUE /*!< Recommended setting is ON */
-#define DEFAULT_COMPACTHEADERS FALSE /*!< Send compact (one-character) SIP headers. Default off */
-#define DEFAULT_TOS_SIP 0 /*!< Call signalling packets should be marked as DSCP CS3, but the default is 0 to be compatible with previous versions. */
-#define DEFAULT_TOS_AUDIO 0 /*!< Audio packets should be marked as DSCP EF (Expedited Forwarding), but the default is 0 to be compatible with previous versions. */
-#define DEFAULT_TOS_VIDEO 0 /*!< Video packets should be marked as DSCP AF41, but the default is 0 to be compatible with previous versions. */
-#define DEFAULT_TOS_TEXT 0 /*!< Text packets should be marked as XXXX XXXX, but the default is 0 to be compatible with previous versions. */
-#define DEFAULT_COS_SIP 4 /*!< Level 2 class of service for SIP signalling */
-#define DEFAULT_COS_AUDIO 5 /*!< Level 2 class of service for audio media */
-#define DEFAULT_COS_VIDEO 6 /*!< Level 2 class of service for video media */
-#define DEFAULT_COS_TEXT 5 /*!< Level 2 class of service for text media (T.140) */
-#define DEFAULT_ALLOW_EXT_DOM TRUE /*!< Allow external domains */
-#define DEFAULT_REALM "asterisk" /*!< Realm for HTTP digest authentication */
-#define DEFAULT_DOMAINSASREALM FALSE /*!< Use the domain option to guess the realm for registration and invite requests */
-#define DEFAULT_NOTIFYRINGING NOTIFYRINGING_ENABLED /*!< Notify devicestate system on ringing state */
-#define DEFAULT_NOTIFYCID DISABLED /*!< Include CID with ringing notifications */
-#define DEFAULT_PEDANTIC TRUE /*!< Follow SIP standards for dialog matching */
-#define DEFAULT_AUTOCREATEPEER AUTOPEERS_DISABLED /*!< Don't create peers automagically */
-#define DEFAULT_MATCHEXTERNADDRLOCALLY FALSE /*!< Match extern IP locally default setting */
-#define DEFAULT_QUALIFY FALSE /*!< Don't monitor devices */
-#define DEFAULT_KEEPALIVE 0 /*!< Don't send keep alive packets */
-#define DEFAULT_KEEPALIVE_INTERVAL 60 /*!< Send keep alive packets at 60 second intervals */
-#define DEFAULT_ALWAYSAUTHREJECT TRUE /*!< Don't reject authentication requests always */
-#define DEFAULT_AUTH_OPTIONS FALSE
-#define DEFAULT_AUTH_MESSAGE TRUE
-#define DEFAULT_ACCEPT_OUTOFCALL_MESSAGE TRUE
-#define DEFAULT_REGEXTENONQUALIFY FALSE
-#define DEFAULT_LEGACY_USEROPTION_PARSING FALSE
-#define DEFAULT_SEND_DIVERSION TRUE
-#define DEFAULT_T1MIN 100 /*!< 100 MS for minimal roundtrip time */
-#define DEFAULT_MAX_CALL_BITRATE (384) /*!< Max bitrate for video */
-#ifndef DEFAULT_USERAGENT
-#define DEFAULT_USERAGENT "Asterisk PBX" /*!< Default Useragent: header unless re-defined in sip.conf */
-#define DEFAULT_SDPSESSION "Asterisk PBX" /*!< Default SDP session name, (s=) header unless re-defined in sip.conf */
-#define DEFAULT_SDPOWNER "root" /*!< Default SDP username field in (o=) header unless re-defined in sip.conf */
-#define DEFAULT_ENGINE "asterisk" /*!< Default RTP engine to use for sessions */
-#define DEFAULT_STORE_SIP_CAUSE FALSE /*!< Don't store HASH(SIP_CAUSE,) for channels by default */
-#endif
-
-/*! @} */
-
-/*! \name SIPflags
- Various flags for the flags field in the pvt structure
- Trying to sort these up (one or more of the following):
- D: Dialog
- P: Peer/user
- G: Global flag
- When flags are used by multiple structures, it is important that
- they have a common layout so it is easy to copy them.
- @{
- */
-#define SIP_OUTGOING (1 << 0) /*!< D: Direction of the last transaction in this dialog */
-#define SIP_OFFER_CC (1 << 1) /*!< D: Offer CC on subsequent responses */
-#define SIP_RINGING (1 << 2) /*!< D: Have sent 180 ringing */
-#define SIP_PROGRESS_SENT (1 << 3) /*!< D: Have sent 183 message progress */
-#define SIP_NEEDREINVITE (1 << 4) /*!< D: Do we need to send another reinvite? */
-#define SIP_PENDINGBYE (1 << 5) /*!< D: Need to send bye after we ack? */
-#define SIP_GOTREFER (1 << 6) /*!< D: Got a refer? */
-#define SIP_CALL_LIMIT (1 << 7) /*!< D: Call limit enforced for this call */
-#define SIP_INC_COUNT (1 << 8) /*!< D: Did this dialog increment the counter of in-use calls? */
-#define SIP_INC_RINGING (1 << 9) /*!< D: Did this connection increment the counter of in-use calls? */
-#define SIP_DEFER_BYE_ON_TRANSFER (1 << 10) /*!< D: Do not hangup at first ast_hangup */
-
-#define SIP_PROMISCREDIR (1 << 11) /*!< DP: Promiscuous redirection */
-#define SIP_TRUSTRPID (1 << 12) /*!< DP: Trust RPID headers? */
-#define SIP_USEREQPHONE (1 << 13) /*!< DP: Add user=phone to numeric URI. Default off */
-#define SIP_USECLIENTCODE (1 << 14) /*!< DP: Trust X-ClientCode info message */
-
-/* DTMF flags - see str2dtmfmode() and dtmfmode2str() */
-#define SIP_DTMF (7 << 15) /*!< DP: DTMF Support: five settings, uses three bits */
-#define SIP_DTMF_RFC2833 (0 << 15) /*!< DP: DTMF Support: RTP DTMF - "rfc2833" */
-#define SIP_DTMF_INBAND (1 << 15) /*!< DP: DTMF Support: Inband audio, only for ULAW/ALAW - "inband" */
-#define SIP_DTMF_INFO (2 << 15) /*!< DP: DTMF Support: SIP Info messages - "info" */
-#define SIP_DTMF_AUTO (3 << 15) /*!< DP: DTMF Support: AUTO switch between rfc2833 and in-band DTMF */
-#define SIP_DTMF_SHORTINFO (4 << 15) /*!< DP: DTMF Support: SIP Info messages - "info" - short variant */
-
-/* NAT settings */
-#define SIP_NAT_FORCE_RPORT (1 << 18) /*!< DP: Force rport even if not present in the request */
-#define SIP_NAT_RPORT_PRESENT (1 << 19) /*!< DP: rport was present in the request */
-
-/* re-INVITE related settings */
-#define SIP_REINVITE (7 << 20) /*!< DP: four settings, uses three bits */
-#define SIP_REINVITE_NONE (0 << 20) /*!< DP: no reinvite allowed */
-#define SIP_DIRECT_MEDIA (1 << 20) /*!< DP: allow peers to be reinvited to send media directly p2p */
-#define SIP_DIRECT_MEDIA_NAT (2 << 20) /*!< DP: allow media reinvite when new peer is behind NAT */
-#define SIP_REINVITE_UPDATE (4 << 20) /*!< DP: use UPDATE (RFC3311) when reinviting this peer */
-
-/* "insecure" settings - see insecure2str() */
-#define SIP_INSECURE (3 << 23) /*!< DP: three settings, uses two bits */
-#define SIP_INSECURE_NONE (0 << 23) /*!< DP: secure mode */
-#define SIP_INSECURE_PORT (1 << 23) /*!< DP: don't require matching port for incoming requests */
-#define SIP_INSECURE_INVITE (1 << 24) /*!< DP: don't require authentication for incoming INVITEs */
-
-/* Sending PROGRESS in-band settings */
-#define SIP_PROG_INBAND (3 << 25) /*!< DP: three settings, uses two bits */
-#define SIP_PROG_INBAND_NO (0 << 25)
-#define SIP_PROG_INBAND_NEVER (1 << 25)
-#define SIP_PROG_INBAND_YES (2 << 25)
-
-#define SIP_USEPATH (1 << 27) /*!< GDP: Trust and use incoming Path headers? */
-#define SIP_SENDRPID (3 << 29) /*!< DP: Remote Party-ID Support */
-#define SIP_SENDRPID_NO (0 << 29)
-#define SIP_SENDRPID_PAI (1 << 29) /*!< Use "P-Asserted-Identity" for rpid */
-#define SIP_SENDRPID_RPID (2 << 29) /*!< Use "Remote-Party-ID" for rpid */
-#define SIP_G726_NONSTANDARD (1 << 31) /*!< DP: Use non-standard packing for G726-32 data */
-
-/*! \brief Flags to copy from peer/user to dialog */
-#define SIP_FLAGS_TO_COPY \
- (SIP_PROMISCREDIR | SIP_TRUSTRPID | SIP_SENDRPID | SIP_DTMF | SIP_REINVITE | \
- SIP_PROG_INBAND | SIP_USECLIENTCODE | SIP_NAT_FORCE_RPORT | SIP_G726_NONSTANDARD | \
- SIP_USEREQPHONE | SIP_INSECURE | SIP_USEPATH)
-
-/*! @} */
-
-/*! \name SIPflags2
- a second page of flags (for flags[1]
- @{
- */
-/* realtime flags */
-#define SIP_PAGE2_RTCACHEFRIENDS (1 << 0) /*!< GP: Should we keep RT objects in memory for extended time? */
-#define SIP_PAGE2_RTAUTOCLEAR (1 << 1) /*!< GP: Should we clean memory from peers after expiry? */
-#define SIP_PAGE2_RPID_UPDATE (1 << 2)
-#define SIP_PAGE2_Q850_REASON (1 << 3) /*!< DP: Get/send cause code via Reason header */
-#define SIP_PAGE2_SYMMETRICRTP (1 << 4) /*!< GDP: Whether symmetric RTP is enabled or not */
-#define SIP_PAGE2_STATECHANGEQUEUE (1 << 5) /*!< D: Unsent state pending change exists */
-#define SIP_PAGE2_CONNECTLINEUPDATE_PEND (1 << 6)
-#define SIP_PAGE2_RPID_IMMEDIATE (1 << 7)
-#define SIP_PAGE2_RPORT_PRESENT (1 << 8) /*!< Was rport received in the Via header? */
-#define SIP_PAGE2_PREFERRED_CODEC (1 << 9) /*!< GDP: Only respond with single most preferred joint codec */
-#define SIP_PAGE2_VIDEOSUPPORT (1 << 10) /*!< DP: Video supported if offered? */
-#define SIP_PAGE2_TEXTSUPPORT (1 << 11) /*!< GDP: Global text enable */
-#define SIP_PAGE2_ALLOWSUBSCRIBE (1 << 12) /*!< GP: Allow subscriptions from this peer? */
-
-#define SIP_PAGE2_ALLOWOVERLAP (3 << 13) /*!< DP: Allow overlap dialing ? */
-#define SIP_PAGE2_ALLOWOVERLAP_NO (0 << 13) /*!< No, terminate with 404 Not found */
-#define SIP_PAGE2_ALLOWOVERLAP_YES (1 << 13) /*!< Yes, using the 484 Address Incomplete response */
-#define SIP_PAGE2_ALLOWOVERLAP_DTMF (2 << 13) /*!< Yes, using the DTMF transmission through Early Media */
-#define SIP_PAGE2_ALLOWOVERLAP_SPARE (3 << 13) /*!< Spare (reserved for another dialling transmission mechanisms like KPML) */
-
-#define SIP_PAGE2_SUBSCRIBEMWIONLY (1 << 15) /*!< GP: Only issue MWI notification if subscribed to */
-#define SIP_PAGE2_IGNORESDPVERSION (1 << 16) /*!< GDP: Ignore the SDP session version number we receive and treat all sessions as new */
-
-#define SIP_PAGE2_T38SUPPORT (3 << 17) /*!< GDP: T.38 Fax Support */
-#define SIP_PAGE2_T38SUPPORT_UDPTL (1 << 17) /*!< GDP: T.38 Fax Support (no error correction) */
-#define SIP_PAGE2_T38SUPPORT_UDPTL_FEC (2 << 17) /*!< GDP: T.38 Fax Support (FEC error correction) */
-#define SIP_PAGE2_T38SUPPORT_UDPTL_REDUNDANCY (3 << 17) /*!< GDP: T.38 Fax Support (redundancy error correction) */
-
-#define SIP_PAGE2_CALL_ONHOLD (3 << 19) /*!< D: Call hold states: */
-#define SIP_PAGE2_CALL_ONHOLD_ACTIVE (1 << 19) /*!< D: Active hold */
-#define SIP_PAGE2_CALL_ONHOLD_ONEDIR (2 << 19) /*!< D: One directional hold */
-#define SIP_PAGE2_CALL_ONHOLD_INACTIVE (3 << 19) /*!< D: Inactive hold */
-
-#define SIP_PAGE2_RFC2833_COMPENSATE (1 << 21) /*!< DP: Compensate for buggy RFC2833 implementations */
-#define SIP_PAGE2_BUGGY_MWI (1 << 22) /*!< DP: Buggy CISCO MWI fix */
-#define SIP_PAGE2_DIALOG_ESTABLISHED (1 << 23) /*!< 29: Has a dialog been established? */
-
-#define SIP_PAGE2_FAX_DETECT (3 << 24) /*!< DP: Fax Detection support */
-#define SIP_PAGE2_FAX_DETECT_CNG (1 << 24) /*!< DP: Fax Detection support - detect CNG in audio */
-#define SIP_PAGE2_FAX_DETECT_T38 (2 << 24) /*!< DP: Fax Detection support - detect T.38 reinvite from peer */
-#define SIP_PAGE2_FAX_DETECT_BOTH (3 << 24) /*!< DP: Fax Detection support - detect both */
-
-#define SIP_PAGE2_UDPTL_DESTINATION (1 << 26) /*!< DP: Use source IP of RTP as destination if NAT is enabled */
-#define SIP_PAGE2_VIDEOSUPPORT_ALWAYS (1 << 27) /*!< DP: Always set up video, even if endpoints don't support it */
-#define SIP_PAGE2_HAVEPEERCONTEXT (1 << 28) /*< Are we associated with a configured peer context? */
-#define SIP_PAGE2_USE_SRTP (1 << 29) /*!< DP: Whether we should offer (only) SRTP */
-
-#define SIP_PAGE2_TRUST_ID_OUTBOUND (3 << 30) /*!< DP: Do we trust the peer with private presence information? */
-#define SIP_PAGE2_TRUST_ID_OUTBOUND_LEGACY (0 << 30) /*!< Legacy, Do not provide private presence information, but include PAI/RPID when private */
-#define SIP_PAGE2_TRUST_ID_OUTBOUND_NO (1 << 30) /*!< No, Do not provide private presence information, do not include PAI/RPID when private */
-#define SIP_PAGE2_TRUST_ID_OUTBOUND_YES (2 << 30) /*!< Yes, provide private presence information in PAI/RPID headers */
-
-#define SIP_PAGE2_FLAGS_TO_COPY \
- (SIP_PAGE2_ALLOWSUBSCRIBE | SIP_PAGE2_ALLOWOVERLAP | SIP_PAGE2_IGNORESDPVERSION | \
- SIP_PAGE2_VIDEOSUPPORT | SIP_PAGE2_T38SUPPORT | SIP_PAGE2_RFC2833_COMPENSATE | \
- SIP_PAGE2_BUGGY_MWI | SIP_PAGE2_TEXTSUPPORT | SIP_PAGE2_FAX_DETECT | \
- SIP_PAGE2_UDPTL_DESTINATION | SIP_PAGE2_VIDEOSUPPORT_ALWAYS | SIP_PAGE2_PREFERRED_CODEC | \
- SIP_PAGE2_RPID_IMMEDIATE | SIP_PAGE2_RPID_UPDATE | SIP_PAGE2_SYMMETRICRTP |\
- SIP_PAGE2_Q850_REASON | SIP_PAGE2_HAVEPEERCONTEXT | SIP_PAGE2_USE_SRTP | SIP_PAGE2_TRUST_ID_OUTBOUND)
-
-
-#define SIP_PAGE3_SNOM_AOC (1 << 0) /*!< DPG: Allow snom aoc messages */
-#define SIP_PAGE3_SRTP_TAG_32 (1 << 1) /*!< DP: Use a 32bit auth tag in INVITE not 80bit */
-#define SIP_PAGE3_NAT_AUTO_RPORT (1 << 2) /*!< DGP: Set SIP_NAT_FORCE_RPORT when NAT is detected */
-#define SIP_PAGE3_NAT_AUTO_COMEDIA (1 << 3) /*!< DGP: Set SIP_PAGE2_SYMMETRICRTP when NAT is detected */
-#define SIP_PAGE3_DIRECT_MEDIA_OUTGOING (1 << 4) /*!< DP: Only send direct media reinvites on outgoing calls */
-#define SIP_PAGE3_USE_AVPF (1 << 5) /*!< DGP: Support a minimal AVPF-compatible profile */
-#define SIP_PAGE3_ICE_SUPPORT (1 << 6) /*!< DGP: Enable ICE support */
-#define SIP_PAGE3_IGNORE_PREFCAPS (1 << 7) /*!< DP: Ignore prefcaps when setting up an outgoing call leg */
-#define SIP_PAGE3_DISCARD_REMOTE_HOLD_RETRIEVAL (1 << 8) /*!< DGP: Stop telling the peer to start music on hold */
-#define SIP_PAGE3_FORCE_AVP (1 << 9) /*!< DGP: Force 'RTP/AVP' for all streams, even DTLS */
-#define SIP_PAGE3_RTCP_MUX (1 << 10) /*!< DGP: Attempt to negotiate RFC 5761 RTCP multiplexing */
-
-#define SIP_PAGE3_FLAGS_TO_COPY \
- (SIP_PAGE3_SNOM_AOC | SIP_PAGE3_SRTP_TAG_32 | SIP_PAGE3_NAT_AUTO_RPORT | SIP_PAGE3_NAT_AUTO_COMEDIA | \
- SIP_PAGE3_DIRECT_MEDIA_OUTGOING | SIP_PAGE3_USE_AVPF | SIP_PAGE3_ICE_SUPPORT | SIP_PAGE3_IGNORE_PREFCAPS | \
- SIP_PAGE3_DISCARD_REMOTE_HOLD_RETRIEVAL | SIP_PAGE3_FORCE_AVP | SIP_PAGE3_RTCP_MUX)
-
-#define CHECK_AUTH_BUF_INITLEN 256
-
-/*! @} */
-
-/*----------------------------------------------------------*/
-/*---- ENUMS ----*/
-/*----------------------------------------------------------*/
-
-/*! \brief Authorization scheme for call transfers
- *
- * \note Not a bitfield flag, since there are plans for other modes,
- * like "only allow transfers for authenticated devices"
- */
-enum transfermodes {
- TRANSFER_OPENFORALL, /*!< Allow all SIP transfers */
- TRANSFER_CLOSED, /*!< Allow no SIP transfers */
-};
-
-/*! \brief The result of a lot of functions */
-enum sip_result {
- AST_SUCCESS = 0, /*!< FALSE means success, funny enough */
- AST_FAILURE = -1, /*!< Failure code */
-};
-
-/*! \brief The results from handling an invite request
- *
- * \note Start at these values so we do not conflict with
- * check_auth_results values when returning from
- * handle_request_invite. check_auth_results only returned during
- * authentication routines
- * */
-enum inv_req_result {
- INV_REQ_SUCCESS = 11, /*!< Success code */
- INV_REQ_FAILED = 10, /*!< Failure code */
- INV_REQ_ERROR = 9, /*!< Error code */
-};
-
-/*! \brief States for the INVITE transaction, not the dialog
- * \note this is for the INVITE that sets up the dialog
- */
-enum invitestates {
- INV_NONE = 0, /*!< No state at all, maybe not an INVITE dialog */
- INV_CALLING = 1, /*!< Invite sent, no answer */
- INV_PROCEEDING = 2, /*!< We got/sent 1xx message */
- INV_EARLY_MEDIA = 3, /*!< We got 18x message with to-tag back */
- INV_COMPLETED = 4, /*!< Got final response with error. Wait for ACK, then CONFIRMED */
- INV_CONFIRMED = 5, /*!< Confirmed response - we've got an ack (Incoming calls only) */
- INV_TERMINATED = 6, /*!< Transaction done - either successful (AST_STATE_UP) or failed, but done
- The only way out of this is a BYE from one side */
- INV_CANCELLED = 7, /*!< Transaction cancelled by client or server in non-terminated state */
-};
-
-/*! \brief When sending a SIP message, we can send with a few options, depending on
- * type of SIP request. UNRELIABLE is mostly used for responses to repeated requests,
- * where the original response would be sent RELIABLE in an INVITE transaction
- */
-enum xmittype {
- XMIT_CRITICAL = 2, /*!< Transmit critical SIP message reliably, with re-transmits.
- * If it fails, it's critical and will cause a teardown of the session */
- XMIT_RELIABLE = 1, /*!< Transmit SIP message reliably, with re-transmits */
- XMIT_UNRELIABLE = 0, /*!< Transmit SIP message without bothering with re-transmits */
-};
-
-/*! \brief Results from the parse_register() function */
-enum parse_register_result {
- PARSE_REGISTER_DENIED,
- PARSE_REGISTER_FAILED,
- PARSE_REGISTER_UPDATE,
- PARSE_REGISTER_QUERY,
-};
-
-/*! \brief Type of subscription, based on the packages we do support, see \ref subscription_types */
-enum subscriptiontype {
- NONE = 0,
- XPIDF_XML,
- DIALOG_INFO_XML,
- CPIM_PIDF_XML,
- PIDF_XML,
- MWI_NOTIFICATION,
- CALL_COMPLETION,
-};
-
-/*! \brief The number of media types in enum \ref media_type below. */
-#define OFFERED_MEDIA_COUNT 4
-
-/*! \brief Media types generate different "dummy answers" for not accepting the offer of
- a media stream. We need to add definitions for each RTP profile. Secure RTP is not
- the same as normal RTP and will require a new definition */
-enum media_type {
- SDP_AUDIO, /*!< RTP/AVP Audio */
- SDP_VIDEO, /*!< RTP/AVP Video */
- SDP_IMAGE, /*!< Image udptl, not TCP or RTP */
- SDP_TEXT, /*!< RTP/AVP Realtime Text */
- SDP_UNKNOWN, /*!< Unknown media type */
-};
-
-/*! \brief Authentication types - proxy or www authentication
- * \note Endpoints, like Asterisk, should always use WWW authentication to
- * allow multiple authentications in the same call - to the proxy and
- * to the end point.
- */
-enum sip_auth_type {
- PROXY_AUTH = 407,
- WWW_AUTH = 401,
-};
-
-/*! \brief Result from get_destination function */
-enum sip_get_dest_result {
- SIP_GET_DEST_EXTEN_MATCHMORE = 1,
- SIP_GET_DEST_EXTEN_FOUND = 0,
- SIP_GET_DEST_EXTEN_NOT_FOUND = -1,
- SIP_GET_DEST_REFUSED = -2,
- SIP_GET_DEST_INVALID_URI = -3,
-};
-
-/*! \brief Authentication result from check_auth* functions */
-enum check_auth_result {
- AUTH_DONT_KNOW = -100, /*!< no result, need to check further */
- /* XXX maybe this is the same as AUTH_NOT_FOUND */
- AUTH_SUCCESSFUL = 0,
- AUTH_CHALLENGE_SENT = 1,
- AUTH_SECRET_FAILED = -1,
- AUTH_USERNAME_MISMATCH = -2,
- AUTH_NOT_FOUND = -3, /*!< returned by register_verify */
- AUTH_UNKNOWN_DOMAIN = -5,
- AUTH_PEER_NOT_DYNAMIC = -6,
- AUTH_ACL_FAILED = -7,
- AUTH_BAD_TRANSPORT = -8,
- AUTH_RTP_FAILED = -9,
- AUTH_SESSION_LIMIT = -10,
-};
-
-/*! \brief States for outbound registrations (with register= lines in sip.conf */
-enum sipregistrystate {
- REG_STATE_UNREGISTERED = 0, /*!< We are not registered
- * \note Initial state. We should have a timeout scheduled for the initial
- * (or next) registration transmission, calling sip_reregister
- */
-
- REG_STATE_REGSENT, /*!< Registration request sent
- * \note sent initial request, waiting for an ack or a timeout to
- * retransmit the initial request.
- */
-
- REG_STATE_AUTHSENT, /*!< We have tried to authenticate
- * \note entered after transmit_register with auth info,
- * waiting for an ack.
- */
-
- REG_STATE_REGISTERED, /*!< Registered and done */
-
- REG_STATE_REJECTED, /*!< Registration rejected
- * \note only used when the remote party has an expire larger than
- * our max-expire. This is a final state from which we do not
- * recover (not sure how correctly).
- */
-
- REG_STATE_TIMEOUT, /*!< Registration about to expire, renewing registration */
-
- REG_STATE_NOAUTH, /*!< We have no accepted credentials
- * \note fatal - no chance to proceed */
-
- REG_STATE_FAILED, /*!< Registration failed after several tries
- * \note fatal - no chance to proceed */
-};
-
-/*! \brief Modes in which Asterisk can be configured to run SIP Session-Timers */
-enum st_mode {
- SESSION_TIMER_MODE_INVALID = 0, /*!< Invalid value */
- SESSION_TIMER_MODE_ACCEPT, /*!< Honor inbound Session-Timer requests */
- SESSION_TIMER_MODE_ORIGINATE, /*!< Originate outbound and honor inbound requests */
- SESSION_TIMER_MODE_REFUSE /*!< Ignore inbound Session-Timers requests */
-};
-
-/*! \brief The entity playing the refresher role for Session-Timers */
-enum st_refresher {
- SESSION_TIMER_REFRESHER_AUTO, /*!< Negotiated */
- SESSION_TIMER_REFRESHER_US, /*!< Initially prefer session refresh by Asterisk */
- SESSION_TIMER_REFRESHER_THEM, /*!< Initially prefer session refresh by the other side */
-};
-
-enum st_refresher_param {
- SESSION_TIMER_REFRESHER_PARAM_UNKNOWN,
- SESSION_TIMER_REFRESHER_PARAM_UAC,
- SESSION_TIMER_REFRESHER_PARAM_UAS,
-};
-
-/*! \brief Automatic peer registration behavior
-*/
-enum autocreatepeer_mode {
- AUTOPEERS_DISABLED = 0, /*!< Automatic peer creation disabled */
- AUTOPEERS_VOLATILE, /*!< Automatic peers dropped on sip reload (pre-1.8 behavior) */
- AUTOPEERS_PERSIST /*!< Automatic peers survive sip configuration reload */
-};
-
-/*! \brief States whether a SIP message can create a dialog in Asterisk. */
-enum can_create_dialog {
- CAN_NOT_CREATE_DIALOG,
- CAN_CREATE_DIALOG,
- CAN_CREATE_DIALOG_UNSUPPORTED_METHOD,
-};
-
-/*! \brief SIP Request methods known by Asterisk
- *
- * \note Do _NOT_ make any changes to this enum, or the array following it;
- * if you think you are doing the right thing, you are probably
- * not doing the right thing. If you think there are changes
- * needed, get someone else to review them first _before_
- * submitting a patch. If these two lists do not match properly
- * bad things will happen.
- */
-enum sipmethod {
- SIP_UNKNOWN, /*!< Unknown response */
- SIP_RESPONSE, /*!< Not request, response to outbound request */
- SIP_REGISTER, /*!< Registration to the mothership, tell us where you are located */
- SIP_OPTIONS, /*!< Check capabilities of a device, used for "ping" too */
- SIP_NOTIFY, /*!< Status update, Part of the event package standard, result of a SUBSCRIBE or a REFER */
- SIP_INVITE, /*!< Set up a session */
- SIP_ACK, /*!< End of a three-way handshake started with INVITE. */
- SIP_PRACK, /*!< Reliable pre-call signalling. Not supported in Asterisk. */
- SIP_BYE, /*!< End of a session */
- SIP_REFER, /*!< Refer to another URI (transfer) */
- SIP_SUBSCRIBE, /*!< Subscribe for updates (voicemail, session status, device status, presence) */
- SIP_MESSAGE, /*!< Text messaging */
- SIP_UPDATE, /*!< Update a dialog. We can send UPDATE; but not accept it */
- SIP_INFO, /*!< Information updates during a session */
- SIP_CANCEL, /*!< Cancel an INVITE */
- SIP_PUBLISH, /*!< Not supported in Asterisk */
- SIP_PING, /*!< Not supported at all, no standard but still implemented out there */
-};
-
-/*! \brief Setting for the 'notifyringing' option, see sip.conf.sample for details. */
-enum notifyringing_setting {
- NOTIFYRINGING_DISABLED = 0,
- NOTIFYRINGING_ENABLED = 1,
- NOTIFYRINGING_NOTINUSE = 2,
-};
-
-/*! \brief Settings for the 'notifycid' option, see sip.conf.sample for details. */
-enum notifycid_setting {
- DISABLED = 0,
- ENABLED = 1,
- IGNORE_CONTEXT = 2,
-};
-
-/*! \brief Modes for SIP domain handling in the PBX */
-enum domain_mode {
- SIP_DOMAIN_AUTO, /*!< This domain is auto-configured */
- SIP_DOMAIN_CONFIG, /*!< This domain is from configuration */
-};
-
-/*! \brief debugging state
- * We store separately the debugging requests from the config file
- * and requests from the CLI. Debugging is enabled if either is set
- * (which means that if sipdebug is set in the config file, we can
- * only turn it off by reloading the config).
- */
-enum sip_debug_e {
- sip_debug_none = 0,
- sip_debug_config = 1,
- sip_debug_console = 2,
-};
-
-/*! \brief T38 States for a call */
-enum t38state {
- T38_DISABLED = 0, /*!< Not enabled */
- T38_LOCAL_REINVITE, /*!< Offered from local - REINVITE */
- T38_PEER_REINVITE, /*!< Offered from peer - REINVITE */
- T38_ENABLED, /*!< Negotiated (enabled) */
- T38_REJECTED /*!< Refused */
-};
-
-/*! \brief Parameters to know status of transfer */
-enum referstatus {
- REFER_IDLE, /*!< No REFER is in progress */
- REFER_SENT, /*!< Sent REFER to transferee */
- REFER_RECEIVED, /*!< Received REFER from transferrer */
- REFER_CONFIRMED, /*!< Refer confirmed with a 100 TRYING (unused) */
- REFER_ACCEPTED, /*!< Accepted by transferee */
- REFER_RINGING, /*!< Target Ringing */
- REFER_200OK, /*!< Answered by transfer target */
- REFER_FAILED, /*!< REFER declined - go on */
- REFER_NOAUTH /*!< We had no auth for REFER */
-};
-
-enum sip_peer_type {
- SIP_TYPE_PEER = (1 << 0),
- SIP_TYPE_USER = (1 << 1),
-};
-
-enum t38_action_flag {
- SDP_T38_NONE = 0, /*!< Do not modify T38 information at all */
- SDP_T38_INITIATE, /*!< Remote side has requested T38 with us */
- SDP_T38_ACCEPT, /*!< Remote side accepted our T38 request */
-};
-
-enum sip_tcptls_alert {
- TCPTLS_ALERT_DATA, /*!< \brief There is new data to be sent out */
- TCPTLS_ALERT_STOP, /*!< \brief A request to stop the tcp_handler thread */
-};
-
-enum digest_keys {
- K_RESP,
- K_URI,
- K_USER,
- K_NONCE,
- K_LAST
-};
-
-/*----------------------------------------------------------*/
-/*---- STRUCTS ----*/
-/*----------------------------------------------------------*/
-
-/*! \brief definition of a sip proxy server
- *
- * For outbound proxies, a sip_peer will contain a reference to a
- * dynamically allocated instance of a sip_proxy. A sip_pvt may also
- * contain a reference to a peer's outboundproxy, or it may contain
- * a reference to the sip_cfg.outboundproxy.
- */
-struct sip_proxy {
- char name[MAXHOSTNAMELEN]; /*!< DNS name of domain/host or IP */
- struct ast_sockaddr ip; /*!< Currently used IP address and port */
- int port;
- time_t last_dnsupdate; /*!< When this was resolved */
- enum ast_transport transport;
- int force; /*!< If it's an outbound proxy, Force use of this outbound proxy for all outbound requests */
- /* Room for a SRV record chain based on the name */
-};
-
-/*! \brief argument for the 'show channels|subscriptions' callback. */
-struct __show_chan_arg {
- int fd;
- int subscriptions;
- int numchans; /* return value */
-};
-
-/*! \name GlobalSettings
- Global settings apply to the channel (often settings you can change in the general section
- of sip.conf
- @{
- */
-/*! \brief a place to store all global settings for the sip channel driver
-
- These are settings that will be possibly to apply on a group level later on.
- \note Do not add settings that only apply to the channel itself and can't
- be applied to devices (trunks, services, phones)
-*/
-struct sip_settings {
- int peer_rtupdate; /*!< G: Update database with registration data for peer? */
- int rtsave_sysname; /*!< G: Save system name at registration? */
- int rtsave_path; /*!< G: Save path header on registration */
- int ignore_regexpire; /*!< G: Ignore expiration of peer */
- int rtautoclear; /*!< Realtime ?? */
- int directrtpsetup; /*!< Enable support for Direct RTP setup (no re-invites) */
- int pedanticsipchecking; /*!< Extra checking ? Default off */
- enum autocreatepeer_mode autocreatepeer; /*!< Auto creation of peers at registration? Default off. */
- int srvlookup; /*!< SRV Lookup on or off. Default is on */
- int allowguest; /*!< allow unauthenticated peers to connect? */
- int alwaysauthreject; /*!< Send 401 Unauthorized for all failing requests */
- int auth_options_requests; /*!< Authenticate OPTIONS requests */
- int auth_message_requests; /*!< Authenticate MESSAGE requests */
- int accept_outofcall_message; /*!< Accept MESSAGE outside of a call */
- int compactheaders; /*!< send compact sip headers */
- int allow_external_domains; /*!< Accept calls to external SIP domains? */
- int regextenonqualify; /*!< Whether to add/remove regexten when qualifying peers */
- int legacy_useroption_parsing; /*!< Whether to strip useroptions in URI via semicolons */
- int send_diversion; /*!< Whether to Send SIP Diversion headers */
- int matchexternaddrlocally; /*!< Match externaddr/externhost setting against localnet setting */
- char regcontext[AST_MAX_CONTEXT]; /*!< Context for auto-extensions */
- char messagecontext[AST_MAX_CONTEXT]; /*!< Default context for out of dialog msgs. */
- unsigned int disallowed_methods; /*!< methods that we should never try to use */
- int notifyringing; /*!< Send notifications on ringing */
- int notifyhold; /*!< Send notifications on hold */
- enum notifycid_setting notifycid; /*!< Send CID with ringing notifications */
- enum transfermodes allowtransfer; /*!< SIP Refer restriction scheme */
- int allowsubscribe; /*!< Flag for disabling ALL subscriptions, this is FALSE only if all peers are FALSE
- the global setting is in globals_flags[1] */
- char realm[MAXHOSTNAMELEN]; /*!< Default realm */
- int domainsasrealm; /*!< Use domains lists as realms */
- struct sip_proxy outboundproxy; /*!< Outbound proxy */
- char default_context[AST_MAX_CONTEXT];
- char default_subscribecontext[AST_MAX_CONTEXT];
- char default_record_on_feature[AST_FEATURE_MAX_LEN];
- char default_record_off_feature[AST_FEATURE_MAX_LEN];
- struct ast_acl_list *contact_acl; /*! \brief Global list of addresses dynamic peers are not allowed to use */
- struct ast_format_cap *caps; /*!< Supported codecs */
- int tcp_enabled;
- int default_max_forwards; /*!< Default max forwards (SIP Anti-loop) */
- int websocket_write_timeout; /*!< Socket write timeout for websocket transports, in ms */
- int websocket_enabled; /*!< Are websockets enabled? */
-};
-
-/*! @} */
-
-struct ast_websocket;
-
-/*! \brief The SIP socket definition */
-struct sip_socket {
- enum ast_transport type; /*!< UDP, TCP or TLS */
- int fd; /*!< Filed descriptor, the actual socket */
- uint16_t unused; /* since 1.6.2, retained not to change order/size of struct */
- struct ast_tcptls_session_instance *tcptls_session; /* If tcp or tls, a socket manager */
- struct ast_websocket *ws_session; /*! If ws or wss, a WebSocket session */
-};
-
-/*! \brief sip_request: The data grabbed from the UDP socket
- *
- * \verbatim
- * Incoming messages: we first store the data from the socket in data[],
- * adding a trailing \0 to make string parsing routines happy.
- * Then call parse_request() and req.method = find_sip_method();
- * to initialize the other fields. The \r\n at the end of each line is
- * replaced by \0, so that data[] is not a conforming SIP message anymore.
- * After this processing, rlpart1 is set to non-NULL to remember
- * that we can run get_header() on this kind of packet.
- *
- * parse_request() splits the first line as follows:
- * Requests have in the first line method uri SIP/2.0
- * rlpart1 = method; rlpart2 = uri;
- * Responses have in the first line SIP/2.0 NNN description
- * rlpart1 = SIP/2.0; rlpart2 = NNN + description;
- *
- * For outgoing packets, we initialize the fields with init_req() or init_resp()
- * (which fills the first line to "METHOD uri SIP/2.0" or "SIP/2.0 code text"),
- * and then fill the rest with add_header() and add_line().
- * The \r\n at the end of the line are still there, so the get_header()
- * and similar functions don't work on these packets.
- * \endverbatim
- */
-struct sip_request {
- ptrdiff_t rlpart1; /*!< Offset of the SIP Method Name or "SIP/2.0" protocol version */
- ptrdiff_t rlpart2; /*!< Offset of the Request URI or Response Status */
- int headers; /*!< # of SIP Headers */
- int method; /*!< Method of this request */
- int lines; /*!< Body Content */
- unsigned int sdp_start; /*!< the line number where the SDP begins */
- unsigned int sdp_count; /*!< the number of lines of SDP */
- char debug; /*!< print extra debugging if non zero */
- char has_to_tag; /*!< non-zero if packet has To: tag */
- char ignore; /*!< if non-zero This is a re-transmit, ignore it */
- char authenticated; /*!< non-zero if this request was authenticated */
- ptrdiff_t header[SIP_MAX_HEADERS]; /*!< Array of offsets into the request string of each SIP header*/
- ptrdiff_t line[SIP_MAX_LINES]; /*!< Array of offsets into the request string of each SDP line*/
- struct ast_str *data;
- struct ast_str *content;
- /* XXX Do we need to unref socket.ser when the request goes away? */
- struct sip_socket socket; /*!< The socket used for this request */
- AST_LIST_ENTRY(sip_request) next;
- unsigned int reqsipoptions; /*!< Items needed for Required header in responses */
-};
-
-/*! \brief given a sip_request and an offset, return the char * that resides there
- *
- * It used to be that rlpart1, rlpart2, and the header and line arrays were character
- * pointers. They are now offsets into the ast_str portion of the sip_request structure.
- * To avoid adding a bunch of redundant pointer arithmetic to the code, this macro is
- * provided to retrieve the string at a particular offset within the request's buffer
- */
-#define REQ_OFFSET_TO_STR(req,offset) (ast_str_buffer((req)->data) + ((req)->offset))
-
-/*! \brief Parameters to the transmit_invite function */
-struct sip_invite_param {
- int addsipheaders; /*!< Add extra SIP headers */
- const char *uri_options; /*!< URI options to add to the URI */
- const char *vxml_url; /*!< VXML url for Cisco phones */
- char *auth; /*!< Authentication */
- char *authheader; /*!< Auth header */
- enum sip_auth_type auth_type; /*!< Authentication type */
- const char *replaces; /*!< Replaces header for call transfers */
- int transfer; /*!< Flag - is this Invite part of a SIP transfer? (invite/replaces) */
- struct sip_proxy *outboundproxy; /*!< Outbound proxy URI */
-};
-
-/*! \brief Structure to store Via information */
-struct sip_via {
- char *via;
- const char *protocol;
- const char *sent_by;
- const char *branch;
- const char *maddr;
- unsigned int port;
- unsigned char ttl;
-};
-
-/*! \brief Domain data structure.
- \note In the future, we will connect this to a configuration tree specific
- for this domain
-*/
-struct domain {
- char domain[MAXHOSTNAMELEN]; /*!< SIP domain we are responsible for */
- char context[AST_MAX_EXTENSION]; /*!< Incoming context for this domain */
- enum domain_mode mode; /*!< How did we find this domain? */
- AST_LIST_ENTRY(domain) list; /*!< List mechanics */
-};
-
-/*! \brief sip_history: Structure for saving transactions within a SIP dialog */
-struct sip_history {
- AST_LIST_ENTRY(sip_history) list;
- char event[0]; /* actually more, depending on needs */
-};
-
-/*! \brief sip_auth: Credentials for authentication to other SIP services */
-struct sip_auth {
- AST_LIST_ENTRY(sip_auth) node;
- char realm[AST_MAX_EXTENSION]; /*!< Realm in which these credentials are valid */
- char username[256]; /*!< Username */
- char secret[256]; /*!< Secret */
- char md5secret[256]; /*!< MD5Secret */
-};
-
-/*! \brief Container of SIP authentication credentials. */
-struct sip_auth_container {
- AST_LIST_HEAD_NOLOCK(, sip_auth) list;
-};
-
-/*! \brief T.38 channel settings (at some point we need to make this alloc'ed */
-struct t38properties {
- enum t38state state; /*!< T.38 state */
- struct ast_control_t38_parameters our_parms;
- struct ast_control_t38_parameters their_parms;
-};
-
-/*! \brief generic struct to map between strings and integers.
- * Fill it with x-s pairs, terminate with an entry with s = NULL;
- * Then you can call map_x_s(...) to map an integer to a string,
- * and map_s_x() for the string -> integer mapping.
- */
-struct _map_x_s {
- int x;
- const char *s;
-};
-
-/*! \brief Structure to handle SIP transfers. Dynamically allocated when needed */
-struct sip_refer {
- AST_DECLARE_STRING_FIELDS(
- AST_STRING_FIELD(refer_to); /*!< Place to store REFER-TO extension */
- AST_STRING_FIELD(refer_to_domain); /*!< Place to store REFER-TO domain */
- AST_STRING_FIELD(refer_to_urioption); /*!< Place to store REFER-TO uri options */
- AST_STRING_FIELD(refer_to_context); /*!< Place to store REFER-TO context */
- AST_STRING_FIELD(referred_by); /*!< Place to store REFERRED-BY extension */
- AST_STRING_FIELD(refer_contact); /*!< Place to store Contact info from a REFER extension */
- AST_STRING_FIELD(replaces_callid); /*!< Replace info: callid */
- AST_STRING_FIELD(replaces_callid_totag); /*!< Replace info: to-tag */
- AST_STRING_FIELD(replaces_callid_fromtag); /*!< Replace info: from-tag */
- );
- int attendedtransfer; /*!< Attended or blind transfer? */
- int localtransfer; /*!< Transfer to local domain? */
- enum referstatus status; /*!< REFER status */
-};
-
-/*! \brief Struct to handle custom SIP notify requests. Dynamically allocated when needed */
-struct sip_notify {
- struct ast_variable *headers;
- struct ast_str *content;
-};
-
-/*! \brief Structure that encapsulates all attributes related to running
- * SIP Session-Timers feature on a per dialog basis.
- */
-struct sip_st_dlg {
- int st_active; /*!< Session-Timers on/off */
- int st_interval; /*!< Session-Timers negotiated session refresh interval */
- enum st_refresher st_ref; /*!< Session-Timers cached refresher */
- int st_schedid; /*!< Session-Timers ast_sched scheduler id */
- int st_active_peer_ua; /*!< Session-Timers on/off in peer UA */
- int st_cached_min_se; /*!< Session-Timers cached Min-SE */
- int st_cached_max_se; /*!< Session-Timers cached Session-Expires */
- enum st_mode st_cached_mode; /*!< Session-Timers cached M.O. */
- enum st_refresher st_cached_ref; /*!< Session-Timers session refresher */
-};
-
-
-/*! \brief Structure that encapsulates all attributes related to configuration
- * of SIP Session-Timers feature on a per user/peer basis.
- */
-struct sip_st_cfg {
- enum st_mode st_mode_oper; /*!< Mode of operation for Session-Timers */
- enum st_refresher_param st_ref; /*!< Session-Timer refresher */
- int st_min_se; /*!< Lowest threshold for session refresh interval */
- int st_max_se; /*!< Highest threshold for session refresh interval */
-};
-
-/*! \brief Structure for remembering offered media in an INVITE, to make sure we reply
- to all media streams. */
-struct offered_media {
- enum media_type type; /*!< The type of media that was offered */
- char *decline_m_line; /*!< Used if the media type is unknown/unused or a media stream is declined */
- AST_LIST_ENTRY(offered_media) next;
-};
-
-/*! Additional headers to send with MESSAGE method packet. */
-struct sip_msg_hdr {
- AST_LIST_ENTRY(sip_msg_hdr) next;
- /*! Name of header to stick in MESSAGE */
- const char *name;
- /*! Value of header to stick in MESSAGE */
- const char *value;
- /*! The name and value strings are stuffed here in that order. */
- char stuff[0];
-};
-
-/*! \brief Structure used for each SIP dialog, ie. a call, a registration, a subscribe.
- * Created and initialized by sip_alloc(), the descriptor goes into the list of
- * descriptors (dialoglist).
- */
-struct sip_pvt {
- struct sip_pvt *next; /*!< Next dialog in chain */
- enum invitestates invitestate; /*!< Track state of SIP_INVITEs */
- ast_callid logger_callid; /*!< Identifier for call used in log messages */
- int method; /*!< SIP method that opened this dialog */
- AST_DECLARE_STRING_FIELDS(
- AST_STRING_FIELD(callid); /*!< Global CallID */
- AST_STRING_FIELD(initviabranch); /*!< The branch ID from the topmost Via header in the initial request */
- AST_STRING_FIELD(initviasentby); /*!< The sent-by from the topmost Via header in the initial request */
- AST_STRING_FIELD(accountcode); /*!< Account code */
- AST_STRING_FIELD(realm); /*!< Authorization realm */
- AST_STRING_FIELD(nonce); /*!< Authorization nonce */
- AST_STRING_FIELD(opaque); /*!< Opaque nonsense */
- AST_STRING_FIELD(qop); /*!< Quality of Protection, since SIP wasn't complicated enough yet. */
- AST_STRING_FIELD(domain); /*!< Authorization domain */
- AST_STRING_FIELD(from); /*!< The From: header */
- AST_STRING_FIELD(useragent); /*!< User agent in SIP request */
- AST_STRING_FIELD(exten); /*!< Extension where to start */
- AST_STRING_FIELD(context); /*!< Context for this call */
- AST_STRING_FIELD(messagecontext); /*!< Default context for outofcall messages. */
- AST_STRING_FIELD(subscribecontext); /*!< Subscribecontext */
- AST_STRING_FIELD(subscribeuri); /*!< Subscribecontext */
- AST_STRING_FIELD(fromdomain); /*!< Domain to show in the from field */
- AST_STRING_FIELD(fromuser); /*!< User to show in the user field */
- AST_STRING_FIELD(fromname); /*!< Name to show in the user field */
- AST_STRING_FIELD(tohost); /*!< Host we should put in the "to" field */
- AST_STRING_FIELD(todnid); /*!< DNID of this call (overrides host) */
- AST_STRING_FIELD(language); /*!< Default language for this call */
- AST_STRING_FIELD(mohinterpret); /*!< MOH class to use when put on hold */
- AST_STRING_FIELD(mohsuggest); /*!< MOH class to suggest when putting a peer on hold */
- AST_STRING_FIELD(rdnis); /*!< Referring DNIS */
- AST_STRING_FIELD(redircause); /*!< Referring cause */
- AST_STRING_FIELD(theirtag); /*!< Their tag */
- AST_STRING_FIELD(theirprovtag); /*!< Provisional their tag, used when evaluating responses to invites */
- AST_STRING_FIELD(tag); /*!< Our tag for this session */
- AST_STRING_FIELD(username); /*!< [user] name */
- AST_STRING_FIELD(peername); /*!< [peer] name, not set if [user] */
- AST_STRING_FIELD(authname); /*!< Who we use for authentication */
- AST_STRING_FIELD(uri); /*!< Original requested URI */
- AST_STRING_FIELD(okcontacturi); /*!< URI from the 200 OK on INVITE */
- AST_STRING_FIELD(peersecret); /*!< Password */
- AST_STRING_FIELD(peermd5secret);
- AST_STRING_FIELD(cid_num); /*!< Caller*ID number */
- AST_STRING_FIELD(cid_name); /*!< Caller*ID name */
- AST_STRING_FIELD(cid_tag); /*!< Caller*ID tag */
- AST_STRING_FIELD(mwi_from); /*!< Name to place in the From header in outgoing NOTIFY requests */
- AST_STRING_FIELD(fullcontact); /*!< The Contact: that the UA registers with us */
- /* we only store the part in in this field. */
- AST_STRING_FIELD(our_contact); /*!< Our contact header */
- AST_STRING_FIELD(url); /*!< URL to be sent with next message to peer */
- AST_STRING_FIELD(parkinglot); /*!< Parkinglot */
- AST_STRING_FIELD(engine); /*!< RTP engine to use */
- AST_STRING_FIELD(dialstring); /*!< The dialstring used to call this SIP endpoint */
- AST_STRING_FIELD(last_presence_subtype); /*!< The last presence subtype sent for a subscription. */
- AST_STRING_FIELD(last_presence_message); /*!< The last presence message for a subscription */
- AST_STRING_FIELD(msg_body); /*!< Text for a MESSAGE body */
- AST_STRING_FIELD(tel_phone_context); /*!< The phone-context portion of a TEL URI */
- AST_STRING_FIELD(sessionunique_remote); /*!< Remote UA's SDP Session unique parts */
- );
- char via[128]; /*!< Via: header */
- int maxforwards; /*!< SIP Loop prevention */
- struct sip_socket socket; /*!< The socket used for this dialog */
- uint32_t ocseq; /*!< Current outgoing seqno */
- uint32_t icseq; /*!< Current incoming seqno */
- uint32_t init_icseq; /*!< Initial incoming seqno from first request */
- ast_group_t callgroup; /*!< Call group */
- ast_group_t pickupgroup; /*!< Pickup group */
- struct ast_namedgroups *named_callgroups; /*!< Named call group */
- struct ast_namedgroups *named_pickupgroups; /*!< Named pickup group */
- uint32_t lastinvite; /*!< Last seqno of invite */
- struct ast_flags flags[3]; /*!< SIP_ flags */
-
- /* boolean flags that don't belong in flags */
- unsigned short do_history:1; /*!< Set if we want to record history */
- unsigned short alreadygone:1; /*!< the peer has sent a message indicating termination of the dialog */
- unsigned short needdestroy:1; /*!< this dialog needs to be destroyed by the monitor thread */
- unsigned short final_destruction_scheduled:1; /*!< final dialog destruction is scheduled. Keep dialog
- * around until then to handle retransmits. */
- unsigned short outgoing_call:1; /*!< this is an outgoing call */
- unsigned short answered_elsewhere:1; /*!< This call is cancelled due to answer on another channel */
- unsigned short novideo:1; /*!< Didn't get video in invite, don't offer */
- unsigned short notext:1; /*!< Text not supported (?) */
- unsigned short session_modify:1; /*!< Session modification request true/false */
- unsigned short route_persistent:1; /*!< Is this the "real" route? */
- unsigned short autoframing:1; /*!< Whether to use our local configuration for frame sizes (off)
- * or respect the other endpoint's request for frame sizes (on)
- * for incoming calls
- */
- unsigned short req_secure_signaling:1;/*!< Whether we are required to have secure signaling or not */
- unsigned short natdetected:1; /*!< Whether we detected a NAT when processing the Via */
- int timer_t1; /*!< SIP timer T1, ms rtt */
- int timer_b; /*!< SIP timer B, ms */
- unsigned int sipoptions; /*!< Supported SIP options on the other end */
- unsigned int reqsipoptions; /*!< Required SIP options on the other end */
- struct ast_format_cap *caps; /*!< Special capability (codec) */
- struct ast_format_cap *jointcaps; /*!< Supported capability at both ends (codecs) */
- struct ast_format_cap *peercaps; /*!< Supported peer capability */
- struct ast_format_cap *redircaps; /*!< Redirect codecs */
- struct ast_format_cap *prefcaps; /*!< Preferred codec (outbound only) */
- int noncodeccapability; /*!< DTMF RFC2833 telephony-event */
- int jointnoncodeccapability; /*!< Joint Non codec capability */
- int maxcallbitrate; /*!< Maximum Call Bitrate for Video Calls */
- int t38_maxdatagram; /*!< T.38 FaxMaxDatagram override */
- int request_queue_sched_id; /*!< Scheduler ID of any scheduled action to process queued requests */
- int provisional_keepalive_sched_id; /*!< Scheduler ID for provisional responses that need to be sent out to avoid cancellation */
- const char *last_provisional; /*!< The last successfully transmitted provisional response message */
- int authtries; /*!< Times we've tried to authenticate */
- struct sip_proxy *outboundproxy; /*!< Outbound proxy for this dialog. Use ref_proxy to set this instead of setting it directly*/
- struct t38properties t38; /*!< T38 settings */
- struct ast_sockaddr udptlredirip; /*!< Where our T.38 UDPTL should be going if not to us */
- struct ast_udptl *udptl; /*!< T.38 UDPTL session */
- char zone[MAX_TONEZONE_COUNTRY]; /*!< Default tone zone for channels created by this dialog */
- int callingpres; /*!< Calling presentation */
- int expiry; /*!< How long we take to expire */
- int sessionversion; /*!< SDP Session Version */
- int sessionid; /*!< SDP Session ID */
- long branch; /*!< The branch identifier of this session */
- long invite_branch; /*!< The branch used when we sent the initial INVITE */
- int64_t sessionversion_remote; /*!< Remote UA's SDP Session Version */
- unsigned int portinuri:1; /*!< Non zero if a port has been specified, will also disable srv lookups */
- struct ast_sockaddr sa; /*!< Our peer */
- struct ast_sockaddr redirip; /*!< Where our RTP should be going if not to us */
- struct ast_sockaddr vredirip; /*!< Where our Video RTP should be going if not to us */
- struct ast_sockaddr tredirip; /*!< Where our Text RTP should be going if not to us */
- time_t lastrtprx; /*!< Last RTP received */
- time_t lastrtptx; /*!< Last RTP sent */
- int rtptimeout; /*!< RTP timeout time */
- int rtpholdtimeout; /*!< RTP timeout time on hold*/
- int rtpkeepalive; /*!< RTP send packets for keepalive */
- struct ast_acl_list *directmediaacl; /*!< Which IPs are allowed to interchange direct media with this peer - copied from sip_peer */
- struct ast_sockaddr recv; /*!< Received as */
- struct ast_sockaddr ourip; /*!< Our IP (as seen from the outside) */
- enum transfermodes allowtransfer; /*!< REFER: restriction scheme */
- struct ast_channel *owner; /*!< Who owns us (if we have an owner) */
- struct sip_route route; /*!< List of routing steps (fm Record-Route) */
- struct sip_notify *notify; /*!< Custom notify type */
- struct sip_auth_container *peerauth;/*!< Realm authentication credentials */
- int noncecount; /*!< Nonce-count */
- unsigned int stalenonce:1; /*!< Marks the current nonce as responded too */
- unsigned int ongoing_reinvite:1; /*!< There is a reinvite in progress that might need to be cleaned up */
- char lastmsg[256]; /*!< Last Message sent/received */
- int amaflags; /*!< AMA Flags */
- uint32_t pendinginvite; /*!< Any pending INVITE or state NOTIFY (in subscribe pvt's) ? (seqno of this) */
- uint32_t glareinvite; /*!< A invite received while a pending invite is already present is stored here. Its seqno is the
- value. Since this glare invite's seqno is not the same as the pending invite's, it must be
- held in order to properly process acknowledgements for our 491 response. */
- struct sip_request initreq; /*!< Latest request that opened a new transaction
- within this dialog.
- NOT the request that opened the dialog */
-
- int initid; /*!< Auto-congest ID if appropriate (scheduler) */
- int waitid; /*!< Wait ID for scheduler after 491 or other delays */
- int reinviteid; /*!< Reinvite in case of provisional, but no final response */
- int autokillid; /*!< Auto-kill ID (scheduler) */
- int t38id; /*!< T.38 Response ID */
- struct sip_refer *refer; /*!< REFER: SIP transfer data structure */
- enum subscriptiontype subscribed; /*!< SUBSCRIBE: Is this dialog a subscription? */
- int stateid; /*!< SUBSCRIBE: ID for devicestate subscriptions */
- int laststate; /*!< SUBSCRIBE: Last known extension state */
- struct ao2_container *last_device_state_info; /*!< SUBSCRIBE: last known extended extension state (take care of refs)*/
- struct timeval last_ringing_channel_time; /*!< SUBSCRIBE: channel timestamp of the channel which caused the last early-state notification */
- int last_presence_state; /*!< SUBSCRIBE: Last known presence state */
- uint32_t dialogver; /*!< SUBSCRIBE: Version for subscription dialog-info */
-
- struct ast_dsp *dsp; /*!< Inband DTMF or Fax CNG tone Detection dsp */
-
- struct sip_peer *relatedpeer; /*!< If this dialog is related to a peer, which one
- Used in peerpoke, mwi subscriptions */
- struct sip_registry *registry; /*!< If this is a REGISTER dialog, to which registry */
- struct ast_rtp_instance *rtp; /*!< RTP Session */
- struct ast_rtp_instance *vrtp; /*!< Video RTP session */
- struct ast_rtp_instance *trtp; /*!< Text RTP session */
- struct sip_pkt *packets; /*!< Packets scheduled for re-transmission */
- struct sip_history_head *history; /*!< History of this SIP dialog */
- size_t history_entries; /*!< Number of entires in the history */
- struct ast_variable *chanvars; /*!< Channel variables to set for inbound call */
- AST_LIST_HEAD_NOLOCK(, sip_msg_hdr) msg_headers; /*!< Additional MESSAGE headers to send. */
- AST_LIST_HEAD_NOLOCK(request_queue, sip_request) request_queue; /*!< Requests that arrived but could not be processed immediately */
- struct sip_invite_param *options; /*!< Options for INVITE */
- struct sip_st_dlg *stimer; /*!< SIP Session-Timers */
- struct ast_sdp_srtp *srtp; /*!< Structure to hold Secure RTP session data for audio */
- struct ast_sdp_srtp *vsrtp; /*!< Structure to hold Secure RTP session data for video */
- struct ast_sdp_srtp *tsrtp; /*!< Structure to hold Secure RTP session data for text */
-
- int red; /*!< T.140 RTP Redundancy */
- int hangupcause; /*!< Storage of hangupcause copied from our owner before we disconnect from the AST channel (only used at hangup) */
-
- struct sip_subscription_mwi *mwi; /*!< If this is a subscription MWI dialog, to which subscription */
- /*! The SIP methods supported by this peer. We get this information from the Allow header of the first
- * message we receive from an endpoint during a dialog.
- */
- unsigned int allowed_methods;
- /*! Some peers are not trustworthy with their Allow headers, and so we need to override their wicked
- * ways through configuration. This is a copy of the peer's disallowed_methods, so that we can apply them
- * to the sip_pvt at various stages of dialog establishment
- */
- unsigned int disallowed_methods;
- /*! When receiving an SDP offer, it is important to take note of what media types were offered.
- * By doing this, even if we don't want to answer a particular media stream with something meaningful, we can
- * still put an m= line in our answer with the port set to 0.
- *
- * The reason for the length being 4 (OFFERED_MEDIA_COUNT) is that in this branch of Asterisk, the only media types supported are
- * image, audio, text, and video. Therefore we need to keep track of which types of media were offered.
- * Note that secure RTP defines new types of SDP media.
- *
- * If we wanted to be 100% correct, we would keep a list of all media streams offered. That way we could respond
- * even to unknown media types, and we could respond to multiple streams of the same type. Such large-scale changes
- * are not a good idea for released branches, though, so we're compromising by just making sure that for the common cases:
- * audio and video, audio and T.38, and audio and text, we give the appropriate response to both media streams.
- *
- * The large-scale changes would be a good idea for implementing during an SDP rewrite.
- */
- AST_LIST_HEAD_NOLOCK(, offered_media) offered_media;
- struct ast_cc_config_params *cc_params;
- struct sip_epa_entry *epa_entry;
- int fromdomainport; /*!< Domain port to show in from field */
-
- struct ast_rtp_dtls_cfg dtls_cfg;
-};
-
-/*! \brief sip packet - raw format for outbound packets that are sent or scheduled for transmission
- * Packets are linked in a list, whose head is in the struct sip_pvt they belong to.
- * Each packet holds a reference to the parent struct sip_pvt.
- * This structure is allocated in __sip_reliable_xmit() and only for packets that
- * require retransmissions.
- */
-struct sip_pkt {
- struct sip_pkt *next; /*!< Next packet in linked list */
- int retrans; /*!< Retransmission number */
- int method; /*!< SIP method for this packet */
- uint32_t seqno; /*!< Sequence number */
- char is_resp; /*!< 1 if this is a response packet (e.g. 200 OK), 0 if it is a request */
- char is_fatal; /*!< non-zero if there is a fatal error */
- int response_code; /*!< If this is a response, the response code */
- struct sip_pvt *owner; /*!< Owner AST call */
- int retransid; /*!< Retransmission ID */
- int timer_a; /*!< SIP timer A, retransmission timer */
- int timer_t1; /*!< SIP Timer T1, estimated RTT or 500 ms */
- struct timeval time_sent; /*!< When pkt was sent */
- int64_t retrans_stop_time; /*!< Time in ms after 'now' that retransmission must stop */
- int retrans_stop; /*!< Timeout is reached, stop retransmission */
- struct ast_str *data;
-};
-
-enum sip_mailbox_status {
- SIP_MAILBOX_STATUS_UNKNOWN = 0,
- SIP_MAILBOX_STATUS_EXISTING,
- SIP_MAILBOX_STATUS_NEW,
-};
-
-/*!
- * \brief A peer's mailbox
- *
- * We could use STRINGFIELDS here, but for only one string, its
- * too much effort ...
- */
-struct sip_mailbox {
- /*! Associated MWI subscription */
- struct ast_mwi_subscriber *event_sub;
- AST_LIST_ENTRY(sip_mailbox) entry;
- struct sip_peer *peer;
- enum sip_mailbox_status status;
- char id[1];
-};
-
-/*! \brief Structure for SIP peer data, we place calls to peers if registered or fixed IP address (host)
-*/
-/* XXX field 'name' must be first otherwise sip_addrcmp() will fail, as will astobj2 hashing of the structure */
-struct sip_peer {
- char name[80]; /*!< the unique name of this object */
- AST_DECLARE_STRING_FIELDS(
- AST_STRING_FIELD(secret); /*!< Password for inbound auth */
- AST_STRING_FIELD(md5secret); /*!< Password in MD5 */
- AST_STRING_FIELD(description); /*!< Description of this peer */
- AST_STRING_FIELD(remotesecret); /*!< Remote secret (trunks, remote devices) */
- AST_STRING_FIELD(context); /*!< Default context for incoming calls */
- AST_STRING_FIELD(messagecontext); /*!< Default context for outofcall messages. */
- AST_STRING_FIELD(subscribecontext); /*!< Default context for subscriptions */
- AST_STRING_FIELD(username); /*!< Temporary username until registration */
- AST_STRING_FIELD(accountcode); /*!< Account code */
- AST_STRING_FIELD(tohost); /*!< If not dynamic, IP address */
- AST_STRING_FIELD(regexten); /*!< Extension to register (if regcontext is used) */
- AST_STRING_FIELD(fromuser); /*!< From: user when calling this peer */
- AST_STRING_FIELD(fromdomain); /*!< From: domain when calling this peer */
- AST_STRING_FIELD(fullcontact); /*!< Contact registered with us (not in sip.conf) */
- AST_STRING_FIELD(cid_num); /*!< Caller ID num */
- AST_STRING_FIELD(cid_name); /*!< Caller ID name */
- AST_STRING_FIELD(cid_tag); /*!< Caller ID tag */
- AST_STRING_FIELD(vmexten); /*!< Dialplan extension for MWI notify message*/
- AST_STRING_FIELD(language); /*!< Default language for prompts */
- AST_STRING_FIELD(mohinterpret); /*!< Music on Hold class */
- AST_STRING_FIELD(mohsuggest); /*!< Music on Hold class */
- AST_STRING_FIELD(parkinglot); /*!< Parkinglot */
- AST_STRING_FIELD(useragent); /*!< User agent in SIP request (saved from registration) */
- AST_STRING_FIELD(mwi_from); /*!< Name to place in From header for outgoing NOTIFY requests */
- AST_STRING_FIELD(engine); /*!< RTP Engine to use */
- AST_STRING_FIELD(unsolicited_mailbox); /*!< Mailbox to store received unsolicited MWI NOTIFY messages information in */
- AST_STRING_FIELD(zone); /*!< Tonezone for this device */
- AST_STRING_FIELD(record_on_feature); /*!< Feature to use when receiving INFO with record: on during a call */
- AST_STRING_FIELD(record_off_feature); /*!< Feature to use when receiving INFO with record: off during a call */
- AST_STRING_FIELD(callback); /*!< Callback extension */
- );
- struct sip_socket socket; /*!< Socket used for this peer */
- enum ast_transport default_outbound_transport; /*!< Peer Registration may change the default outbound transport.
- If register expires, default should be reset. to this value */
- /* things that don't belong in flags */
- unsigned short transports:5; /*!< Transports (enum ast_transport) that are acceptable for this peer */
- unsigned short is_realtime:1; /*!< this is a 'realtime' peer */
- unsigned short rt_fromcontact:1;/*!< copy fromcontact from realtime */
- unsigned short host_dynamic:1; /*!< Dynamic Peers register with Asterisk */
- unsigned short selfdestruct:1; /*!< Automatic peers need to destruct themselves */
- unsigned short the_mark:1; /*!< That which bears the_mark should be deleted! */
- unsigned short autoframing:1; /*!< Whether to use our local configuration for frame sizes (off)
- * or respect the other endpoint's request for frame sizes (on)
- * for incoming calls
- */
- unsigned short deprecated_username:1; /*!< If it's a realtime peer, are they using the deprecated "username" instead of "defaultuser" */
- struct sip_auth_container *auth;/*!< Realm authentication credentials */
- int amaflags; /*!< AMA Flags (for billing) */
- int callingpres; /*!< Calling id presentation */
- int inuse; /*!< Number of calls in use */
- int ringing; /*!< Number of calls ringing */
- int onhold; /*!< Peer has someone on hold */
- int call_limit; /*!< Limit of concurrent calls */
- unsigned int t38_maxdatagram; /*!< T.38 FaxMaxDatagram override */
- int busy_level; /*!< Level of active channels where we signal busy */
- int maxforwards; /*!< SIP Loop prevention */
- enum transfermodes allowtransfer; /*! SIP Refer restriction scheme */
- int lastmsgssent; /*!< The last known VM message counts (new/old) */
- unsigned int sipoptions; /*!< Supported SIP options */
- struct ast_flags flags[3]; /*!< SIP_ flags */
-
- /*! Mailboxes that this peer cares about */
- AST_LIST_HEAD_NOLOCK(, sip_mailbox) mailboxes;
-
- int maxcallbitrate; /*!< Maximum Bitrate for a video call */
- int expire; /*!< When to expire this peer registration */
- struct ast_format_cap *caps; /*!< Codec capability */
- int rtptimeout; /*!< RTP timeout */
- int rtpholdtimeout; /*!< RTP Hold Timeout */
- int rtpkeepalive; /*!< Send RTP packets for keepalive */
- ast_group_t callgroup; /*!< Call group */
- ast_group_t pickupgroup; /*!< Pickup group */
- struct ast_namedgroups *named_callgroups; /*!< Named call group */
- struct ast_namedgroups *named_pickupgroups; /*!< Named pickup group */
- struct sip_proxy *outboundproxy;/*!< Outbound proxy for this peer */
- struct ast_dnsmgr_entry *dnsmgr;/*!< DNS refresh manager for peer */
- struct ast_sockaddr addr; /*!< IP address of peer */
- unsigned int portinuri:1; /*!< Whether the port should be included in the URI */
- struct sip_pvt *call; /*!< Call pointer */
- int pokeexpire; /*!< Qualification: When to expire poke (qualify= checking) */
- int lastms; /*!< Qualification: How long last response took (in ms), or -1 for no response */
- int maxms; /*!< Qualification: Max ms we will accept for the host to be up, 0 to not monitor */
- int qualifyfreq; /*!< Qualification: Qualification: How often to check for the host to be up */
- struct timeval ps; /*!< Qualification: Time for sending SIP OPTION in sip_pke_peer() */
- int keepalive; /*!< Keepalive: How often to send keep alive packet */
- int keepalivesend; /*!< Keepalive: Scheduled item for sending keep alive packet */
- struct ast_sockaddr defaddr; /*!< Default IP address, used until registration */
- struct ast_acl_list *acl; /*!< Access control list */
- struct ast_acl_list *contactacl; /*!< Restrict what IPs are allowed in the Contact header (for registration) */
- struct ast_acl_list *directmediaacl; /*!< Restrict what IPs are allowed to interchange direct media with */
- struct ast_variable *chanvars; /*!< Variables to set for channel created by user */
- struct sip_pvt *mwipvt; /*!< Subscription for MWI */
- struct sip_st_cfg stimer; /*!< SIP Session-Timers */
- int timer_t1; /*!< The maximum T1 value for the peer */
- int timer_b; /*!< The maximum timer B (transaction timeouts) */
- int fromdomainport; /*!< The From: domain port */
- struct sip_route path; /*!< List of out-of-dialog outgoing routing steps (fm Path headers) */
-
- /*XXX Seems like we suddenly have two flags with the same content. Why? To be continued... */
- enum sip_peer_type type; /*!< Distinguish between "user" and "peer" types. This is used solely for CLI and manager commands */
- unsigned int disallowed_methods;
- struct ast_cc_config_params *cc_params;
-
- struct ast_endpoint *endpoint;
-
- struct ast_rtp_dtls_cfg dtls_cfg;
-};
-
-/*!
- * \brief Registrations with other SIP proxies
- *
- * Created by sip_register(), the entry is linked in the 'regl' list,
- * and never deleted (other than at 'sip reload' or module unload times).
- * The entry always has a pending timeout, either waiting for an ACK to
- * the REGISTER message (in which case we have to retransmit the request),
- * or waiting for the next REGISTER message to be sent (either the initial one,
- * or once the previously completed registration one expires).
- * The registration can be in one of many states, though at the moment
- * the handling is a bit mixed.
- */
-struct sip_registry {
- AST_DECLARE_STRING_FIELDS(
- AST_STRING_FIELD(configvalue);/*!< register string from config */
- AST_STRING_FIELD(callid); /*!< Global Call-ID */
- AST_STRING_FIELD(realm); /*!< Authorization realm */
- AST_STRING_FIELD(nonce); /*!< Authorization nonce */
- AST_STRING_FIELD(opaque); /*!< Opaque nonsense */
- AST_STRING_FIELD(qop); /*!< Quality of Protection, since SIP wasn't complicated enough yet. */
- AST_STRING_FIELD(authdomain); /*!< Authorization domain */
- AST_STRING_FIELD(regdomain); /*!< Registration doamin */
- AST_STRING_FIELD(username); /*!< Who we are registering as */
- AST_STRING_FIELD(authuser); /*!< Who we *authenticate* as */
- AST_STRING_FIELD(hostname); /*!< Domain or host we register to */
- AST_STRING_FIELD(secret); /*!< Password in clear text */
- AST_STRING_FIELD(md5secret); /*!< Password in md5 */
- AST_STRING_FIELD(callback); /*!< Contact extension */
- AST_STRING_FIELD(peername); /*!< Peer registering to */
- AST_STRING_FIELD(localtag); /*!< Local tag generated same time as callid */
- );
- enum ast_transport transport; /*!< Transport for this registration UDP, TCP or TLS */
- int portno; /*!< Optional port override */
- int regdomainport; /*!< Port override for domainport */
- int expire; /*!< Sched ID of expiration */
- int configured_expiry; /*!< Configured value to use for the Expires header */
- int expiry; /*!< Negotiated value used for the Expires header */
- int regattempts; /*!< Number of attempts (since the last success) */
- int timeout; /*!< sched id of sip_reg_timeout */
- int refresh; /*!< How often to refresh */
- struct sip_pvt *call; /*!< create a sip_pvt structure for each outbound "registration dialog" in progress */
- enum sipregistrystate regstate; /*!< Registration state (see above) */
- struct timeval regtime; /*!< Last successful registration time */
- int callid_valid; /*!< 0 means we haven't chosen callid for this registry yet. */
- uint32_t ocseq; /*!< Sequence number we got to for REGISTERs for this registry */
- struct ast_dnsmgr_entry *dnsmgr; /*!< DNS refresh manager for register */
- struct ast_sockaddr us; /*!< Who the server thinks we are */
- int noncecount; /*!< Nonce-count */
- char lastmsg[256]; /*!< Last Message sent/received */
-};
-
-struct tcptls_packet {
- AST_LIST_ENTRY(tcptls_packet) entry;
- struct ast_str *data;
- size_t len;
-};
-/*! \brief Definition of a thread that handles a socket */
-struct sip_threadinfo {
- /*! TRUE if the thread needs to kill itself. (The module is being unloaded.) */
- int stop;
- int alert_pipe[2]; /*! Used to alert tcptls thread when packet is ready to be written */
- pthread_t threadid;
- struct ast_tcptls_session_instance *tcptls_session;
- enum ast_transport type; /*!< We keep a copy of the type here so we can display it in the connection list */
- AST_LIST_HEAD_NOLOCK(, tcptls_packet) packet_q;
-};
-
-/*!
- * \brief Definition of an MWI subscription to another server
- */
-struct sip_subscription_mwi {
- AST_DECLARE_STRING_FIELDS(
- AST_STRING_FIELD(username); /*!< Who we are sending the subscription as */
- AST_STRING_FIELD(authuser); /*!< Who we *authenticate* as */
- AST_STRING_FIELD(hostname); /*!< Domain or host we subscribe to */
- AST_STRING_FIELD(secret); /*!< Password in clear text */
- AST_STRING_FIELD(mailbox); /*!< Mailbox store to put MWI into */
- );
- enum ast_transport transport; /*!< Transport to use */
- int portno; /*!< Optional port override */
- int resub; /*!< Sched ID of resubscription */
- unsigned int subscribed:1; /*!< Whether we are currently subscribed or not */
- struct sip_pvt *call; /*!< Outbound subscription dialog */
- struct ast_dnsmgr_entry *dnsmgr; /*!< DNS refresh manager for subscription */
- struct ast_sockaddr us; /*!< Who the server thinks we are */
-};
-
-/*!
- * SIP PUBLISH support!
- * PUBLISH support was added to chan_sip due to its use in the call-completion
- * event package. In order to suspend and unsuspend monitoring of a called party,
- * a PUBLISH message must be sent. Rather than try to hack in PUBLISH transmission
- * and reception solely for the purposes of handling call-completion-related messages,
- * an effort has been made to create a generic framework for handling PUBLISH messages.
- *
- * There are two main components to the effort, the event publication agent (EPA) and
- * the event state compositor (ESC). Both of these terms appear in RFC 3903, and the
- * implementation in Asterisk conforms to the defintions there. An EPA is a UAC that
- * transmits PUBLISH requests. An ESC is a UAS that receives PUBLISH requests and
- * acts appropriately based on the content of those requests.
- *
- * ESC:
- * The main structure in chan_sip is the event_state_compositor. There is an
- * event_state_compositor structure for each event package supported (as of Nov 2009
- * this is only the call-completion package). The structure contains data which is
- * intrinsic to the event package itself, such as the name of the package and a set
- * of callbacks for handling incoming PUBLISH requests. In addition, the
- * event_state_compositor struct contains an ao2_container of sip_esc_entries.
- *
- * A sip_esc_entry corresponds to an entity which has sent a PUBLISH to Asterisk. We are
- * able to match the incoming PUBLISH to a sip_esc_entry using the Sip-If-Match header
- * of the message. Of course, if none is present, then a new sip_esc_entry will be created.
- *
- * Once it is determined what type of PUBLISH request has come in (from RFC 3903, it may
- * be an initial, modify, refresh, or remove), then the event package-specific callbacks
- * may be called. If your event package doesn't need to take any specific action for a
- * specific PUBLISH type, it is perfectly safe to not define the callback at all. The callback
- * only needs to take care of application-specific information. If there is a problem, it is
- * up to the callback to take care of sending an appropriate 4xx or 5xx response code. In such
- * a case, the callback should return -1. This will tell the function that called the handler
- * that an appropriate error response has been sent. If the callback returns 0, however, then
- * the caller of the callback will generate a new entity tag and send a 200 OK response.
- *
- * ESC entries are reference-counted, however as an implementor of a specific event package,
- * this should be transparent, since the reference counts are handled by the general ESC
- * framework.
- *
- * EPA:
- * The event publication agent in chan_sip is structured quite a bit differently than the
- * ESC. With an ESC, an appropriate entry has to be found based on the contents of an incoming
- * PUBLISH message. With an EPA, the application interested in sending the PUBLISH can maintain
- * a reference to the appropriate EPA entry instead. Similarly, when matching a PUBLISH response
- * to an appropriate EPA entry, the sip_pvt can maintain a reference to the corresponding
- * EPA entry. The result of this train of thought is that there is no compelling reason to
- * maintain a container of these entries.
- *
- * Instead, there is only the sip_epa_entry structure. Every sip_epa_entry has an entity tag
- * that it maintains so that subsequent PUBLISH requests will be identifiable by the ESC on
- * the far end. In addition, there is a static_data field which contains information that is
- * common to all sip_epa_entries for a specific event package. This static data includes the
- * name of the event package and callbacks for handling specific responses for outgoing PUBLISHes.
- * Also, there is a field for pointing to instance-specific data. This can include the current
- * published state or other identifying information that is specific to an instance of an EPA
- * entry of a particular event package.
- *
- * When an application wishes to send a PUBLISH request, it simply will call create_epa_entry,
- * followed by transmit_publish in order to send the PUBLISH. That's all that is necessary.
- * Like with ESC entries, sip_epa_entries are reference counted. Unlike ESC entries, though,
- * sip_epa_entries reference counts have to be maintained to some degree by the application making
- * use of the sip_epa_entry. The application will acquire a reference to the EPA entry when it
- * calls create_epa_entry. When the application has finished using the EPA entry (which may not
- * be until after several PUBLISH transactions have taken place) it must use ao2_ref to decrease
- * the reference count by 1.
- */
-
-/*!
- * \brief The states that can be represented in a SIP call-completion PUBLISH
- */
-enum sip_cc_publish_state {
- /*! Closed, i.e. unavailable */
- CC_CLOSED,
- /*! Open, i.e. available */
- CC_OPEN,
-};
-
-/*!
- * \brief The states that can be represented in a SIP call-completion NOTIFY
- */
-enum sip_cc_notify_state {
- /*! Queued, i.e. unavailable */
- CC_QUEUED,
- /*! Ready, i.e. available */
- CC_READY,
-};
-
-/*!
- * \brief The types of PUBLISH messages defined in RFC 3903
- */
-enum sip_publish_type {
- /*!
- * \brief Unknown
- *
- * \details
- * This actually is not defined in RFC 3903. We use this as a constant
- * to indicate that an incoming PUBLISH does not fit into any of the
- * other categories and is thus invalid.
- */
- SIP_PUBLISH_UNKNOWN,
- /*!
- * \brief Initial
- *
- * \details
- * The first PUBLISH sent. This will contain a non-zero Expires header
- * as well as a body that indicates the current state of the endpoint
- * that has sent the message. The initial PUBLISH is the only type
- * of PUBLISH to not contain a Sip-If-Match header in it.
- */
- SIP_PUBLISH_INITIAL,
- /*!
- * \brief Refresh
- *
- * \details
- * Used to keep a published state from expiring. This will contain a
- * non-zero Expires header but no body since its purpose is not to
- * update state.
- */
- SIP_PUBLISH_REFRESH,
- /*!
- * \brief Modify
- *
- * \details
- * Used to change state from its previous value. This will contain
- * a body updating the published state. May or may not contain an
- * Expires header.
- */
- SIP_PUBLISH_MODIFY,
- /*!
- * \brief Remove
- *
- * \details
- * Used to remove published state from an ESC. This will contain
- * an Expires header set to 0 and likely no body.
- */
- SIP_PUBLISH_REMOVE,
-};
-
-/*!
- * Data which is the same for all instances of an EPA for a
- * particular event package
- */
-struct epa_static_data {
- /*! The event type */
- enum subscriptiontype event;
- /*!
- * The name of the event as it would
- * appear in a SIP message
- */
- const char *name;
- /*!
- * The callback called when a 200 OK is received on an outbound PUBLISH
- */
- void (*handle_ok)(struct sip_pvt *, struct sip_request *, struct sip_epa_entry *);
- /*!
- * The callback called when an error response is received on an outbound PUBLISH
- */
- void (*handle_error)(struct sip_pvt *, const int resp, struct sip_request *, struct sip_epa_entry *);
- /*!
- * Destructor to call to clean up instance data
- */
- void (*destructor)(void *instance_data);
-};
-
-/*!
- * \brief backend for an event publication agent
- */
-struct epa_backend {
- const struct epa_static_data *static_data;
- AST_LIST_ENTRY(epa_backend) next;
-};
-
-struct sip_epa_entry {
- /*!
- * When we are going to send a publish, we need to
- * know the type of PUBLISH to send.
- */
- enum sip_publish_type publish_type;
- /*!
- * When we send a PUBLISH, we have to be
- * sure to include the entity tag that we
- * received in the previous response.
- */
- char entity_tag[SIPBUFSIZE];
- /*!
- * The destination to which this EPA should send
- * PUBLISHes. This may be the name of a SIP peer
- * or a hostname.
- */
- char destination[SIPBUFSIZE];
- /*!
- * The body of the most recently-sent PUBLISH message.
- * This is useful for situations such as authentication,
- * in which we must send a message identical to the
- * one previously sent
- */
- char body[SIPBUFSIZE];
- /*!
- * Every event package has some constant data and
- * callbacks that all instances will share. This
- * data resides in this field.
- */
- const struct epa_static_data *static_data;
- /*!
- * In addition to the static data that all instances
- * of sip_epa_entry will have, each instance will
- * require its own instance-specific data.
- */
- void *instance_data;
-};
-
-/*!
- * \brief Instance data for a Call completion EPA entry
- */
-struct cc_epa_entry {
- /*!
- * The core ID of the CC transaction
- * for which this EPA entry belongs. This
- * essentially acts as a unique identifier
- * for the entry and is used in the hash
- * and comparison functions
- */
- int core_id;
- /*!
- * We keep the last known state of the
- * device in question handy in case
- * it needs to be known by a third party.
- * Also, in the case where for some reason
- * we get asked to transmit state that we
- * already sent, we can just ignore the
- * request.
- */
- enum sip_cc_publish_state current_state;
-};
-
-struct event_state_compositor;
-
-/*!
- * \brief common ESC items for all event types
- *
- * The entity_id field serves as a means by which
- * A specific entry may be found.
- */
-struct sip_esc_entry {
- /*!
- * The name of the party who
- * sent us the PUBLISH. This will more
- * than likely correspond to a peer name.
- *
- * This field's utility isn't really that
- * great. It's mainly just a user-recognizable
- * handle that can be printed in debug messages.
- */
- const char *device_name;
- /*!
- * The event package for which this esc_entry
- * exists. Most of the time this isn't really
- * necessary since you'll have easy access to the
- * ESC which contains this entry. However, in
- * some circumstances, we won't have the ESC
- * available.
- */
- const char *event;
- /*!
- * The entity ID used when corresponding
- * with the EPA on the other side. As the
- * ESC, we generate an entity ID for each
- * received PUBLISH and store it in this
- * structure.
- */
- char entity_tag[30];
- /*!
- * The ID for the scheduler. We schedule
- * destruction of a sip_esc_entry when we
- * receive a PUBLISH. The destruction is
- * scheduled for the duration received in
- * the Expires header.
- */
- int sched_id;
- /*!
- * Each ESC entry will be for a specific
- * event type. Those entries will need to
- * carry data which is intrinsic to the
- * ESC entry but which is specific to
- * the event package
- */
- void *event_specific_data;
-};
-
-typedef int (* const esc_publish_callback)(struct sip_pvt *, struct sip_request *, struct event_state_compositor *, struct sip_esc_entry *);
-
-/*!
- * \brief Callbacks for SIP ESCs
- *
- * \details
- * The names of the callbacks are self-explanatory. The
- * corresponding handler is called whenever the specific
- * type of PUBLISH is received.
- */
-struct sip_esc_publish_callbacks {
- const esc_publish_callback initial_handler;
- const esc_publish_callback refresh_handler;
- const esc_publish_callback modify_handler;
- const esc_publish_callback remove_handler;
-};
-
-struct sip_cc_agent_pvt {
- int offer_timer_id;
- /* A copy of the original call's Call-ID.
- * We use this as a search key when attempting
- * to find a particular sip_pvt.
- */
- char original_callid[SIPBUFSIZE];
- /* A copy of the exten called originally.
- * We use this to set the proper extension
- * to dial during the recall since the incoming
- * request URI is one that was generated just
- * for the recall
- */
- char original_exten[SIPBUFSIZE];
- /* A reference to the dialog which we will
- * be sending a NOTIFY on when it comes time
- * to send one
- */
- struct sip_pvt *subscribe_pvt;
- /* When we send a NOTIFY, we include a URI
- * that should be used by the caller when he
- * wishes to send a PUBLISH or INVITE to us.
- * We store that URI here.
- */
- char notify_uri[SIPBUFSIZE];
- /* When we advertise call completion to a caller,
- * we provide a URI for the caller to use when
- * he sends us a SUBSCRIBE. We store it for matching
- * purposes when we receive the SUBSCRIBE from the
- * caller.
- */
- char subscribe_uri[SIPBUFSIZE];
- char is_available;
-};
-
-struct sip_monitor_instance {
- AST_DECLARE_STRING_FIELDS(
- AST_STRING_FIELD(subscribe_uri);
- AST_STRING_FIELD(notify_uri);
- AST_STRING_FIELD(peername);
- AST_STRING_FIELD(device_name);
- );
- int core_id;
- struct sip_pvt *subscription_pvt;
- struct sip_epa_entry *suspension_entry;
-};
-
-/*! \brief List of well-known SIP options. If we get this in a require,
- we should check the list and answer accordingly. */
-static const struct cfsip_options {
- int id; /*!< Bitmap ID */
- int supported; /*!< Supported by Asterisk ? */
- char * const text; /*!< Text id, as in standard */
-} sip_options[] = { /* XXX used in 3 places */
- /* RFC3262: PRACK 100% reliability */
- { SIP_OPT_100REL, NOT_SUPPORTED, "100rel" },
- /* RFC3959: SIP Early session support */
- { SIP_OPT_EARLY_SESSION, NOT_SUPPORTED, "early-session" },
- /* SIMPLE events: RFC4662 */
- { SIP_OPT_EVENTLIST, NOT_SUPPORTED, "eventlist" },
- /* RFC 4916- Connected line ID updates */
- { SIP_OPT_FROMCHANGE, NOT_SUPPORTED, "from-change" },
- /* GRUU: Globally Routable User Agent URI's */
- { SIP_OPT_GRUU, NOT_SUPPORTED, "gruu" },
- /* RFC4244 History info */
- { SIP_OPT_HISTINFO, NOT_SUPPORTED, "histinfo" },
- /* RFC3911: SIP Join header support */
- { SIP_OPT_JOIN, NOT_SUPPORTED, "join" },
- /* Disable the REFER subscription, RFC 4488 */
- { SIP_OPT_NOREFERSUB, NOT_SUPPORTED, "norefersub" },
- /* SIP outbound - the final NAT battle - draft-sip-outbound */
- { SIP_OPT_OUTBOUND, NOT_SUPPORTED, "outbound" },
- /* RFC3327: Path support */
- { SIP_OPT_PATH, NOT_SUPPORTED, "path" },
- /* RFC3840: Callee preferences */
- { SIP_OPT_PREF, NOT_SUPPORTED, "pref" },
- /* RFC3312: Precondition support */
- { SIP_OPT_PRECONDITION, NOT_SUPPORTED, "precondition" },
- /* RFC3323: Privacy with proxies*/
- { SIP_OPT_PRIVACY, NOT_SUPPORTED, "privacy" },
- /* RFC-ietf-sip-uri-list-conferencing-02.txt conference invite lists */
- { SIP_OPT_RECLISTINV, NOT_SUPPORTED, "recipient-list-invite" },
- /* RFC-ietf-sip-uri-list-subscribe-02.txt - subscription lists */
- { SIP_OPT_RECLISTSUB, NOT_SUPPORTED, "recipient-list-subscribe" },
- /* RFC3891: Replaces: header for transfer */
- { SIP_OPT_REPLACES, SUPPORTED, "replaces" },
- /* One version of Polycom firmware has the wrong label */
- { SIP_OPT_REPLACES, SUPPORTED, "replace" },
- /* RFC4412 Resource priorities */
- { SIP_OPT_RESPRIORITY, NOT_SUPPORTED, "resource-priority" },
- /* RFC3329: Security agreement mechanism */
- { SIP_OPT_SEC_AGREE, NOT_SUPPORTED, "sec_agree" },
- /* RFC4092: Usage of the SDP ANAT Semantics in the SIP */
- { SIP_OPT_SDP_ANAT, NOT_SUPPORTED, "sdp-anat" },
- /* RFC4028: SIP Session-Timers */
- { SIP_OPT_TIMER, SUPPORTED, "timer" },
- /* RFC4538: Target-dialog */
- { SIP_OPT_TARGET_DIALOG,NOT_SUPPORTED, "tdialog" },
-};
-
-struct digestkeys {
- const char *key;
- const char *s;
-};
-
-AST_THREADSTORAGE(check_auth_buf);
-
-/*----------------------------------------------------------*/
-/*---- FUNCTIONS ----*/
-/*----------------------------------------------------------*/
-
-struct sip_peer *sip_find_peer(const char *peer, struct ast_sockaddr *addr, int realtime, int which_objects, int devstate_only, int transport);
-void sip_auth_headers(enum sip_auth_type code, char **header, char **respheader);
-const char *sip_get_header(const struct sip_request *req, const char *name);
-const char *sip_get_transport(enum ast_transport t);
-
-#define sip_ref_peer(peer, tag) ao2_t_bump(peer, tag)
-#define sip_unref_peer(peer, tag) ({ ao2_t_cleanup(peer, tag); (NULL); })
-
-#endif
diff --git a/channels/sip/include/sip_utils.h b/channels/sip/include/sip_utils.h
deleted file mode 100644
index 01207e7741..0000000000
--- a/channels/sip/include/sip_utils.h
+++ /dev/null
@@ -1,89 +0,0 @@
-/*
- * Asterisk -- An open source telephony toolkit.
- *
- * Copyright (C) 2010, Digium, Inc.
- *
- * See http://www.asterisk.org for more information about
- * the Asterisk project. Please do not directly contact
- * any of the maintainers of this project for assistance;
- * the project provides a web site, mailing lists and IRC
- * channels for your use.
- *
- * This program is free software, distributed under the terms of
- * the GNU General Public License Version 2. See the LICENSE file
- * at the top of the source tree.
- */
-
-/*!
- * \file
- * \brief sip utils header file
- */
-
-#ifndef _SIP_UTILS_H
-#define _SIP_UTILS_H
-
-/* wrapper macro to tell whether t points to one of the sip_tech descriptors */
-#define IS_SIP_TECH(t) ((t) == &sip_tech || (t) == &sip_tech_info)
-
-/*!
- * \brief converts ascii port to int representation.
- *
- * \arg pt[in] string that contains a port.
- * \arg standard[in] port to return in case the port string input is NULL
- * or if there is a parsing error.
- *
- * \return An integer port representation.
- */
-unsigned int port_str2int(const char *pt, unsigned int standard);
-
-/*! \brief Locate closing quote in a string, skipping escaped quotes.
- * optionally with a limit on the search.
- * start must be past the first quote.
- */
-const char *find_closing_quote(const char *start, const char *lim);
-
-
-/*! \brief Convert SIP hangup causes to Asterisk hangup causes */
-int hangup_sip2cause(int cause);
-
-/*! \brief Convert Asterisk hangup causes to SIP codes
-\verbatim
- Possible values from causes.h
- AST_CAUSE_NOTDEFINED AST_CAUSE_NORMAL AST_CAUSE_BUSY
- AST_CAUSE_FAILURE AST_CAUSE_CONGESTION AST_CAUSE_UNALLOCATED
-
- In addition to these, a lot of PRI codes is defined in causes.h
- ...should we take care of them too ?
-
- Quote RFC 3398
-
- ISUP Cause value SIP response
- ---------------- ------------
- 1 unallocated number 404 Not Found
- 2 no route to network 404 Not found
- 3 no route to destination 404 Not found
- 16 normal call clearing --- (*)
- 17 user busy 486 Busy here
- 18 no user responding 408 Request Timeout
- 19 no answer from the user 480 Temporarily unavailable
- 20 subscriber absent 480 Temporarily unavailable
- 21 call rejected 403 Forbidden (+)
- 22 number changed (w/o diagnostic) 410 Gone
- 22 number changed (w/ diagnostic) 301 Moved Permanently
- 23 redirection to new destination 410 Gone
- 26 non-selected user clearing 404 Not Found (=)
- 27 destination out of order 502 Bad Gateway
- 28 address incomplete 484 Address incomplete
- 29 facility rejected 501 Not implemented
- 31 normal unspecified 480 Temporarily unavailable
-\endverbatim
-*/
-const char *hangup_cause2sip(int cause);
-
-/*! \brief Return a string describing the force_rport value for the given flags */
-const char *force_rport_string(struct ast_flags *flags);
-
-/*! \brief Return a string describing the comedia value for the given flags */
-const char *comedia_string(struct ast_flags *flags);
-
-#endif
diff --git a/channels/sip/reqresp_parser.c b/channels/sip/reqresp_parser.c
deleted file mode 100644
index c72c322325..0000000000
--- a/channels/sip/reqresp_parser.c
+++ /dev/null
@@ -1,2686 +0,0 @@
-/*
- * Asterisk -- An open source telephony toolkit.
- *
- * Copyright (C) 2010, Digium, Inc.
- *
- * See http://www.asterisk.org for more information about
- * the Asterisk project. Please do not directly contact
- * any of the maintainers of this project for assistance;
- * the project provides a web site, mailing lists and IRC
- * channels for your use.
- *
- * This program is free software, distributed under the terms of
- * the GNU General Public License Version 2. See the LICENSE file
- * at the top of the source tree.
- */
-
-/*!
- * \file
- * \brief sip request parsing functions and unit tests
- */
-
-/*** MODULEINFO
- deprecated
- ***/
-
-#include "asterisk.h"
-
-#include "include/sip.h"
-#include "include/sip_utils.h"
-#include "include/reqresp_parser.h"
-
-#ifdef HAVE_XLOCALE_H
-locale_t c_locale;
-#endif
-
-/*! \brief * parses a URI in its components.*/
-int parse_uri_full(char *uri, const char *scheme, char **user, char **pass,
- char **hostport, struct uriparams *params, char **headers,
- char **residue)
-{
- char *userinfo = NULL;
- char *parameters = NULL;
- char *endparams = NULL;
- char *c = NULL;
- int error = 0;
- int teluri_scheme = 0;
-
- /*
- * Initialize requested strings - some functions don't care if parse_uri fails
- * and will attempt to use string pointers passed into parse_uri even after a
- * parse_uri failure
- */
- if (user) {
- *user = "";
- }
- if (pass) {
- *pass = "";
- }
- if (hostport) {
- *hostport = "";
- }
- if (headers) {
- *headers = "";
- }
- if (residue) {
- *residue = "";
- }
-
- /* check for valid input */
- if (ast_strlen_zero(uri)) {
- return -1;
- }
-
- if (scheme) {
- int l;
- char *scheme2 = ast_strdupa(scheme);
- char *cur = strsep(&scheme2, ",");
- for (; !ast_strlen_zero(cur); cur = strsep(&scheme2, ",")) {
- l = strlen(cur);
- if (!strncasecmp(uri, cur, l)) {
- teluri_scheme = !strncasecmp(uri, "tel:", 4); /* TEL URI */
- uri += l;
- break;
- }
- }
- if (ast_strlen_zero(cur)) {
- ast_debug(1, "No supported scheme found in '%s' using the scheme[s] %s\n", uri, scheme);
- error = -1;
- }
- }
-
- if (!hostport) {
- /* if we don't want to split around hostport, keep everything as a
- * userinfo - cos thats how old parse_uri operated*/
- userinfo = uri;
- } else if (teluri_scheme) {
- /*
- * tel: TEL URI INVITE RFC 3966 patch
- * See http://www.ietf.org/rfc/rfc3966.txt
- *
- * Once the full RFC 3966 parsing is implemented,
- * the ext= or isub= parameters would be extracted from userinfo.
- * When this kind of subaddressing would be implemented, the userinfo must be further parsed.
- * Those parameters would be used for ISDN or PSTN local extensions.
- *
- * Current restrictions:
- * We currently consider the ";isub=" or the ";ext=" as part of the userinfo (unparsed).
- */
-
- if ((c = strstr(uri, ";phone-context="))) {
- /*
- * Local number with context or domain.
- * ext= or isub= TEL URI parameters should be upfront.
- * All other parameters should come after the ";phone-context=" parameter.
- * If other parameters would occur before ";phone-context=" they will be ignored.
- */
-
- *c = '\0';
- userinfo = uri;
- uri = c + 15;
- *hostport = uri;
- } else if ('+' == uri[0]) {
- /* Global number without context or domain; possibly followed by RFC 3966 and optional other parameters. */
-
- userinfo = uri;
- *hostport = uri;
- } else {
- ast_debug(1, "No RFC 3966 global number or context found in '%s'; returning local number anyway\n", uri);
- userinfo = uri; /* Return local number anyway */
- error = -1;
- }
- } else {
- char *dom = "";
- if ((c = strchr(uri, '@'))) {
- *c++ = '\0';
- dom = c;
- userinfo = uri;
- uri = c; /* userinfo can contain ? and ; chars so step forward before looking for params and headers */
- } else {
- /* domain-only URI, according to the SIP RFC. */
- dom = uri;
- userinfo = "";
- }
-
- *hostport = dom;
- }
-
- if (pass && (c = strchr(userinfo, ':'))) { /* user:password */
- *c++ = '\0';
- *pass = c;
- } else if (pass) {
- *pass = "";
- }
-
- if (user) {
- *user = userinfo;
- }
-
- parameters = uri;
- /* strip [?headers] from end of uri - even if no header pointer exists*/
- if ((c = strrchr(uri, '?'))) {
- *c++ = '\0';
- uri = c;
- if (headers) {
- *headers = c;
- }
- if ((c = strrchr(uri, ';'))) {
- *c++ = '\0';
- } else {
- c = strrchr(uri, '\0');
- }
- uri = c; /* residue */
-
-
- } else if (headers) {
- *headers = "";
- }
-
- /* parse parameters */
- endparams = strchr(parameters,'\0');
- if ((c = strchr(parameters, ';'))) {
- *c++ = '\0';
- parameters = c;
- } else {
- parameters = endparams;
- }
-
- if (params) {
- char *rem = parameters; /* unparsed or unrecognised remainder */
- char *label;
- char *value;
- int lr = 0;
-
- params->transport = "";
- params->user = "";
- params->method = "";
- params->ttl = "";
- params->maddr = "";
- params->lr = 0;
-
- rem = parameters;
-
- while ((value = strchr(parameters, '=')) || (lr = !strncmp(parameters, "lr", 2))) {
- /* The while condition will not continue evaluation to set lr if it matches "lr=" */
- if (lr) {
- value = parameters;
- } else {
- *value++ = '\0';
- }
- label = parameters;
- if ((c = strchr(value, ';'))) {
- *c++ = '\0';
- parameters = c;
- } else {
- parameters = endparams;
- }
-
- if (!strcmp(label, "transport")) {
- params->transport = value;
- rem = parameters;
- } else if (!strcmp(label, "user")) {
- params->user = value;
- rem = parameters;
- } else if (!strcmp(label, "method")) {
- params->method = value;
- rem = parameters;
- } else if (!strcmp(label, "ttl")) {
- params->ttl = value;
- rem = parameters;
- } else if (!strcmp(label, "maddr")) {
- params->maddr = value;
- rem = parameters;
- /* Treat "lr", "lr=yes", "lr=on", "lr=1", "lr=almostanything" as lr enabled and "", "lr=no", "lr=off", "lr=0", "lr=" and "lranything" as lr disabled */
- } else if ((!strcmp(label, "lr") && strcmp(value, "no") && strcmp(value, "off") && strcmp(value, "0") && strcmp(value, "")) || ((lr) && strcmp(value, "lr"))) {
- params->lr = 1;
- rem = parameters;
- } else {
- value--;
- *value = '=';
- if (c) {
- c--;
- *c = ';';
- }
- }
- }
- if (rem > uri) { /* no headers */
- uri = rem;
- }
-
- }
-
- if (residue) {
- *residue = uri;
- }
-
- return error;
-}
-
-#ifdef TEST_FRAMEWORK
-AST_TEST_DEFINE(sip_parse_uri_full_test)
-{
- int res = AST_TEST_PASS;
- char uri[1024];
- char *user, *pass, *hostport, *headers, *residue;
- struct uriparams params;
-
- struct testdata {
- char *desc;
- char *uri;
- char *user;
- char *pass;
- char *hostport;
- char *headers;
- char *residue;
- struct uriparams params;
- AST_LIST_ENTRY(testdata) list;
- };
-
-
- struct testdata *testdataptr;
-
- static AST_LIST_HEAD_NOLOCK(testdataliststruct, testdata) testdatalist;
-
- struct testdata td1 = {
- .desc = "no headers",
- .uri = "sip:user:secret@host:5060;param=discard;transport=tcp;param2=residue",
- .user = "user",
- .pass = "secret",
- .hostport = "host:5060",
- .headers = "",
- .residue = "param2=residue",
- .params.transport = "tcp",
- .params.lr = 0,
- .params.user = ""
- };
-
- struct testdata td2 = {
- .desc = "with headers",
- .uri = "sip:user:secret@host:5060;param=discard;transport=tcp;param2=discard2?header=blah&header2=blah2;param3=residue",
- .user = "user",
- .pass = "secret",
- .hostport = "host:5060",
- .headers = "header=blah&header2=blah2",
- .residue = "param3=residue",
- .params.transport = "tcp",
- .params.lr = 0,
- .params.user = ""
- };
-
- struct testdata td3 = {
- .desc = "difficult user",
- .uri = "sip:-_.!~*'()&=+$,;?/:secret@host:5060;transport=tcp",
- .user = "-_.!~*'()&=+$,;?/",
- .pass = "secret",
- .hostport = "host:5060",
- .headers = "",
- .residue = "",
- .params.transport = "tcp",
- .params.lr = 0,
- .params.user = ""
- };
-
- struct testdata td4 = {
- .desc = "difficult pass",
- .uri = "sip:user:-_.!~*'()&=+$,@host:5060;transport=tcp",
- .user = "user",
- .pass = "-_.!~*'()&=+$,",
- .hostport = "host:5060",
- .headers = "",
- .residue = "",
- .params.transport = "tcp",
- .params.lr = 0,
- .params.user = ""
- };
-
- struct testdata td5 = {
- .desc = "difficult host",
- .uri = "sip:user:secret@1-1.a-1.:5060;transport=tcp",
- .user = "user",
- .pass = "secret",
- .hostport = "1-1.a-1.:5060",
- .headers = "",
- .residue = "",
- .params.transport = "tcp",
- .params.lr = 0,
- .params.user = ""
- };
-
- struct testdata td6 = {
- .desc = "difficult params near transport",
- .uri = "sip:user:secret@host:5060;-_.!~*'()[]/:&+$=-_.!~*'()[]/:&+$;transport=tcp",
- .user = "user",
- .pass = "secret",
- .hostport = "host:5060",
- .headers = "",
- .residue = "",
- .params.transport = "tcp",
- .params.lr = 0,
- .params.user = ""
- };
-
- struct testdata td7 = {
- .desc = "difficult params near headers",
- .uri = "sip:user:secret@host:5060;-_.!~*'()[]/:&+$=-_.!~*'()[]/:&+$?header=blah&header2=blah2;-_.!~*'()[]/:&+$=residue",
- .user = "user",
- .pass = "secret",
- .hostport = "host:5060",
- .headers = "header=blah&header2=blah2",
- .residue = "-_.!~*'()[]/:&+$=residue",
- .params.transport = "",
- .params.lr = 0,
- .params.user = ""
- };
-
- struct testdata td8 = {
- .desc = "lr parameter",
- .uri = "sip:user:secret@host:5060;param=discard;lr?header=blah",
- .user = "user",
- .pass = "secret",
- .hostport = "host:5060",
- .headers = "header=blah",
- .residue = "",
- .params.transport = "",
- .params.lr = 1,
- .params.user = ""
- };
-
- struct testdata td9 = {
- .desc = "alternative lr parameter",
- .uri = "sip:user:secret@host:5060;param=discard;lr=yes?header=blah",
- .user = "user",
- .pass = "secret",
- .hostport = "host:5060",
- .headers = "header=blah",
- .residue = "",
- .params.transport = "",
- .params.lr = 1,
- .params.user = ""
- };
-
- struct testdata td10 = {
- .desc = "no lr parameter",
- .uri = "sip:user:secret@host:5060;paramlr=lr;lr=no;lr=off;lr=0;lr=;=lr;lrextra;lrparam2=lr?header=blah",
- .user = "user",
- .pass = "secret",
- .hostport = "host:5060",
- .headers = "header=blah",
- .residue = "",
- .params.transport = "",
- .params.lr = 0,
- .params.user = ""
- };
-
- /* RFC 3966 TEL URI INVITE */
- struct testdata td11 = {
- .desc = "tel local number",
- .uri = "tel:0987654321;phone-context=+32987654321",
- .user = "0987654321",
- .pass = "",
- .hostport = "+32987654321",
- .headers = "",
- .residue = "",
- .params.transport = "",
- .params.lr = 0,
- .params.user = ""
- };
-
- struct testdata td12 = {
- .desc = "tel global number",
- .uri = "tel:+32987654321",
- .user = "+32987654321",
- .pass = "",
- .hostport = "+32987654321",
- .headers = "",
- .residue = "",
- .params.transport = "",
- .params.lr = 0,
- .params.user = ""
- };
-
- /*
- * Once the full RFC 3966 parsing is implemented,
- * only the ext= or isub= parameters would be extracted from .user
- * Then the ;param=discard would be ignored,
- * and the .user would only contain "0987654321"
- */
- struct testdata td13 = {
- .desc = "tel local number",
- .uri = "tel:0987654321;ext=1234;param=discard;phone-context=+32987654321;transport=udp;param2=discard2?header=blah&header2=blah2;param3=residue",
- .user = "0987654321;ext=1234;param=discard",
- .pass = "",
- .hostport = "+32987654321",
- .headers = "header=blah&header2=blah2",
- .residue = "param3=residue",
- .params.transport = "udp",
- .params.lr = 0,
- .params.user = ""
- };
-
- AST_LIST_HEAD_SET_NOLOCK(&testdatalist, &td1);
- AST_LIST_INSERT_TAIL(&testdatalist, &td2, list);
- AST_LIST_INSERT_TAIL(&testdatalist, &td3, list);
- AST_LIST_INSERT_TAIL(&testdatalist, &td4, list);
- AST_LIST_INSERT_TAIL(&testdatalist, &td5, list);
- AST_LIST_INSERT_TAIL(&testdatalist, &td6, list);
- AST_LIST_INSERT_TAIL(&testdatalist, &td7, list);
- AST_LIST_INSERT_TAIL(&testdatalist, &td8, list);
- AST_LIST_INSERT_TAIL(&testdatalist, &td9, list);
- AST_LIST_INSERT_TAIL(&testdatalist, &td10, list);
- AST_LIST_INSERT_TAIL(&testdatalist, &td11, list);
- AST_LIST_INSERT_TAIL(&testdatalist, &td12, list);
- AST_LIST_INSERT_TAIL(&testdatalist, &td13, list);
-
- switch (cmd) {
- case TEST_INIT:
- info->name = "sip_uri_full_parse_test";
- info->category = "/channels/chan_sip/";
- info->summary = "tests sip full uri parsing";
- info->description =
- "Tests full parsing of various URIs "
- "Verifies output matches expected behavior.";
- return AST_TEST_NOT_RUN;
- case TEST_EXECUTE:
- break;
- }
-
- AST_LIST_TRAVERSE(&testdatalist, testdataptr, list) {
- user = pass = hostport = headers = residue = NULL;
- params.transport = params.user = params.method = params.ttl = params.maddr = NULL;
- params.lr = 0;
-
- ast_copy_string(uri,testdataptr->uri,sizeof(uri));
- if (parse_uri_full(uri, "sip:,sips:,tel:", &user,
- &pass, &hostport,
- ¶ms,
- &headers,
- &residue) ||
- (user && strcmp(testdataptr->user, user)) ||
- (pass && strcmp(testdataptr->pass, pass)) ||
- (hostport && strcmp(testdataptr->hostport, hostport)) ||
- (headers && strcmp(testdataptr->headers, headers)) ||
- (residue && strcmp(testdataptr->residue, residue)) ||
- (strcmp(testdataptr->params.transport,params.transport)) ||
- (testdataptr->params.lr != params.lr) ||
- (strcmp(testdataptr->params.user,params.user))
- ) {
- ast_test_status_update(test, "Sub-Test: %s, failed.\n", testdataptr->desc);
- res = AST_TEST_FAIL;
- }
- }
-
-
- return res;
-}
-#endif
-
-int parse_uri(char *uri, const char *scheme, char **user, char **pass,
- char **hostport, char **transport) {
- int ret;
- char *headers;
- struct uriparams params;
-
- headers = NULL;
- ret = parse_uri_full(uri, scheme, user, pass, hostport, ¶ms, &headers, NULL);
- if (transport) {
- *transport=params.transport;
- }
- return ret;
-}
-
-#ifdef TEST_FRAMEWORK
-AST_TEST_DEFINE(sip_parse_uri_test)
-{
- int res = AST_TEST_PASS;
- char *name, *pass, *hostport, *transport;
- char uri1[] = "sip:name@host";
- char uri2[] = "sip:name@host;transport=tcp";
- char uri3[] = "sip:name:secret@host;transport=tcp";
- char uri4[] = "sip:name:secret@host:port;transport=tcp?headers=%40%40testblah&headers2=blah%20blah";
- /* test 5 is for NULL input */
- char uri6[] = "sip:name:secret@host:port;transport=tcp?headers=%40%40testblah&headers2=blah%20blah";
- char uri7[] = "sip:name:secret@host:port;transport=tcp?headers=%40%40testblah&headers2=blah%20blah";
- char uri8[] = "sip:host";
- char uri9[] = "sip:host:port;transport=tcp?headers=%40%40testblah&headers2=blah%20blah";
- char uri10[] = "host:port;transport=tcp?headers=%40%40testblah&headers2=blah%20blah";
- char uri11[] = "host";
- char uri12[] = "tel:911"; /* TEL URI Local number without context or global number */
-
- switch (cmd) {
- case TEST_INIT:
- info->name = "sip_uri_parse_test";
- info->category = "/channels/chan_sip/";
- info->summary = "tests sip uri parsing";
- info->description =
- "Tests parsing of various URIs "
- "Verifies output matches expected behavior.";
- return AST_TEST_NOT_RUN;
- case TEST_EXECUTE:
- break;
- }
-
- /* Test 1, simple URI */
- name = pass = hostport = transport = NULL;
- if (parse_uri(uri1, "sip:,sips:", &name, &pass, &hostport, &transport) ||
- strcmp(name, "name") ||
- !ast_strlen_zero(pass) ||
- strcmp(hostport, "host") ||
- !ast_strlen_zero(transport)) {
- ast_test_status_update(test, "Test 1: simple uri failed. \n");
- res = AST_TEST_FAIL;
- }
-
- /* Test 2, add tcp transport */
- name = pass = hostport = transport = NULL;
- if (parse_uri(uri2, "sip:,sips:", &name, &pass, &hostport, &transport) ||
- strcmp(name, "name") ||
- !ast_strlen_zero(pass) ||
- strcmp(hostport, "host") ||
- strcmp(transport, "tcp")) {
- ast_test_status_update(test, "Test 2: uri with addtion of tcp transport failed. \n");
- res = AST_TEST_FAIL;
- }
-
- /* Test 3, add secret */
- name = pass = hostport = transport = NULL;
- if (parse_uri(uri3, "sip:,sips:", &name, &pass, &hostport, &transport) ||
- strcmp(name, "name") ||
- strcmp(pass, "secret") ||
- strcmp(hostport, "host") ||
- strcmp(transport, "tcp")) {
- ast_test_status_update(test, "Test 3: uri with addition of secret failed.\n");
- res = AST_TEST_FAIL;
- }
-
- /* Test 4, add port and unparsed header field*/
- name = pass = hostport = transport = NULL;
- if (parse_uri(uri4, "sip:,sips:", &name, &pass, &hostport, &transport) ||
- strcmp(name, "name") ||
- strcmp(pass, "secret") ||
- strcmp(hostport, "host:port") ||
- strcmp(transport, "tcp")) {
- ast_test_status_update(test, "Test 4: add port and unparsed header field failed.\n");
- res = AST_TEST_FAIL;
- }
-
- /* Test 5, verify parse_uri does not crash when given a NULL uri */
- name = pass = hostport = transport = NULL;
- if (!parse_uri(NULL, "sip:,sips:", &name, &pass, &hostport, &transport)) {
- ast_test_status_update(test, "Test 5: passing a NULL uri failed.\n");
- res = AST_TEST_FAIL;
- }
-
- /* Test 6, verify parse_uri does not crash when given a NULL output parameters */
- name = pass = hostport = transport = NULL;
- if (parse_uri(uri6, "sip:,sips:", NULL, NULL, NULL, NULL)) {
- ast_test_status_update(test, "Test 6: passing NULL output parameters failed.\n");
- res = AST_TEST_FAIL;
- }
-
- /* Test 7, verify parse_uri returns user:secret and hostport when no port or secret output parameters are supplied. */
- name = pass = hostport = transport = NULL;
- if (parse_uri(uri7, "sip:,sips:", &name, NULL, &hostport, NULL) ||
- strcmp(name, "name:secret") ||
- strcmp(hostport, "host:port")) {
-
- ast_test_status_update(test, "Test 7: providing no port and secret output parameters failed.\n");
- res = AST_TEST_FAIL;
- }
-
- /* Test 8, verify parse_uri can handle a hostport only uri */
- name = pass = hostport = transport = NULL;
- if (parse_uri(uri8, "sip:,sips:", &name, &pass, &hostport, &transport) ||
- strcmp(hostport, "host") ||
- !ast_strlen_zero(name)) {
- ast_test_status_update(test, "Test 8: add port and unparsed header field failed.\n");
- res = AST_TEST_FAIL;
- }
-
- /* Test 9, add port and unparsed header field with hostport only uri*/
- name = pass = hostport = transport = NULL;
- if (parse_uri(uri9, "sip:,sips:", &name, &pass, &hostport, &transport) ||
- !ast_strlen_zero(name) ||
- !ast_strlen_zero(pass) ||
- strcmp(hostport, "host:port") ||
- strcmp(transport, "tcp")) {
- ast_test_status_update(test, "Test 9: hostport only uri failed \n");
- res = AST_TEST_FAIL;
- }
-
- /* Test 10, handle invalid/missing "sip:,sips:" scheme
- * we expect parse_uri to return an error, but still parse
- * the results correctly here */
- name = pass = hostport = transport = NULL;
- if (!parse_uri(uri10, "sip:,sips:", &name, &pass, &hostport, &transport) ||
- !ast_strlen_zero(name) ||
- !ast_strlen_zero(pass) ||
- strcmp(hostport, "host:port") ||
- strcmp(transport, "tcp")) {
- ast_test_status_update(test, "Test 10: missing \"sip:sips:\" scheme failed\n");
- res = AST_TEST_FAIL;
- }
-
- /* Test 11, simple hostport only URI with missing scheme
- * we expect parse_uri to return an error, but still parse
- * the results correctly here */
- name = pass = hostport = transport = NULL;
- if (!parse_uri(uri11, "sip:,sips:", &name, &pass, &hostport, &transport) ||
- !ast_strlen_zero(name) ||
- !ast_strlen_zero(pass) ||
- strcmp(hostport, "host") ||
- !ast_strlen_zero(transport)) {
- ast_test_status_update(test, "Test 11: simple uri with missing scheme failed. \n");
- res = AST_TEST_FAIL;
- }
-
- /* Test 12, simple URI */
- name = pass = hostport = transport = NULL;
- if (!parse_uri(uri12, "sip:,sips:,tel:", &name, &pass, &hostport, &transport) ||
- strcmp(name, "911") || /* We return local number anyway */
- !ast_strlen_zero(pass) ||
- !ast_strlen_zero(hostport) || /* No global number nor context */
- !ast_strlen_zero(transport)) {
- ast_test_status_update(test, "Test 12: TEL URI INVITE failed.\n");
- res = AST_TEST_FAIL;
- }
-
- return res;
-}
-#endif
-
-/*! \brief Get caller id name from SIP headers, copy into output buffer
- *
- * \return input string pointer placed after display-name field if possible
- */
-const char *get_calleridname(const char *input, char *output, size_t outputsize)
-{
- /* From RFC3261:
- *
- * From = ( "From" / "f" ) HCOLON from-spec
- * from-spec = ( name-addr / addr-spec ) *( SEMI from-param )
- * name-addr = [ display-name ] LAQUOT addr-spec RAQUOT
- * display-name = *(token LWS)/ quoted-string
- * token = 1*(alphanum / "-" / "." / "!" / "%" / "*"
- * / "_" / "+" / "`" / "'" / "~" )
- * quoted-string = SWS DQUOTE *(qdtext / quoted-pair ) DQUOTE
- * qdtext = LWS / %x21 / %x23-5B / %x5D-7E
- * / UTF8-NONASCII
- * quoted-pair = "\" (%x00-09 / %x0B-0C / %x0E-7F)
- *
- * HCOLON = *WSP ":" SWS
- * SWS = [LWS]
- * LWS = *[*WSP CRLF] 1*WSP
- * WSP = (SP / HTAB)
- *
- * Deviations from it:
- * - following CRLF's in LWS is not done (here at least)
- * - ascii NUL is never legal as it terminates the C-string
- * - utf8-nonascii is not checked for validity
- */
- char *orig_output = output;
- const char *orig_input = input;
-
- if (!output || !outputsize) {
- /* Bad output parameters. Should never happen. */
- return input;
- }
-
- /* clear any empty characters in the beginning */
- input = ast_skip_blanks(input);
-
- /* make sure the output buffer is initialized */
- *orig_output = '\0';
-
- /* make room for '\0' at the end of the output buffer */
- --outputsize;
-
- /* no data at all or no display name? */
- if (!input || *input == '<') {
- return input;
- }
-
- /* quoted-string rules */
- if (input[0] == '"') {
- input++; /* skip the first " */
-
- for (; *input; ++input) {
- if (*input == '"') { /* end of quoted-string */
- break;
- } else if (*input == 0x5c) { /* quoted-pair = "\" (%x00-09 / %x0B-0C / %x0E-7F) */
- ++input;
- if (!*input) {
- break;
- }
- if ((unsigned char) *input > 0x7f || *input == 0xa || *input == 0xd) {
- continue; /* not a valid quoted-pair, so skip it */
- }
- } else if ((*input != 0x9 && (unsigned char) *input < 0x20)
- || *input == 0x7f) {
- continue; /* skip this invalid character. */
- }
-
- if (0 < outputsize) {
- /* We still have room for the output display-name. */
- *output++ = *input;
- --outputsize;
- }
- }
-
- /* if this is successful, input should be at the ending quote */
- if (*input != '"') {
- ast_log(LOG_WARNING, "No ending quote for display-name was found\n");
- *orig_output = '\0';
- return orig_input;
- }
-
- /* make sure input is past the last quote */
- ++input;
-
- /* terminate output */
- *output = '\0';
- } else { /* either an addr-spec or tokenLWS-combo */
- for (; *input; ++input) {
- /* token or WSP (without LWS) */
- if ((*input >= '0' && *input <= '9') || (*input >= 'A' && *input <= 'Z')
- || (*input >= 'a' && *input <= 'z') || *input == '-' || *input == '.'
- || *input == '!' || *input == '%' || *input == '*' || *input == '_'
- || *input == '+' || *input == '`' || *input == '\'' || *input == '~'
- || *input == 0x9 || *input == ' ') {
- if (0 < outputsize) {
- /* We still have room for the output display-name. */
- *output++ = *input;
- --outputsize;
- }
- } else if (*input == '<') { /* end of tokenLWS-combo */
- /* we could assert that the previous char is LWS, but we don't care */
- break;
- } else if (*input == ':') {
- /* This invalid character which indicates this is addr-spec rather than display-name. */
- *orig_output = '\0';
- return orig_input;
- } else { /* else, invalid character we can skip. */
- continue; /* skip this character */
- }
- }
-
- if (*input != '<') { /* if we never found the start of addr-spec then this is invalid */
- *orig_output = '\0';
- return orig_input;
- }
-
- /* terminate output while trimming any trailing whitespace */
- do {
- *output-- = '\0';
- } while (orig_output <= output && (*output == 0x9 || *output == ' '));
- }
-
- return input;
-}
-
-#ifdef TEST_FRAMEWORK
-AST_TEST_DEFINE(get_calleridname_test)
-{
- int res = AST_TEST_PASS;
- const char *in1 = " \" quoted-text internal \\\" quote \"";
- const char *in2 = " token text with no quotes ";
- const char *overflow1 = " \"quoted-text overflow 1234567890123456789012345678901234567890\" ";
- const char *overflow2 = " non-quoted text overflow 1234567890123456789012345678901234567890 ";
- const char *noendquote = " \"quoted-text no end ";
- const char *addrspec = " sip:blah@blah";
- const char *no_quotes_no_brackets = "blah@blah";
- const char *after_dname;
- char dname[40];
-
- switch (cmd) {
- case TEST_INIT:
- info->name = "sip_get_calleridname_test";
- info->category = "/channels/chan_sip/";
- info->summary = "decodes callerid name from sip header";
- info->description = "Decodes display-name field of sip header. Checks for valid output and expected failure cases.";
- return AST_TEST_NOT_RUN;
- case TEST_EXECUTE:
- break;
- }
-
- /* quoted-text with backslash escaped quote */
- after_dname = get_calleridname(in1, dname, sizeof(dname));
- ast_test_status_update(test, "display-name1: %s\nafter: %s\n", dname, after_dname);
- if (strcmp(dname, " quoted-text internal \" quote ")) {
- ast_test_status_update(test, "display-name1 test failed\n");
- res = AST_TEST_FAIL;
- }
-
- /* token text */
- after_dname = get_calleridname(in2, dname, sizeof(dname));
- ast_test_status_update(test, "display-name2: %s\nafter: %s\n", dname, after_dname);
- if (strcmp(dname, "token text with no quotes")) {
- ast_test_status_update(test, "display-name2 test failed\n");
- res = AST_TEST_FAIL;
- }
-
- /* quoted-text buffer overflow */
- after_dname = get_calleridname(overflow1, dname, sizeof(dname));
- ast_test_status_update(test, "overflow display-name1: %s\nafter: %s\n", dname, after_dname);
- if (strcmp(dname, "quoted-text overflow 123456789012345678")) {
- ast_test_status_update(test, "overflow display-name1 test failed\n");
- res = AST_TEST_FAIL;
- }
-
- /* non-quoted-text buffer overflow */
- after_dname = get_calleridname(overflow2, dname, sizeof(dname));
- ast_test_status_update(test, "overflow display-name2: %s\nafter: %s\n", dname, after_dname);
- if (strcmp(dname, "non-quoted text overflow 12345678901234")) {
- ast_test_status_update(test, "overflow display-name2 test failed\n");
- res = AST_TEST_FAIL;
- }
-
- /* quoted-text buffer with no terminating end quote */
- after_dname = get_calleridname(noendquote, dname, sizeof(dname));
- ast_test_status_update(test, "noendquote display-name1: %s\nafter: %s\n", dname, after_dname);
- if (*dname != '\0' && after_dname != noendquote) {
- ast_test_status_update(test, "no end quote for quoted-text display-name failed\n");
- res = AST_TEST_FAIL;
- }
-
- /* addr-spec rather than display-name. */
- after_dname = get_calleridname(addrspec, dname, sizeof(dname));
- ast_test_status_update(test, "addr-spec display-name1: %s\nafter: %s\n", dname, after_dname);
- if (*dname != '\0' && after_dname != addrspec) {
- ast_test_status_update(test, "detection of addr-spec failed\n");
- res = AST_TEST_FAIL;
- }
-
- /* no quotes, no brackets */
- after_dname = get_calleridname(no_quotes_no_brackets, dname, sizeof(dname));
- ast_test_status_update(test, "no_quotes_no_brackets display-name1: %s\nafter: %s\n", dname, after_dname);
- if (*dname != '\0' && after_dname != no_quotes_no_brackets) {
- ast_test_status_update(test, "detection of addr-spec failed\n");
- res = AST_TEST_FAIL;
- }
-
- return res;
-}
-#endif
-
-int get_name_and_number(const char *hdr, char **name, char **number)
-{
- char header[256];
- char tmp_name[256];
- char *tmp_number = NULL;
- char *hostport = NULL;
- char *dummy = NULL;
-
- if (!name || !number || ast_strlen_zero(hdr)) {
- return -1;
- }
-
- *number = NULL;
- *name = NULL;
- ast_copy_string(header, hdr, sizeof(header));
-
- /* strip the display-name portion off the beginning of the header. */
- get_calleridname(header, tmp_name, sizeof(tmp_name));
-
- /* get uri within < > brackets */
- tmp_number = get_in_brackets(header);
-
- /* parse out the number here */
- if (parse_uri(tmp_number, "sip:,sips:", &tmp_number, &dummy, &hostport, NULL) || ast_strlen_zero(tmp_number)) {
- ast_log(LOG_ERROR, "can not parse name and number from sip header.\n");
- return -1;
- }
-
- /* number is not option, and must be present at this point */
- *number = ast_strdup(tmp_number);
- ast_uri_decode(*number, ast_uri_sip_user);
-
- /* name is optional and may not be present at this point */
- if (!ast_strlen_zero(tmp_name)) {
- *name = ast_strdup(tmp_name);
- }
-
- return 0;
-}
-
-#ifdef TEST_FRAMEWORK
-AST_TEST_DEFINE(get_name_and_number_test)
-{
- int res = AST_TEST_PASS;
- char *name = NULL;
- char *number = NULL;
- const char *in1 = "NAME ";
- const char *in2 = "\"NA>";
- const char *in3 = "NAME";
- const char *in4 = "";
- const char *in5 = "This is a screwed up string @place>";
-
- switch (cmd) {
- case TEST_INIT:
- info->name = "sip_get_name_and_number_test";
- info->category = "/channels/chan_sip/";
- info->summary = "Tests getting name and number from sip header";
- info->description =
- "Runs through various test situations in which a name and "
- "and number can be retrieved from a sip header.";
- return AST_TEST_NOT_RUN;
- case TEST_EXECUTE:
- break;
- }
-
- /* Test 1. get name and number */
- number = name = NULL;
- if ((get_name_and_number(in1, &name, &number)) ||
- strcmp(name, "NAME") ||
- strcmp(number, "NUMBER")) {
-
- ast_test_status_update(test, "Test 1, simple get name and number failed.\n");
- res = AST_TEST_FAIL;
- }
- ast_free(name);
- ast_free(number);
-
- /* Test 2. get quoted name and number */
- number = name = NULL;
- if ((get_name_and_number(in2, &name, &number)) ||
- strcmp(name, "NA>= first_bracket) {
- break; /* no need to look at quoted part */
- }
- /* the bracket is within quotes, so ignore it */
- parse = find_closing_quote(first_quote + 1, NULL);
- if (!*parse) {
- ast_log(LOG_WARNING, "No closing quote found in '%s'\n", src);
- return -1;
- }
- parse++;
- }
-
- /* Require a first bracket. Unlike get_in_brackets_full, this procedure is passed a const,
- * so it can expect a pointer to an original value */
- if (!first_bracket) {
- ast_log(LOG_WARNING, "No opening bracket found in '%s'\n", src);
- return 1;
- }
-
- if ((second_bracket = strchr(first_bracket, '>'))) {
- *start = first_bracket;
- *length = second_bracket - first_bracket;
- return 0;
- }
- ast_log(LOG_WARNING, "No closing bracket found in '%s'\n", src);
- return -1;
-}
-
-int get_in_brackets_full(char *tmp,char **out,char **residue)
-{
- const char *parse = tmp;
- char *first_bracket;
- char *second_bracket;
-
- if (out) {
- *out = "";
- }
- if (residue) {
- *residue = "";
- }
-
- if (ast_strlen_zero(tmp)) {
- return 1;
- }
-
- /*
- * Skip any quoted text until we find the part in brackets.
- * On any error give up and return -1
- */
- while ( (first_bracket = strchr(parse, '<')) ) {
- char *first_quote = strchr(parse, '"');
- first_bracket++;
- if (!first_quote || first_quote >= first_bracket) {
- break; /* no need to look at quoted part */
- }
- /* the bracket is within quotes, so ignore it */
- parse = find_closing_quote(first_quote + 1, NULL);
- if (!*parse) {
- ast_log(LOG_WARNING, "No closing quote found in '%s'\n", tmp);
- return -1;
- }
- parse++;
- }
-
- /* If no first bracket then still look for a second bracket as some other parsing functions
- may overwrite first bracket with NULL when terminating a token based display-name. As this
- only affects token based display-names there is no danger of brackets being in quotes */
- if (first_bracket) {
- parse = first_bracket;
- } else {
- parse = tmp;
- }
-
- if ((second_bracket = strchr(parse, '>'))) {
- *second_bracket++ = '\0';
- if (out) {
- *out = (char *) parse;
- }
- if (residue) {
- *residue = second_bracket;
- }
- return 0;
- }
-
- if ((first_bracket)) {
- ast_log(LOG_WARNING, "No closing bracket found in '%s'\n", tmp);
- return -1;
- }
-
- if (out) {
- *out = tmp;
- }
-
- return 1;
-}
-
-char *get_in_brackets(char *tmp)
-{
- char *out;
-
- if ((get_in_brackets_full(tmp, &out, NULL))) {
- return tmp;
- }
- return out;
-}
-
-#ifdef TEST_FRAMEWORK
-AST_TEST_DEFINE(get_in_brackets_test)
-{
- int res = AST_TEST_PASS;
- char in_brackets[] = "sip:name:secret@host:port;transport=tcp?headers=testblah&headers2=blahblah";
- char no_name[] = "";
- char quoted_string[] = "\"I'm a quote stri>";
- char missing_end_quote[] = "\"I'm a quote string ";
- char name_no_quotes[] = "name not in quotes ";
- char no_end_bracket[] = "name not in quotes name = "sip_get_in_brackets_test";
- info->category = "/channels/chan_sip/";
- info->summary = "Tests getting a sip uri in <> brackets within a sip header.";
- info->description =
- "Runs through various test situations in which a sip uri "
- "in angle brackets needs to be retrieved";
- return AST_TEST_NOT_RUN;
- case TEST_EXECUTE:
- break;
- }
-
- /* Test 1, simple get in brackets */
- if (!(uri = get_in_brackets(no_name)) || strcmp(uri, in_brackets)) {
- ast_test_status_update(test, "Test 1, simple get in brackets failed. %s\n", uri);
- res = AST_TEST_FAIL;
- }
-
- /* Test 2, starts with quoted string */
- if (!(uri = get_in_brackets(quoted_string)) || strcmp(uri, in_brackets)) {
- ast_test_status_update(test, "Test 2, get in brackets with quoted string in front failed. %s\n", uri);
- res = AST_TEST_FAIL;
- }
-
- /* Test 3, missing end quote */
- if (!(uri = get_in_brackets(missing_end_quote)) || !strcmp(uri, in_brackets)) {
- ast_test_status_update(test, "Test 3, missing end quote failed. %s\n", uri);
- res = AST_TEST_FAIL;
- }
-
- /* Test 4, starts with a name not in quotes */
- if (!(uri = get_in_brackets(name_no_quotes)) || strcmp(uri, in_brackets)) {
- ast_test_status_update(test, "Test 4, passing name not in quotes failed. %s\n", uri);
- res = AST_TEST_FAIL;
- }
-
- /* Test 5, no end bracket, should just return everything after the first '<' */
- if (!(uri = get_in_brackets(no_end_bracket)) || !strcmp(uri, in_brackets)) {
- ast_test_status_update(test, "Test 5, no end bracket failed. %s\n", uri);
- res = AST_TEST_FAIL;
- }
-
- /* Test 6, NULL input */
- if (get_in_brackets(NULL)) {
- ast_test_status_update(test, "Test 6, NULL input failed.\n");
- res = AST_TEST_FAIL;
- }
-
- /* Test 7, no name, and no brackets. */
- if (!(uri = get_in_brackets(no_name_no_brackets)) || strcmp(uri, "sip:name@host")) {
- ast_test_status_update(test, "Test 7 failed. %s\n", uri);
- res = AST_TEST_FAIL;
- }
-
- /* Test 8, no start bracket, but with ending bracket. */
- if (!(uri = get_in_brackets(missing_start_bracket)) || strcmp(uri, in_brackets)) {
- ast_test_status_update(test, "Test 8 failed. %s\n", uri);
- res = AST_TEST_FAIL;
- }
-
- return res;
-}
-#endif
-
-int parse_name_andor_addr(char *uri, const char *scheme, char **name,
- char **user, char **pass, char **hostport,
- struct uriparams *params, char **headers,
- char **residue)
-{
- char buf[1024];
- char **residue2 = residue;
- char *orig_uri = uri;
- int ret;
-
- buf[0] = '\0';
- if (name) {
- uri = (char *) get_calleridname(uri, buf, sizeof(buf));
- }
- ret = get_in_brackets_full(uri, &uri, residue);
- if (ret == 0) {
- /*
- * The uri is in brackets so do not treat unknown trailing uri
- * parameters as potential message header parameters.
- */
- if (residue && **residue) {
- /* step over the first semicolon as per parse_uri_full residue */
- *residue = *residue + 1;
- }
- residue2 = NULL;
- }
-
- if (name) {
- if (buf[0]) {
- /*
- * There is always room at orig_uri for the display-name because
- * at least one character has always been removed. A '"' or '<'
- * has been removed.
- */
- strcpy(orig_uri, buf);
- *name = orig_uri;
- } else {
- *name = "";
- }
- }
-
- return parse_uri_full(uri, scheme, user, pass, hostport, params, headers, residue2);
-}
-
-#ifdef TEST_FRAMEWORK
-AST_TEST_DEFINE(parse_name_andor_addr_test)
-{
- int res = AST_TEST_PASS;
- char uri[1024];
- char *name, *user, *pass, *hostport, *headers, *residue;
- struct uriparams params;
-
- struct testdata {
- char *desc;
- char *uri;
- char *name;
- char *user;
- char *pass;
- char *hostport;
- char *headers;
- char *residue;
- struct uriparams params;
- AST_LIST_ENTRY(testdata) list;
- };
-
- struct testdata *testdataptr;
-
- static AST_LIST_HEAD_NOLOCK(testdataliststruct, testdata) testdatalist;
-
- struct testdata td1 = {
- .desc = "quotes and brackets",
- .uri = "\"name :@ \" ;tag=tag",
- .name = "name :@ ",
- .user = "user",
- .pass = "secret",
- .hostport = "host:5060",
- .headers = "",
- .residue = "tag=tag",
- .params.transport = "tcp",
- .params.lr = 0,
- .params.user = ""
- };
-
- struct testdata td2 = {
- .desc = "no quotes",
- .uri = "givenname familyname ;expires=3600",
- .name = "givenname familyname",
- .user = "user",
- .pass = "secret",
- .hostport = "host:5060",
- .headers = "",
- .residue = "expires=3600",
- .params.transport = "tcp",
- .params.lr = 0,
- .params.user = ""
- };
-
- struct testdata td3 = {
- .desc = "no brackets",
- .uri = "sip:user:secret@host:5060;param=discard;transport=tcp;q=1",
- .name = "",
- .user = "user",
- .pass = "secret",
- .hostport = "host:5060",
- .headers = "",
- .residue = "q=1",
- .params.transport = "tcp",
- .params.lr = 0,
- .params.user = ""
- };
-
- struct testdata td4 = {
- .desc = "just host",
- .uri = "sips:host",
- .name = "",
- .user = "",
- .pass = "",
- .hostport = "host",
- .headers = "",
- .residue = "",
- .params.transport = "",
- .params.lr = 0,
- .params.user = ""
- };
-
-
- AST_LIST_HEAD_SET_NOLOCK(&testdatalist, &td1);
- AST_LIST_INSERT_TAIL(&testdatalist, &td2, list);
- AST_LIST_INSERT_TAIL(&testdatalist, &td3, list);
- AST_LIST_INSERT_TAIL(&testdatalist, &td4, list);
-
-
- switch (cmd) {
- case TEST_INIT:
- info->name = "parse_name_andor_addr_test";
- info->category = "/channels/chan_sip/";
- info->summary = "tests parsing of name_andor_addr abnf structure";
- info->description =
- "Tests parsing of abnf name-andor-addr = name-addr / addr-spec "
- "Verifies output matches expected behavior.";
- return AST_TEST_NOT_RUN;
- case TEST_EXECUTE:
- break;
- }
-
- AST_LIST_TRAVERSE(&testdatalist, testdataptr, list) {
- name = user = pass = hostport = headers = residue = NULL;
- params.transport = params.user = params.method = params.ttl = params.maddr = NULL;
- params.lr = 0;
- ast_copy_string(uri,testdataptr->uri,sizeof(uri));
- if (parse_name_andor_addr(uri, "sip:,sips:",
- &name,
- &user,
- &pass,
- &hostport,
- ¶ms,
- &headers,
- &residue) ||
- (name && strcmp(testdataptr->name, name)) ||
- (user && strcmp(testdataptr->user, user)) ||
- (pass && strcmp(testdataptr->pass, pass)) ||
- (hostport && strcmp(testdataptr->hostport, hostport)) ||
- (headers && strcmp(testdataptr->headers, headers)) ||
- (residue && strcmp(testdataptr->residue, residue)) ||
- (strcmp(testdataptr->params.transport,params.transport)) ||
- (strcmp(testdataptr->params.user,params.user))
- ) {
- ast_test_status_update(test, "Sub-Test: %s,failed.\n", testdataptr->desc);
- res = AST_TEST_FAIL;
- }
- }
-
- return res;
-}
-#endif
-
-int get_comma(char *in, char **out)
-{
- char *c;
- char *parse = in;
- if (out) {
- *out = in;
- }
-
- /* Skip any quoted text */
- while (*parse) {
- if ((c = strchr(parse, '"'))) {
- in = (char *)find_closing_quote((const char *)c + 1, NULL);
- if (!*in) {
- ast_log(LOG_WARNING, "No closing quote found in '%s'\n", c);
- return -1;
- } else {
- break;
- }
- } else {
- break;
- }
- parse++;
- }
- parse = in;
-
- /* Skip any userinfo components of a uri as they may contain commas */
- if ((c = strchr(parse,'@'))) {
- parse = c+1;
- }
- if ((out) && (c = strchr(parse,','))) {
- *c++ = '\0';
- *out = c;
- return 0;
- }
- return 1;
-}
-
-int parse_contact_header(char *contactheader, struct contactliststruct *contactlist)
-{
- int res;
- int last;
- char *comma;
- char *residue;
- char *param;
- char *value;
- struct contact *split_contact = NULL;
-
- if (*contactheader == '*') {
- return 1;
- }
-
- split_contact = ast_calloc(1, sizeof(*split_contact));
-
- AST_LIST_HEAD_SET_NOLOCK(contactlist, split_contact);
- while ((last = get_comma(contactheader, &comma)) != -1) {
- res = parse_name_andor_addr(contactheader, "sip:,sips:",
- &split_contact->name, &split_contact->user,
- &split_contact->pass, &split_contact->hostport,
- &split_contact->params, &split_contact->headers,
- &residue);
- if (res == -1) {
- return res;
- }
-
- /* parse contact params */
- split_contact->expires = split_contact->q = "";
-
- while ((value = strchr(residue,'='))) {
- *value++ = '\0';
-
- param = residue;
- if ((residue = strchr(value,';'))) {
- *residue++ = '\0';
- } else {
- residue = "";
- }
-
- if (!strcmp(param,"expires")) {
- split_contact->expires = value;
- } else if (!strcmp(param,"q")) {
- split_contact->q = value;
- }
- }
-
- if (last) {
- return 0;
- }
- contactheader = comma;
-
- split_contact = ast_calloc(1, sizeof(*split_contact));
- AST_LIST_INSERT_TAIL(contactlist, split_contact, list);
- }
- return last;
-}
-
-#ifdef TEST_FRAMEWORK
-AST_TEST_DEFINE(parse_contact_header_test)
-{
- int res = AST_TEST_PASS;
- char contactheader[1024];
- int star;
- struct contactliststruct contactlist;
- struct contactliststruct *contactlistptr=&contactlist;
-
- struct testdata {
- char *desc;
- char *contactheader;
- int star;
- struct contactliststruct *contactlist;
-
- AST_LIST_ENTRY(testdata) list;
- };
-
- struct testdata *testdataptr;
- struct contact *tdcontactptr;
- struct contact *contactptr;
-
- static AST_LIST_HEAD_NOLOCK(testdataliststruct, testdata) testdatalist;
- struct contactliststruct contactlist1, contactlist2;
-
- struct testdata td1 = {
- .desc = "single contact",
- .contactheader = "\"name :@;?&,\" ;expires=3600",
- .contactlist = &contactlist1,
- .star = 0
- };
- struct contact contact11 = {
- .name = "name :@;?&,",
- .user = "user",
- .pass = "secret",
- .hostport = "host:5082",
- .params.transport = "tcp",
- .params.ttl = "",
- .params.lr = 0,
- .headers = "",
- .expires = "3600",
- .q = ""
- };
-
- struct testdata td2 = {
- .desc = "multiple contacts",
- .contactheader = "sip:,user1,:,secret1,@host1;ttl=7;q=1;expires=3600,sips:host2",
- .contactlist = &contactlist2,
- .star = 0,
- };
- struct contact contact21 = {
- .name = "",
- .user = ",user1,",
- .pass = ",secret1,",
- .hostport = "host1",
- .params.transport = "",
- .params.ttl = "7",
- .params.lr = 0,
- .headers = "",
- .expires = "3600",
- .q = "1"
- };
- struct contact contact22 = {
- .name = "",
- .user = "",
- .pass = "",
- .hostport = "host2",
- .params.transport = "",
- .params.ttl = "",
- .params.lr = 0,
- .headers = "",
- .expires = "",
- .q = ""
- };
-
- struct testdata td3 = {
- .desc = "star - all contacts",
- .contactheader = "*",
- .star = 1,
- .contactlist = NULL
- };
-
- AST_LIST_HEAD_SET_NOLOCK(&testdatalist, &td1);
- AST_LIST_INSERT_TAIL(&testdatalist, &td2, list);
- AST_LIST_INSERT_TAIL(&testdatalist, &td3, list);
-
- AST_LIST_HEAD_SET_NOLOCK(&contactlist1, &contact11);
-
- AST_LIST_HEAD_SET_NOLOCK(&contactlist2, &contact21);
- AST_LIST_INSERT_TAIL(&contactlist2, &contact22, list);
-
-
- switch (cmd) {
- case TEST_INIT:
- info->name = "parse_contact_header_test";
- info->category = "/channels/chan_sip/";
- info->summary = "tests parsing of sip contact header";
- info->description =
- "Tests parsing of a contact header including those with multiple contacts "
- "Verifies output matches expected behavior.";
- return AST_TEST_NOT_RUN;
- case TEST_EXECUTE:
- break;
- }
-
- AST_LIST_TRAVERSE(&testdatalist, testdataptr, list) {
- ast_copy_string(contactheader,testdataptr->contactheader,sizeof(contactheader));
- star = parse_contact_header(contactheader,contactlistptr);
- if (testdataptr->star) {
- /* expecting star rather than list of contacts */
- if (!star) {
- ast_test_status_update(test, "Sub-Test: %s,failed.\n", testdataptr->desc);
- res = AST_TEST_FAIL;
- break;
- }
- } else {
- contactptr = AST_LIST_FIRST(contactlistptr);
- AST_LIST_TRAVERSE(testdataptr->contactlist, tdcontactptr, list) {
- if (!contactptr ||
- strcmp(tdcontactptr->name, contactptr->name) ||
- strcmp(tdcontactptr->user, contactptr->user) ||
- strcmp(tdcontactptr->pass, contactptr->pass) ||
- strcmp(tdcontactptr->hostport, contactptr->hostport) ||
- strcmp(tdcontactptr->headers, contactptr->headers) ||
- strcmp(tdcontactptr->expires, contactptr->expires) ||
- strcmp(tdcontactptr->q, contactptr->q) ||
- strcmp(tdcontactptr->params.transport, contactptr->params.transport) ||
- strcmp(tdcontactptr->params.ttl, contactptr->params.ttl) ||
- (tdcontactptr->params.lr != contactptr->params.lr)
- ) {
- ast_test_status_update(test, "Sub-Test: %s,failed.\n", testdataptr->desc);
- res = AST_TEST_FAIL;
- break;
- }
-
- contactptr = AST_LIST_NEXT(contactptr,list);
- }
-
- while ((contactptr = AST_LIST_REMOVE_HEAD(contactlistptr,list))) {
- ast_free(contactptr);
- }
- }
- }
-
- return res;
-}
-#endif
-
-/*!
- * \brief Parse supported header in incoming packet
- *
- * \details This function parses through the options parameters and
- * builds a bit field representing all the SIP options in that field. When an
- * item is found that is not supported, it is copied to the unsupported
- * out buffer.
- */
-unsigned int parse_sip_options(const char *options, char *unsupported, size_t unsupported_len)
-{
- char *next, *sep;
- char *temp;
- int i, found, supported;
- unsigned int profile = 0;
-
- char *out = unsupported;
- size_t outlen = unsupported_len;
- char *cur_out = out;
-
- if (ast_strlen_zero(options) )
- return 0;
-
- temp = ast_strdupa(options);
-
- ast_debug(3, "Begin: parsing SIP \"Supported: %s\"\n", options);
- for (next = temp; next; next = sep) {
- found = FALSE;
- supported = FALSE;
- if ((sep = strchr(next, ',')) != NULL) {
- *sep++ = '\0';
- }
-
- /* trim leading and trailing whitespace */
- next = ast_strip(next);
-
- if (ast_strlen_zero(next)) {
- continue; /* if there is a blank argument in there just skip it */
- }
-
- ast_debug(3, "Found SIP option: -%s-\n", next);
- for (i = 0; i < ARRAY_LEN(sip_options); i++) {
- if (!strcasecmp(next, sip_options[i].text)) {
- profile |= sip_options[i].id;
- if (sip_options[i].supported == SUPPORTED) {
- supported = TRUE;
- }
- found = TRUE;
- ast_debug(3, "Matched SIP option: %s\n", next);
- break;
- }
- }
-
- /* If option is not supported, add to unsupported out buffer */
- if (!supported && out && outlen) {
- size_t copylen = strlen(next);
- size_t cur_outlen = strlen(out);
- /* Check to see if there is enough room to store this option.
- * Copy length is string length plus 2 for the ',' and '\0' */
- if ((cur_outlen + copylen + 2) < outlen) {
- /* if this isn't the first item, add the ',' */
- if (cur_outlen) {
- *cur_out = ',';
- cur_out++;
- cur_outlen++;
- }
- ast_copy_string(cur_out, next, (outlen - cur_outlen));
- cur_out += copylen;
- }
- }
-
- if (!found) {
- profile |= SIP_OPT_UNKNOWN;
- if (!strncasecmp(next, "x-", 2))
- ast_debug(3, "Found private SIP option, not supported: %s\n", next);
- else
- ast_debug(3, "Found no match for SIP option: %s (Please file bug report!)\n", next);
- }
- }
-
- return profile;
-}
-
-#ifdef TEST_FRAMEWORK
-AST_TEST_DEFINE(sip_parse_options_test)
-{
- int res = AST_TEST_PASS;
- char unsupported[64];
- unsigned int option_profile = 0;
- struct testdata {
- char *name;
- char *input_options;
- char *expected_unsupported;
- unsigned int expected_profile;
- AST_LIST_ENTRY(testdata) list;
- };
-
- struct testdata *testdataptr;
- static AST_LIST_HEAD_NOLOCK(testdataliststruct, testdata) testdatalist;
-
- struct testdata test1 = {
- .name = "test_all_unsupported",
- .input_options = "unsupported1,,, ,unsupported2,unsupported3,unsupported4",
- .expected_unsupported = "unsupported1,unsupported2,unsupported3,unsupported4",
- .expected_profile = SIP_OPT_UNKNOWN,
- };
- struct testdata test2 = {
- .name = "test_all_unsupported_one_supported",
- .input_options = " unsupported1, replaces, unsupported3 , , , ,unsupported4",
- .expected_unsupported = "unsupported1,unsupported3,unsupported4",
- .expected_profile = SIP_OPT_UNKNOWN | SIP_OPT_REPLACES
- };
- struct testdata test3 = {
- .name = "test_two_supported_two_unsupported",
- .input_options = ",, timer ,replaces ,unsupported3,unsupported4",
- .expected_unsupported = "unsupported3,unsupported4",
- .expected_profile = SIP_OPT_UNKNOWN | SIP_OPT_REPLACES | SIP_OPT_TIMER,
- };
-
- struct testdata test4 = {
- .name = "test_all_supported",
- .input_options = "timer,replaces",
- .expected_unsupported = "",
- .expected_profile = SIP_OPT_REPLACES | SIP_OPT_TIMER,
- };
-
- struct testdata test5 = {
- .name = "test_all_supported_redundant",
- .input_options = "timer,replaces,timer,replace,timer,replaces",
- .expected_unsupported = "",
- .expected_profile = SIP_OPT_REPLACES | SIP_OPT_TIMER,
- };
- struct testdata test6 = {
- .name = "test_buffer_overflow",
- .input_options = "unsupported1,replaces,timer,unsupported4,unsupported_huge____"
- "____________________________________,__________________________________________"
- "________________________________________________",
- .expected_unsupported = "unsupported1,unsupported4",
- .expected_profile = SIP_OPT_UNKNOWN | SIP_OPT_REPLACES | SIP_OPT_TIMER,
- };
- struct testdata test7 = {
- .name = "test_null_input",
- .input_options = NULL,
- .expected_unsupported = "",
- .expected_profile = 0,
- };
- struct testdata test8 = {
- .name = "test_whitespace_input",
- .input_options = " ",
- .expected_unsupported = "",
- .expected_profile = 0,
- };
- struct testdata test9 = {
- .name = "test_whitespace_plus_option_input",
- .input_options = " , , ,timer , , , , , ",
- .expected_unsupported = "",
- .expected_profile = SIP_OPT_TIMER,
- };
-
- switch (cmd) {
- case TEST_INIT:
- info->name = "sip_parse_options_test";
- info->category = "/channels/chan_sip/";
- info->summary = "Tests parsing of sip options";
- info->description =
- "Tests parsing of SIP options from supported and required "
- "header fields. Verifies when unsupported options are encountered "
- "that they are appended to the unsupported out buffer and that the "
- "correct bit field representnig the option profile is returned.";
- return AST_TEST_NOT_RUN;
- case TEST_EXECUTE:
- break;
- }
-
- AST_LIST_HEAD_SET_NOLOCK(&testdatalist, &test1);
- AST_LIST_INSERT_TAIL(&testdatalist, &test2, list);
- AST_LIST_INSERT_TAIL(&testdatalist, &test3, list);
- AST_LIST_INSERT_TAIL(&testdatalist, &test4, list);
- AST_LIST_INSERT_TAIL(&testdatalist, &test5, list);
- AST_LIST_INSERT_TAIL(&testdatalist, &test6, list);
- AST_LIST_INSERT_TAIL(&testdatalist, &test7, list);
- AST_LIST_INSERT_TAIL(&testdatalist, &test8, list);
- AST_LIST_INSERT_TAIL(&testdatalist, &test9, list);
-
- /* Test with unsupported char buffer */
- AST_LIST_TRAVERSE(&testdatalist, testdataptr, list) {
- memset(unsupported, 0, sizeof(unsupported));
- option_profile = parse_sip_options(testdataptr->input_options, unsupported, ARRAY_LEN(unsupported));
- if (option_profile != testdataptr->expected_profile ||
- strcmp(unsupported, testdataptr->expected_unsupported)) {
- ast_test_status_update(test, "Test with output buffer \"%s\", expected unsupported: %s actual unsupported:"
- "%s expected bit profile: %x actual bit profile: %x\n",
- testdataptr->name,
- testdataptr->expected_unsupported,
- unsupported,
- testdataptr->expected_profile,
- option_profile);
- res = AST_TEST_FAIL;
- } else {
- ast_test_status_update(test, "\"%s\" passed got expected unsupported: %s and bit profile: %x\n",
- testdataptr->name,
- unsupported,
- option_profile);
- }
-
- option_profile = parse_sip_options(testdataptr->input_options, NULL, 0);
- if (option_profile != testdataptr->expected_profile) {
- ast_test_status_update(test, "NULL output test \"%s\", expected bit profile: %x actual bit profile: %x\n",
- testdataptr->name,
- testdataptr->expected_profile,
- option_profile);
- res = AST_TEST_FAIL;
- } else {
- ast_test_status_update(test, "\"%s\" with NULL output buf passed, bit profile: %x\n",
- testdataptr->name,
- option_profile);
- }
- }
-
- return res;
-}
-#endif
-
-/*! \brief helper routine for sip_uri_cmp to compare URI parameters
- *
- * This takes the parameters from two SIP URIs and determines
- * if the URIs match. The rules for parameters *suck*. Here's a breakdown
- * 1. If a parameter appears in both URIs, then they must have the same value
- * in order for the URIs to match
- * 2. If one URI has a user, maddr, ttl, or method parameter, then the other
- * URI must also have that parameter and must have the same value
- * in order for the URIs to match
- * 3. All other headers appearing in only one URI are not considered when
- * determining if URIs match
- *
- * \param input1 Parameters from URI 1
- * \param input2 Parameters from URI 2
- * \retval 0 URIs' parameters match
- * \retval nonzero URIs' parameters do not match
- */
-static int sip_uri_params_cmp(const char *input1, const char *input2)
-{
- char *params1 = NULL;
- char *params2 = NULL;
- char *pos1;
- char *pos2;
- int zerolength1 = 0;
- int zerolength2 = 0;
- int maddrmatch = 0;
- int ttlmatch = 0;
- int usermatch = 0;
- int methodmatch = 0;
-
- if (ast_strlen_zero(input1)) {
- zerolength1 = 1;
- } else {
- params1 = ast_strdupa(input1);
- }
- if (ast_strlen_zero(input2)) {
- zerolength2 = 1;
- } else {
- params2 = ast_strdupa(input2);
- }
-
- /* Quick optimization. If both params are zero-length, then
- * they match
- */
- if (zerolength1 && zerolength2) {
- return 0;
- }
-
- for (pos1 = strsep(¶ms1, ";"); pos1; pos1 = strsep(¶ms1, ";")) {
- char *value1 = pos1;
- char *name1 = strsep(&value1, "=");
- char *params2dup = NULL;
- int matched = 0;
- if (!value1) {
- value1 = "";
- }
- /* Checkpoint reached. We have the name and value parsed for param1
- * We have to duplicate params2 each time through this loop
- * or else the inner loop below will not work properly.
- */
- if (!zerolength2) {
- params2dup = ast_strdupa(params2);
- }
- for (pos2 = strsep(¶ms2dup, ";"); pos2; pos2 = strsep(¶ms2dup, ";")) {
- char *name2 = pos2;
- char *value2 = strchr(pos2, '=');
- if (!value2) {
- value2 = "";
- } else {
- *value2++ = '\0';
- }
- if (!strcasecmp(name1, name2)) {
- if (strcasecmp(value1, value2)) {
- goto fail;
- } else {
- matched = 1;
- break;
- }
- }
- }
- /* Check to see if the parameter is one of the 'must-match' parameters */
- if (!strcasecmp(name1, "maddr")) {
- if (matched) {
- maddrmatch = 1;
- } else {
- goto fail;
- }
- } else if (!strcasecmp(name1, "ttl")) {
- if (matched) {
- ttlmatch = 1;
- } else {
- goto fail;
- }
- } else if (!strcasecmp(name1, "user")) {
- if (matched) {
- usermatch = 1;
- } else {
- goto fail;
- }
- } else if (!strcasecmp(name1, "method")) {
- if (matched) {
- methodmatch = 1;
- } else {
- goto fail;
- }
- }
- }
-
- /* We've made it out of that horrible O(m*n) construct and there are no
- * failures yet. We're not done yet, though, because params2 could have
- * an maddr, ttl, user, or method header and params1 did not.
- */
- for (pos2 = strsep(¶ms2, ";"); pos2; pos2 = strsep(¶ms2, ";")) {
- char *value2 = pos2;
- char *name2 = strsep(&value2, "=");
- if (!value2) {
- value2 = "";
- }
- if ((!strcasecmp(name2, "maddr") && !maddrmatch) ||
- (!strcasecmp(name2, "ttl") && !ttlmatch) ||
- (!strcasecmp(name2, "user") && !usermatch) ||
- (!strcasecmp(name2, "method") && !methodmatch)) {
- goto fail;
- }
- }
- return 0;
-
-fail:
- return 1;
-}
-
-/*! \brief helper routine for sip_uri_cmp to compare URI headers
- *
- * This takes the headers from two SIP URIs and determines
- * if the URIs match. The rules for headers is simple. If a header
- * appears in one URI, then it must also appear in the other URI. The
- * order in which the headers appear does not matter.
- *
- * \param input1 Headers from URI 1
- * \param input2 Headers from URI 2
- * \retval 0 URI headers match
- * \retval nonzero URI headers do not match
- */
-static int sip_uri_headers_cmp(const char *input1, const char *input2)
-{
- char *headers1 = NULL;
- char *headers2 = NULL;
- int zerolength1 = 0;
- int zerolength2 = 0;
- int different = 0;
- char *header1;
-
- if (ast_strlen_zero(input1)) {
- zerolength1 = 1;
- } else {
- headers1 = ast_strdupa(input1);
- }
-
- if (ast_strlen_zero(input2)) {
- zerolength2 = 1;
- } else {
- headers2 = ast_strdupa(input2);
- }
-
- /* If one URI contains no headers and the other
- * does, then they cannot possibly match
- */
- if (zerolength1 != zerolength2) {
- return 1;
- }
-
- if (zerolength1 && zerolength2)
- return 0;
-
- /* At this point, we can definitively state that both inputs are
- * not zero-length. First, one more optimization. If the length
- * of the headers is not equal, then we definitely have no match
- */
- if (strlen(headers1) != strlen(headers2)) {
- return 1;
- }
-
- for (header1 = strsep(&headers1, "&"); header1; header1 = strsep(&headers1, "&")) {
- if (!strcasestr(headers2, header1)) {
- different = 1;
- break;
- }
- }
-
- return different;
-}
-
-/*!
- * \brief Compare domain sections of SIP URIs
- *
- * For hostnames, a case insensitive string comparison is
- * used. For IP addresses, a binary comparison is used. This
- * is mainly because IPv6 addresses have many ways of writing
- * the same address.
- *
- * For specifics about IP address comparison, see the following
- * document: http://tools.ietf.org/html/draft-ietf-sip-ipv6-abnf-fix-05
- *
- * \param host1 The domain from the first URI
- * \param host2 THe domain from the second URI
- * \retval 0 The domains match
- * \retval nonzero The domains do not match
- */
-static int sip_uri_domain_cmp(const char *host1, const char *host2)
-{
- struct ast_sockaddr addr1;
- struct ast_sockaddr addr2;
- int addr1_parsed;
- int addr2_parsed;
-
- addr1_parsed = ast_sockaddr_parse(&addr1, host1, 0);
- addr2_parsed = ast_sockaddr_parse(&addr2, host2, 0);
-
- if (addr1_parsed != addr2_parsed) {
- /* One domain was an IP address and the other had
- * a host name. FAIL!
- */
- return 1;
- }
-
- /* Both are host names. A string comparison will work
- * perfectly here. Specifying the "C" locale ensures that
- * The LC_CTYPE conventions use those defined in ANSI C,
- * i.e. ASCII.
- */
- if (!addr1_parsed) {
-#ifdef HAVE_XLOCALE_H
- if(!c_locale) {
- return strcasecmp(host1, host2);
- } else {
- return strcasecmp_l(host1, host2, c_locale);
- }
-#else
- return strcasecmp(host1, host2);
-#endif
- }
-
- /* Both contain IP addresses */
- return ast_sockaddr_cmp(&addr1, &addr2);
-}
-
-int sip_uri_cmp(const char *input1, const char *input2)
-{
- char *uri1;
- char *uri2;
- char *uri_scheme1;
- char *uri_scheme2;
- char *host1;
- char *host2;
- char *params1;
- char *params2;
- char *headers1;
- char *headers2;
-
- /* XXX It would be really nice if we could just use parse_uri_full() here
- * to separate the components of the URI, but unfortunately it is written
- * in a way that can cause URI parameters to be discarded.
- */
-
- if (!input1 || !input2) {
- return 1;
- }
-
- uri1 = ast_strdupa(input1);
- uri2 = ast_strdupa(input2);
-
- ast_uri_decode(uri1, ast_uri_sip_user);
- ast_uri_decode(uri2, ast_uri_sip_user);
-
- uri_scheme1 = strsep(&uri1, ":");
- uri_scheme2 = strsep(&uri2, ":");
-
- if (strcmp(uri_scheme1, uri_scheme2)) {
- return 1;
- }
-
- /* This function is tailored for SIP and SIPS URIs. There's no
- * need to check uri_scheme2 since we have determined uri_scheme1
- * and uri_scheme2 are equivalent already.
- */
- if (strcmp(uri_scheme1, "sip") && strcmp(uri_scheme1, "sips")) {
- return 1;
- }
-
- if (ast_strlen_zero(uri1) || ast_strlen_zero(uri2)) {
- return 1;
- }
-
- if ((host1 = strchr(uri1, '@'))) {
- *host1++ = '\0';
- }
- if ((host2 = strchr(uri2, '@'))) {
- *host2++ = '\0';
- }
-
- /* Check for mismatched username and passwords. This is the
- * only case-sensitive comparison of a SIP URI
- */
- if ((host1 && !host2) ||
- (host2 && !host1) ||
- (host1 && host2 && strcmp(uri1, uri2))) {
- return 1;
- }
-
- if (!host1) {
- host1 = uri1;
- }
- if (!host2) {
- host2 = uri2;
- }
-
- /* Strip off the parameters and headers so we can compare
- * host and port
- */
-
- if ((params1 = strchr(host1, ';'))) {
- *params1++ = '\0';
- }
- if ((params2 = strchr(host2, ';'))) {
- *params2++ = '\0';
- }
-
- /* Headers come after parameters, but there may be headers without
- * parameters, thus the S_OR
- */
- if ((headers1 = strchr(S_OR(params1, host1), '?'))) {
- *headers1++ = '\0';
- }
- if ((headers2 = strchr(S_OR(params2, host2), '?'))) {
- *headers2++ = '\0';
- }
-
- if (sip_uri_domain_cmp(host1, host2)) {
- return 1;
- }
-
- /* Headers have easier rules to follow, so do those first */
- if (sip_uri_headers_cmp(headers1, headers2)) {
- return 1;
- }
-
- /* And now the parameters. Ugh */
- return sip_uri_params_cmp(params1, params2);
-}
-
-#define URI_CMP_MATCH 0
-#define URI_CMP_NOMATCH 1
-
-#ifdef TEST_FRAMEWORK
-AST_TEST_DEFINE(sip_uri_cmp_test)
-{
- static const struct {
- const char *uri1;
- const char *uri2;
- int expected_result;
- } uri_cmp_tests [] = {
- /* These are identical, so they match */
- { "sip:bob@example.com", "sip:bob@example.com", URI_CMP_MATCH },
- /* Different usernames. No match */
- { "sip:alice@example.com", "sip:bob@example.com", URI_CMP_NOMATCH },
- /* Different hosts. No match */
- { "sip:bob@example.com", "sip:bob@examplez.com", URI_CMP_NOMATCH },
- /* Now start using IP addresses. Identical, so they match */
- { "sip:bob@1.2.3.4", "sip:bob@1.2.3.4", URI_CMP_MATCH },
- /* Two identical IPv4 addresses represented differently. Match */
- { "sip:bob@1.2.3.4", "sip:bob@001.002.003.004", URI_CMP_MATCH },
- /* Logically equivalent IPv4 Address and hostname. No Match */
- { "sip:bob@127.0.0.1", "sip:bob@localhost", URI_CMP_NOMATCH },
- /* Logically equivalent IPv6 address and hostname. No Match */
- { "sip:bob@[::1]", "sip:bob@localhost", URI_CMP_NOMATCH },
- /* Try an IPv6 one as well */
- { "sip:bob@[2001:db8::1234]", "sip:bob@[2001:db8::1234]", URI_CMP_MATCH },
- /* Two identical IPv6 addresses represented differently. Match */
- { "sip:bob@[2001:db8::1234]", "sip:bob@[2001:0db8::1234]", URI_CMP_MATCH },
- /* Different ports. No match */
- { "sip:bob@1.2.3.4:5060", "sip:bob@1.2.3.4:5061", URI_CMP_NOMATCH },
- /* Same port logically, but only one address specifies it. No match */
- { "sip:bob@1.2.3.4:5060", "sip:bob@1.2.3.4", URI_CMP_NOMATCH },
- /* And for safety, try with IPv6 */
- { "sip:bob@[2001:db8:1234]:5060", "sip:bob@[2001:db8:1234]", URI_CMP_NOMATCH },
- /* User comparison is case sensitive. No match */
- { "sip:bob@example.com", "sip:BOB@example.com", URI_CMP_NOMATCH },
- /* Host comparison is case insensitive. Match */
- { "sip:bob@example.com", "sip:bob@EXAMPLE.COM", URI_CMP_MATCH },
- /* Add headers to the URI. Identical, so they match */
- { "sip:bob@example.com?header1=value1&header2=value2", "sip:bob@example.com?header1=value1&header2=value2", URI_CMP_MATCH },
- /* Headers in URI 1 are not in URI 2. No Match */
- { "sip:bob@example.com?header1=value1&header2=value2", "sip:bob@example.com", URI_CMP_NOMATCH },
- /* Header present in both URIs does not have matching values. No match */
- { "sip:bob@example.com?header1=value1&header2=value2", "sip:bob@example.com?header1=value1&header2=value3", URI_CMP_NOMATCH },
- /* Add parameters to the URI. Identical so they match */
- { "sip:bob@example.com;param1=value1;param2=value2", "sip:bob@example.com;param1=value1;param2=value2", URI_CMP_MATCH },
- /* Same parameters in both URIs but appear in different order. Match */
- { "sip:bob@example.com;param2=value2;param1=value1", "sip:bob@example.com;param1=value1;param2=value2", URI_CMP_MATCH },
- /* params in URI 1 are not in URI 2. Match */
- { "sip:bob@example.com;param1=value1;param2=value2", "sip:bob@example.com", URI_CMP_MATCH },
- /* param present in both URIs does not have matching values. No match */
- { "sip:bob@example.com;param1=value1;param2=value2", "sip:bob@example.com;param1=value1;param2=value3", URI_CMP_NOMATCH },
- /* URI 1 has a maddr param but URI 2 does not. No match */
- { "sip:bob@example.com;param1=value1;maddr=192.168.0.1", "sip:bob@example.com;param1=value1", URI_CMP_NOMATCH },
- /* URI 1 and URI 2 both have identical maddr params. Match */
- { "sip:bob@example.com;param1=value1;maddr=192.168.0.1", "sip:bob@example.com;param1=value1;maddr=192.168.0.1", URI_CMP_MATCH },
- /* URI 1 is a SIPS URI and URI 2 is a SIP URI. No Match */
- { "sips:bob@example.com", "sip:bob@example.com", URI_CMP_NOMATCH },
- /* No URI schemes. No match */
- { "bob@example.com", "bob@example.com", URI_CMP_NOMATCH },
- /* Crashiness tests. Just an address scheme. No match */
- { "sip", "sips", URI_CMP_NOMATCH },
- /* Still just an address scheme. Even though they're the same, No match */
- { "sip", "sip", URI_CMP_NOMATCH },
- /* Empty strings. No match */
- { "", "", URI_CMP_NOMATCH },
- /* An empty string and a NULL. No match */
- { "", NULL, URI_CMP_NOMATCH },
- };
- int i;
- int test_res = AST_TEST_PASS;
- switch (cmd) {
- case TEST_INIT:
- info->name = "sip_uri_cmp_test";
- info->category = "/channels/chan_sip/";
- info->summary = "Tests comparison of SIP URIs";
- info->description = "Several would-be tricky URI comparisons are performed";
- return AST_TEST_NOT_RUN;
- case TEST_EXECUTE:
- break;
- }
-
- for (i = 0; i < ARRAY_LEN(uri_cmp_tests); ++i) {
- int cmp_res1;
- int cmp_res2;
- if ((cmp_res1 = sip_uri_cmp(uri_cmp_tests[i].uri1, uri_cmp_tests[i].uri2))) {
- /* URI comparison may return -1 or +1 depending on the failure. Standardize
- * the return value to be URI_CMP_NOMATCH on any failure
- */
- cmp_res1 = URI_CMP_NOMATCH;
- }
- if (cmp_res1 != uri_cmp_tests[i].expected_result) {
- ast_test_status_update(test, "Unexpected comparison result for URIs %s and %s. "
- "Expected %s but got %s\n", uri_cmp_tests[i].uri1, uri_cmp_tests[i].uri2,
- uri_cmp_tests[i].expected_result == URI_CMP_MATCH ? "Match" : "No Match",
- cmp_res1 == URI_CMP_MATCH ? "Match" : "No Match");
- test_res = AST_TEST_FAIL;
- }
-
- /* All URI comparisons are commutative, so for the sake of being thorough, we'll
- * rerun the comparison with the parameters reversed
- */
- if ((cmp_res2 = sip_uri_cmp(uri_cmp_tests[i].uri2, uri_cmp_tests[i].uri1))) {
- /* URI comparison may return -1 or +1 depending on the failure. Standardize
- * the return value to be URI_CMP_NOMATCH on any failure
- */
- cmp_res2 = URI_CMP_NOMATCH;
- }
- if (cmp_res2 != uri_cmp_tests[i].expected_result) {
- ast_test_status_update(test, "Unexpected comparison result for URIs %s and %s. "
- "Expected %s but got %s\n", uri_cmp_tests[i].uri2, uri_cmp_tests[i].uri1,
- uri_cmp_tests[i].expected_result == URI_CMP_MATCH ? "Match" : "No Match",
- cmp_res2 == URI_CMP_MATCH ? "Match" : "No Match");
- test_res = AST_TEST_FAIL;
- }
- }
-
- return test_res;
-}
-#endif
-
-void free_via(struct sip_via *v)
-{
- if (!v) {
- return;
- }
-
- ast_free(v->via);
- ast_free(v);
-}
-
-struct sip_via *parse_via(const char *header)
-{
- struct sip_via *v = ast_calloc(1, sizeof(*v));
- char *via, *parm;
-
- if (!v) {
- return NULL;
- }
-
- v->via = ast_strdup(header);
- v->ttl = 1;
-
- via = v->via;
-
- if (ast_strlen_zero(via)) {
- ast_log(LOG_ERROR, "received request without a Via header\n");
- free_via(v);
- return NULL;
- }
-
- /* seperate the first via-parm */
- via = strsep(&via, ",");
-
- /* chop off sent-protocol */
- v->protocol = strsep(&via, " \t\r\n");
- if (ast_strlen_zero(v->protocol)) {
- ast_log(LOG_ERROR, "missing sent-protocol in Via header\n");
- free_via(v);
- return NULL;
- }
- v->protocol = ast_skip_blanks(v->protocol);
-
- if (via) {
- via = ast_skip_blanks(via);
- }
-
- /* chop off sent-by */
- v->sent_by = strsep(&via, "; \t\r\n");
- if (ast_strlen_zero(v->sent_by)) {
- ast_log(LOG_ERROR, "missing sent-by in Via header\n");
- free_via(v);
- return NULL;
- }
- v->sent_by = ast_skip_blanks(v->sent_by);
-
- /* store the port, we have to handle ipv6 addresses containing ':'
- * characters gracefully */
- if (((parm = strchr(v->sent_by, ']')) && *(++parm) == ':') || (!(parm = strchr(v->sent_by, ']')) && (parm = strchr(v->sent_by, ':')))) {
- char *endptr;
-
- v->port = strtol(++parm, &endptr, 10);
- }
-
- /* evaluate any via-parms */
- while ((parm = strsep(&via, "; \t\r\n"))) {
- char *c;
- if ((c = strstr(parm, "maddr="))) {
- v->maddr = ast_skip_blanks(c + sizeof("maddr=") - 1);
- } else if ((c = strstr(parm, "branch="))) {
- v->branch = ast_skip_blanks(c + sizeof("branch=") - 1);
- } else if ((c = strstr(parm, "ttl="))) {
- char *endptr;
- c = ast_skip_blanks(c + sizeof("ttl=") - 1);
- v->ttl = strtol(c, &endptr, 10);
-
- /* make sure we got a valid ttl value */
- if (c == endptr) {
- v->ttl = 1;
- }
- }
- }
-
- return v;
-}
-
-#ifdef TEST_FRAMEWORK
-AST_TEST_DEFINE(parse_via_test)
-{
- int res = AST_TEST_PASS;
- int i = 1;
- struct sip_via *via;
- struct testdata {
- char *in;
- char *expected_protocol;
- char *expected_branch;
- char *expected_sent_by;
- char *expected_maddr;
- unsigned int expected_port;
- unsigned char expected_ttl;
- int expected_null;
- AST_LIST_ENTRY(testdata) list;
- };
- struct testdata *testdataptr;
- static AST_LIST_HEAD_NOLOCK(testdataliststruct, testdata) testdatalist;
- struct testdata t1 = {
- .in = "SIP/2.0/UDP host:port;branch=thebranch",
- .expected_protocol = "SIP/2.0/UDP",
- .expected_sent_by = "host:port",
- .expected_branch = "thebranch",
- };
- struct testdata t2 = {
- .in = "SIP/2.0/UDP host:port",
- .expected_protocol = "SIP/2.0/UDP",
- .expected_sent_by = "host:port",
- .expected_branch = "",
- };
- struct testdata t3 = {
- .in = "SIP/2.0/UDP",
- .expected_null = 1,
- };
- struct testdata t4 = {
- .in = "BLAH/BLAH/BLAH host:port;branch=",
- .expected_protocol = "BLAH/BLAH/BLAH",
- .expected_sent_by = "host:port",
- .expected_branch = "",
- };
- struct testdata t5 = {
- .in = "SIP/2.0/UDP host:5060;branch=thebranch;maddr=224.0.0.1;ttl=1",
- .expected_protocol = "SIP/2.0/UDP",
- .expected_sent_by = "host:5060",
- .expected_port = 5060,
- .expected_branch = "thebranch",
- .expected_maddr = "224.0.0.1",
- .expected_ttl = 1,
- };
- struct testdata t6 = {
- .in = "SIP/2.0/UDP host:5060;\n branch=thebranch;\r\n maddr=224.0.0.1; ttl=1",
- .expected_protocol = "SIP/2.0/UDP",
- .expected_sent_by = "host:5060",
- .expected_port = 5060,
- .expected_branch = "thebranch",
- .expected_maddr = "224.0.0.1",
- .expected_ttl = 1,
- };
- struct testdata t7 = {
- .in = "SIP/2.0/UDP [::1]:5060",
- .expected_protocol = "SIP/2.0/UDP",
- .expected_sent_by = "[::1]:5060",
- .expected_port = 5060,
- .expected_branch = "",
- };
- switch (cmd) {
- case TEST_INIT:
- info->name = "parse_via_test";
- info->category = "/channels/chan_sip/";
- info->summary = "Tests parsing the Via header";
- info->description =
- "Runs through various test situations in which various "
- " parameters parameter must be extracted from a VIA header";
- return AST_TEST_NOT_RUN;
- case TEST_EXECUTE:
- break;
- }
-
- AST_LIST_HEAD_SET_NOLOCK(&testdatalist, &t1);
- AST_LIST_INSERT_TAIL(&testdatalist, &t2, list);
- AST_LIST_INSERT_TAIL(&testdatalist, &t3, list);
- AST_LIST_INSERT_TAIL(&testdatalist, &t4, list);
- AST_LIST_INSERT_TAIL(&testdatalist, &t5, list);
- AST_LIST_INSERT_TAIL(&testdatalist, &t6, list);
- AST_LIST_INSERT_TAIL(&testdatalist, &t7, list);
-
-
- AST_LIST_TRAVERSE(&testdatalist, testdataptr, list) {
- via = parse_via(testdataptr->in);
- if (!via) {
- if (!testdataptr->expected_null) {
- ast_test_status_update(test, "TEST#%d FAILED: VIA = \"%s\"\n"
- "failed to parse header\n",
- i, testdataptr->in);
- res = AST_TEST_FAIL;
- }
- i++;
- continue;
- }
-
- if (testdataptr->expected_null) {
- ast_test_status_update(test, "TEST#%d FAILED: VIA = \"%s\"\n"
- "successfully parased invalid via header\n",
- i, testdataptr->in);
- res = AST_TEST_FAIL;
- free_via(via);
- i++;
- continue;
- }
-
- if ((ast_strlen_zero(via->protocol) && !ast_strlen_zero(testdataptr->expected_protocol))
- || (!ast_strlen_zero(via->protocol) && strcmp(via->protocol, testdataptr->expected_protocol))) {
-
- ast_test_status_update(test, "TEST#%d FAILED: VIA = \"%s\"\n"
- "parsed protocol = \"%s\"\n"
- "expected = \"%s\"\n"
- "failed to parse protocol\n",
- i, testdataptr->in, via->protocol, testdataptr->expected_protocol);
- res = AST_TEST_FAIL;
- }
-
- if ((ast_strlen_zero(via->sent_by) && !ast_strlen_zero(testdataptr->expected_sent_by))
- || (!ast_strlen_zero(via->sent_by) && strcmp(via->sent_by, testdataptr->expected_sent_by))) {
-
- ast_test_status_update(test, "TEST#%d FAILED: VIA = \"%s\"\n"
- "parsed sent_by = \"%s\"\n"
- "expected = \"%s\"\n"
- "failed to parse sent-by\n",
- i, testdataptr->in, via->sent_by, testdataptr->expected_sent_by);
- res = AST_TEST_FAIL;
- }
-
- if (testdataptr->expected_port && testdataptr->expected_port != via->port) {
- ast_test_status_update(test, "TEST#%d FAILED: VIA = \"%s\"\n"
- "parsed port = \"%u\"\n"
- "expected = \"%u\"\n"
- "failed to parse port\n",
- i, testdataptr->in, via->port, testdataptr->expected_port);
- res = AST_TEST_FAIL;
- }
-
- if ((ast_strlen_zero(via->branch) && !ast_strlen_zero(testdataptr->expected_branch))
- || (!ast_strlen_zero(via->branch) && strcmp(via->branch, testdataptr->expected_branch))) {
-
- ast_test_status_update(test, "TEST#%d FAILED: VIA = \"%s\"\n"
- "parsed branch = \"%s\"\n"
- "expected = \"%s\"\n"
- "failed to parse branch\n",
- i, testdataptr->in, via->branch, testdataptr->expected_branch);
- res = AST_TEST_FAIL;
- }
-
- if ((ast_strlen_zero(via->maddr) && !ast_strlen_zero(testdataptr->expected_maddr))
- || (!ast_strlen_zero(via->maddr) && strcmp(via->maddr, testdataptr->expected_maddr))) {
-
- ast_test_status_update(test, "TEST#%d FAILED: VIA = \"%s\"\n"
- "parsed maddr = \"%s\"\n"
- "expected = \"%s\"\n"
- "failed to parse maddr\n",
- i, testdataptr->in, via->maddr, testdataptr->expected_maddr);
- res = AST_TEST_FAIL;
- }
-
- if (testdataptr->expected_ttl && testdataptr->expected_ttl != via->ttl) {
- ast_test_status_update(test, "TEST#%d FAILED: VIA = \"%s\"\n"
- "parsed ttl = \"%d\"\n"
- "expected = \"%d\"\n"
- "failed to parse ttl\n",
- i, testdataptr->in, via->ttl, testdataptr->expected_ttl);
- res = AST_TEST_FAIL;
- }
-
- free_via(via);
- i++;
- }
- return res;
-}
-#endif
-
-void sip_request_parser_register_tests(void)
-{
- AST_TEST_REGISTER(get_calleridname_test);
- AST_TEST_REGISTER(sip_parse_uri_test);
- AST_TEST_REGISTER(get_in_brackets_test);
- AST_TEST_REGISTER(get_name_and_number_test);
- AST_TEST_REGISTER(sip_parse_uri_full_test);
- AST_TEST_REGISTER(parse_name_andor_addr_test);
- AST_TEST_REGISTER(parse_contact_header_test);
- AST_TEST_REGISTER(sip_parse_options_test);
- AST_TEST_REGISTER(sip_uri_cmp_test);
- AST_TEST_REGISTER(parse_via_test);
-}
-void sip_request_parser_unregister_tests(void)
-{
- AST_TEST_UNREGISTER(sip_parse_uri_test);
- AST_TEST_UNREGISTER(get_calleridname_test);
- AST_TEST_UNREGISTER(get_in_brackets_test);
- AST_TEST_UNREGISTER(get_name_and_number_test);
- AST_TEST_UNREGISTER(sip_parse_uri_full_test);
- AST_TEST_UNREGISTER(parse_name_andor_addr_test);
- AST_TEST_UNREGISTER(parse_contact_header_test);
- AST_TEST_UNREGISTER(sip_parse_options_test);
- AST_TEST_UNREGISTER(sip_uri_cmp_test);
- AST_TEST_UNREGISTER(parse_via_test);
-}
-
-int sip_reqresp_parser_init(void)
-{
-#ifdef HAVE_XLOCALE_H
- c_locale = newlocale(LC_CTYPE_MASK, "C", NULL);
- if (!c_locale) {
- return -1;
- }
-#endif
- return 0;
-}
-
-void sip_reqresp_parser_exit(void)
-{
-#ifdef HAVE_XLOCALE_H
- if (c_locale) {
- freelocale(c_locale);
- c_locale = NULL;
- }
-#endif
-}
diff --git a/channels/sip/route.c b/channels/sip/route.c
deleted file mode 100644
index 916c3afe43..0000000000
--- a/channels/sip/route.c
+++ /dev/null
@@ -1,203 +0,0 @@
-/*
- * Asterisk -- An open source telephony toolkit.
- *
- * Copyright (C) 2013, Digium, Inc.
- *
- * See http://www.asterisk.org for more information about
- * the Asterisk project. Please do not directly contact
- * any of the maintainers of this project for assistance;
- * the project provides a web site, mailing lists and IRC
- * channels for your use.
- *
- * This program is free software, distributed under the terms of
- * the GNU General Public License Version 2. See the LICENSE file
- * at the top of the source tree.
- */
-
-/*!
- * \file
- * \brief sip_route functions
- */
-
-/*** MODULEINFO
- deprecated
- ***/
-
-#include "asterisk.h"
-
-#include "asterisk/utils.h"
-
-#include "include/route.h"
-#include "include/reqresp_parser.h"
-
-/*!
- * \brief Traverse route hops
- */
-#define sip_route_traverse(route,elem) AST_LIST_TRAVERSE(&(route)->list, elem, list)
-#define sip_route_first(route) AST_LIST_FIRST(&(route)->list)
-
-/*!
- * \brief Structure to save a route hop
- */
-struct sip_route_hop {
- AST_LIST_ENTRY(sip_route_hop) list;
- char uri[0];
-};
-
-const char *sip_route_add(struct sip_route *route, const char *uri, size_t len, int inserthead)
-{
- struct sip_route_hop *hop;
-
- if (!uri || len < 1 || uri[0] == '\0') {
- return NULL;
- }
-
- /* Expand len to include null terminator */
- len++;
-
- /* ast_calloc is not needed because all fields are initialized in this block */
- hop = ast_malloc(sizeof(struct sip_route_hop) + len);
- if (!hop) {
- return NULL;
- }
- ast_copy_string(hop->uri, uri, len);
-
- if (inserthead) {
- AST_LIST_INSERT_HEAD(&route->list, hop, list);
- route->type = route_invalidated;
- } else {
- if (sip_route_empty(route)) {
- route->type = route_invalidated;
- }
- AST_LIST_INSERT_TAIL(&route->list, hop, list);
- hop->list.next = NULL;
- }
-
- return hop->uri;
-}
-
-void sip_route_process_header(struct sip_route *route, const char *header, int inserthead)
-{
- const char *hop;
- int len = 0;
- const char *uri;
-
- if (!route) {
- ast_log(LOG_ERROR, "sip_route_process_header requires non-null route");
- ast_do_crash();
- return;
- }
-
- while (!get_in_brackets_const(header, &uri, &len)) {
- header = strchr(header, ',');
- if (header >= uri && header <= (uri + len)) {
- /* comma inside brackets */
- const char *next_br = strchr(header, '<');
- if (next_br && next_br <= (uri + len)) {
- header++;
- continue;
- }
- continue;
- }
- if ((hop = sip_route_add(route, uri, len, inserthead))) {
- ast_debug(2, "sip_route_process_header: <%s>\n", hop);
- }
- header = strchr(uri + len + 1, ',');
- if (header == NULL) {
- /* No more field-values, we're done with this header */
- break;
- }
- /* Advance past comma */
- header++;
- }
-}
-
-void sip_route_copy(struct sip_route *dst, const struct sip_route *src)
-{
- struct sip_route_hop *hop;
-
- /* make sure dst is empty */
- sip_route_clear(dst);
-
- sip_route_traverse(src, hop) {
- const char *uri = sip_route_add(dst, hop->uri, strlen(hop->uri), 0);
- if (uri) {
- ast_debug(2, "sip_route_copy: copied hop: <%s>\n", uri);
- }
- }
-
- dst->type = src->type;
-}
-
-void sip_route_clear(struct sip_route *route)
-{
- struct sip_route_hop *hop;
-
- while ((hop = AST_LIST_REMOVE_HEAD(&route->list, list))) {
- ast_free(hop);
- }
-
- route->type = route_loose;
-}
-
-void sip_route_dump(const struct sip_route *route)
-{
- if (sip_route_empty(route)) {
- ast_verbose("sip_route_dump: no route/path\n");
- } else {
- struct sip_route_hop *hop;
- sip_route_traverse(route, hop) {
- ast_verbose("sip_route_dump: route/path hop: <%s>\n", hop->uri);
- }
- }
-}
-
-struct ast_str *sip_route_list(const struct sip_route *route, int formatcli, int skip)
-{
- struct sip_route_hop *hop;
- const char *comma;
- struct ast_str *buf;
- int i = 0 - skip;
-
- buf = ast_str_create(64);
- if (!buf) {
- return NULL;
- }
-
- comma = formatcli ? ", " : ",";
-
- sip_route_traverse(route, hop) {
- if (i >= 0) {
- ast_str_append(&buf, 0, "%s<%s>", i ? comma : "", hop->uri);
- }
- i++;
- }
-
- if (formatcli && i <= 0) {
- ast_str_append(&buf, 0, "N/A");
- }
-
- return buf;
-}
-
-int sip_route_is_strict(struct sip_route *route)
-{
- if (!route) {
- return 0;
- }
-
- if (route->type == route_invalidated) {
- struct sip_route_hop *hop = sip_route_first(route);
- int ret = hop && (strstr(hop->uri, ";lr") == NULL);
- route->type = ret ? route_strict : route_loose;
- return ret;
- }
-
- return (route->type == route_strict) ? 1 : 0;
-}
-
-const char *sip_route_first_uri(const struct sip_route *route)
-{
- struct sip_route_hop *hop = sip_route_first(route);
- return hop ? hop->uri : NULL;
-}
diff --git a/channels/sip/security_events.c b/channels/sip/security_events.c
deleted file mode 100644
index eabc9045f4..0000000000
--- a/channels/sip/security_events.c
+++ /dev/null
@@ -1,358 +0,0 @@
-/*
- * Asterisk -- An open source telephony toolkit.
- *
- * Copyright (C) 2012, Digium, Inc.
- *
- * Michael L. Young
- *
- * See http://www.asterisk.org for more information about
- * the Asterisk project. Please do not directly contact
- * any of the maintainers of this project for assistance;
- * the project provides a web site, mailing lists and IRC
- * channels for your use.
- *
- * This program is free software, distributed under the terms of
- * the GNU General Public License Version 2. See the LICENSE file
- * at the top of the source tree.
- */
-
-/*!
- * \file
- *
- * \brief Generate security events in the SIP channel
- *
- * \author Michael L. Young
- */
-
-/*** MODULEINFO
- deprecated
- ***/
-
-#include "asterisk.h"
-
-#include "include/sip.h"
-#include "include/security_events.h"
-
-/*! \brief Determine transport type used to receive request*/
-
-static enum ast_transport security_event_get_transport(const struct sip_pvt *p)
-{
- return p->socket.type;
-}
-
-void sip_report_invalid_peer(const struct sip_pvt *p)
-{
- char session_id[32];
-
- struct ast_security_event_inval_acct_id inval_acct_id = {
- .common.event_type = AST_SECURITY_EVENT_INVAL_ACCT_ID,
- .common.version = AST_SECURITY_EVENT_INVAL_ACCT_ID_VERSION,
- .common.service = "SIP",
- .common.account_id = p->exten,
- .common.local_addr = {
- .addr = &p->ourip,
- .transport = security_event_get_transport(p)
- },
- .common.remote_addr = {
- .addr = &p->sa,
- .transport = security_event_get_transport(p)
- },
- .common.session_id = session_id,
- };
-
- snprintf(session_id, sizeof(session_id), "%p", p);
-
- ast_security_event_report(AST_SEC_EVT(&inval_acct_id));
-}
-
-void sip_report_failed_acl(const struct sip_pvt *p, const char *aclname)
-{
- char session_id[32];
-
- struct ast_security_event_failed_acl failed_acl_event = {
- .common.event_type = AST_SECURITY_EVENT_FAILED_ACL,
- .common.version = AST_SECURITY_EVENT_FAILED_ACL_VERSION,
- .common.service = "SIP",
- .common.account_id = p->exten,
- .common.local_addr = {
- .addr = &p->ourip,
- .transport = security_event_get_transport(p)
- },
- .common.remote_addr = {
- .addr = &p->sa,
- .transport = security_event_get_transport(p)
- },
- .common.session_id = session_id,
- .acl_name = aclname,
- };
-
- snprintf(session_id, sizeof(session_id), "%p", p);
-
- ast_security_event_report(AST_SEC_EVT(&failed_acl_event));
-}
-
-void sip_report_inval_password(const struct sip_pvt *p, const char *response_challenge, const char *response_hash)
-{
- char session_id[32];
-
- struct ast_security_event_inval_password inval_password = {
- .common.event_type = AST_SECURITY_EVENT_INVAL_PASSWORD,
- .common.version = AST_SECURITY_EVENT_INVAL_PASSWORD_VERSION,
- .common.service = "SIP",
- .common.account_id = p->exten,
- .common.local_addr = {
- .addr = &p->ourip,
- .transport = security_event_get_transport(p)
- },
- .common.remote_addr = {
- .addr = &p->sa,
- .transport = security_event_get_transport(p)
- },
- .common.session_id = session_id,
-
- .challenge = p->nonce,
- .received_challenge = response_challenge,
- .received_hash = response_hash,
- };
-
- snprintf(session_id, sizeof(session_id), "%p", p);
-
- ast_security_event_report(AST_SEC_EVT(&inval_password));
-}
-
-void sip_report_auth_success(const struct sip_pvt *p, uint32_t using_password)
-{
- char session_id[32];
-
- struct ast_security_event_successful_auth successful_auth = {
- .common.event_type = AST_SECURITY_EVENT_SUCCESSFUL_AUTH,
- .common.version = AST_SECURITY_EVENT_SUCCESSFUL_AUTH_VERSION,
- .common.service = "SIP",
- .common.account_id = p->exten,
- .common.local_addr = {
- .addr = &p->ourip,
- .transport = security_event_get_transport(p)
- },
- .common.remote_addr = {
- .addr = &p->sa,
- .transport = security_event_get_transport(p)
- },
- .common.session_id = session_id,
- .using_password = using_password,
- };
-
- snprintf(session_id, sizeof(session_id), "%p", p);
-
- ast_security_event_report(AST_SEC_EVT(&successful_auth));
-}
-
-void sip_report_session_limit(const struct sip_pvt *p)
-{
- char session_id[32];
-
- struct ast_security_event_session_limit session_limit = {
- .common.event_type = AST_SECURITY_EVENT_SESSION_LIMIT,
- .common.version = AST_SECURITY_EVENT_SESSION_LIMIT_VERSION,
- .common.service = "SIP",
- .common.account_id = p->exten,
- .common.local_addr = {
- .addr = &p->ourip,
- .transport = security_event_get_transport(p)
- },
- .common.remote_addr = {
- .addr = &p->sa,
- .transport = security_event_get_transport(p)
- },
- .common.session_id = session_id,
- };
-
- snprintf(session_id, sizeof(session_id), "%p", p);
-
- ast_security_event_report(AST_SEC_EVT(&session_limit));
-}
-
-void sip_report_failed_challenge_response(const struct sip_pvt *p, const char *response, const char *expected_response)
-{
- char session_id[32];
- char account_id[256];
-
- struct ast_security_event_chal_resp_failed chal_resp_failed = {
- .common.event_type = AST_SECURITY_EVENT_CHAL_RESP_FAILED,
- .common.version = AST_SECURITY_EVENT_CHAL_RESP_FAILED_VERSION,
- .common.service = "SIP",
- .common.account_id = account_id,
- .common.local_addr = {
- .addr = &p->ourip,
- .transport = security_event_get_transport(p)
- },
- .common.remote_addr = {
- .addr = &p->sa,
- .transport = security_event_get_transport(p)
- },
- .common.session_id = session_id,
-
- .challenge = p->nonce,
- .response = response,
- .expected_response = expected_response,
- };
-
- if (!ast_strlen_zero(p->from)) { /* When dialing, show account making call */
- ast_copy_string(account_id, p->from, sizeof(account_id));
- } else {
- ast_copy_string(account_id, p->exten, sizeof(account_id));
- }
-
- snprintf(session_id, sizeof(session_id), "%p", p);
-
- ast_security_event_report(AST_SEC_EVT(&chal_resp_failed));
-}
-
-void sip_report_chal_sent(const struct sip_pvt *p)
-{
- char session_id[32];
- char account_id[256];
-
- struct ast_security_event_chal_sent chal_sent = {
- .common.event_type = AST_SECURITY_EVENT_CHAL_SENT,
- .common.version = AST_SECURITY_EVENT_CHAL_SENT_VERSION,
- .common.service = "SIP",
- .common.account_id = account_id,
- .common.local_addr = {
- .addr = &p->ourip,
- .transport = security_event_get_transport(p)
- },
- .common.remote_addr = {
- .addr = &p->sa,
- .transport = security_event_get_transport(p)
- },
- .common.session_id = session_id,
-
- .challenge = p->nonce,
- };
-
- if (!ast_strlen_zero(p->from)) { /* When dialing, show account making call */
- ast_copy_string(account_id, p->from, sizeof(account_id));
- } else {
- ast_copy_string(account_id, p->exten, sizeof(account_id));
- }
-
- snprintf(session_id, sizeof(session_id), "%p", p);
-
- ast_security_event_report(AST_SEC_EVT(&chal_sent));
-}
-
-void sip_report_inval_transport(const struct sip_pvt *p, const char *transport)
-{
- char session_id[32];
-
- struct ast_security_event_inval_transport inval_transport = {
- .common.event_type = AST_SECURITY_EVENT_INVAL_TRANSPORT,
- .common.version = AST_SECURITY_EVENT_INVAL_TRANSPORT_VERSION,
- .common.service = "SIP",
- .common.account_id = p->exten,
- .common.local_addr = {
- .addr = &p->ourip,
- .transport = security_event_get_transport(p)
- },
- .common.remote_addr = {
- .addr = &p->sa,
- .transport = security_event_get_transport(p)
- },
- .common.session_id = session_id,
-
- .transport = transport,
- };
-
- snprintf(session_id, sizeof(session_id), "%p", p);
-
- ast_security_event_report(AST_SEC_EVT(&inval_transport));
-}
-
-int sip_report_security_event(const char *peer, struct ast_sockaddr *addr, const struct sip_pvt *p,
- const struct sip_request *req, const int res)
-{
-
- struct sip_peer *peer_report;
- enum check_auth_result res_report = res;
- struct ast_str *buf;
- char *c;
- const char *authtoken;
- char *reqheader, *respheader;
- int result = 0;
- char aclname[256];
- struct digestkeys keys[] = {
- [K_RESP] = { "response=", "" },
- [K_URI] = { "uri=", "" },
- [K_USER] = { "username=", "" },
- [K_NONCE] = { "nonce=", "" },
- [K_LAST] = { NULL, NULL}
- };
-
- peer_report = sip_find_peer(peer, addr, TRUE, FINDPEERS, FALSE, p->socket.type);
-
- switch(res_report) {
- case AUTH_DONT_KNOW:
- break;
- case AUTH_SUCCESSFUL:
- if (peer_report) {
- if (ast_strlen_zero(peer_report->secret) && ast_strlen_zero(peer_report->md5secret)) {
- sip_report_auth_success(p, 0);
- } else {
- sip_report_auth_success(p, 1);
- }
- }
- break;
- case AUTH_CHALLENGE_SENT:
- sip_report_chal_sent(p);
- break;
- case AUTH_SECRET_FAILED:
- case AUTH_USERNAME_MISMATCH:
- sip_auth_headers(WWW_AUTH, &respheader, &reqheader);
- authtoken = sip_get_header(req, reqheader);
- buf = ast_str_thread_get(&check_auth_buf, CHECK_AUTH_BUF_INITLEN);
- ast_str_set(&buf, 0, "%s", authtoken);
- c = ast_str_buffer(buf);
-
- sip_digest_parser(c, keys);
-
- if (res_report == AUTH_SECRET_FAILED) {
- sip_report_inval_password(p, keys[K_NONCE].s, keys[K_RESP].s);
- } else {
- if (peer_report) {
- sip_report_failed_challenge_response(p, keys[K_USER].s, peer_report->username);
- }
- }
- break;
- case AUTH_NOT_FOUND:
- /* with sip_cfg.alwaysauthreject on, generates 2 events */
- sip_report_invalid_peer(p);
- break;
- case AUTH_UNKNOWN_DOMAIN:
- snprintf(aclname, sizeof(aclname), "domain_must_match");
- sip_report_failed_acl(p, aclname);
- break;
- case AUTH_PEER_NOT_DYNAMIC:
- snprintf(aclname, sizeof(aclname), "peer_not_dynamic");
- sip_report_failed_acl(p, aclname);
- break;
- case AUTH_ACL_FAILED:
- /* with sip_cfg.alwaysauthreject on, generates 2 events */
- snprintf(aclname, sizeof(aclname), "device_must_match_acl");
- sip_report_failed_acl(p, aclname);
- break;
- case AUTH_BAD_TRANSPORT:
- sip_report_inval_transport(p, sip_get_transport(req->socket.type));
- break;
- case AUTH_RTP_FAILED:
- break;
- case AUTH_SESSION_LIMIT:
- sip_report_session_limit(p);
- break;
- }
-
- if (peer_report) {
- sip_unref_peer(peer_report, "sip_report_security_event: sip_unref_peer: from handle_incoming");
- }
-
- return result;
-}
diff --git a/channels/sip/utils.c b/channels/sip/utils.c
deleted file mode 100644
index d53316cf15..0000000000
--- a/channels/sip/utils.c
+++ /dev/null
@@ -1,49 +0,0 @@
-/*
- * Asterisk -- An open source telephony toolkit.
- *
- * Copyright (C) 1999 - 2012, Digium, Inc.
- *
- * See http://www.asterisk.org for more information about
- * the Asterisk project. Please do not directly contact
- * any of the maintainers of this project for assistance;
- * the project provides a web site, mailing lists and IRC
- * channels for your use.
- *
- * This program is free software, distributed under the terms of
- * the GNU General Public License Version 2. See the LICENSE file
- * at the top of the source tree.
- */
-
-/*!
- * \file
- * \brief Utility functions for chan_sip
- *
- * \author Terry Wilson
- */
-
-/*** MODULEINFO
- deprecated
- ***/
-
-#include "asterisk.h"
-
-#include "asterisk/utils.h"
-#include "asterisk/cli.h"
-#include "include/sip.h"
-#include "include/sip_utils.h"
-
-const char *force_rport_string(struct ast_flags *flags)
-{
- if (ast_test_flag(&flags[2], SIP_PAGE3_NAT_AUTO_RPORT)) {
- return ast_test_flag(&flags[0], SIP_NAT_FORCE_RPORT) ? "Auto (Yes)" : "Auto (No)";
- }
- return AST_CLI_YESNO(ast_test_flag(&flags[0], SIP_NAT_FORCE_RPORT));
-}
-
-const char *comedia_string(struct ast_flags *flags)
-{
- if (ast_test_flag(&flags[2], SIP_PAGE3_NAT_AUTO_COMEDIA)) {
- return ast_test_flag(&flags[1], SIP_PAGE2_SYMMETRICRTP) ? "Auto (Yes)" : "Auto (No)";
- }
- return AST_CLI_YESNO(ast_test_flag(&flags[1], SIP_PAGE2_SYMMETRICRTP));
-}
diff --git a/configs/samples/cli.conf.sample b/configs/samples/cli.conf.sample
index 0ddd92c998..f6a546ac82 100644
--- a/configs/samples/cli.conf.sample
+++ b/configs/samples/cli.conf.sample
@@ -7,6 +7,5 @@
; Any commands listed in this section will get automatically executed
; when Asterisk starts as a daemon or foreground process (-c).
;
-;sip set debug on = yes
;core set verbose 3 = yes
;core set debug 1 = yes
diff --git a/configs/samples/cli_aliases.conf.sample b/configs/samples/cli_aliases.conf.sample
index ba85594f5a..4b98f18c8e 100644
--- a/configs/samples/cli_aliases.conf.sample
+++ b/configs/samples/cli_aliases.conf.sample
@@ -137,8 +137,6 @@ show queue=queue show
add queue member=queue add member
remove queue member=queue remove member
ael no debug=ael nodebug
-sip debug=sip set debug
-sip no debug=sip set debug off
show voicemail users=voicemail show users
show voicemail zones=voicemail show zones
iax2 trunk debug=iax2 set debug trunk
@@ -173,7 +171,6 @@ core set chanvar=dialplan set chanvar
agi dumphtml=agi dump html
ael debug=ael set debug
funcdevstate list=devstate list
-sip history=sip set history on
abort shutdown=core abort shutdown
stop now=core stop now
stop gracefully=core stop gracefully
diff --git a/configs/samples/cli_permissions.conf.sample b/configs/samples/cli_permissions.conf.sample
index 9f69e1c9ab..8632a72c0e 100644
--- a/configs/samples/cli_permissions.conf.sample
+++ b/configs/samples/cli_permissions.conf.sample
@@ -30,7 +30,7 @@ default_perm=permit ; To leave asterisk working as normal
; This list is read in the sequence that is being written, so
; In this example the user 'eliel' is allow to run only the following
; commands:
-; sip show peer
+; pjsip show endpoints
; core set debug
; core set verbose
; If the user is not specified, the default_perm option will be apply to
@@ -39,14 +39,14 @@ default_perm=permit ; To leave asterisk working as normal
; Notice that you can also use regular expressions to allow or deny access to a
; certain command like: 'core show application D*'. In this example the user will be
; allowed to view the documentation for all the applications starting with 'D'.
-; Another regular expression could be: 'channel originate SIP/[0-9]* extension *'
-; allowing the user to use 'channel originate' on a sip channel and with the 'extension'
+; Another regular expression could be: 'channel originate PJSIP/[0-9]* extension *'
+; allowing the user to use 'channel originate' on a pjsip channel and with the 'extension'
; parameter and avoiding the use of the 'application' parameter.
;
; We can also use the templates syntax:
; [supportTemplate](!)
; deny=all
-; permit=sip show ; all commands starting with 'sip show' will be allowed
+; permit=pjsip show ; all commands starting with 'pjsip show' will be allowed
; permit=core show
;
; You can specify permissions for a local group instead of a user,
@@ -55,20 +55,20 @@ default_perm=permit ; To leave asterisk working as normal
;
;[@adm]
;deny=all
-;permit=sip
+;permit=pjsip
;permit=core
;
;
;[eliel]
;deny=all
-;permit=sip show peer
-;deny=sip show peers
+;permit=pjsip show endpoint
+;deny=pjsip show endpoints
;permit=core set
;
;
;User 'tommy' inherits from template 'supportTemplate':
; deny=all
-; permit=sip show
+; permit=pjsip show
; permit=core show
;[tommy](supportTemplate)
;permit=core set debug
diff --git a/configs/samples/codecs.conf.sample b/configs/samples/codecs.conf.sample
index ef5a2f8cf5..c1fd077c7b 100644
--- a/configs/samples/codecs.conf.sample
+++ b/configs/samples/codecs.conf.sample
@@ -83,10 +83,10 @@ genericplc_on_equal_codecs => false
; Once this config file is loaded, silk8 can be used anywhere a
; peer's codec capabilities are defined.
;
-; In sip.conf 'silk8' can be defined as a capability for a peer.
-; [peer1]
+; In pjsip.conf 'silk8' can be defined as a capability for an endpoint.
+; [endpoint1]
; type=peer
-; host=dynamic
+; aor=endpoint1
; disallow=all
; allow=silk8 ;custom codec defined in codecs.conf
;
diff --git a/configs/samples/extconfig.conf.sample b/configs/samples/extconfig.conf.sample
index 0e53823ddf..f5de687325 100644
--- a/configs/samples/extconfig.conf.sample
+++ b/configs/samples/extconfig.conf.sample
@@ -76,8 +76,6 @@
;
;iaxusers => odbc,asterisk
;iaxpeers => odbc,asterisk
-;sippeers => odbc,asterisk
-;sipregs => odbc,asterisk ; (avoid sipregs if possible, e.g. by using a view)
;ps_endpoints => odbc,asterisk
;ps_auths => odbc,asterisk
;ps_aors => odbc,asterisk
diff --git a/configs/samples/extensions.ael.sample b/configs/samples/extensions.ael.sample
index ba8ce7eb9f..ffa3cdd25d 100644
--- a/configs/samples/extensions.ael.sample
+++ b/configs/samples/extensions.ael.sample
@@ -129,7 +129,7 @@ context ael-dundi-e164-customers {
//
// If you are an ITSP or Reseller, list your customers here.
//
- //_12564286000 => Dial(SIP/customer1);
+ //_12564286000 => Dial(PJSIP/customer1);
//_12564286001 => Dial(IAX2/customer2);
};
@@ -143,7 +143,7 @@ context ael-dundi-e164-via-pstn {
context ael-dundi-e164-local {
//
- // Context to put your dundi IAX2 or SIP user in for
+ // Context to put your dundi or IAX2 user in for
// full access
//
includes {
@@ -396,11 +396,8 @@ context ael-default {
};
//
// Extensions like the two below can be used for FWD, Nikotel, sipgate etc.
-// Note that you must have a [sipprovider] section in sip.conf whereas
-// the otherprovider.net example does not require such a peer definition
//
-//_41X. => Dial(SIP/${EXTEN:2}@sipprovider,,r);
-//_42X. => Dial(SIP/user:passwd@${EXTEN:2}@otherprovider.net,30,rT);
+//_42X. => Dial(PJSIP/user:passwd@${EXTEN:2}@otherprovider.net,30,rT);
// Real extensions would go here. Generally you want real extensions to be
// 4 or 5 digits long (although there is no such requirement) and start with a
@@ -409,8 +406,8 @@ context ael-default {
// them with names, too, and use global variables
// 6245 => {
-// hint(SIP/Grandstream1&SIP/Xlite1,Joe Schmoe); // Channel hints for presence
-// Dial(SIP/Grandstream1,20,rt); // permit transfer
+// hint(PJSIP/Grandstream1&PJSIP/Xlite1,Joe Schmoe); // Channel hints for presence
+// Dial(PJSIP/Grandstream1,20,rt); // permit transfer
// Dial(${HINT}/5245},20,rtT); // Use hint as listed
// switch(${DIALSTATUS}) {
// case BUSY:
diff --git a/configs/samples/extensions.conf.sample b/configs/samples/extensions.conf.sample
index e5fc5e1cf8..893c3ea223 100644
--- a/configs/samples/extensions.conf.sample
+++ b/configs/samples/extensions.conf.sample
@@ -426,14 +426,14 @@ exten => _X!,1,Verbose(2,Performing ISN lookup for ${EXTEN})
same => n,GotoIf($["${FILTER(0-9,${SUFFIX})}" != "${SUFFIX}"]?fn-CONGESTION,1)
; filter out bad characters per the README-SERIOUSLY.best-practices.txt document
same => n,Set(TIMEOUT(absolute)=10800)
- same => n,Set(isnresult=${ENUMLOOKUP(${EXTEN},sip,,1,freenum.org)}) ; perform our lookup with freenum.org
+ same => n,Set(isnresult=${ENUMLOOKUP(${EXTEN},pjsip,,1,freenum.org)}) ; perform our lookup with freenum.org
same => n,GotoIf($["${isnresult}" != ""]?from)
same => n,Set(DIALSTATUS=CONGESTION)
same => n,Goto(fn-CONGESTION,1)
same => n(from),Set(__SIPFROMUSER=${CALLERID(num)})
same => n,GotoIf($["${GLOBAL(FREENUMDOMAIN)}" = ""]?dial) ; check if we set the FREENUMDOMAIN global variable in [global]
same => n,Set(__SIPFROMDOMAIN=${GLOBAL(FREENUMDOMAIN)}) ; if we did set it, then we'll use it for our outbound dialing domain
- same => n(dial),Dial(SIP/${isnresult},40)
+ same => n(dial),Dial(PJSIP/${isnresult},40)
same => n,Goto(fn-${DIALSTATUS},1)
exten => fn-BUSY,1,Busy()
@@ -661,9 +661,7 @@ exten => _X.,1,Gosub(sub-page,s,1(SIP/${EXTEN}))
[public]
;
-; ATTENTION: If your Asterisk is connected to the internet and you do
-; not have allowguest=no in sip.conf, everybody out there may use your
-; public context without authentication. In that case you want to
+; ATTENTION: If your Asterisk is connected to the internet,
; double check which services you offer to the world.
;
include => demo
@@ -675,20 +673,14 @@ include => demo
;
include => demo
-;
-; An extension like the one below can be used for FWD, Nikotel, sipgate etc.
-; Note that you must have a [sipprovider] section in sip.conf
-;
-;exten => _41X.,1,Dial(SIP/${FILTER(0-9,${EXTEN:2})}@sipprovider,,r)
-
; Real extensions would go here. Generally you want real extensions to be
; 4 or 5 digits long (although there is no such requirement) and start with a
; single digit that is fairly large (like 6 or 7) so that you have plenty of
; room to overlap extensions and menu options without conflict. You can alias
; them with names, too, and use global variables
-;exten => 6245,hint,SIP/Grandstream1&SIP/Xlite1(Joe Schmoe) ; Channel hints for presence
-;exten => 6245,1,Dial(SIP/Grandstream1,20,rt) ; permit transfer
+;exten => 6245,hint,PJSIP/Grandstream1&PJSIP/Xlite1(Joe Schmoe) ; Channel hints for presence
+;exten => 6245,1,Dial(PJSIP/Grandstream1,20,rt) ; permit transfer
;exten => 6245,n(dial),Dial(${HINT},20,rtT) ; Use hint as listed
;exten => 6245,n,VoiceMail(6245,u) ; Voicemail (unavailable)
;exten => 6245,s+1,Hangup ; s+1, same as n
@@ -708,7 +700,7 @@ include => demo
;exten => wil,1,Goto(6236,1)
;If you want to subscribe to the status of a parking space, this is
-;how you do it. Subscribe to extension 6600 in sip, and you will see
+;how you do it. Subscribe to extension 6600, and you will see
;the status of the first parking lot with this extensions' help
;exten => 6600,hint,park:701@parkedcalls
;exten => 6600,1,noop
@@ -758,7 +750,7 @@ include => demo
;
;exten => t,1,Goto(s,goodbye)
;
-; this is the context our internal SIP hardphones use (see sip.conf)
+; this is the context our internal SIP hardphones use
;
;[acme-internal]
;exten => s,1,Answer()
@@ -799,28 +791,6 @@ include => demo
; ...
; include => time
;
-; Note: if you're geographically spread out, you can have SIP extensions
-; specify their own local timezone in sip.conf as:
-;
-; [boi]
-; type=friend
-; context=acme-internal
-; callerid="Boise Ofc. <2083451111>"
-; ...
-; ; use system-wide default timezone of MST7MDT
-;
-; [lws]
-; type=friend
-; context=acme-internal
-; callerid="Lewiston Ofc. <2087431111>"
-; ...
-; setvar=timezone=PST8PDT
-;
-; "timezone" isn't a 'reserved' name in any way, and other places where
-; the timezone is significant (e.g. calls to "SayUnixTime()", etc) will
-; require modification as well. Note that voicemail.conf already has
-; a mechanism for timezones.
-;
[time]
exten => _X.,30000(time),NoOp(Time: ${EXTEN} ${timezone})
@@ -831,7 +801,6 @@ exten => _X.,30000(time),NoOp(Time: ${EXTEN} ${timezone})
same => n,Set(FUTURETIME=$[${EPOCH} + 12])
same => n,SayUnixTime(${FUTURETIME},Zulu,HNS)
same => n,SayPhonetic(z)
-; use the timezone associated with the extension (sip only), or system-wide
; default if one hasn't been set.
same => n,SayUnixTime(${FUTURETIME},${timezone},HNS)
same => n,Playback(spy-local)
diff --git a/configs/samples/extensions.lua.sample b/configs/samples/extensions.lua.sample
index e9d84e01c3..c5ed2926ed 100644
--- a/configs/samples/extensions.lua.sample
+++ b/configs/samples/extensions.lua.sample
@@ -215,9 +215,7 @@ extensions = {
};
public = {
- -- ATTENTION: If your Asterisk is connected to the internet and you do
- -- not have allowguest=no in sip.conf, everybody out there may use your
- -- public context without authentication. In that case you want to
+ -- ATTENTION: If your Asterisk is connected to the internet
-- double check which services you offer to the world.
--
include = {"demo"};
diff --git a/configs/samples/hep.conf.sample b/configs/samples/hep.conf.sample
index db39bed26e..5b5bb45b1e 100644
--- a/configs/samples/hep.conf.sample
+++ b/configs/samples/hep.conf.sample
@@ -27,9 +27,8 @@ capture_id = 1234 ; A unique integer identifier for this
; with each packet from this server.
uuid_type = call-id ; Specify the preferred source for the Homer
; correlation UUID. Valid options are:
- ; - 'call-id' for the PJSIP or chan_sip SIP
- ; Call-ID
+ ; - 'call-id' for the PJSIP
; - 'channel' for the Asterisk channel name
; Note: If 'call-id' is specified but the
- ; channel is not PJSIP or chan_sip then the
- ; Asterisk channel name will be used instead.
+ ; channel is not PJSIP then the Asterisk
+ ; channel name will be used instead.
diff --git a/configs/samples/modules.conf.sample b/configs/samples/modules.conf.sample
index 41cf9056ef..433f7ed8ac 100644
--- a/configs/samples/modules.conf.sample
+++ b/configs/samples/modules.conf.sample
@@ -43,9 +43,6 @@ noload = res_hep.so
noload = res_hep_pjsip.so
noload = res_hep_rtcp.so
;
-; Do not load chan_sip by default, it may conflict with res_pjsip.
-noload = chan_sip.so
-;
; Load one of the voicemail modules as they are mutually exclusive.
; By default, load app_voicemail only (automatically).
;
diff --git a/configs/samples/phoneprov.conf.sample b/configs/samples/phoneprov.conf.sample
index df3058fe3a..fc2c486851 100644
--- a/configs/samples/phoneprov.conf.sample
+++ b/configs/samples/phoneprov.conf.sample
@@ -1,16 +1,15 @@
[general]
-; This section applies only to the default sip.conf/users.conf config provider
+; This section applies only to the default users.conf config provider
; embedded in res_phoneprov. Other providers may provide their own default settings.
; The default behavior of res_phoneprov will be to set the SERVER template variable to
-; the IP address that the phone uses to contact the provisioning server and the
-; SERVER_PORT variable to the bindport setting in sip.conf. Unless you have a very
-; unusual setup, you should not need to set serveraddr, serveriface, or serverport.
+; the IP address that the phone uses to contact the provisioning server. Unless you have
+; an unusual setup, you should not need to set serveraddr, serveriface, or serverport.
;serveraddr=192.168.1.1 ; Override address to send to the phone to use as server address.
;serveriface=eth0 ; Same as above, except an ethernet interface.
; Useful for when the interface uses DHCP and the asterisk http
- ; server listens on a different IP than chan_sip.
+ ; server listens on a different IP than sip.
;serverport=5060 ; Override port to send to the phone to use as server port.
default_profile=polycom ; The default profile to use if none specified in users.conf
diff --git a/configs/samples/res_config_mysql.conf.sample b/configs/samples/res_config_mysql.conf.sample
index 7d9c78e67f..b17599e26a 100644
--- a/configs/samples/res_config_mysql.conf.sample
+++ b/configs/samples/res_config_mysql.conf.sample
@@ -17,7 +17,7 @@
; and writes should be performed to the same database.
;
; For example, in extconfig.conf, you could specify a line like:
-; sippeers => mysql,readhost.asterisk/writehost.asterisk,sippeers
+; queue_members => mysql,readhost.asterisk/writehost.asterisk,queue_members
; and then define the contexts [readhost.asterisk] and [writehost.asterisk]
; below.
;
diff --git a/configs/samples/res_ldap.conf.sample b/configs/samples/res_ldap.conf.sample
index c8d1286c39..92a5073897 100644
--- a/configs/samples/res_ldap.conf.sample
+++ b/configs/samples/res_ldap.conf.sample
@@ -7,9 +7,7 @@
; In order to use this module, you start
; in extconfig.conf with a configuration like this:
;
-; sippeers = ldap,"dc=myDomain,dc=myDomainExt",sip
; extensions = ldap,"dc=myDomain,dc=myDomainExt",extensions
-; sip.conf = ldap,"dc=myDomain,dc=myDomainExt",config
;
; In the case of LDAP the last keyword in each line above specifies
; a section in this file.
@@ -70,60 +68,6 @@ app = AstExtensionApplication
appdata = AstExtensionApplicationData
additionalFilter=(objectClass=AstExtension)
-;
-; Sip Users Table
-;
-[sip]
-name = cn ; We use the "cn" as the default value for name on the line above
- ; because objectClass=AsteriskSIPUser does not include a uid as an allowed field
- ; If your entry combines other objectClasses and uid is available, you may
- ; prefer to change the line to be name = uid, especially if your LDAP entries
- ; contain spaces in the cn field.
- ; You may also find it appropriate to use something completely different.
- ; This is possible by changing the line above to name = AstAccountName (or whatever you
- ; prefer).
- ;
-amaflags = AstAccountAMAFlags
-callgroup = AstAccountCallGroup
-callerid = AstAccountCallerID
-directmedia = AstAccountDirectMedia
-context = AstAccountContext
-dtmfmode = AstAccountDTMFMode
-fromuser = AstAccountFromUser
-fromdomain = AstAccountFromDomain
-fullcontact = AstAccountFullContact
-fullcontact = gecos
-host = AstAccountHost
-insecure = AstAccountInsecure
-mailbox = AstAccountMailbox
-md5secret = AstAccountRealmedPassword ; Must be an MD5 hash. Field value can start with
- ; {md5} but it is not required.
- ; Generate the password via the md5sum command, e.g.
- ; echo "my_password" | md5sum
-nat = AstAccountNAT
-deny = AstAccountDeny
-permit = AstAccountPermit
-pickupgroup = AstAccountPickupGroup
-port = AstAccountPort
-qualify = AstAccountQualify
-restrictcid = AstAccountRestrictCID
-rtptimeout = AstAccountRTPTimeout
-rtpholdtimeout = AstAccountRTPHoldTimeout
-type = AstAccountType
-disallow = AstAccountDisallowedCodec
-allow = AstAccountAllowedCodec
-MusicOnHold = AstAccountMusicOnHold
-regseconds = AstAccountExpirationTimestamp
-regcontext = AstAccountRegistrationContext
-regexten = AstAccountRegistrationExten
-CanCallForward = AstAccountCanCallForward
-ipaddr = AstAccountIPAddress
-defaultuser = AstAccountDefaultUser
-regserver = AstAccountRegistrationServer
-lastms = AstAccountLastQualifyMilliseconds
-supportpath = AstAccountPathSupport
-additionalFilter=(objectClass=AsteriskSIPUser)
-
;
; IAX Users Table
;
diff --git a/configs/samples/res_stun_monitor.conf.sample b/configs/samples/res_stun_monitor.conf.sample
index 12d32a4cd4..27502d3ab8 100644
--- a/configs/samples/res_stun_monitor.conf.sample
+++ b/configs/samples/res_stun_monitor.conf.sample
@@ -6,11 +6,10 @@
; provided by the STUN server an event is sent out internally within Asterisk
; to alert all listeners to that event of the change.
-; The current default listeners for the network change event include chan_sip
-; and chan_iax. Both of these channel drivers by default react to this event
-; by renewing all outbound registrations. This allows the endpoints Asterisk
-; is registering with to become aware of the address change and know the new
-; location.
+; The current default listeners for the network change event include chan_iax.
+; Both of these channel drivers by default react to this event by renewing all
+; outbound registrations. This allows the endpoints Asterisk is registering with
+; to become aware of the address change and know the new location.
;
[general]
;
diff --git a/configs/samples/sip.conf.sample b/configs/samples/sip.conf.sample
deleted file mode 100644
index 4947754900..0000000000
--- a/configs/samples/sip.conf.sample
+++ /dev/null
@@ -1,1621 +0,0 @@
-;
-; SIP Configuration example for Asterisk
-;
-; Note: Please read the security documentation for Asterisk in order to
-; understand the risks of installing Asterisk with the sample
-; configuration. If your Asterisk is installed on a public
-; IP address connected to the Internet, you will want to learn
-; about the various security settings BEFORE you start
-; Asterisk.
-;
-; Especially note the following settings:
-; - allowguest (default enabled)
-; - permit/deny/acl - IP address filters
-; - contactpermit/contactdeny/contactacl - IP address filters for registrations
-; - context - Which set of services you offer various users
-;
-; SIP dial strings
-; ----------------------------------------------------------
-; In the dialplan (extensions.conf) you can use several
-; syntaxes for dialing SIP devices.
-; SIP/devicename
-; SIP/username@domain (SIP uri)
-; SIP/username[:password[:md5secret[:authname[:transport]]]]@host[:port]
-; SIP/devicename/extension
-; SIP/devicename/extension/IPorHost
-; SIP/username@domain//IPorHost
-; And to alter the To: or the From: header, you can additionally append
-; the following to any of the above strings:
-; [![touser[@todomain]][![fromuser][@fromdomain]]]
-;
-;
-; Devicename
-; devicename is defined as a peer in a section below.
-;
-; username@domain
-; Call any SIP user on the Internet
-; (Don't forget to enable DNS SRV records if you want to use this)
-;
-; devicename/extension
-; If you define a SIP proxy as a peer below, you may call
-; SIP/proxyhostname/user or SIP/user@proxyhostname
-; where the proxyhostname is defined in a section below
-; This syntax also works with ATA's with FXO ports
-;
-; SIP/username[:password[:md5secret[:authname]]]@host[:port]
-; This form allows you to specify password or md5secret and authname
-; without altering any authentication data in config.
-; Examples:
-;
-; SIP/*98@mysipproxy
-; SIP/sales:topsecret::account02@domain.com:5062
-; SIP/12345678::bc53f0ba8ceb1ded2b70e05c3f91de4f:myname@192.168.0.1
-;
-; IPorHost
-; The next server for this call regardless of domain/peer
-;
-; All of these dial strings specify the SIP request URI.
-; In addition, you can specify a specific To: header by adding an
-; exclamation mark after the dial string, like
-;
-; SIP/sales@mysipproxy!sales@edvina.net
-;
-; (Specifying only @todomain without touser will create an invalid SIP
-; request.)
-;
-; Similarly, you can specify the From header as well, after a second
-; exclamation mark:
-;
-; SIP/customer@mysipproxy!!customersupport@wearespindle.com
-;
-; A new feature for 1.8 allows one to specify a host or IP address to use
-; when routing the call. This is typically used in tandem with func_srv if
-; multiple methods of reaching the same domain exist. The host or IP address
-; is specified after the third slash in the dialstring. Examples:
-;
-; SIP/devicename/extension/IPorHost
-; SIP/username@domain//IPorHost
-;
-; CLI Commands
-; -------------------------------------------------------------
-; Useful CLI commands to check peers/users:
-; sip show peers Show all SIP peers (including friends)
-; sip show registry Show status of hosts we register with
-;
-; sip set debug on Show all SIP messages
-;
-; sip reload Reload configuration file
-; sip show settings Show the current channel configuration
-;
-; ------ Naming devices ------------------------------------------------------
-;
-; When naming devices, make sure you understand how Asterisk matches calls
-; that come in.
-; 1. Asterisk checks the SIP From: address username and matches against
-; names of devices with type=user
-; The name is the text between square brackets [name]
-; 2. Asterisk checks the From: addres and matches the list of devices
-; with a type=peer
-; 3. Asterisk checks the IP address (and port number) that the INVITE
-; was sent from and matches against any devices with type=peer
-;
-; Don't mix extensions with the names of the devices. Devices need a unique
-; name. The device name is *not* used as phone numbers. Phone numbers are
-; anything you declare as an extension in the dialplan (extensions.conf).
-;
-; When setting up trunks, make sure there's no risk that any From: username
-; (caller ID) will match any of your device names, because then Asterisk
-; might match the wrong device.
-;
-; Note: The parameter "username" is not the username and in most cases is
-; not needed at all. Check below. In later releases, it's renamed
-; to "defaultuser" which is a better name, since it is used in
-; combination with the "defaultip" setting.
-; ----------------------------------------------------------------------------
-
-; ** Old configuration options **
-; The "call-limit" configuation option is considered old is replaced
-; by new functionality. To enable callcounters, you use the new
-; "callcounter" setting (for extension states in queue and subscriptions)
-; You are encouraged to use the dialplan groupcount functionality
-; to enforce call limits instead of using this channel-specific method.
-; You can still set limits per device in sip.conf or in a database by using
-; "setvar" to set variables that can be used in the dialplan for various limits.
-
-[general]
-context=public ; Default context for incoming calls. Defaults to 'default'
-;allowguest=no ; Allow or reject guest calls (default is yes)
- ; If your Asterisk is connected to the Internet
- ; and you have allowguest=yes
- ; you want to check which services you offer everyone
- ; out there, by enabling them in the default context (see below).
-;match_auth_username=yes ; if available, match user entry using the
- ; 'username' field from the authentication line
- ; instead of the From: field.
-allowoverlap=no ; Disable overlap dialing support. (Default is yes)
-;allowoverlap=yes ; Enable RFC3578 overlap dialing support.
- ; Can use the Incomplete application to collect the
- ; needed digits from an ambiguous dialplan match.
-;allowoverlap=dtmf ; Enable overlap dialing support using DTMF delivery
- ; methods (inband, RFC2833, SIP INFO) in the early
- ; media phase. Uses the Incomplete application to
- ; collect the needed digits.
-;allowtransfer=no ; Disable all transfers (unless enabled in peers or users)
- ; Default is enabled. The Dial() options 't' and 'T' are not
- ; related as to whether SIP transfers are allowed or not.
-;realm=mydomain.tld ; Realm for digest authentication
- ; defaults to "asterisk". If you set a system name in
- ; asterisk.conf, it defaults to that system name
- ; Realms MUST be globally unique according to RFC 3261
- ; Set this to your host name or domain name
-;domainsasrealm=no ; Use domains list as realms
- ; You can serve multiple Realms specifying several
- ; 'domain=...' directives (see below).
- ; In this case Realm will be based on request 'From'/'To' header
- ; and should match one of domain names.
- ; Otherwise default 'realm=...' will be used.
-;recordonfeature=automixmon ; Default feature to use when receiving 'Record: on' header
- ; from an INFO message. Defaults to 'automon'. Works with
- ; dynamic features. Feature must be usable on requesting
- ; channel for it to work. Setting this value to a blank
- ; will disable it.
-;recordofffeature=automixmon ; Default feature to use when receiving 'Record: off' header
- ; from an INFO message. Defaults to 'automon'. Works with
- ; dynamic features. Feature must be usable on requesting
- ; channel for it to work. Setting this value to a blank
- ; will disable it.
-
-; With the current situation, you can do one of four things:
-; a) Listen on a specific IPv4 address. Example: bindaddr=192.0.2.1
-; b) Listen on a specific IPv6 address. Example: bindaddr=2001:db8::1
-; c) Listen on the IPv4 wildcard. Example: bindaddr=0.0.0.0
-; d) Listen on the IPv4 and IPv6 wildcards. Example: bindaddr=::
-; (You can choose independently for UDP, TCP, and TLS, by specifying different values for
-; "udpbindaddr", "tcpbindaddr", and "tlsbindaddr".)
-; (Note that using bindaddr=:: will show only a single IPv6 socket in netstat.
-; IPv4 is supported at the same time using IPv4-mapped IPv6 addresses.)
-;
-; You may optionally add a port number. (The default is port 5060 for UDP and TCP, 5061
-; for TLS).
-; IPv4 example: bindaddr=0.0.0.0:5062
-; IPv6 example: bindaddr=[::]:5062
-;
-; The address family of the bound UDP address is used to determine how Asterisk performs
-; DNS lookups. In cases a) and c) above, only A records are considered. In case b), only
-; AAAA records are considered. In case d), both A and AAAA records are considered. Note,
-; however, that Asterisk ignores all records except the first one. In case d), when both A
-; and AAAA records are available, either an A or AAAA record will be first, and which one
-; depends on the operating system. On systems using glibc, AAAA records are given
-; priority.
-
-udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all)
- ; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)
-
-;rtpbindaddr=172.16.42.1 ; IP address to bind RTP listen sock to (default is disabled). When
- ; disabled the udpbindaddr is used.
-
-; When a dialog is started with another SIP endpoint, the other endpoint
-; should include an Allow header telling us what SIP methods the endpoint
-; implements. However, some endpoints either do not include an Allow header
-; or lie about what methods they implement. In the former case, Asterisk
-; makes the assumption that the endpoint supports all known SIP methods.
-; If you know that your SIP endpoint does not provide support for a specific
-; method, then you may provide a comma-separated list of methods that your
-; endpoint does not implement in the disallowed_methods option. Note that
-; if your endpoint is truthful with its Allow header, then there is no need
-; to set this option. This option may be set in the general section or may
-; be set per endpoint. If this option is set both in the general section and
-; in a peer section, then the peer setting completely overrides the general
-; setting (i.e. the result is *not* the union of the two options).
-;
-; Note also that while Asterisk currently will parse an Allow header to learn
-; what methods an endpoint supports, the only actual use for this currently
-; is for determining if Asterisk may send connected line UPDATE requests and
-; MESSAGE requests. Its use may be expanded in the future.
-;
-; disallowed_methods = UPDATE
-
-;
-; Note that the TCP and TLS support for chan_sip is currently considered
-; experimental. Since it is new, all of the related configuration options are
-; subject to change in any release. If they are changed, the changes will
-; be reflected in this sample configuration file, as well as in the UPGRADE.txt file.
-;
-tcpenable=no ; Enable server for incoming TCP connections (default is no)
-tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces)
- ; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)
-
-;tlsenable=no ; Enable server for incoming TLS (secure) connections (default is no)
-;tlsbindaddr=0.0.0.0 ; IP address for TLS server to bind to (0.0.0.0) binds to all interfaces)
- ; Optionally add a port number, 192.168.1.1:5063 (default is port 5061)
- ; Remember that the IP address must match the common name (hostname) in the
- ; certificate, so you don't want to bind a TLS socket to multiple IP addresses.
- ; For details how to construct a certificate for SIP see
- ; http://tools.ietf.org/html/draft-ietf-sip-domain-certs
-
-;tcpauthtimeout = 30 ; tcpauthtimeout specifies the maximum number
- ; of seconds a client has to authenticate. If
- ; the client does not authenticate beofre this
- ; timeout expires, the client will be
- ; disconnected. (default: 30 seconds)
-
-;tcpauthlimit = 100 ; tcpauthlimit specifies the maximum number of
- ; unauthenticated sessions that will be allowed
- ; to connect at any given time. (default: 100)
-
-;websocket_enabled = true ; Set to false to prevent chan_sip from listening to websockets. This
- ; is neeeded when using chan_sip and res_pjsip_transport_websockets on
- ; the same system.
-
-;websocket_write_timeout = 100 ; Default write timeout to set on websocket transports.
- ; This value may need to be adjusted for connections where
- ; Asterisk must write a substantial amount of data and the
- ; receiving clients are slow to process the received information.
- ; Value is in milliseconds; default is 100 ms.
-
-transport=udp ; Set the default transports. The order determines the primary default transport.
- ; If tcpenable=no and the transport set is tcp, we will fallback to UDP.
-
-srvlookup=yes ; Enable DNS SRV lookups on outbound calls
- ; Note: Asterisk only uses the first host
- ; in SRV records
- ; Disabling DNS SRV lookups disables the
- ; ability to place SIP calls based on domain
- ; names to some other SIP users on the Internet
- ; Specifying a port in a SIP peer definition or
- ; when dialing outbound calls will supress SRV
- ; lookups for that peer or call.
-
-;pedantic=yes ; Enable checking of tags in headers,
- ; international character conversions in URIs
- ; and multiline formatted headers for strict
- ; SIP compatibility (defaults to "yes")
-
-; See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for a description of these parameters.
-;tos_sip=cs3 ; Sets TOS for SIP packets.
-;tos_audio=ef ; Sets TOS for RTP audio packets.
-;tos_video=af41 ; Sets TOS for RTP video packets.
-;tos_text=af41 ; Sets TOS for RTP text packets.
-
-;cos_sip=3 ; Sets 802.1p priority for SIP packets.
-;cos_audio=5 ; Sets 802.1p priority for RTP audio packets.
-;cos_video=4 ; Sets 802.1p priority for RTP video packets.
-;cos_text=3 ; Sets 802.1p priority for RTP text packets.
-
-;maxexpiry=3600 ; Maximum allowed time of incoming registrations (seconds)
-;minexpiry=60 ; Minimum length of registrations (default 60)
-;defaultexpiry=120 ; Default length of incoming/outgoing registration
-;submaxexpiry=3600 ; Maximum allowed time of incoming subscriptions (seconds), default: maxexpiry
-;subminexpiry=60 ; Minimum length of subscriptions, default: minexpiry
-;mwiexpiry=3600 ; Expiry time for outgoing MWI subscriptions
-;maxforwards=70 ; Setting for the SIP Max-Forwards: header (loop prevention)
- ; Default value is 70
-;qualifyfreq=60 ; Qualification: How often to check for the host to be up in seconds
- ; and reported in milliseconds with sip show settings.
- ; Set to low value if you use low timeout for NAT of UDP sessions
- ; Default: 60
-;qualifygap=100 ; Number of milliseconds between each group of peers being qualified
- ; Default: 100
-;qualifypeers=1 ; Number of peers in a group to be qualified at the same time
- ; Default: 1
-;keepalive=60 ; Interval at which keepalive packets should be sent to a peer
- ; Valid options are yes (60 seconds), no, or the number of seconds.
- ; Default: 0
-;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY
-;buggymwi=no ; Cisco SIP firmware doesn't support the MWI RFC
- ; fully. Enable this option to not get error messages
- ; when sending MWI to phones with this bug.
-;mwi_from=asterisk ; When sending MWI NOTIFY requests, use this setting in
- ; the From: header as the "name" portion. Also fill the
- ; "user" portion of the URI in the From: header with this
- ; value if no fromuser is set
- ; Default: empty
-;vmexten=voicemail ; dialplan extension to reach mailbox sets the
- ; Message-Account in the MWI notify message
- ; defaults to "asterisk"
-
-; Codec negotiation
-;
-; When Asterisk is receiving a call, the codec will initially be set to the
-; first codec in the allowed codecs defined for the user receiving the call
-; that the caller also indicates that it supports. But, after the caller
-; starts sending RTP, Asterisk will switch to using whatever codec the caller
-; is sending.
-;
-; When Asterisk is placing a call, the codec used will be the first codec in
-; the allowed codecs that the callee indicates that it supports. Asterisk will
-; *not* switch to whatever codec the callee is sending.
-;
-;preferred_codec_only=yes ; Respond to a SIP invite with the single most preferred codec
- ; rather than advertising all joint codec capabilities. This
- ; limits the other side's codec choice to exactly what we prefer.
-
-;disallow=all ; First disallow all codecs
-;allow=ulaw ; Allow codecs in order of preference
-;allow=ilbc ; see https://wiki.asterisk.org/wiki/display/AST/RTP+Packetization
- ; for framing options
-;autoframing=yes ; Set packetization based on the remote endpoint's (ptime)
- ; preferences. Defaults to no.
-;
-; This option specifies a preference for which music on hold class this channel
-; should listen to when put on hold if the music class has not been set on the
-; channel with Set(CHANNEL(musicclass)=whatever) in the dialplan, and the peer
-; channel putting this one on hold did not suggest a music class.
-;
-; This option may be specified globally, or on a per-user or per-peer basis.
-;
-;mohinterpret=default
-;
-; This option specifies which music on hold class to suggest to the peer channel
-; when this channel places the peer on hold. It may be specified globally or on
-; a per-user or per-peer basis.
-;
-;mohsuggest=default
-;
-;parkinglot=plaza ; Sets the default parking lot for call parking
- ; This may also be set for individual users/peers
- ; Parkinglots are configured in features.conf
-;language=en ; Default language setting for all users/peers
- ; This may also be set for individual users/peers
-;tonezone=se ; Default tonezone for all users/peers
- ; This may also be set for individual users/peers
-
-;relaxdtmf=yes ; Relax dtmf handling
-;trustrpid = no ; If Remote-Party-ID should be trusted
-;sendrpid = yes ; If Remote-Party-ID should be sent (defaults to no)
-;sendrpid = rpid ; Use the "Remote-Party-ID" header
- ; to send the identity of the remote party
- ; This is identical to sendrpid=yes
-;sendrpid = pai ; Use the "P-Asserted-Identity" header
- ; to send the identity of the remote party
-;rpid_update = no ; In certain cases, the only method by which a connected line
- ; change may be immediately transmitted is with a SIP UPDATE request.
- ; If communicating with another Asterisk server, and you wish to be able
- ; transmit such UPDATE messages to it, then you must enable this option.
- ; Otherwise, we will have to wait until we can send a reinvite to
- ; transmit the information.
-;trust_id_outbound = no ; Controls whether or not we trust this peer with private identity
- ; information (when the remote party has callingpres=prohib or equivalent).
- ; no - RPID/PAI headers will not be included for private peer information
- ; yes - RPID/PAI headers will include the private peer information. Privacy
- ; requirements will be indicated in a Privacy header for sendrpid=pai
- ; legacy - RPID/PAI will be included for private peer information. In the
- ; case of sendrpid=pai, private data that would be included in them
- ; will be anonymized. For sendrpid=rpid, private data may be included
- ; but the remote party's domain will be anonymized. The way legacy
- ; behaves may violate RFC-3325, but it follows historic behavior.
- ; This option is set to 'legacy' by default
-;prematuremedia=no ; Some ISDN links send empty media frames before
- ; the call is in ringing or progress state. The SIP
- ; channel will then send 183 indicating early media
- ; which will be empty - thus users get no ring signal.
- ; Setting this to "yes" will stop any media before we have
- ; call progress (meaning the SIP channel will not send 183 Session
- ; Progress for early media). Default is "yes". Also make sure that
- ; the SIP peer is configured with progressinband=never.
- ;
- ; In order for "noanswer" applications to work, you need to run
- ; the progress() application in the priority before the app.
-
-;progressinband=no ; If we should generate in-band ringing. Always
- ; use 'never' to never use in-band signalling, even in cases
- ; where some buggy devices might not render it
- ; Valid values: yes, no, never Default: no
-;useragent=Asterisk PBX ; Allows you to change the user agent string
- ; The default user agent string also contains the Asterisk
- ; version. If you don't want to expose this, change the
- ; useragent string.
-;promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP address
- ; Note that promiscredir when redirects are made to the
- ; local system will cause loops since Asterisk is incapable
- ; of performing a "hairpin" call.
-;usereqphone = no ; If yes, ";user=phone" is added to uri that contains
- ; a valid phone number
-;dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833
- ; Other options:
- ; info : SIP INFO messages (application/dtmf-relay)
- ; shortinfo : SIP INFO messages (application/dtmf)
- ; inband : Inband audio (requires 64 kbit codec -alaw, ulaw)
- ; auto : Use rfc2833 if offered, inband otherwise
-
-;compactheaders = yes ; send compact sip headers.
-;
-;videosupport=yes ; Turn on support for SIP video. You need to turn this
- ; on in this section to get any video support at all.
- ; You can turn it off on a per peer basis if the general
- ; video support is enabled, but you can't enable it for
- ; one peer only without enabling in the general section.
- ; If you set videosupport to "always", then RTP ports will
- ; always be set up for video, even on clients that don't
- ; support it. This assists callfile-derived calls and
- ; certain transferred calls to use always use video when
- ; available. [yes|NO|always]
-
-;textsupport=no ; Support for ITU-T T.140 realtime text.
- ; The default value is "no".
-
-;maxcallbitrate=384 ; Maximum bitrate for video calls (default 384 kb/s)
- ; Videosupport and maxcallbitrate is settable
- ; for peers and users as well
-;authfailureevents=no ; generate manager "peerstatus" events when peer can't
- ; authenticate with Asterisk. Peerstatus will be "rejected".
-;alwaysauthreject = yes ; When an incoming INVITE or REGISTER is to be rejected,
- ; for any reason, always reject with an identical response
- ; equivalent to valid username and invalid password/hash
- ; instead of letting the requester know whether there was
- ; a matching user or peer for their request. This reduces
- ; the ability of an attacker to scan for valid SIP usernames.
- ; This option is set to "yes" by default.
-
-;auth_options_requests = yes ; Enabling this option will authenticate OPTIONS requests just like
- ; INVITE requests are. By default this option is disabled.
-
-;accept_outofcall_message = no ; Disable this option to reject all MESSAGE requests outside of a
- ; call. By default, this option is enabled. When enabled, MESSAGE
- ; requests are passed in to the dialplan.
-
-;outofcall_message_context = messages ; Context all out of dialog msgs are sent to. When this
- ; option is not set, the context used during peer matching
- ; is used. This option can be defined at both the peer and
- ; global level.
-
-;auth_message_requests = yes ; Enabling this option will authenticate MESSAGE requests.
- ; By default this option is enabled. However, it can be disabled
- ; should an application desire to not load the Asterisk server with
- ; doing authentication and implement end to end security in the
- ; message body.
-
-;g726nonstandard = yes ; If the peer negotiates G726-32 audio, use AAL2 packing
- ; order instead of RFC3551 packing order (this is required
- ; for Sipura and Grandstream ATAs, among others). This is
- ; contrary to the RFC3551 specification, the peer _should_
- ; be negotiating AAL2-G726-32 instead :-(
-;outboundproxy=proxy.provider.domain ; send outbound signaling to this proxy, not directly to the devices
-;outboundproxy=proxy.provider.domain:8080 ; send outbound signaling to this proxy, not directly to the devices
-;outboundproxy=proxy.provider.domain,force ; Send ALL outbound signalling to proxy, ignoring route: headers
-;outboundproxy=tls://proxy.provider.domain ; same as '=proxy.provider.domain' except we try to connect with tls
-;outboundproxy=192.0.2.1 ; IPv4 address literal (default port is 5060)
-;outboundproxy=2001:db8::1 ; IPv6 address literal (default port is 5060)
-;outboundproxy=192.168.0.2.1:5062 ; IPv4 address literal with explicit port
-;outboundproxy=[2001:db8::1]:5062 ; IPv6 address literal with explicit port
-; ; (could also be tcp,udp) - defining transports on the proxy line only
-; ; applies for the global proxy, otherwise use the transport= option
-
-;supportpath=yes ; This activates parsing and handling of Path header as defined in RFC 3327. This enables
- ; Asterisk to route outgoing out-of-dialog requests via a set of proxies by using a pre-loaded
- ; route-set defined by the Path headers in the REGISTER request.
- ; NOTE: There are multiple things to consider with this setting:
- ; * As this influences routing of SIP requests make sure to not trust Path headers provided
- ; by the user's SIP client (the proxy in front of Asterisk should remove existing user
- ; provided Path headers).
- ; * When a peer has both a path and outboundproxy set, the path will be added to Route: header
- ; but routing to next hop is done using the outboundproxy.
- ; * If set globally, not only will all peers use the Path header, but outbound REGISTER
- ; requests from Asterisk will add path to the Supported header.
-
-;rtsavepath=yes ; If using dynamic realtime, store the path headers
-
-;matchexternaddrlocally = yes ; Only substitute the externaddr or externhost setting if it matches
- ; your localnet setting. Unless you have some sort of strange network
- ; setup you will not need to enable this.
-
-;dynamic_exclude_static = yes ; Disallow all dynamic hosts from registering
- ; as any IP address used for staticly defined
- ; hosts. This helps avoid the configuration
- ; error of allowing your users to register at
- ; the same address as a SIP provider.
-
-;contactdeny=0.0.0.0/0.0.0.0 ; Use contactpermit and contactdeny to
-;contactpermit=172.16.0.0/255.255.0.0 ; restrict at what IPs your users may
- ; register their phones.
-;contactacl=named_acl_example ; Use named ACLs defined in acl.conf
-
-;rtp_engine=asterisk ; RTP engine to use when communicating with the device
-
-;
-; If regcontext is specified, Asterisk will dynamically create and destroy a
-; NoOp priority 1 extension for a given peer who registers or unregisters with
-; us and have a "regexten=" configuration item.
-; Multiple contexts may be specified by separating them with '&'. The
-; actual extension is the 'regexten' parameter of the registering peer or its
-; name if 'regexten' is not provided. If more than one context is provided,
-; the context must be specified within regexten by appending the desired
-; context after '@'. More than one regexten may be supplied if they are
-; separated by '&'. Patterns may be used in regexten.
-;
-;regcontext=sipregistrations
-;regextenonqualify=yes ; Default "no"
- ; If you have qualify on and the peer becomes unreachable
- ; this setting will enforce inactivation of the regexten
- ; extension for the peer
-;legacy_useroption_parsing=yes ; Default "no" ; If you have this option enabled and there are semicolons
- ; in the user field of a sip URI, the field be truncated
- ; at the first semicolon seen. This effectively makes
- ; semicolon a non-usable character for peer names, extensions,
- ; and maybe other, less tested things. This can be useful
- ; for improving compatability with devices that like to use
- ; user options for whatever reason. The behavior is similar to
- ; how SIP URI's were typically handled in 1.6.2, hence the name.
-
-;send_diversion=no ; Default "yes" ; Asterisk normally sends Diversion headers with certain SIP
- ; invites to relay data about forwarded calls. If this option
- ; is disabled, Asterisk won't send Diversion headers unless
- ; they are added manually.
-
-; The shrinkcallerid function removes '(', ' ', ')', non-trailing '.', and '-' not
-; in square brackets. For example, the caller id value 555.5555 becomes 5555555
-; when this option is enabled. Disabling this option results in no modification
-; of the caller id value, which is necessary when the caller id represents something
-; that must be preserved. This option can only be used in the [general] section.
-; By default this option is on.
-;
-;shrinkcallerid=yes ; on by default
-
-
-;use_q850_reason = no ; Default "no"
- ; Set to yes add Reason header and use Reason header if it is available.
-
-; When the Transfer() application sends a REFER SIP message, extra headers specified in
-; the dialplan by way of SIPAddHeader are sent out with that message. 1.8 and earlier did not
-; add the extra headers. To revert to 1.8- behavior, call SIPRemoveHeader with no arguments
-; before calling Transfer() to remove all additional headers from the channel. The setting
-; below is for transitional compatibility only.
-;
-;refer_addheaders=yes ; on by default
-
-;autocreatepeer=no ; Allow any UAC not explicitly defined to register
- ; WITHOUT AUTHENTICATION. Enabling this options poses a high
- ; potential security risk and should be avoided unless the
- ; server is behind a trusted firewall.
- ; If set to "yes", then peers created in this fashion
- ; are purged during SIP reloads.
- ; When set to "persist", the peers created in this fashion
- ; are not purged during SIP reloads.
-
-;
-; ----------------------- TLS settings ------------------------------------------------------------
-;tlscertfile= ; Certificate chain (*.pem format only) to use for TLS connections
- ; The certificates must be sorted starting with the subject's certificate
- ; and followed by intermediate CA certificates if applicable. If the
- ; file name ends in _rsa, for example "asterisk_rsa.pem", the files
- ; "asterisk_dsa.pem" and/or "asterisk_ecc.pem" are loaded
- ; (certificate, intermediates, private key), to support multiple
- ; algorithms for server authentication (RSA, DSA, ECDSA). If the chains
- ; are different, at least OpenSSL 1.0.2 is required.
- ; Default is to look for "asterisk.pem" in current directory
-
-;tlsprivatekey= ; Private key file (*.pem format only) for TLS connections.
- ; If no tlsprivatekey is specified, tlscertfile is searched for
- ; for both public and private key.
-
-;tlscafile=
-; If the server your connecting to uses a self signed certificate
-; you should have their certificate installed here so the code can
-; verify the authenticity of their certificate.
-
-;tlscapath=
-; A directory full of CA certificates. The files must be named with
-; the CA subject name hash value.
-; (see man SSL_CTX_load_verify_locations for more info)
-
-;tlsdontverifyserver=[yes|no]
-; If set to yes, don't verify the servers certificate when acting as
-; a client. If you don't have the server's CA certificate you can
-; set this and it will connect without requiring tlscafile to be set.
-; Default is no.
-
-;tlscipher=
-; A string specifying which SSL ciphers to use or not use
-; A list of valid SSL cipher strings can be found at:
-; http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
-;
-;tlsclientmethod=tlsv1 ; values include tlsv1, sslv3, sslv2.
- ; Specify protocol for outbound client connections.
- ; If left unspecified, the default is the general-
- ; purpose version-flexible SSL/TLS method (sslv23).
- ; With that, the actual protocol version used will
- ; be negotiated to the highest version mutually
- ; supported by Asterisk and the remote server, i.e.
- ; TLSv1.2. The supported protocols are listed at
- ; http://www.openssl.org/docs/ssl/SSL_CTX_new.html
- ; SSLv2 and SSLv3 are disabled within Asterisk.
- ; Your distribution might have changed that list
- ; further.
-;
-; -------------------------- SIP timers ----------------------------------------------------
-; These timers are used primarily in INVITE transactions.
-; The default for Timer T1 is 500 ms or the measured run-trip time between
-; Asterisk and the device if you have qualify=yes for the device.
-;
-;t1min=100 ; Minimum roundtrip time for messages to monitored hosts
- ; Defaults to 100 ms
-;timert1=500 ; Default T1 timer
- ; Defaults to 500 ms or the measured round-trip
- ; time to a peer (qualify=yes).
-;timerb=32000 ; Call setup timer. If a provisional response is not received
- ; in this amount of time, the call will autocongest
- ; Defaults to 64*timert1
-
-; -------------------------- RTP timers ----------------------------------------------------
-; These timers are currently used for both audio and video streams. The RTP timeouts
-; are only applied to the audio channel.
-; The settings are settable in the global section as well as per device
-;
-;rtptimeout=60 ; Terminate call if 60 seconds of no RTP or RTCP activity
- ; on the audio channel
- ; when we're not on hold. This is to be able to hangup
- ; a call in the case of a phone disappearing from the net,
- ; like a powerloss or grandma tripping over a cable.
-;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP or RTCP activity
- ; on the audio channel
- ; when we're on hold (must be > rtptimeout)
-;rtpkeepalive= ; Send keepalives in the RTP stream to keep NAT open
- ; (default is off - zero)
-
-; -------------------------- SIP Session-Timers (RFC 4028)------------------------------------
-; SIP Session-Timers provide an end-to-end keep-alive mechanism for active SIP sessions.
-; This mechanism can detect and reclaim SIP channels that do not terminate through normal
-; signaling procedures. Session-Timers can be configured globally or at a user/peer level.
-; The operation of Session-Timers is driven by the following configuration parameters:
-;
-; * session-timers - Session-Timers feature operates in the following three modes:
-; originate : Request and run session-timers always
-; accept : Run session-timers only when requested by other UA
-; refuse : Do not run session timers in any case
-; The default mode of operation is 'accept'.
-; * session-expires - Maximum session refresh interval in seconds. Defaults to 1800 secs.
-; * session-minse - Minimum session refresh interval in seconds. Defualts to 90 secs.
-; * session-refresher - The session refresher (uac|uas). Defaults to 'uas'.
-; uac - Default to the caller initially refreshing when possible
-; uas - Default to the callee initially refreshing when possible
-;
-; Note that, due to recommendations in RFC 4028, Asterisk will always honor the other
-; endpoint's preference for who will handle refreshes. Asterisk will never override the
-; preferences of the other endpoint. Doing so could result in Asterisk and the endpoint
-; fighting over who sends the refreshes. This holds true for the initiation of session
-; timers and subsequent re-INVITE requests whether Asterisk is the caller or callee, or
-; whether Asterisk is currently the refresher or not.
-;
-;session-timers=originate
-;session-expires=600
-;session-minse=90
-;session-refresher=uac
-;
-; -------------------------- SIP DEBUGGING ---------------------------------------------------
-;sipdebug = yes ; Turn on SIP debugging by default, from
- ; the moment the channel loads this configuration.
- ; NOTE: You cannot use the CLI to turn it off. You'll
- ; need to edit this and reload the config.
-;recordhistory=yes ; Record SIP history by default
- ; (see sip history / sip no history)
-;dumphistory=yes ; Dump SIP history at end of SIP dialogue
- ; SIP history is output to the DEBUG logging channel
-
-
-; -------------------------- STATUS NOTIFICATIONS (SUBSCRIPTIONS) ----------------------------
-; You can subscribe to the status of extensions with a "hint" priority
-; (See extensions.conf.sample for examples)
-; chan_sip support two major formats for notifications: dialog-info and SIMPLE
-;
-; You will get more detailed reports (busy etc) if you have a call counter enabled
-; for a device.
-;
-; If you set the busylevel, we will indicate busy when we have a number of calls that
-; matches the busylevel treshold.
-;
-; For queues, you will need this level of detail in status reporting, regardless
-; if you use SIP subscriptions. Queues and manager use the same internal interface
-; for reading status information.
-;
-; Note: Subscriptions does not work if you have a realtime dialplan and use the
-; realtime switch.
-;
-;allowsubscribe=no ; Disable support for subscriptions. (Default is yes)
-;subscribecontext = default ; Set a specific context for SUBSCRIBE requests
- ; Useful to limit subscriptions to local extensions
- ; Settable per peer/user also
-;notifyringing = no ; Control when subscriptions get notified of ringing state.
- ; Specify 'no' to not send any ringing notifications.
- ; Specify 'yes' to always send ringing notifications (default).
- ; Specify 'notinuse' to only send ringing notifications for
- ; extensions that are not currently in use. This is useful as a
- ; visual indication of who is available to pick up an incoming call
-;notifyhold = yes ; Notify subscriptions on HOLD state (default: no)
- ; Turning on notifyringing and notifyhold will add a lot
- ; more database transactions if you are using realtime.
-;notifycid = yes ; Control whether caller ID information is sent along with
- ; dialog-info+xml notifications (supported by snom phones).
- ; Note that this feature will only work properly when the
- ; incoming call is using the same extension and context that
- ; is being used as the hint for the called extension. This means
- ; that it won't work when using subscribecontext for your sip
- ; user or peer (if subscribecontext is different than context).
- ; This is also limited to a single caller, meaning that if an
- ; extension is ringing because multiple calls are incoming,
- ; only one will be used as the source of caller ID. Specify
- ; 'ignore-context' to ignore the called context when looking
- ; for the caller's channel. The default value is 'no.' Setting
- ; notifycid to 'ignore-context' also causes call-pickups attempted
- ; via SNOM's NOTIFY mechanism to set the context for the call pickup
- ; to PICKUPMARK.
-;callcounter = yes ; Enable call counters on devices. This can be set per
- ; device too.
-
-; ---------------------------------------- T.38 FAX SUPPORT ----------------------------------
-;
-; This setting is available in the [general] section as well as in device configurations.
-; Setting this to yes enables T.38 FAX (UDPTL) on SIP calls; it defaults to off.
-;
-; t38pt_udptl = yes ; Enables T.38 with FEC error correction.
-; t38pt_udptl = yes,fec ; Enables T.38 with FEC error correction.
-; t38pt_udptl = yes,redundancy ; Enables T.38 with redundancy error correction.
-; t38pt_udptl = yes,none ; Enables T.38 with no error correction.
-;
-; In some cases, T.38 endpoints will provide a T38FaxMaxDatagram value (during T.38 setup) that
-; is based on an incorrect interpretation of the T.38 recommendation, and results in failures
-; because Asterisk does not believe it can send T.38 packets of a reasonable size to that
-; endpoint (Cisco media gateways are one example of this situation). In these cases, during a
-; T.38 call you will see warning messages on the console/in the logs from the Asterisk UDPTL
-; stack complaining about lack of buffer space to send T.38 FAX packets. If this occurs, you
-; can set an override (globally, or on a per-device basis) to make Asterisk ignore the
-; T38FaxMaxDatagram value specified by the other endpoint, and use a configured value instead.
-; This can be done by appending 'maxdatagram=' to the t38pt_udptl configuration option,
-; like this:
-;
-; t38pt_udptl = yes,fec,maxdatagram=400 ; Enables T.38 with FEC error correction and overrides
-; ; the other endpoint's provided value to assume we can
-; ; send 400 byte T.38 FAX packets to it.
-;
-; FAX detection will cause the SIP channel to jump to the 'fax' extension (if it exists)
-; based one or more events being detected. The events that can be detected are an incoming
-; CNG tone or an incoming T.38 re-INVITE request.
-;
-; faxdetect = yes ; Default 'no', 'yes' enables both CNG and T.38 detection
-; faxdetect = cng ; Enables only CNG detection
-; faxdetect = t38 ; Enables only T.38 detection
-;
-; ---------------------------------------- OUTBOUND SIP REGISTRATIONS ------------------------
-; Asterisk can register as a SIP user agent to a SIP proxy (provider)
-; Format for the register statement is:
-; register => [peer?][transport://]user[@domain][:secret[:authuser]]@host[:port][/extension][~expiry]
-;
-;
-;
-; domain is either
-; - domain in DNS
-; - host name in DNS
-; - the name of a peer defined below or in realtime
-; The domain is where you register your username, so your SIP uri you are registering to
-; is username@domain
-;
-; If no extension is given, the 's' extension is used. The extension needs to
-; be defined in extensions.conf to be able to accept calls from this SIP proxy
-; (provider).
-;
-; A similar effect can be achieved by adding a "callbackextension" option in a peer section.
-; this is equivalent to having the following line in the general section:
-;
-; register => fromuser:secret:username@host/callbackextension
-;
-; and more readable because you don't have to write the parameters in two places
-; (note that the "port" is ignored - this is a bug that should be fixed).
-;
-; Note that a register= line doesn't mean that we will match the incoming call in any
-; other way than described above. If you want to control where the call enters your
-; dialplan, which context, you want to define a peer with the hostname of the provider's
-; server. If the provider has multiple servers to place calls to your system, you need
-; a peer for each server.
-;
-; Beginning with Asterisk version 1.6.2, the "user" portion of the register line may
-; contain a port number. Since the logical separator between a host and port number is a
-; ':' character, and this character is already used to separate between the optional "secret"
-; and "authuser" portions of the line, there is a bit of a hoop to jump through if you wish
-; to use a port here. That is, you must explicitly provide a "secret" and "authuser" even if
-; they are blank. See the third example below for an illustration.
-;
-;
-; Examples:
-;
-;register => 1234:password@mysipprovider.com
-;
-; This will pass incoming calls to the 's' extension
-;
-;
-;register => 2345:password@sip_proxy/1234
-;
-; Register 2345 at sip provider 'sip_proxy'. Calls from this provider
-; connect to local extension 1234 in extensions.conf, default context,
-; unless you configure a [sip_proxy] section below, and configure a
-; context.
-; Tip 1: Avoid assigning hostname to a sip.conf section like [provider.com]
-; Tip 2: Use separate inbound and outbound sections for SIP providers
-; (instead of type=friend) if you have calls in both directions
-;
-;register => 3456@mydomain:5082::@mysipprovider.com
-;
-; Note that in this example, the optional authuser and secret portions have
-; been left blank because we have specified a port in the user section
-;
-;register => tls://username:xxxxxx@sip-tls-proxy.example.org
-;
-; The 'transport' part defaults to 'udp' but may also be 'tcp' or 'tls'.
-; Using 'udp://' explicitly is also useful in case the username part
-; contains a '/' ('user/name').
-
-;registertimeout=20 ; retry registration calls every 20 seconds (default)
-;registerattempts=10 ; Number of registration attempts before we give up
- ; 0 = continue forever, hammering the other server
- ; until it accepts the registration
- ; Default is 0 tries, continue forever
-;register_retry_403=yes ; Treat 403 responses to registrations as if they were
- ; 401 responses and continue retrying according to normal
- ; retry rules.
-
-; ---------------------------------------- OUTBOUND MWI SUBSCRIPTIONS -------------------------
-; Asterisk can subscribe to receive the MWI from another SIP server and store it locally for retrieval
-; by other phones. At this time, you can only subscribe using UDP as the transport.
-; Format for the mwi register statement is:
-; mwi => user[:secret[:authuser]]@host[:port]/mailbox
-;
-; Examples:
-;mwi => 1234:password@mysipprovider.com/1234
-;mwi => 1234:password@myportprovider.com:6969/1234
-;mwi => 1234:password:authuser@myauthprovider.com/1234
-;mwi => 1234:password:authuser@myauthportprovider.com:6969/1234
-;
-; MWI received will be stored in the 1234 mailbox of the SIP_Remote context.
-; It can be used by other phones by following the below:
-; mailbox=1234@SIP_Remote
-; ---------------------------------------- NAT SUPPORT ------------------------
-;
-; WARNING: SIP operation behind a NAT is tricky and you really need
-; to read and understand well the following section.
-;
-; When Asterisk is behind a NAT device, the "local" address (and port) that
-; a socket is bound to has different values when seen from the inside or
-; from the outside of the NATted network. Unfortunately this address must
-; be communicated to the outside (e.g. in SIP and SDP messages), and in
-; order to determine the correct value Asterisk needs to know:
-;
-; + whether it is talking to someone "inside" or "outside" of the NATted network.
-; This is configured by assigning the "localnet" parameter with a list
-; of network addresses that are considered "inside" of the NATted network.
-; IF LOCALNET IS NOT SET, THE EXTERNAL ADDRESS WILL NOT BE SET CORRECTLY.
-; Multiple entries are allowed, e.g. a reasonable set is the following:
-;
-; localnet=192.168.0.0/255.255.0.0 ; RFC 1918 addresses
-; localnet=10.0.0.0/255.0.0.0 ; Also RFC1918
-; localnet=172.16.0.0/12 ; Another RFC1918 with CIDR notation
-; localnet=169.254.0.0/255.255.0.0 ; Zero conf local network
-;
-; + the "externally visible" address and port number to be used when talking
-; to a host outside the NAT. This information is derived by one of the
-; following (mutually exclusive) config file parameters:
-;
-; a. "externaddr = hostname[:port]" specifies a static address[:port] to
-; be used in SIP and SDP messages.
-; The hostname is looked up only once, when [re]loading sip.conf .
-; If a port number is not present, use the port specified in the "udpbindaddr"
-; (which is not guaranteed to work correctly, because a NAT box might remap the
-; port number as well as the address).
-; This approach can be useful if you have a NAT device where you can
-; configure the mapping statically. Examples:
-;
-; externaddr = 12.34.56.78 ; use this address.
-; externaddr = 12.34.56.78:9900 ; use this address and port.
-; externaddr = mynat.my.org:12600 ; Public address of my nat box.
-; externtcpport = 9900 ; The externally mapped tcp port, when Asterisk is behind a static NAT or PAT.
-; ; externtcpport will default to the externaddr or externhost port if either one is set.
-; externtlsport = 12600 ; The externally mapped tls port, when Asterisk is behind a static NAT or PAT.
-; ; externtlsport port will default to the RFC designated port of 5061.
-;
-; b. "externhost = hostname[:port]" is similar to "externaddr" except
-; that the hostname is looked up every "externrefresh" seconds
-; (default 10s). This can be useful when your NAT device lets you choose
-; the port mapping, but the IP address is dynamic.
-; Beware, you might suffer from service disruption when the name server
-; resolution fails. Examples:
-;
-; externhost=foo.dyndns.net ; refreshed periodically
-; externrefresh=180 ; change the refresh interval
-;
-; Note that at the moment all these mechanism work only for the SIP socket.
-; The IP address discovered with externaddr/externhost is reused for
-; media sessions as well, but the port numbers are not remapped so you
-; may still experience problems.
-;
-; NOTE 1: in some cases, NAT boxes will use different port numbers in
-; the internal<->external mapping. In these cases, the "externaddr" and
-; "externhost" might not help you configure addresses properly.
-;
-; NOTE 2: when using "externaddr" or "externhost", the address part is
-; also used as the external address for media sessions. Thus, the port
-; information in the SDP may be wrong!
-;
-; In addition to the above, Asterisk has an additional "nat" parameter to
-; address NAT-related issues in incoming SIP or media sessions.
-; In particular, depending on the 'nat= ' settings described below, Asterisk
-; may override the address/port information specified in the SIP/SDP messages,
-; and use the information (sender address) supplied by the network stack instead.
-; However, this is only useful if the external traffic can reach us.
-; The following settings are allowed (both globally and in individual sections):
-;
-; nat = no ; Do no special NAT handling other than RFC3581
-; nat = force_rport ; Pretend there was an rport parameter even if there wasn't
-; nat = comedia ; Send media to the port Asterisk received it from regardless
-; ; of where the SDP says to send it.
-; nat = auto_force_rport ; Set the force_rport option if Asterisk detects NAT (default)
-; nat = auto_comedia ; Set the comedia option if Asterisk detects NAT
-;
-; The nat settings can be combined. For example, to set both force_rport and comedia
-; one would set nat=force_rport,comedia. If any of the comma-separated options is 'no',
-; Asterisk will ignore any other settings and set nat=no. If one of the "auto" settings
-; is used in conjunction with its non-auto counterpart (nat=comedia,auto_comedia), then
-; the non-auto option will be ignored.
-;
-; The RFC 3581-defined 'rport' parameter allows a client to request that Asterisk send
-; SIP responses to it via the source IP and port from which the request originated
-; instead of the address/port listed in the top-most Via header. This is useful if a
-; client knows that it is behind a NAT and therefore cannot guess from what address/port
-; its request will be sent. Asterisk will always honor the 'rport' parameter if it is
-; sent. The force_rport setting causes Asterisk to always send responses back to the
-; address/port from which it received requests; even if the other side doesn't support
-; adding the 'rport' parameter.
-;
-; 'comedia RTP handling' refers to the technique of sending RTP to the port that the
-; the other endpoint's RTP arrived from, and means 'connection-oriented media'. This is
-; only partially related to RFC 4145 which was referred to as COMEDIA while it was in
-; draft form. This method is used to accomodate endpoints that may be located behind
-; NAT devices, and as such the address/port they tell Asterisk to send RTP packets to
-; for their media streams is not the actual address/port that will be used on the nearer
-; side of the NAT.
-;
-; IT IS IMPORTANT TO NOTE that if the nat setting in the general section differs from
-; the nat setting in a peer definition, then the peer username will be discoverable
-; by outside parties as Asterisk will respond to different ports for defined and
-; undefined peers. For this reason it is recommended to ONLY DEFINE NAT SETTINGS IN THE
-; GENERAL SECTION. Specifically, if nat=force_rport in one section and nat=no in the
-; other, then valid peers with settings differing from those in the general section will
-; be discoverable.
-;
-; In addition to these settings, Asterisk *always* uses 'symmetric RTP' mode as defined by
-; RFC 4961; Asterisk will always send RTP packets from the same port number it expects
-; to receive them on.
-;
-; The IP address used for media (audio, video, and text) in the SDP can also be overridden by using
-; the media_address configuration option. This is only applicable to the general section and
-; can not be set per-user or per-peer.
-;
-; Note that this does not change the listen address for RTP, it only changes the
-; advertised address in the SDP. The Asterisk RTP engine will still listen on
-; the standard IP address.
-;
-; media_address = 172.16.42.1
-;
-; Through the use of the res_stun_monitor module, Asterisk has the ability to detect when the
-; perceived external network address has changed. When the stun_monitor is installed and
-; configured, chan_sip will renew all outbound registrations when the monitor detects any sort
-; of network change has occurred. By default this option is enabled, but only takes effect once
-; res_stun_monitor is configured. If res_stun_monitor is enabled and you wish to not
-; generate all outbound registrations on a network change, use the option below to disable
-; this feature.
-;
-; subscribe_network_change_event = yes ; on by default
-;
-; ICE/STUN/TURN usage can be enabled globally or on a per-peer basis using the icesupport
-; configuration option. When set to yes ICE support is enabled. When set to no it is disabled.
-; It is disabled by default.
-;
-; icesupport = yes
-
-; ---------------------------------- MEDIA HANDLING --------------------------------
-; By default, Asterisk tries to re-invite media streams to an optimal path. If there's
-; no reason for Asterisk to stay in the media path, the media will be redirected.
-; This does not really work well in the case where Asterisk is outside and the
-; clients are on the inside of a NAT. In that case, you want to set directmedia=nonat.
-;
-;directmedia=yes ; Asterisk by default tries to redirect the
- ; RTP media stream to go directly from
- ; the caller to the callee. Some devices do not
- ; support this (especially if one of them is behind a NAT).
- ; The default setting is YES. If you have all clients
- ; behind a NAT, or for some other reason want Asterisk to
- ; stay in the audio path, you may want to turn this off.
-
- ; This setting also affect direct RTP
- ; at call setup (a new feature in 1.4 - setting up the
- ; call directly between the endpoints instead of sending
- ; a re-INVITE).
-
- ; Additionally this option does not disable all reINVITE operations.
- ; It only controls Asterisk generating reINVITEs for the specific
- ; purpose of setting up a direct media path. If a reINVITE is
- ; needed to switch a media stream to inactive (when placed on
- ; hold) or to T.38, it will still be done, regardless of this
- ; setting. Note that direct T.38 is not supported.
-
-;directmedia=nonat ; An additional option is to allow media path redirection
- ; (reinvite) but only when the peer where the media is being
- ; sent is known to not be behind a NAT (as the RTP core can
- ; determine it based on the apparent IP address the media
- ; arrives from).
-
-;directmedia=update ; Yet a third option... use UPDATE for media path redirection,
- ; instead of INVITE. This can be combined with 'nonat', as
- ; 'directmedia=update,nonat'. It implies 'yes'.
-
-;directmedia=outgoing ; When sending directmedia reinvites, do not send an immediate
- ; reinvite on an incoming call leg. This option is useful when
- ; peered with another SIP user agent that is known to send
- ; immediate direct media reinvites upon call establishment. Setting
- ; the option in this situation helps to prevent potential glares.
- ; Setting this option implies 'yes'.
-
-;directrtpsetup=yes ; Enable the new experimental direct RTP setup. This sets up
- ; the call directly with media peer-2-peer without re-invites.
- ; Will not work for video and cases where the callee sends
- ; RTP payloads and fmtp headers in the 200 OK that does not match the
- ; callers INVITE. This will also fail if directmedia is enabled when
- ; the device is actually behind NAT.
-
-;directmediadeny=0.0.0.0/0 ; Use directmediapermit and directmediadeny to restrict
-;directmediapermit=172.16.0.0/16; which RTP source IPs should be able to pass directmedia to
- ; each other. Note that directmedia ACLs are not a global
- ; setting, but must be defined per peer.
- ; (There is no default setting, this is just an example)
- ; Use this if some of your phones are on IP addresses that
- ; can not reach each other directly. This way you can force
- ; RTP to always flow through asterisk in such cases.
-;directmediaacl=acl_example ; Use named ACLs defined in acl.conf
-
-;ignoresdpversion=yes ; By default, Asterisk will honor the session version
- ; number in SDP packets and will only modify the SDP
- ; session if the version number changes. This option will
- ; force asterisk to ignore the SDP session version number
- ; and treat all SDP data as new data. This is required
- ; for devices that send us non standard SDP packets
- ; (observed with Microsoft OCS). By default this option is
- ; off.
-
-;sdpsession=Asterisk PBX ; Allows you to change the SDP session name string, (s=)
- ; Like the useragent parameter, the default user agent string
- ; also contains the Asterisk version.
-;sdpowner=root ; Allows you to change the username field in the SDP owner string, (o=)
- ; This field MUST NOT contain spaces
-;encryption=no ; Whether to offer SRTP encrypted media (and only SRTP encrypted media)
- ; on outgoing calls to a peer. Calls will fail with HANGUPCAUSE=58 if
- ; the peer does not support SRTP. Defaults to no.
-;encryption_taglen=80 ; Set the auth tag length offered in the INVITE either 32/80 default 80
-;
-;avpf=yes ; Enable inter-operability with media streams using the AVPF RTP profile.
- ; This will cause all offers and answers to use AVPF (or SAVPF). This
- ; option may be specified at the global or peer scope.
-;force_avp=yes ; Force 'RTP/AVP', 'RTP/AVPF', 'RTP/SAVP', and 'RTP/SAVPF' to be used for
- ; media streams when appropriate, even if a DTLS stream is present.
-;rtcp_mux=yes ; Enable support for RFC 5761 RTCP multiplexing which is required for
- ; WebRTC support
-; ---------------------------------------- REALTIME SUPPORT ------------------------
-; For additional information on ARA, the Asterisk Realtime Architecture,
-; please read https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration
-;
-;rtcachefriends=yes ; Cache realtime friends by adding them to the internal list
- ; just like friends added from the config file only on a
- ; as-needed basis? (yes|no)
-
-;rtsavesysname=yes ; Save systemname in realtime database at registration
- ; Default= no
-
-;rtupdate=yes ; Send registry updates to database using realtime? (yes|no)
- ; If set to yes, when a SIP UA registers successfully, the ip address,
- ; the origination port, the registration period, and the username of
- ; the UA will be set to database via realtime.
- ; If not present, defaults to 'yes'. Note: realtime peers will
- ; probably not function across reloads in the way that you expect, if
- ; you turn this option off.
-;rtautoclear=yes ; Auto-Expire friends created on the fly on the same schedule
- ; as if it had just registered? (yes|no|)
- ; If set to yes, when the registration expires, the friend will
- ; vanish from the configuration until requested again. If set
- ; to an integer, friends expire within this number of seconds
- ; instead of the registration interval.
-
-;ignoreregexpire=yes ; Enabling this setting has two functions:
- ;
- ; For non-realtime peers, when their registration expires, the
- ; information will _not_ be removed from memory or the Asterisk database
- ; if you attempt to place a call to the peer, the existing information
- ; will be used in spite of it having expired
- ;
- ; For realtime peers, when the peer is retrieved from realtime storage,
- ; the registration information will be used regardless of whether
- ; it has expired or not; if it expires while the realtime peer
- ; is still in memory (due to caching or other reasons), the
- ; information will not be removed from realtime storage
-
-; ---------------------------------------- SIP DOMAIN SUPPORT ------------------------
-; Incoming INVITE and REFER messages can be matched against a list of 'allowed'
-; domains, each of which can direct the call to a specific context if desired.
-; By default, all domains are accepted and sent to the default context or the
-; context associated with the user/peer placing the call.
-; REGISTER to non-local domains will be automatically denied if a domain
-; list is configured.
-;
-; Domains can be specified using:
-; domain=[,]
-; Examples:
-; domain=myasterisk.dom
-; domain=customer.com,customer-context
-;
-; In addition, all the 'default' domains associated with a server should be
-; added if incoming request filtering is desired.
-; autodomain=yes
-;
-; To disallow requests for domains not serviced by this server:
-; allowexternaldomains=no
-
-;domain=mydomain.tld,mydomain-incoming
- ; Add domain and configure incoming context
- ; for external calls to this domain
-;domain=1.2.3.4 ; Add IP address as local domain
- ; You can have several "domain" settings
-;allowexternaldomains=no ; Disable INVITE and REFER to non-local domains
- ; Default is yes
-;autodomain=yes ; Turn this on to have Asterisk add local host
- ; name and local IP to domain list.
-
-; fromdomain=mydomain.tld ; When making outbound SIP INVITEs to
- ; non-peers, use your primary domain "identity"
- ; for From: headers instead of just your IP
- ; address. This is to be polite and
- ; it may be a mandatory requirement for some
- ; destinations which do not have a prior
- ; account relationship with your server.
-
-; ----------------------------- Advice of Charge CONFIGURATION --------------------------
-; snom_aoc_enabled = yes; ; This options turns on and off support for sending AOC-D and
- ; AOC-E to snom endpoints. This option can be used both in the
- ; peer and global scope. The default for this option is off.
-
-
-; ----------------------------- JITTER BUFFER CONFIGURATION --------------------------
-; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
- ; SIP channel. Defaults to "no". An enabled jitterbuffer will
- ; be used only if the sending side can create and the receiving
- ; side can not accept jitter. The SIP channel can accept jitter,
- ; thus a jitterbuffer on the receive SIP side will be used only
- ; if it is forced and enabled.
-
-; jbforce = no ; Forces the use of a jitterbuffer on the receive side of a SIP
- ; channel. Defaults to "no".
-
-; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
-
-; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
- ; resynchronized. Useful to improve the quality of the voice, with
- ; big jumps in/broken timestamps, usually sent from exotic devices
- ; and programs. Defaults to 1000.
-
-; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a SIP
- ; channel. Two implementations are currently available - "fixed"
- ; (with size always equals to jbmaxsize) and "adaptive" (with
- ; variable size, actually the new jb of IAX2). Defaults to fixed.
-
-; jbtargetextra = 40 ; This option only affects the jb when 'jbimpl = adaptive' is set.
- ; The option represents the number of milliseconds by which the new jitter buffer
- ; will pad its size. the default is 40, so without modification, the new
- ; jitter buffer will set its size to the jitter value plus 40 milliseconds.
- ; increasing this value may help if your network normally has low jitter,
- ; but occasionally has spikes.
-
-; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
-
-; ----------------------------------------------------------------------------------
-
-[authentication]
-; Global credentials for outbound calls, i.e. when a proxy challenges your
-; Asterisk server for authentication. These credentials override
-; any credentials in peer/register definition if realm is matched.
-;
-; This way, Asterisk can authenticate for outbound calls to other
-; realms. We match realm on the proxy challenge and pick an set of
-; credentials from this list
-; Syntax:
-; auth = :@
-; auth = #@
-; Example:
-;auth=mark:topsecret@digium.com
-;
-; You may also add auth= statements to [peer] definitions
-; Peer auth= override all other authentication settings if we match on realm
-
-; -----------------------------------------------------------------------------
-; DEVICE CONFIGURATION
-;
-; SIP entities have a 'type' which determines their roles within Asterisk.
-; * For entities with 'type=peer':
-; Peers handle both inbound and outbound calls and are matched by ip/port, so for
-; The case of incoming calls from the peer, the IP address must match in order for
-; The invitation to work. This means calls made from either direction won't work if
-; The peer is unregistered while host=dynamic or if the host is otherise not set to
-; the correct IP of the sender.
-; * For entities with 'type=user':
-; Asterisk users handle inbound calls only (meaning they call Asterisk, Asterisk can't
-; call them) and are matched by their authorization information (authname and secret).
-; Asterisk doesn't rely on their IP and will accept calls regardless of the host setting
-; as long as the incoming SIP invite authorizes successfully.
-; * For entities with 'type=friend':
-; Asterisk will create the entity as both a friend and a peer. Asterisk will accept
-; calls from friends like it would for users, requiring only that the authorization
-; matches rather than the IP address. Since it is also a peer, a friend entity can
-; be called as long as its IP is known to Asterisk. In the case of host=dynamic,
-; this means it is necessary for the entity to register before Asterisk can call it.
-;
-; Use remotesecret for outbound authentication, and secret for authenticating
-; inbound requests. For historical reasons, if no remotesecret is supplied for an
-; outbound registration or call, the secret will be used.
-;
-; For device names, we recommend using only a-z, numerics (0-9) and underscore
-;
-; For local phones, type=friend works most of the time
-;
-; If you have one-way audio, you probably have NAT problems.
-; If Asterisk is on a public IP, and the phone is inside of a NAT device
-; you will need to configure nat option for those phones.
-; Also, turn on qualify=yes to keep the nat session open
-;
-; Configuration options available
-; --------------------
-; context
-; callingpres
-; permit
-; deny
-; secret
-; md5secret
-; remotesecret
-; transport
-; dtmfmode
-; directmedia
-; nat
-; callgroup
-; pickupgroup
-; language
-; allow
-; disallow
-; autoframing
-; insecure
-; trustrpid
-; trust_id_outbound
-; progressinband
-; promiscredir
-; useclientcode
-; accountcode
-; setvar
-; callerid
-; amaflags
-; callcounter
-; busylevel
-; allowoverlap
-; allowsubscribe
-; allowtransfer
-; ignoresdpversion
-; subscribecontext
-; template
-; videosupport
-; maxcallbitrate
-; rfc2833compensate
-; Note: app_voicemail mailboxes must be in the form of mailbox@context.
-; mailbox
-; session-timers
-; session-expires
-; session-minse
-; session-refresher
-; t38pt_usertpsource
-; regexten
-; fromdomain
-; fromuser
-; host
-; port
-; qualify
-; keepalive
-; defaultip
-; defaultuser
-; rtptimeout
-; rtpholdtimeout
-; sendrpid
-; outboundproxy
-; rfc2833compensate
-; callbackextension
-; timert1
-; timerb
-; qualifyfreq
-; t38pt_usertpsource
-; contactpermit ; Limit what a host may register as (a neat trick
-; contactdeny ; is to register at the same IP as a SIP provider,
-; contactacl ; then call oneself, and get redirected to that
-; ; same location).
-; directmediapermit
-; directmediadeny
-; directmediaacl
-; unsolicited_mailbox
-; use_q850_reason
-; maxforwards
-; encryption
-; description ; Used to provide a description of the peer in console output
-; dtlsenable
-; dtlsautogeneratecert
-; dtlsverify
-; dtlsrekey
-; dtlscertfile
-; dtlsprivatekey
-; dtlscipher
-; dtlscafile
-; dtlscapath
-; dtlssetup
-; dtlsfingerprint
-; ignore_requested_pref ; Ignore the requested codec and determine the preferred codec
-; ; from the peer's configuration.
-;
-
-; -----------------------------------------------------------------------------
-; DTLS-SRTP CONFIGURATION
-;
-; DTLS-SRTP support is available if the underlying RTP engine in use supports it.
-;
-; Note that all configuration options except dtlsenable can be set at the general level.
-; If set they will be present on the user or peer unless overridden with a different value.
-;
-; dtlsenable = yes ; Enable or disable DTLS-SRTP support
-; dtlsverify = yes ; Verify that provided peer certificate and fingerprint are valid
-; ; A value of 'yes' will perform both certificate and fingerprint verification
-; ; A value of 'no' will perform no certificate or fingerprint verification
-; ; A value of 'fingerprint' will perform ONLY fingerprint verification
-; ; A value of 'certificate' will perform ONLY certficiate verification
-; dtlsrekey = 60 ; Interval at which to renegotiate the TLS session and rekey the SRTP session
-; ; If this is not set or the value provided is 0 rekeying will be disabled
-; dtlsautogeneratecert = yes ; Enable ephemeral DTLS certificate generation. The default is 'no.'
-; dtlscertfile = file ; Path to certificate file to present
-; dtlsprivatekey = file ; Path to private key for certificate file
-; dtlscipher = ; Cipher to use for TLS negotiation
-; ; A list of valid SSL cipher strings can be found at:
-; ; http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
-; dtlscafile = file ; Path to certificate authority certificate
-; dtlscapath = path ; Path to a directory containing certificate authority certificates
-; dtlssetup = actpass ; Whether we are willing to accept connections, connect to the other party, or both.
-; ; Valid options are active (we want to connect to the other party), passive (we want to
-; ; accept connections only), and actpass (we will do both). This value will be used in
-; ; the outgoing SDP when offering and for incoming SDP offers when the remote party sends
-; ; actpass
-; dtlsfingerprint = sha-1 ; The hash to use for the fingerprint in SDP (valid options are sha-1 and sha-256)
-
-;[sip_proxy]
-; For incoming calls only. Example: FWD (Free World Dialup)
-; We match on IP address of the proxy for incoming calls
-; since we can not match on username (caller id)
-;type=peer
-;context=from-fwd
-;host=fwd.pulver.com
-
-;[sip_proxy-out]
-;type=peer ; we only want to call out, not be called
-;remotesecret=guessit ; Our password to their service
-;defaultuser=yourusername ; Authentication user for outbound proxies
-;fromuser=yourusername ; Many SIP providers require this!
-;fromdomain=provider.sip.domain
-;host=box.provider.com
-;transport=udp,tcp ; This sets the default transport type to udp for outgoing, and will
-; ; accept both tcp and udp. The default transport type is only used for
-; ; outbound messages until a Registration takes place. During the
-; ; peer Registration the transport type may change to another supported
-; ; type if the peer requests so.
-
-;usereqphone=yes ; This provider requires ";user=phone" on URI
-;callcounter=yes ; Enable call counter
-;busylevel=2 ; Signal busy at 2 or more calls
-;outboundproxy=proxy.provider.domain ; send outbound signaling to this proxy, not directly to the peer
-;port=80 ; The port number we want to connect to on the remote side
- ; Also used as "defaultport" in combination with "defaultip" settings
-
-; -- sample definition for a provider
-;[provider1]
-;type=peer
-;host=sip.provider1.com
-;fromuser=4015552299 ; how your provider knows you
-;remotesecret=youwillneverguessit ; The password we use to authenticate to them
-;secret=gissadetdu ; The password they use to contact us
-;callbackextension=123 ; Register with this server and require calls coming back to this extension
-;transport=udp,tcp ; This sets the transport type to udp for outgoing, and will
-; ; accept both tcp and udp. Default is udp. The first transport
-; ; listed will always be used for outgoing connections.
-;unsolicited_mailbox=4015552299 ; If the remote SIP server sends an unsolicited MWI NOTIFY message the new/old
-; ; message count will be stored in the configured virtual mailbox. It can be used
-; ; by any device supporting MWI by specifying @SIP_Remote as the
-; ; mailbox.
-
-;
-; Because you might have a large number of similar sections, it is generally
-; convenient to use templates for the common parameters, and add them
-; the the various sections. Examples are below, and we can even leave
-; the templates uncommented as they will not harm:
-
-[basic-options](!) ; a template
- dtmfmode=rfc2833
- context=from-office
- type=friend
-
-[natted-phone](!,basic-options) ; another template inheriting basic-options
- directmedia=no
- host=dynamic
-
-[public-phone](!,basic-options) ; another template inheriting basic-options
- directmedia=yes
-
-[my-codecs](!) ; a template for my preferred codecs
- disallow=all
- allow=ilbc
- allow=g729
- allow=gsm
- allow=g723
- allow=ulaw
- ; Or, more simply:
- ;allow=!all,ilbc,g729,gsm,g723,ulaw
-
-[ulaw-phone](!) ; and another one for ulaw-only
- disallow=all
- allow=ulaw
- ; Again, more simply:
- ;allow=!all,ulaw
-
-; and finally instantiate a few phones
-;
-; [2133](natted-phone,my-codecs)
-; secret = peekaboo
-; [2134](natted-phone,ulaw-phone)
-; secret = not_very_secret
-; [2136](public-phone,ulaw-phone)
-; secret = not_very_secret_either
-; ...
-;
-
-; Standard configurations not using templates look like this:
-;
-;[grandstream1]
-;type=friend
-;context=from-sip ; Where to start in the dialplan when this phone calls
-;recordonfeature=dynamicfeature1 ; Feature to use when INFO with Record: on is received.
-;recordofffeature=dynamicfeature2 ; Feature to use when INFO with Record: off is received.
-;callerid=John Doe <1234> ; Full caller ID, to override the phones config
- ; on incoming calls to Asterisk
-;description=Courtesy Phone ; Description of the peer. Shown when doing 'sip show peers'.
-;host=192.168.0.23 ; we have a static but private IP address
- ; No registration allowed
-;directmedia=yes ; allow RTP voice traffic to bypass Asterisk
-;dtmfmode=info ; either RFC2833 or INFO for the BudgeTone
-;call-limit=1 ; permit only 1 outgoing call and 1 incoming call at a time
- ; from the phone to asterisk (deprecated)
- ; 1 for the explicit peer, 1 for the explicit user,
- ; remember that a friend equals 1 peer and 1 user in
- ; memory
- ; There is no combined call counter for a "friend"
- ; so there's currently no way in sip.conf to limit
- ; to one inbound or outbound call per phone. Use
- ; the group counters in the dial plan for that.
- ;
-;mailbox=1234@default ; mailbox 1234 in voicemail context "default"
-;disallow=all ; need to disallow=all before we can use allow=
-;allow=ulaw ; Note: In user sections the order of codecs
- ; listed with allow= does NOT matter!
-;allow=alaw
-;allow=g723.1 ; Asterisk only supports g723.1 pass-thru!
-;allow=g729 ; Pass-thru only unless g729 license obtained
-;callingpres=allowed_passed_screen ; Set caller ID presentation
- ; See function CALLERPRES documentation for possible
- ; values.
-
-;[xlite1]
-; Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!
-; Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
-;type=friend
-;regexten=1234 ; When they register, create extension 1234
-;callerid="Jane Smith" <5678>
-;host=dynamic ; This device needs to register
-;directmedia=no ; Typically set to NO if behind NAT
-;disallow=all
-;allow=gsm ; GSM consumes far less bandwidth than ulaw
-;allow=ulaw
-;allow=alaw
-;mailbox=1234@default,1233@default ; Subscribe to status of multiple mailboxes
-
-;[snom]
-;type=friend ; Friends place calls and receive calls
-;context=from-sip ; Context for incoming calls from this user
-;secret=blah
-;subscribecontext=localextensions ; Only allow SUBSCRIBE for local extensions
-;language=de ; Use German prompts for this user
-;host=dynamic ; This peer register with us
-;dtmfmode=inband ; Choices are inband, rfc2833, or info
-;defaultip=192.168.0.59 ; IP used until peer registers
-;mailbox=1234@context,2345@context ; Mailbox(-es) for message waiting indicator
-;subscribemwi=yes ; Only send notifications if this phone
- ; subscribes for mailbox notification
-;vmexten=voicemail ; dialplan extension to reach mailbox
- ; sets the Message-Account in the MWI notify message
- ; defaults to global vmexten which defaults to "asterisk"
-;disallow=all
-;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!
-
-
-;[polycom]
-;type=friend ; Friends place calls and receive calls
-;context=from-sip ; Context for incoming calls from this user
-;secret=blahpoly
-;host=dynamic ; This peer register with us
-;dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
-;defaultuser=polly ; Username to use in INVITE until peer registers
-;defaultip=192.168.40.123
- ; Normally you do NOT need to set this parameter
-;disallow=all
-;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!
-;progressinband=no ; Polycom phones don't work properly with "never"
-
-
-;[pingtel]
-;type=friend
-;secret=blah
-;host=dynamic
-;insecure=port ; Allow matching of peer by IP address without
- ; matching port number
-;insecure=invite ; Do not require authentication of incoming INVITEs
-;insecure=port,invite ; (both)
-;qualify=1000 ; Consider it down if it's 1 second to reply
- ; Helps with NAT session
- ; qualify=yes uses default value
-;qualifyfreq=60 ; Qualification: How often to check for the
- ; host to be up in seconds
- ; Set to low value if you use low timeout for
- ; NAT of UDP sessions
-;
-; Call group and Pickup group should be in the range from 0 to 63
-;
-;callgroup=1,3-4 ; We are in caller groups 1,3,4
-;pickupgroup=1,3-5 ; We can do call pick-p for call group 1,3,4,5
-;namedcallgroup=engineering,sales,netgroup,protgroup ; We are in named call groups engineering,sales,netgroup,protgroup
-;namedpickupgroup=sales ; We can do call pick-p for named call group sales
-;defaultip=192.168.0.60 ; IP address to use if peer has not registered
-;deny=0.0.0.0/0.0.0.0 ; ACL: Control access to this account based on IP address
-;permit=192.168.0.60/255.255.255.0
-;permit=192.168.0.60/24 ; we can also use CIDR notation for subnet masks
-;permit=2001:db8::/32 ; IPv6 ACLs can be specified if desired. IPv6 ACLs
- ; apply only to IPv6 addresses, and IPv4 ACLs apply
- ; only to IPv4 addresses.
-;acl=named_acl_example ; Use named ACLs defined in acl.conf
-
-;[cisco1]
-;type=friend
-;secret=blah
-;qualify=200 ; Qualify peer is no more than 200ms away
-;host=dynamic ; This device registers with us
-;directmedia=no ; Asterisk by default tries to redirect the
- ; RTP media stream (audio) to go directly from
- ; the caller to the callee. Some devices do not
- ; support this (especially if one of them is
- ; behind a NAT).
-;defaultip=192.168.0.4 ; IP address to use until registration
-;defaultuser=goran ; Username to use when calling this device before registration
- ; Normally you do NOT need to set this parameter
-;setvar=CUSTID=5678 ; Channel variable to be set for all calls from or to this device
-;setvar=ATTENDED_TRANSFER_COMPLETE_SOUND=beep ; This channel variable will
- ; cause the given audio file to
- ; be played upon completion of
- ; an attended transfer to the
- ; target of the transfer.
-
-;[pre14-asterisk]
-;type=friend
-;secret=digium
-;host=dynamic
-;rfc2833compensate=yes ; Compensate for pre-1.4 DTMF transmission from another Asterisk machine.
- ; You must have this turned on or DTMF reception will work improperly.
-;t38pt_usertpsource=yes ; Use the source IP address of RTP as the destination IP address for UDPTL packets
- ; if the nat option is enabled. If a single RTP packet is received Asterisk will know the
- ; external IP address of the remote device. If port forwarding is done at the client side
- ; then UDPTL will flow to the remote device.
diff --git a/configs/samples/sip_notify.conf.sample b/configs/samples/sip_notify.conf.sample
deleted file mode 100644
index 8224ee1ff4..0000000000
--- a/configs/samples/sip_notify.conf.sample
+++ /dev/null
@@ -1,57 +0,0 @@
-; rfc3842
-; put empty "Content=>" at the end to have CRLF after last body line
-
-[clear-mwi]
-Event=>message-summary
-Content-type=>application/simple-message-summary
-Content=>Messages-Waiting: no
-Content=>Message-Account: sip:asterisk@127.0.0.1
-Content=>Voice-Message: 0/0 (0/0)
-Content=>
-
-; Aastra
-
-[aastra-check-cfg]
-Event=>check-sync
-
-[aastra-xml]
-Event=>aastra-xml
-
-; Digium
-
-[digium-check-cfg]
-Event=>check-sync
-
-; Linksys
-
-[linksys-cold-restart]
-Event=>reboot_now
-
-[linksys-warm-restart]
-Event=>restart_now
-
-; Polycom
-
-[polycom-check-cfg]
-Event=>check-sync
-
-; Sipura
-
-[sipura-check-cfg]
-Event=>resync
-
-[sipura-get-report]
-Event=>report
-
-; snom
-
-[snom-check-cfg]
-Event=>check-sync\;reboot=false
-
-[snom-reboot]
-Event=>check-sync\;reboot=true
-
-; Cisco
-
-[cisco-check-cfg]
-Event=>check-sync
diff --git a/configs/samples/users.conf.sample b/configs/samples/users.conf.sample
index 52f02b5ba6..185fc2a5b2 100644
--- a/configs/samples/users.conf.sample
+++ b/configs/samples/users.conf.sample
@@ -4,12 +4,12 @@
; Creating entries in users.conf is a "shorthand" for creating individual
; entries in each configuration file. Using users.conf is not intended to
; provide you with as much flexibility as using the separate configuration
-; files (e.g. sip.conf, iax.conf, etc) but is intended to accelerate the
+; files (e.g. iax.conf, etc) but is intended to accelerate the
; simple task of adding users. Note that creating individual items (e.g.
-; custom SIP peers, IAX friends, etc.) will allow you to override specific
-; parameters within this file. Parameter names here are the same as they
-; appear in the other configuration files. There is no way to change the
-; value of a parameter here for just one subsystem.
+; IAX friends, etc.) will allow you to override specific parameters within
+; this file. Parameter names here are the same as they appear in the
+; other configuration files. There is no way to change the value of a
+; parameter here for just one subsystem.
;
[general]
@@ -30,10 +30,6 @@ hasvoicemail = yes
;
vmsecret = 1234
;
-; Create SIP Peer
-;
-hassip = yes
-;
; Create IAX friend
;
hasiax = yes
@@ -94,7 +90,6 @@ pickupgroup = 1
;dahdichan = 1
;hasvoicemail = yes
;vmsecret = 1234
-;hassip = yes
;hasiax = no
;hash323 = no
;hasmanager = no
diff --git a/contrib/scripts/ast_logescalator b/contrib/scripts/ast_logescalator
index 054ef9b9c0..b5a44ccefe 100755
--- a/contrib/scripts/ast_logescalator
+++ b/contrib/scripts/ast_logescalator
@@ -15,7 +15,7 @@ SYNOPSIS
$prog [ --help ] | [ [ --reset ] | [
[ --uniqueid="" ]
- [ --pjsip-debug= ] [ --sip-debug= ]
+ [ --pjsip-debug= ]
[ --iax2-debug= ]
[ --agi-debug= ] [ --ami-debug= ]
@@ -26,7 +26,7 @@ SYNOPSIS
[ --dtmf-debug= ] [ --fax-debug= ]
[ --security-debug= ]
- [ --pjsip-history= ] [ --sip-history= ]
+ [ --pjsip-history= ]
[ --verbose= ] [ --debug= ]
] ]
@@ -51,7 +51,7 @@ DESCRIPTION
on/off commands will use the same uniqueid. Use the --reset
option to reset it (and everything else).
- --pjsip-debug --sip-debug --iax2-debug --agi-debug --ami-debug
+ --pjsip-debug --iax2-debug --agi-debug --ami-debug
--ari-debug --cdr-debug --channel-debug --rtp-debug --rtcp-debug
Issues the subsystem appropriate command to turn on
or off debugging. These are usually functional debug messages
@@ -62,10 +62,10 @@ DESCRIPTION
These subsystems set up their own log channels so if turned
on, log files will be created in \$astlogdir for them.
- --pjsip-history --sip-history
- The pjsip and sip channels have the ability to output an
- abbreviated, one-line, packet summary. If enabled, the summaries
- will be written to \$astlogdir/pjsip_history.\$UNIQUEID and
+ --pjsip-history
+ The pjsip channels have the ability to output an abbreviated,
+ one-line, packet summary. If enabled, the summaries will be
+ written to \$astlogdir/pjsip_history.\$UNIQUEID and
\$astlogdir/sip_history.\$UNIQUEID.
--verbose-level --debug-level
@@ -114,7 +114,6 @@ RESET=false
declare -A DEBUG_COMMANDS=(
[PJSIP,on]="pjsip set logger on" [PJSIP,off]="pjsip set logger off"
-[SIP,on]="sip set debug on" [SIP,off]="sip set debug off"
[IAX2,on]="iax2 set debug on" [IAX2,off]="iax2 set debug off"
[ARI,on]="ari set debug all on" [ARI,off]="ari set debug all off"
[AMI,on]="manager set debug on" [AMI,off]="manager set debug off"
@@ -152,8 +151,6 @@ for a in "$@" ; do
DEBUGS=true
;;
--pjsip-history=*)
- ;&
- --sip-history=*)
subsystem=${a%-history=*}
subsystem=${subsystem#--}
if [[ ${a#*=} =~ ^[Yy].* ]] ; then
@@ -224,8 +221,6 @@ if $RESET ; then
asterisk -rx "core set debug 0"
asterisk -rx "pjsip set logger off"
asterisk -rx "pjsip set history off"
- asterisk -rx "sip set debug off"
- asterisk -rx "sip set history off"
asterisk -rx "iax2 set debug off"
asterisk -rx "manager set debug off"
asterisk -rx "ari set debug all off"
@@ -259,9 +254,6 @@ if ! grep -q "; --START DEBUG_LOGGING-- ;" $CLI_CONF ; then
[pjsip_debug](!)
pjsip set logger on = yes
- [sip_debug](!)
- sip set debug on = yes
-
[iax2_debug](!)
iax2 set debug on = yes
@@ -299,10 +291,6 @@ if ! grep -q "; --START DEBUG_LOGGING-- ;" $CLI_CONF ; then
logger add channel $PJSIP_HISTORY_LOG PJSIP_HISTORY = yes
pjsip set history on = yes
- [sip_history](!)
- logger add channel $SIP_HISTORY_LOG SIP_HISTORY = yes
- sip set history on = yes
-
[verbose_level](!)
core set verbose 3 = yes
@@ -327,13 +315,12 @@ else
VERBOSE_LOG=$(sed -n -r -e "s@logger add channel ($LOG_DIR/message\..+)\s+NOTICE.*@\1@p" "$CLI_CONF")
DEBUG_LOG=$(sed -n -r -e "s@logger add channel ($LOG_DIR/debug\..+)\s+DEBUG.*@\1@p" "$CLI_CONF")
PJSIP_HISTORY_LOG=$(sed -n -r -e "s@logger add channel ($LOG_DIR/pjsip_history\..+)\s+PJSIP.*@\1@p" "$CLI_CONF")
- SIP_HISTORY_LOG=$(sed -n -r -e "s@logger add channel ($LOG_DIR/sip_history\..+)\s+SIP.*@\1@p" "$CLI_CONF")
DTMF_LOG=$(sed -n -r -e "s@logger add channel ($LOG_DIR/dtmf\..+)\s+DTMF.*@\1@p" "$CLI_CONF")
FAX_LOG=$(sed -n -r -e "s@logger add channel ($LOG_DIR/fax\..+)\s+FAX.*@\1@p" "$CLI_CONF")
SECURITY_LOG=$(sed -n -r -e "s@logger add channel ($LOG_DIR/security\..+)\s+SECURITY.*@\1@p" "$CLI_CONF")
fi
-for x in PJSIP SIP ARI AMI AGI ARI IAX2 CDR RTP RTCP ; do
+for x in PJSIP ARI AMI AGI ARI IAX2 CDR RTP RTCP ; do
if eval \$${x}_DEBUG_SPECIFIED ; then
if eval \$${x}_DEBUG ; then
if $ASTERISK_IS_RUNNING ; then
@@ -367,7 +354,7 @@ for x in DTMF FAX SECURITY ; do
fi
done
-for x in PJSIP SIP ; do
+for x in PJSIP ; do
if eval \$${x}_HISTORY_SPECIFIED ; then
if eval \$${x}_HISTORY ; then
if $ASTERISK_IS_RUNNING ; then
diff --git a/contrib/scripts/autosupport b/contrib/scripts/autosupport
index 414cb49f6a..265638950d 100755
--- a/contrib/scripts/autosupport
+++ b/contrib/scripts/autosupport
@@ -192,7 +192,6 @@ if [ -e /var/run/asterisk.ctl ] || [ -e /var/run/asterisk/asterisk.ctl ]; then
"core show uptime" "core show settings" "core show sysinfo" "core show channels" \
"pri show spans" "dahdi show status" "dahdi show channels" "dahdi show channel 1" \
"pjsip show endpoints" "pjsip show registrations" "pjsip list channels" \
- "sip show peers" "sip show registry" "sip show channels" "sip show subscriptions" "sip show settings" \
"show g729" "g729 show version" "g729 show licenses" "g729 show hostid" \
"digium_phones show version" "digium_phones show alerts" "digium_phones show applications" \
"digium_phones show firmwares" "digium_phones show lines" "digium_phones show networks" \
diff --git a/contrib/scripts/retrieve_sip_conf_from_mysql.pl b/contrib/scripts/retrieve_sip_conf_from_mysql.pl
deleted file mode 100644
index 6bf775eac5..0000000000
--- a/contrib/scripts/retrieve_sip_conf_from_mysql.pl
+++ /dev/null
@@ -1,92 +0,0 @@
-#!/usr/bin/perl -Tw
-# Retrieves the sip user/peer entries from the database
-# Use these commands to create the appropriate tables in MySQL
-#
-#CREATE TABLE sip (id INT(11) DEFAULT -1 NOT NULL,keyword VARCHAR(20) NOT NULL,data VARCHAR(50) NOT NULL, flags INT(1) DEFAULT 0 NOT NULL,PRIMARY KEY (id,keyword));
-#
-# if flags = 1 then the records are not included in the output file
-
-use DBI;
-################### BEGIN OF CONFIGURATION ####################
-
-# the name of the extensions table
-$table_name = "sip";
-# the path to the extensions.conf file
-# WARNING: this file will be substituted by the output of this program
-$sip_conf = "/etc/asterisk/sip_additional.conf";
-# the name of the box the MySQL database is running on
-$hostname = "localhost";
-# the name of the database our tables are kept
-$database = "sip";
-# username to connect to the database
-$username = "root";
-# password to connect to the database
-$password = "";
-
-################### END OF CONFIGURATION #######################
-
-$additional = "";
-
-open EXTEN, ">$sip_conf" || die "Cannot create/overwrite extensions file: $sip_conf\n";
-
-$dbh = DBI->connect("dbi:mysql:dbname=$database;host=$hostname", "$username", "$password");
-$statement = "SELECT keyword,data from $table_name where id=0 and keyword <> 'account' and flags <> 1";
-my $result = $dbh->selectall_arrayref($statement);
-unless ($result) {
- # check for errors after every single database call
- print "dbh->selectall_arrayref($statement) failed!\n";
- print "DBI::err=[$DBI::err]\n";
- print "DBI::errstr=[$DBI::errstr]\n";
- exit;
-}
-my @resultSet = @{$result};
-if ( $#resultSet > -1 ) {
- foreach $row (@{ $result }) {
- my @result = @{ $row };
- $additional .= $result[0]."=".$result[1]."\n";
- }
-}
-
-$statement = "SELECT data,id from $table_name where keyword='account' and flags <> 1 group by data";
-
-$result = $dbh->selectall_arrayref($statement);
-unless ($result) {
- # check for errors after every single database call
- print "dbh->selectall_arrayref($statement) failed!\n";
- print "DBI::err=[$DBI::err]\n";
- print "DBI::errstr=[$DBI::errstr]\n";
-}
-
-@resultSet = @{$result};
-if ( $#resultSet == -1 ) {
- print "No sip accounts defined in $table_name\n";
- exit;
-}
-
-foreach my $row ( @{ $result } ) {
- my $account = @{ $row }[0];
- my $id = @{ $row }[1];
- print EXTEN "[$account]\n";
- $statement = "SELECT keyword,data from $table_name where id=$id and keyword <> 'account' and flags <> 1 order by keyword";
- my $result = $dbh->selectall_arrayref($statement);
- unless ($result) {
- # check for errors after every single database call
- print "dbh->selectall_arrayref($statement) failed!\n";
- print "DBI::err=[$DBI::err]\n";
- print "DBI::errstr=[$DBI::errstr]\n";
- exit;
- }
-
- my @resSet = @{$result};
- if ( $#resSet == -1 ) {
- print "no results\n";
- exit;
- }
- foreach my $row ( @{ $result } ) {
- my @result = @{ $row };
- print EXTEN "$result[0]=$result[1]\n";
- }
- print EXTEN "$additional\n";
-}
-
-exit 0;
diff --git a/contrib/scripts/sip_nat_settings b/contrib/scripts/sip_nat_settings
deleted file mode 100755
index 444fb67c3e..0000000000
--- a/contrib/scripts/sip_nat_settings
+++ /dev/null
@@ -1,67 +0,0 @@
-#!/bin/sh
-
-# sip_nat_settings: generate NAT settings for sip.conf of an Asterisk system
-# that is behind a NAT router.
-#
-# This is a script to generate sane defaults for externip and localnet
-# of sip.conf. The output should be included in the [general] section of
-# sip.conf .
-#
-# Multiple network interfaces: If you have multiple network interfaces,
-# this script will generate a 'localnet' line for each of them that has a
-# broadcast (ipv4) address, except the loopback interface (lo). You can
-# later rem-out all of those you don't need.
-#
-# Alternatively, provide a network interface as a parameter an a localnet
-# line will only be generated for its network.
-#
-# Copyright (C) 2005 by Tzafrir Cohen
-#
-# This program is free software; you can redistribute it and/or modify
-# it under the terms of the GNU General Public License as published by
-# the Free Software Foundation; either version 2 of the License, or
-# (at your option) any later version.
-#
-# This program is distributed in the hope that it will be useful,
-# but WITHOUT ANY WARRANTY; without even the implied warranty of
-# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
-# GNU General Public License for more details.
-#
-# You should have received a copy of the GNU General Public License
-# along with this program; if not, write to the Free Software
-# Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
-
-# see http://unix.stackexchange.com/q/22615
-externip=`dig @resolver1.opendns.com -4 myip.opendns.com A +short`
-
-# optional parameter: network interface to use. By default: none.
-IFACE="$1"
-
-OS=`uname -s`
-case "$OS" in
-Linux)
- echo "externip = $externip"
- if [ -x "${IFACE}" ]; then
- ip --brief -family inet address show scope global up dev $IFACE | awk '{print "localnet = " $3}'
- else
- ip --brief -family inet address show scope global up | awk '{print "localnet = " $3}'
- fi
- ;;
-OpenBSD|FreeBSD)
- if [ "${OS}" = "FreeBSD" ]; then
- VER=`uname -r | cut -d . -f 1`
- if [ ${VER} -lt 7 ]; then
- echo "Unsupported OS"
- exit 1
- fi
- fi
- echo "externip = $externip"
- ip=`/sbin/ifconfig $IFACE | awk '/\tinet .* broadcast/{print $6}'`
- x=`/sbin/ifconfig $IFACE | awk '/\tinet .* broadcast/{print $4}'`
- printf 'localnet = %s/%u.%u.%u.%u\n' $ip $(($x>>24&0xff)) $(($x>>16&0xff)) $(($x>>8&0xff)) $(($x&0xff))
- ;;
-*)
- echo >&2 "$0: Unsupported OS $OS"
- exit 1
- ;;
-esac
diff --git a/contrib/systemd/asterisk.socket b/contrib/systemd/asterisk.socket
index afdca0df78..98b058c91d 100644
--- a/contrib/systemd/asterisk.socket
+++ b/contrib/systemd/asterisk.socket
@@ -17,10 +17,6 @@ ListenStream=0.0.0.0:5039
ListenStream=127.0.0.1:8088
# HTTPS
ListenStream=127.0.0.1:8089
-# chan_sip TCP
-ListenStream=0.0.0.0:5060
-# chan_sip TLS
-ListenStream=0.0.0.0:5061
[Install]
WantedBy=sockets.target
diff --git a/doc/UPGRADE-staging/chan_sip_removal.txt b/doc/UPGRADE-staging/chan_sip_removal.txt
new file mode 100644
index 0000000000..d73e8d6571
--- /dev/null
+++ b/doc/UPGRADE-staging/chan_sip_removal.txt
@@ -0,0 +1,6 @@
+Subject: chan_sip
+Master-Only: True
+
+This module was deprecated in Asterisk 17
+and is now being removed in accordance with
+the Asterisk Module Deprecation policy.
diff --git a/funcs/func_enum.c b/funcs/func_enum.c
index f649e0f297..67a8d3d6ad 100644
--- a/funcs/func_enum.c
+++ b/funcs/func_enum.c
@@ -57,7 +57,7 @@
If no method-type is given, the default will be
- sip .
+ pjsip .
If no zone-suffix is given, the default will be
@@ -96,7 +96,7 @@
If no method-type is given, the default will be
- sip .
+ pjsip .
@@ -185,7 +185,7 @@ static int function_enum(struct ast_channel *chan, const char *cmd, char *data,
if (args.tech && !ast_strlen_zero(args.tech)) {
ast_copy_string(tech,args.tech, sizeof(tech));
} else {
- ast_copy_string(tech,"sip",sizeof(tech));
+ ast_copy_string(tech,"pjsip",sizeof(tech));
}
if (!args.zone) {
@@ -279,7 +279,7 @@ static int enum_query_read(struct ast_channel *chan, const char *cmd, char *data
if (!args.zone)
args.zone = "e164.zone";
- ast_copy_string(tech, args.tech ? args.tech : "sip", sizeof(tech));
+ ast_copy_string(tech, args.tech ? args.tech : "pjsip", sizeof(tech));
if (!(erds = ast_calloc(1, sizeof(*erds))))
goto finish;
diff --git a/include/asterisk/ccss.h b/include/asterisk/ccss.h
index 381781840d..86633c190a 100644
--- a/include/asterisk/ccss.h
+++ b/include/asterisk/ccss.h
@@ -518,7 +518,7 @@ struct ast_cc_monitor {
* \details
* When issuing a CC recall, some technologies will require
* that a name other than the device name is dialed. For instance,
- * with SIP, a specific URI will be used which chan_sip will be able
+ * with SIP, a specific URI will be used which sip will be able
* to recognize as being a CC recall. Similarly, ISDN will need a specific
* dial string to know that the call is a recall.
*/
diff --git a/include/asterisk/channel.h b/include/asterisk/channel.h
index e5613df1fd..d6cb4d43f6 100644
--- a/include/asterisk/channel.h
+++ b/include/asterisk/channel.h
@@ -616,7 +616,7 @@ struct ast_msg_data;
* Structure to describe a channel "technology", ie a channel driver
* See for examples:
* \arg chan_iax2.c - The Inter-Asterisk exchange protocol
- * \arg chan_sip.c - The SIP channel driver
+ * \arg chan_pjsip.c - The SIP channel driver
* \arg chan_dahdi.c - PSTN connectivity (TDM, PRI, T1/E1, FXO, FXS)
*
* \details
diff --git a/include/asterisk/config_options.h b/include/asterisk/config_options.h
index 5460475828..8f7edfa6cd 100644
--- a/include/asterisk/config_options.h
+++ b/include/asterisk/config_options.h
@@ -124,8 +124,8 @@ struct aco_type {
const char *name; /*!< The name of this type (must match XML documentation) */
const char *category; /*!< A regular expression for matching categories to be allowed or denied */
const char *matchfield; /*!< An option name to match for this type (i.e. a 'type'-like column) */
- const char *matchvalue; /*!< The value of the option to require for matching (i.e. 'peer' for type= in sip.conf) */
- aco_matchvalue_func matchfunc; /*!< A function for determining whether the option value matches (i.e. hassip= requires ast_true()) */
+ const char *matchvalue; /*!< The value of the option to require for matching (i.e. 'peer') */
+ aco_matchvalue_func matchfunc; /*!< A function for determining whether the option value matches */
enum aco_category_op category_match; /*!< Whether the following category regex is a whitelist or blacklist */
size_t item_offset; /*!< The offset in the config snapshot for the global config or item config container */
unsigned int hidden; /*!< Type is for internal purposes only and it and all options should not be visible to users */
diff --git a/include/asterisk/doxygen/architecture.h b/include/asterisk/doxygen/architecture.h
index 780b64f778..302edd896c 100644
--- a/include/asterisk/doxygen/architecture.h
+++ b/include/asterisk/doxygen/architecture.h
@@ -497,7 +497,7 @@ Example dialplan:
exten => 5551212,n,Hangup()Call Setup: An incoming SIP INVITE begins this scenario. It is received by
- the SIP channel driver (chan_sip.c). Specifically, the monitor thread in chan_sip
+ the SIP channel driver (chan_pjsip.c). Specifically, the monitor thread in chan_pjsip
is responsible for handling this incoming request. Further, the monitor thread
is responsible for completing any handshake necessary to complete the call setup
process.
@@ -517,8 +517,8 @@ Example dialplan:
code simply executes the ast_answer() API call. This API call operates on an
ast_channel. It handles generic ast_channel hangup processing, as well as executes
the answer callback function defined in the associated ast_channel_tech for the
- active channel. In this case, the sip_answer() function in chan_sip.c will get
- executed to handle the SIP specific operations required to answer a call.
+ active channel. In this case, the chan_pjsip_answer() function in chan_pjsip.c will
+ get executed to handle the SIP specific operations required to answer a call.
-# Play the File: The next step of the dialplan says to play back a %sound file
to the caller. The Playback() application will be executed.
The code for this application is in apps/app_playback.c. The code in the application
@@ -562,7 +562,7 @@ Example dialplan:
exten => 5551212,n,Dial(IAX2/mypeer)Call Setup: An incoming SIP INVITE begins this scenario. It is received by
- the SIP channel driver (chan_sip.c). Specifically, the monitor thread in chan_sip
+ the SIP channel driver (chan_pjsip.c). Specifically, the monitor thread in chan_pjsip
is responsible for handling this incoming request. Further, the monitor thread
is responsible for completing any handshake necessary to complete the call setup
process.
diff --git a/include/asterisk/doxyref.h b/include/asterisk/doxyref.h
index 53f4564712..becfddfd49 100644
--- a/include/asterisk/doxyref.h
+++ b/include/asterisk/doxyref.h
@@ -250,7 +250,7 @@
/*!
* \page Config_rtp RTP configuration
* \arg Implemented in \ref rtp.c
- * Used in \ref chan_sip.c (and various H.323 channels)
+ * Used in various H.323 channels
* \section rtpconf rtp.conf
* \verbinclude rtp.conf.sample
*/
diff --git a/include/asterisk/module.h b/include/asterisk/module.h
index f79dc8eb10..84ff4b9da0 100644
--- a/include/asterisk/module.h
+++ b/include/asterisk/module.h
@@ -242,7 +242,7 @@ int ast_update_module_list_condition(int (*modentry)(const char *module, const c
/*!
* \brief Check if module with the name given is loaded
- * \param name Module name, like "chan_sip.so"
+ * \param name Module name, like "chan_pjsip.so"
* \retval 1 if true
* \retval 0 if false
*/
diff --git a/include/asterisk/stasis_channels.h b/include/asterisk/stasis_channels.h
index 73771864a6..d9ae9e8f04 100644
--- a/include/asterisk/stasis_channels.h
+++ b/include/asterisk/stasis_channels.h
@@ -112,7 +112,7 @@ struct ast_channel_snapshot_base {
);
struct timeval creationtime; /*!< The time of channel creation */
int tech_properties; /*!< Properties of the channel's technology */
- AST_STRING_FIELD_EXTENDED(protocol_id); /*!< Channel driver protocol id (i.e. Call-ID for chan_sip/chan_pjsip) */
+ AST_STRING_FIELD_EXTENDED(protocol_id); /*!< Channel driver protocol id (i.e. Call-ID for chan_pjsip) */
};
/*!
diff --git a/include/asterisk/tcptls.h b/include/asterisk/tcptls.h
index 1b1b56c6c4..1d358a220f 100644
--- a/include/asterisk/tcptls.h
+++ b/include/asterisk/tcptls.h
@@ -26,7 +26,7 @@
* in or out the DO_SSL macro.
*
* TLS/SSL support is basically implemented by reading from a config file
- * (currently manager.conf, http.conf and sip.conf) the names of the certificate
+ * (currently manager.conf, http.conf and pjsip.conf) the names of the certificate
* files and cipher to use, and then run ssl_setup() to create an appropriate
* data structure named ssl_ctx.
*
diff --git a/main/acl.c b/main/acl.c
index 2a40fae51a..fe388da538 100644
--- a/main/acl.c
+++ b/main/acl.c
@@ -273,7 +273,6 @@ static struct ast_ha *ast_duplicate_ha(struct ast_ha *original)
}
/* Create duplicate HA link list */
-/* Used in chan_sip2 templates */
struct ast_ha *ast_duplicate_ha_list(struct ast_ha *original)
{
struct ast_ha *start = original;
diff --git a/main/config.c b/main/config.c
index 1074407967..1522d8f24e 100644
--- a/main/config.c
+++ b/main/config.c
@@ -3053,10 +3053,7 @@ static int reload_module(void)
if (!driver || !database)
continue;
- if (!strcasecmp(v->name, "sipfriends")) {
- ast_log(LOG_WARNING, "The 'sipfriends' table is obsolete, update your config to use sippeers instead.\n");
- ast_realtime_append_mapping("sippeers", driver, database, table ? table : "sipfriends", pri);
- } else if (!strcasecmp(v->name, "iaxfriends")) {
+ if (!strcasecmp(v->name, "iaxfriends")) {
ast_log(LOG_WARNING, "The 'iaxfriends' table is obsolete, update your config to use iaxusers and iaxpeers, though they can point to the same table.\n");
ast_realtime_append_mapping("iaxusers", driver, database, table ? table : "iaxfriends", pri);
ast_realtime_append_mapping("iaxpeers", driver, database, table ? table : "iaxfriends", pri);
diff --git a/main/rtp_engine.c b/main/rtp_engine.c
index d36f70ce05..1b8e9c51ea 100644
--- a/main/rtp_engine.c
+++ b/main/rtp_engine.c
@@ -1760,7 +1760,7 @@ static int find_unused_payload(const struct ast_rtp_codecs *codecs)
* in Asterisk because when Compact Headers are activated, no rtpmap is
* send for those below 35. If you want to use 35 and below
* A) do not use Compact Headers,
- * B) remove that code in chan_sip/res_pjsip, or
+ * B) remove that code in res_pjsip, or
* C) add a flag that this RTP Payload Type got reassigned dynamically
* and requires a rtpmap even with Compact Headers enabled.
*/
diff --git a/main/udptl.c b/main/udptl.c
index 7a0aa841d6..b1b30a59e9 100644
--- a/main/udptl.c
+++ b/main/udptl.c
@@ -36,14 +36,12 @@
* \page T38fax_udptl T.38 support :: UDPTL
*
* Asterisk supports T.38 fax passthrough, origination and termination. It does
- * not support gateway operation. The only channel driver that supports T.38 at
- * this time is chan_sip.
+ * not support gateway operation.
*
* UDPTL is handled very much like RTP. It can be reinvited to go directly between
* the endpoints, without involving Asterisk in the media stream.
*
* \b References:
- * - chan_sip.c
* - udptl.c
* - app_fax.c
*/
@@ -1317,7 +1315,7 @@ static void *udptl_snapshot_alloc(void)
static int removed_options_handler(const struct aco_option *opt, struct ast_variable *var, void *obj)
{
if (!strcasecmp(var->name, "t38faxudpec")) {
- ast_log(LOG_WARNING, "t38faxudpec in udptl.conf is no longer supported; use the t38pt_udptl configuration option in sip.conf instead.\n");
+ ast_log(LOG_WARNING, "t38faxudpec in udptl.conf is no longer supported.\n");
} else if (!strcasecmp(var->name, "t38faxmaxdatagram")) {
ast_log(LOG_WARNING, "t38faxmaxdatagram in udptl.conf is no longer supported; value is now supplied by T.38 applications.\n");
}
diff --git a/menuselect/example_menuselect-tree b/menuselect/example_menuselect-tree
index 81c436a888..68caae5ae2 100644
--- a/menuselect/example_menuselect-tree
+++ b/menuselect/example_menuselect-tree
@@ -195,8 +195,6 @@
-
-
zaptel
diff --git a/menuselect/test/menuselect-tree b/menuselect/test/menuselect-tree
index 6bad4cab0c..b8fe4c7974 100644
--- a/menuselect/test/menuselect-tree
+++ b/menuselect/test/menuselect-tree
@@ -225,9 +225,6 @@
-
- chan_local
-
diff --git a/pbx/ael/ael-test/ael-test18/extensions.ael b/pbx/ael/ael-test/ael-test18/extensions.ael
index c1ea6a1ffb..78ea7c7bdf 100644
--- a/pbx/ael/ael-test/ael-test18/extensions.ael
+++ b/pbx/ael/ael-test/ael-test18/extensions.ael
@@ -6,7 +6,7 @@ context default
switch(${StatusCode}) {
case 1:
- Dial(SIP/706,12);
+ Dial(PJSIP/706,12);
switch(${DIALSTATUS}) {
case BUSY:
Voicemail(b706);
diff --git a/pbx/ael/ael-test/ael-test3/extensions.ael b/pbx/ael/ael-test/ael-test3/extensions.ael
index aae77a2745..0ca904aee7 100755
--- a/pbx/ael/ael-test/ael-test3/extensions.ael
+++ b/pbx/ael/ael-test/ael-test3/extensions.ael
@@ -2004,7 +2004,7 @@ macro ciddial(dialnum, lookup, waittime, dialopts, ddev)
{
BackGround(try_voip);
CALLERID(num)=7075679201;
- Dial(SIP/1${lookup}@tctwest,${waittime},${dialopts});
+ Dial(PJSIP/1${lookup}@tctwest,${waittime},${dialopts});
if( "${DIALSTATUS}" = "CHANUNAVAIL" )
{
BackGround(try_cell);
@@ -2033,7 +2033,7 @@ macro ciddial2(dialnum, lookup, waittime, dialopts, ddev) // give priority to tc
Set(cidnu=${CALLERID(num)});
Set(CALLERID(name)=${cidn});
Set(CALLERID(num)=7075679201);
- Dial(SIP/1${lookup}@tctwest,${waittime},${dialopts});
+ Dial(PJSIP/1${lookup}@tctwest,${waittime},${dialopts});
if( "${DIALSTATUS}" = "CHANUNAVAIL" )
{
Set(CALLERID(num)=${cidnu}); // put the original number back
diff --git a/pbx/ael/ael-test/ael-test5/extensions.ael b/pbx/ael/ael-test/ael-test5/extensions.ael
index e4f703b86d..6057957ae1 100644
--- a/pbx/ael/ael-test/ael-test5/extensions.ael
+++ b/pbx/ael/ael-test/ael-test5/extensions.ael
@@ -116,11 +116,11 @@ context huntsville-calling {
macro dialout( number ) {
Realtime(call_info,exten,${CALLERIDNUM:5},mon_);
if ("${mon_monitor}" = "YES") {
- Dial(SIP/${number}@zgw1.zvbwu.edu,,wW);
- Dial(SIP/${number}@zgw2.zvbwu.edu,,wW);
+ Dial(PJSIP/${number}@zgw1.zvbwu.edu,,wW);
+ Dial(PJSIP/${number}@zgw2.zvbwu.edu,,wW);
} else {
- Dial(SIP/${number}@zgw1.zvbwu.edu);
- Dial(SIP/${number}@zgw2.zvbwu.edu);
+ Dial(PJSIP/${number}@zgw1.zvbwu.edu);
+ Dial(PJSIP/${number}@zgw2.zvbwu.edu);
};
return;
};
@@ -143,9 +143,9 @@ macro stdexten( ext ) {
&checkcf(${ext});
Realtime(call_info,exten,${CALLERIDNUM:5},mon_);
if ("${mon_monitor}" = "YES") {
- Dial(SIP/${info_forwardto},25,wW);
+ Dial(PJSIP/${info_forwardto},25,wW);
} else {
- Dial(SIP/${info_forwardto},25);
+ Dial(PJSIP/${info_forwardto},25);
};
switch ("${DIALSTATUS}") {
case "BUSY":
@@ -183,7 +183,7 @@ macro stdexten( ext ) {
};
macro uvm( ext ) {
- Dial(SIP/u${ext}@ixtlchochitl.zvbwu.edu);
+ Dial(PJSIP/u${ext}@ixtlchochitl.zvbwu.edu);
Playback(im-sorry);
Playback(voice-mail-system);
Playback(down);
@@ -192,7 +192,7 @@ macro uvm( ext ) {
};
macro bvm( ext ) {
- Dial(SIP/b${ext}@ixtlchochitl.zvbwu.edu);
+ Dial(PJSIP/b${ext}@ixtlchochitl.zvbwu.edu);
Playback(im-sorry);
Playback(voice-mail-system);
Playback(down);
@@ -213,7 +213,7 @@ macro checkcf( ext ) {
if ("${ext}" = "43974") {
Set(info_forwardto=${ext}&SCCP/${ext});
} else {
- Set(info_forwardto=${ext}&SIP/${ext}w);
+ Set(info_forwardto=${ext}&PJSIP/${ext}w);
};
return;
};
@@ -344,14 +344,14 @@ context local1 {
context from-scm2 {
_4XXXX => {
- NoOp(DIALING SIP EXTENSION ${EXTEN} - FROM ${CALLERIDNUM});
- Dial(SIP/${EXTEN},20,wW);
+ NoOp(DIALING PJSIP EXTENSION ${EXTEN} - FROM ${CALLERIDNUM});
+ Dial(PJSIP/${EXTEN},20,wW);
Hangup;
};
_6XXXX => {
- NoOp(DIALING SIP EXTENSION ${EXTEN} - FROM ${CALLERIDNUM});
- Dial(SIP/${EXTEN},20,wW);
+ NoOp(DIALING PJSIP EXTENSION ${EXTEN} - FROM ${CALLERIDNUM});
+ Dial(PJSIP/${EXTEN},20,wW);
Hangup;
};
};
@@ -803,7 +803,7 @@ context vm-include {
context vm-direct {
s => {
- Dial(SIP/5555@ixtlchochitl.zvbwu.edu,20);
+ Dial(PJSIP/5555@ixtlchochitl.zvbwu.edu,20);
Playback(im-sorry);
Playback(voice-mail-system);
Playback(down);
@@ -815,7 +815,7 @@ context vm-direct {
context vm-extension {
s => {
- Dial(SIP/62100@ixtlchochitl.zvbwu.edu,20);
+ Dial(PJSIP/62100@ixtlchochitl.zvbwu.edu,20);
Playback(im-sorry);
Playback(voice-mail-system);
Playback(down);
@@ -827,7 +827,7 @@ context vm-extension {
context vm-directory {
5556 => {
- Dial(SIP/5556@ixtlchochitl.zvbwu.edu);
+ Dial(PJSIP/5556@ixtlchochitl.zvbwu.edu);
Playback(im-sorry);
Playback(voice-mail-system);
Playback(down);
diff --git a/pbx/ael/ael-test/ael-test6/extensions.ael b/pbx/ael/ael-test/ael-test6/extensions.ael
index 13ebf67fd9..7f759f2620 100644
--- a/pbx/ael/ael-test/ael-test6/extensions.ael
+++ b/pbx/ael/ael-test/ael-test6/extensions.ael
@@ -116,11 +116,11 @@ context huntsville-calling {
macro dialout( number ) {
Realtime(call_info|exten|${CALLERIDNUM:5}|mon_);
if ("${mon_monitor}" = "YES") {
- Dial(SIP/${number}@sgw1.shsu.edu,,wW);
- Dial(SIP/${number}@sgw2.shsu.edu,,wW);
+ Dial(PJSIP/${number}@sgw1.shsu.edu,,wW);
+ Dial(PJSIP/${number}@sgw2.shsu.edu,,wW);
} else {
- Dial(SIP/${number}@sgw1.shsu.edu);
- Dial(SIP/${number}@sgw2.shsu.edu);
+ Dial(PJSIP/${number}@sgw1.shsu.edu);
+ Dial(PJSIP/${number}@sgw2.shsu.edu);
};
};
@@ -142,9 +142,9 @@ macro stdexten( ext ) {
&checkcf(${ext});
Realtime(call_info|exten|${CALLERIDNUM:5}|mon_);
if ("${mon_monitor}" = "YES") {
- Dial(SIP/${info_forwardto},25,wW);
+ Dial(PJSIP/${info_forwardto},25,wW);
} else {
- Dial(SIP/${info_forwardto},25);
+ Dial(PJSIP/${info_forwardto},25);
};
switch ("${DIALSTATUS}") {
case "BUSY":
@@ -182,7 +182,7 @@ macro stdexten( ext ) {
};
macro uvm( ext ) {
- Dial(SIP/u${ext}@svm1.shsu.edu);
+ Dial(PJSIP/u${ext}@svm1.shsu.edu);
Playback(im-sorry);
Playback(voice-mail-system);
Playback(down);
@@ -191,7 +191,7 @@ macro uvm( ext ) {
};
macro bvm( ext ) {
- Dial(SIP/b${ext}@svm1.shsu.edu);
+ Dial(PJSIP/b${ext}@svm1.shsu.edu);
Playback(im-sorry);
Playback(voice-mail-system);
Playback(down);
@@ -211,7 +211,7 @@ macro checkcf( ext ) {
if ("${ext}" = "43974") {
Set(info_forwardto=${ext}&SCCP/${ext});
} else {
- Set(info_forwardto=${ext}&SIP/${ext}w);
+ Set(info_forwardto=${ext}&PJSIP/${ext}w);
};
};
@@ -340,14 +340,14 @@ context local1 {
context from-scm2 {
_4XXXX => {
- NoOp(DIALING SIP EXTENSION ${EXTEN} - FROM ${CALLERIDNUM});
- Dial(SIP/${EXTEN},20,wW);
+ NoOp(DIALING PJSIP EXTENSION ${EXTEN} - FROM ${CALLERIDNUM});
+ Dial(PJSIP/${EXTEN},20,wW);
Hangup;
};
_6XXXX => {
- NoOp(DIALING SIP EXTENSION ${EXTEN} - FROM ${CALLERIDNUM});
- Dial(SIP/${EXTEN},20,wW);
+ NoOp(DIALING PJSIP EXTENSION ${EXTEN} - FROM ${CALLERIDNUM});
+ Dial(PJSIP/${EXTEN},20,wW);
Hangup;
};
};
@@ -798,7 +798,7 @@ context vm-include {
context vm-direct {
s => {
- Dial(SIP/5555@svm1.shsu.edu,20);
+ Dial(PJSIP/5555@svm1.shsu.edu,20);
Playback(im-sorry);
Playback(voice-mail-system);
Playback(down);
@@ -810,7 +810,7 @@ context vm-direct {
context vm-extension {
s => {
- Dial(SIP/62100@svm1.shsu.edu,20);
+ Dial(PJSIP/62100@svm1.shsu.edu,20);
Playback(im-sorry);
Playback(voice-mail-system);
Playback(down);
@@ -822,7 +822,7 @@ context vm-extension {
context vm-directory {
5556 => {
- Dial(SIP/5556@svm1.shsu.edu);
+ Dial(PJSIP/5556@svm1.shsu.edu);
Playback(im-sorry);
Playback(voice-mail-system);
Playback(down);
diff --git a/pbx/ael/ael-test/ael-test7/extensions.ael b/pbx/ael/ael-test/ael-test7/extensions.ael
index e951bf29af..f0aeb3fa2f 100644
--- a/pbx/ael/ael-test/ael-test7/extensions.ael
+++ b/pbx/ael/ael-test/ael-test7/extensions.ael
@@ -141,9 +141,9 @@ repeat:
Wait(1);
Hangup;
};
- 1 => &stdexten(1,SIP/1);
- 2 => &stdexten(2,SIP/2);
- 3 => &stdexten(3,SIP/3);
+ 1 => &stdexten(1,PJSIP/1);
+ 2 => &stdexten(2,PJSIP/2);
+ 3 => &stdexten(3,PJSIP/3);
2271653 => jump 1;
7322271653 => jump 1;
@@ -166,7 +166,7 @@ repeat:
17322271677 => jump 3;
galka => jump 3;
911 => Dial(${PSTNPROTO}/911@${PSTN},60,);
- 380 => Dial(SIP/topspeen@212.40.38.70,60,T);
+ 380 => Dial(PJSIP/topspeen@212.40.38.70,60,T);
// Fun stuff
100 => {
diff --git a/pbx/ael/ael-test/ael-test8/extensions.ael b/pbx/ael/ael-test/ael-test8/extensions.ael
index 290444aaaf..d5106f97a6 100644
--- a/pbx/ael/ael-test/ael-test8/extensions.ael
+++ b/pbx/ael/ael-test/ael-test8/extensions.ael
@@ -6,7 +6,7 @@ context default
switch(${StatusCode}) {
case 1:
- Dial(SIP/706,12);
+ Dial(PJSIP/706,12);
switch(${DIALSTATUS}) {
case BUSY:
Voicemail(b706);
diff --git a/pbx/ael/ael-test/ael-vtest13/extensions.ael b/pbx/ael/ael-test/ael-vtest13/extensions.ael
index d8870cb4a0..9719c5128c 100755
--- a/pbx/ael/ael-test/ael-vtest13/extensions.ael
+++ b/pbx/ael/ael-test/ael-vtest13/extensions.ael
@@ -2003,7 +2003,7 @@ macro ciddial(dialnum, lookup, waittime, dialopts, ddev)
{
BackGround(try_voip);
CALLERID(num)=7075679201;
- Dial(SIP/1${lookup}@tctwest,${waittime},${dialopts});
+ Dial(PJSIP/1${lookup}@tctwest,${waittime},${dialopts});
if( "${DIALSTATUS}" = "CHANUNAVAIL" )
{
BackGround(try_cell);
@@ -2032,7 +2032,7 @@ macro ciddial2(dialnum, lookup, waittime, dialopts, ddev) // give priority to tc
Set(cidnu=${CALLERID(num)});
Set(CALLERID(name)=${cidn});
Set(CALLERID(num)=7075679201);
- Dial(SIP/1${lookup}@tctwest,${waittime},${dialopts});
+ Dial(PJSIP/1${lookup}@tctwest,${waittime},${dialopts});
if( "${DIALSTATUS}" = "CHANUNAVAIL" )
{
Set(CALLERID(num)=${cidnu}); // put the original number back
diff --git a/pbx/ael/ael-test/ref.ael-vtest13 b/pbx/ael/ael-test/ref.ael-vtest13
index 3c66be2609..244b6796b1 100644
--- a/pbx/ael/ael-test/ref.ael-vtest13
+++ b/pbx/ael/ael-test/ref.ael-vtest13
@@ -2131,7 +2131,7 @@ exten => ~~s~~,9,Dial(${ddev}/${dialnum}|${waittime}|${dialopts})
exten => ~~s~~,10,GotoIf($["${DIALSTATUS}" = "CHANUNAVAIL" ]?11:19)
exten => ~~s~~,11,BackGround(try_voip)
exten => ~~s~~,12,Set(CALLERID(num)=$[7075679201])
-exten => ~~s~~,13,Dial(SIP/1${lookup}@tctwest,${waittime},${dialopts})
+exten => ~~s~~,13,Dial(PJSIP/1${lookup}@tctwest,${waittime},${dialopts})
exten => ~~s~~,14,GotoIf($["${DIALSTATUS}" = "CHANUNAVAIL" ]?15:18)
exten => ~~s~~,15,BackGround(try_cell)
exten => ~~s~~,16,Set(CALLERID(num)=$[${cidnu}])
@@ -2168,7 +2168,7 @@ exten => ~~s~~,6,Set(cidn=${DB(cidname/${lookup})})
exten => ~~s~~,7,Set(cidnu=${CALLERID(num)})
exten => ~~s~~,8,Set(CALLERID(name)=${cidn})
exten => ~~s~~,9,Set(CALLERID(num)=7075679201)
-exten => ~~s~~,10,Dial(SIP/1${lookup}@tctwest,${waittime},${dialopts})
+exten => ~~s~~,10,Dial(PJSIP/1${lookup}@tctwest,${waittime},${dialopts})
exten => ~~s~~,11,GotoIf($["${DIALSTATUS}" = "CHANUNAVAIL" ]?12:19)
exten => ~~s~~,12,Set(CALLERID(num)=${cidnu})
exten => ~~s~~,13,BackGround(try_zap)
diff --git a/pbx/pbx_config.c b/pbx/pbx_config.c
index 648d751278..f301d57174 100644
--- a/pbx/pbx_config.c
+++ b/pbx/pbx_config.c
@@ -1998,10 +1998,6 @@ static void pbx_load_users(void)
if (!strcasecmp(cat, "general"))
continue;
iface[0] = '\0';
- if (ast_true(ast_config_option(cfg, cat, "hassip"))) {
- snprintf(tmp, sizeof(tmp), "SIP/%s", cat);
- append_interface(iface, sizeof(iface), tmp);
- }
if (ast_true(ast_config_option(cfg, cat, "hasiax"))) {
snprintf(tmp, sizeof(tmp), "IAX2/%s", cat);
append_interface(iface, sizeof(iface), tmp);
diff --git a/res/ari/resource_endpoints.h b/res/ari/resource_endpoints.h
index 4391b36e6e..54af8f78aa 100644
--- a/res/ari/resource_endpoints.h
+++ b/res/ari/resource_endpoints.h
@@ -52,9 +52,9 @@ struct ast_ari_endpoints_list_args {
void ast_ari_endpoints_list(struct ast_variable *headers, struct ast_ari_endpoints_list_args *args, struct ast_ari_response *response);
/*! Argument struct for ast_ari_endpoints_send_message() */
struct ast_ari_endpoints_send_message_args {
- /*! The endpoint resource or technology specific URI to send the message to. Valid resources are sip, pjsip, and xmpp. */
+ /*! The endpoint resource or technology specific URI to send the message to. Valid resources are pjsip, and xmpp. */
const char *to;
- /*! The endpoint resource or technology specific identity to send this message from. Valid resources are sip, pjsip, and xmpp. */
+ /*! The endpoint resource or technology specific identity to send this message from. Valid resources are pjsip, and xmpp. */
const char *from;
/*! The body of the message */
const char *body;
@@ -81,7 +81,7 @@ int ast_ari_endpoints_send_message_parse_body(
void ast_ari_endpoints_send_message(struct ast_variable *headers, struct ast_ari_endpoints_send_message_args *args, struct ast_ari_response *response);
/*! Argument struct for ast_ari_endpoints_list_by_tech() */
struct ast_ari_endpoints_list_by_tech_args {
- /*! Technology of the endpoints (sip,iax2,...) */
+ /*! Technology of the endpoints (iax2,...) */
const char *tech;
};
/*!
@@ -113,7 +113,7 @@ struct ast_ari_endpoints_send_message_to_endpoint_args {
const char *tech;
/*! ID of the endpoint */
const char *resource;
- /*! The endpoint resource or technology specific identity to send this message from. Valid resources are sip, pjsip, and xmpp. */
+ /*! The endpoint resource or technology specific identity to send this message from. Valid resources are pjsip, and xmpp. */
const char *from;
/*! The body of the message */
const char *body;
diff --git a/res/res_phoneprov.c b/res/res_phoneprov.c
index a050047a06..5a4efa092b 100644
--- a/res/res_phoneprov.c
+++ b/res/res_phoneprov.c
@@ -29,7 +29,7 @@
* \author George Joseph
*/
-/*! \li \ref res_phoneprov.c uses the configuration file \ref phoneprov.conf and \ref users.conf and \ref sip.conf
+/*! \li \ref res_phoneprov.c uses the configuration file \ref phoneprov.conf and \ref users.conf
* \addtogroup configuration_file Configuration Files
*/
@@ -1234,11 +1234,6 @@ static struct varshead *get_defaults(void)
}
value = ast_variable_retrieve(phoneprov_cfg, "general", pp_general_lookup[AST_PHONEPROV_STD_SERVER_PORT]);
- if (!value) {
- if ((cfg = ast_config_load("sip.conf", config_flags)) && cfg != CONFIG_STATUS_FILEINVALID) {
- value = ast_variable_retrieve(cfg, "general", "bindport");
- }
- }
var = ast_var_assign(variable_lookup[AST_PHONEPROV_STD_SERVER_PORT], S_OR(value, "5060"));
if (cfg && cfg != CONFIG_STATUS_FILEINVALID) {
ast_config_destroy(cfg);
@@ -1384,7 +1379,7 @@ static int unload_module(void)
ast_custom_function_unregister(&pp_each_extension_function);
ast_cli_unregister_multiple(pp_cli, ARRAY_LEN(pp_cli));
- /* This cleans up the sip.conf/users.conf provider (called specifically for clarity) */
+ /* This cleans up the users.conf provider (called specifically for clarity) */
ast_phoneprov_provider_unregister(SIPUSERS_PROVIDER_NAME);
/* This cleans up the framework which also cleans up the providers. */
@@ -1449,9 +1444,9 @@ static int load_module(void)
goto error;
}
- /* Register ourselves as the provider for sip.conf/users.conf */
+ /* Register ourselves as the provider for users.conf */
if (ast_phoneprov_provider_register(SIPUSERS_PROVIDER_NAME, load_users)) {
- ast_log(LOG_WARNING, "Unable register sip/users config provider. Others may succeed.\n");
+ ast_log(LOG_WARNING, "Unable register users config provider. Others may succeed.\n");
}
ast_http_uri_link(&phoneprovuri);
diff --git a/res/res_pjsip/pjsip_config.xml b/res/res_pjsip/pjsip_config.xml
index 99ca64571e..8c1eb81389 100644
--- a/res/res_pjsip/pjsip_config.xml
+++ b/res/res_pjsip/pjsip_config.xml
@@ -439,9 +439,7 @@
This setting allows to choose the DTMF mode for endpoint communication.
- DTMF is sent out of band of the main audio stream. This
- supercedes the older RFC-2833 used within
- the older chan_sip .
+ DTMF is sent out of band of the main audio stream.
DTMF is sent as part of audio stream.
diff --git a/res/res_pjsip/pjsip_options.c b/res/res_pjsip/pjsip_options.c
index 220a947762..e395ba0d7d 100644
--- a/res/res_pjsip/pjsip_options.c
+++ b/res/res_pjsip/pjsip_options.c
@@ -242,8 +242,7 @@ static pj_status_t send_options_response(pjsip_rx_data *rdata, int code)
/*
* XXX TODO: pjsip doesn't care a lot about either of these headers -
* while it provides specific methods to create them, they are defined
- * to be the standard string header creation. We never did add them
- * in chan_sip, although RFC 3261 says they SHOULD. Hard coded here.
+ * to be the standard string header creation. Hard coded here.
*/
ast_sip_add_header(tdata, "Accept-Encoding", DEFAULT_ENCODING);
ast_sip_add_header(tdata, "Accept-Language", DEFAULT_LANGUAGE);
diff --git a/res/res_pjsip_dlg_options.c b/res/res_pjsip_dlg_options.c
index d07373495f..47c4bc6c39 100644
--- a/res/res_pjsip_dlg_options.c
+++ b/res/res_pjsip_dlg_options.c
@@ -63,8 +63,7 @@ static int options_incoming_request(struct ast_sip_session *session, pjsip_rx_da
/*
* XXX TODO: pjsip doesn't care a lot about either of these headers -
* while it provides specific methods to create them, they are defined
- * to be the standard string header creation. We never did add them
- * in chan_sip, although RFC 3261 says they SHOULD. Hard coded here.
+ * to be the standard string header creation. Hard coded here.
*/
ast_sip_add_header(tdata, "Accept-Encoding", DEFAULT_ENCODING);
ast_sip_add_header(tdata, "Accept-Language", DEFAULT_LANGUAGE);
diff --git a/rest-api/api-docs/channels.json b/rest-api/api-docs/channels.json
index 269976dfa4..e3b4fcbbba 100644
--- a/rest-api/api-docs/channels.json
+++ b/rest-api/api-docs/channels.json
@@ -2129,7 +2129,7 @@
"protocol_id": {
"required": true,
"type": "string",
- "description": "Protocol id from underlying channel driver (i.e. Call-ID for chan_sip/chan_pjsip; will be empty if not applicable or not implemented by driver)."
+ "description": "Protocol id from underlying channel driver (i.e. Call-ID for chan_pjsip; will be empty if not applicable or not implemented by driver)."
},
"name": {
"required": true,
diff --git a/rest-api/api-docs/endpoints.json b/rest-api/api-docs/endpoints.json
index 1f77d3705f..80baf97bcc 100644
--- a/rest-api/api-docs/endpoints.json
+++ b/rest-api/api-docs/endpoints.json
@@ -31,7 +31,7 @@
"parameters": [
{
"name": "to",
- "description": "The endpoint resource or technology specific URI to send the message to. Valid resources are sip, pjsip, and xmpp.",
+ "description": "The endpoint resource or technology specific URI to send the message to. Valid resources are pjsip, and xmpp.",
"paramType": "query",
"required": true,
"allowMultiple": false,
@@ -39,7 +39,7 @@
},
{
"name": "from",
- "description": "The endpoint resource or technology specific identity to send this message from. Valid resources are sip, pjsip, and xmpp.",
+ "description": "The endpoint resource or technology specific identity to send this message from. Valid resources are pjsip, and xmpp.",
"paramType": "query",
"required": true,
"allowMultiple": false,
@@ -55,7 +55,7 @@
},
{
"name": "variables",
- "descriptioni": "The \"variables\" key in the body object holds technology specific key/value pairs to append to the message. These can be interpreted and used by the various resource types; for example, pjsip and sip resource types will add the key/value pairs as SIP headers,",
+ "descriptioni": "The \"variables\" key in the body object holds technology specific key/value pairs to append to the message. These can be interpreted and used by the various resource types; for example, pjsip resource types will add the key/value pairs as SIP headers,",
"paramType": "body",
"required": false,
"dataType": "containers",
@@ -87,7 +87,7 @@
"parameters": [
{
"name": "tech",
- "description": "Technology of the endpoints (sip,iax2,...)",
+ "description": "Technology of the endpoints (pjsip,iax2,...)",
"paramType": "path",
"dataType": "string"
}
@@ -161,7 +161,7 @@
},
{
"name": "from",
- "description": "The endpoint resource or technology specific identity to send this message from. Valid resources are sip, pjsip, and xmpp.",
+ "description": "The endpoint resource or technology specific identity to send this message from. Valid resources are pjsip and xmpp.",
"paramType": "query",
"required": true,
"allowMultiple": false,
@@ -177,7 +177,7 @@
},
{
"name": "variables",
- "descriptioni": "The \"variables\" key in the body object holds technology specific key/value pairs to append to the message. These can be interpreted and used by the various resource types; for example, pjsip and sip resource types will add the key/value pairs as SIP headers,",
+ "descriptioni": "The \"variables\" key in the body object holds technology specific key/value pairs to append to the message. These can be interpreted and used by the various resource types; for example, pjsip resource types will add the key/value pairs as SIP headers,",
"paramType": "body",
"required": false,
"dataType": "containers",
@@ -239,12 +239,12 @@
"properties": {
"from": {
"type": "string",
- "description": "A technology specific URI specifying the source of the message. For sip and pjsip technologies, any SIP URI can be specified. For xmpp, the URI must correspond to the client connection being used to send the message.",
+ "description": "A technology specific URI specifying the source of the message. For pjsip technology, any SIP URI can be specified. For xmpp, the URI must correspond to the client connection being used to send the message.",
"required": true
},
"to": {
"type": "string",
- "description": "A technology specific URI specifying the destination of the message. Valid technologies include sip, pjsip, and xmp. The destination of a message should be an endpoint.",
+ "description": "A technology specific URI specifying the destination of the message. Valid technologies include pjsip, and xmp. The destination of a message should be an endpoint.",
"required": true
},
"body": {
diff --git a/tests/CI/gateTestGroups.json b/tests/CI/gateTestGroups.json
index 415a7d709b..7b73a6cd77 100644
--- a/tests/CI/gateTestGroups.json
+++ b/tests/CI/gateTestGroups.json
@@ -36,18 +36,6 @@
"runTestsuiteOptions": "--test-timeout=180",
"testcmd": "--test-regex=tests/channels/pjsip/[s-z]"
},
- {
- "name": "sip1",
- "dir": "tests/CI/output/sip1",
- "runTestsuiteOptions": "--test-timeout=240",
- "testcmd": "--test-regex=tests/channels/SIP/[Sa-r]"
- },
- {
- "name": "sip2",
- "dir": "tests/CI/output/sip2",
- "runTestsuiteOptions": "--test-timeout=240",
- "testcmd": "--test-regex=tests/channels/SIP/[s-z]"
- },
{
"name": "iax ",
"dir": "tests/CI/output/iax2_local",
diff --git a/tests/CI/periodic-dailyTestGroups.json b/tests/CI/periodic-dailyTestGroups.json
index cc837f69c7..6bf0ef7065 100644
--- a/tests/CI/periodic-dailyTestGroups.json
+++ b/tests/CI/periodic-dailyTestGroups.json
@@ -17,12 +17,6 @@
"runTestsuiteOptions": "--test-timeout=180",
"testcmd": "-t tests/channels/pjsip"
},
- {
- "name": "sip ",
- "dir": "tests/CI/output/sip",
- "runTestsuiteOptions": "--test-timeout=240",
- "testcmd": "-t tests/channels/SIP"
- },
{
"name": "iax ",
"dir": "tests/CI/output/iax2_local",
diff --git a/tests/CI/ref_debugTestGroups.json b/tests/CI/ref_debugTestGroups.json
index b8c8e9b8ef..8f4034077f 100644
--- a/tests/CI/ref_debugTestGroups.json
+++ b/tests/CI/ref_debugTestGroups.json
@@ -9,11 +9,6 @@
"dir": "tests/CI/output/pjsip",
"testcmd": "-t tests/channels/pjsip"
},
- {
- "name": "sip ",
- "dir": "tests/CI/output/sip",
- "testcmd": "-t tests/channels/SIP"
- },
{
"name": "iax ",
"dir": "tests/CI/output/iax2_local",