asterisk/main/dial.c

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/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 1999 - 2007, Digium, Inc.
*
* Joshua Colp <jcolp@digium.com>
*
* 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 Dialing API
*
* \author Joshua Colp <jcolp@digium.com>
*/
/*** MODULEINFO
<support_level>core</support_level>
***/
#include "asterisk.h"
#include <sys/time.h>
#include <signal.h>
#include "asterisk/channel.h"
#include "asterisk/utils.h"
#include "asterisk/lock.h"
#include "asterisk/linkedlists.h"
#include "asterisk/dial.h"
#include "asterisk/pbx.h"
#include "asterisk/musiconhold.h"
#include "asterisk/app.h"
#include "asterisk/causes.h"
#include "asterisk/stasis_channels.h"
Detect potential forwarding loops based on count. A potential problem that can arise is the following: * Bob's phone is programmed to automatically forward to Carol. * Carol's phone is programmed to automatically forward to Bob. * Alice calls Bob. If left unchecked, this results in an endless loops of call forwards that would eventually result in some sort of fiery crash. Asterisk's method of solving this issue was to track which interfaces had been dialed. If a destination were dialed a second time, then the attempt to call that destination would fail since a loop was detected. The problem with this method is that call forwarding has evolved. Some SIP phones allow for a user to manually forward an incoming call to an ad-hoc destination. This can mean that: * There are legitimate use cases where a device may be dialed multiple times, or * There can be human error when forwarding calls. This change removes the old method of detecting forwarding loops in favor of keeping a count of the number of destinations a channel has dialed on a particular branch of a call. If the number exceeds the set number of max forwards, then the call fails. This approach has the following advantages over the old: * It is much simpler. * It can detect loops involving local channels. * It is user configurable. The only disadvantage it has is that in the case where there is a legitimate forwarding loop present, it takes longer to detect it. However, the forwarding loop is still properly detected and the call is cleaned up as it should be. Address review feedback on gerrit. * Correct "mfgium" to "Digium" * Decrement max forwards by one in the case where allocation of the max forwards datastore is required. * Remove irrelevant code change from pjsip_global_headers.c ASTERISK-24958 #close Change-Id: Ia7e4b7cd3bccfbd34d9a859838356931bba56c23
2015-04-15 15:38:02 +00:00
#include "asterisk/max_forwards.h"
/*! \brief Main dialing structure. Contains global options, channels being dialed, and more! */
struct ast_dial {
int num; /*!< Current number to give to next dialed channel */
int timeout; /*!< Maximum time allowed for dial attempts */
int actual_timeout; /*!< Actual timeout based on all factors (ie: channels) */
enum ast_dial_result state; /*!< Status of dial */
void *options[AST_DIAL_OPTION_MAX]; /*!< Global options */
ast_dial_state_callback state_callback; /*!< Status callback */
void *user_data; /*!< Attached user data */
AST_LIST_HEAD(, ast_dial_channel) channels; /*!< Channels being dialed */
pthread_t thread; /*!< Thread (if running in async) */
ast_callid callid; /*!< callid (if running in async) */
ast_mutex_t lock; /*! Lock to protect the thread information above */
};
/*! \brief Dialing channel structure. Contains per-channel dialing options, asterisk channel, and more! */
struct ast_dial_channel {
int num; /*!< Unique number for dialed channel */
int timeout; /*!< Maximum time allowed for attempt */
char *tech; /*!< Technology being dialed */
char *device; /*!< Device being dialed */
void *options[AST_DIAL_OPTION_MAX]; /*!< Channel specific options */
int cause; /*!< Cause code in case of failure */
unsigned int is_running_app:1; /*!< Is this running an application? */
char *assignedid1; /*!< UniqueID to assign channel */
char *assignedid2; /*!< UniqueID to assign 2nd channel */
struct ast_channel *owner; /*!< Asterisk channel */
AST_LIST_ENTRY(ast_dial_channel) list; /*!< Linked list information */
};
/*! \brief Typedef for dial option enable */
typedef void *(*ast_dial_option_cb_enable)(void *data);
/*! \brief Typedef for dial option disable */
typedef int (*ast_dial_option_cb_disable)(void *data);
/*! \brief Structure for 'ANSWER_EXEC' option */
struct answer_exec_struct {
char app[AST_MAX_APP]; /*!< Application name */
char *args; /*!< Application arguments */
};
/*! \brief Enable function for 'ANSWER_EXEC' option */
static void *answer_exec_enable(void *data)
{
struct answer_exec_struct *answer_exec = NULL;
char *app = ast_strdupa((char*)data), *args = NULL;
/* Not giving any data to this option is bad, mmmk? */
if (ast_strlen_zero(app))
return NULL;
/* Create new data structure */
if (!(answer_exec = ast_calloc(1, sizeof(*answer_exec))))
return NULL;
/* Parse out application and arguments */
if ((args = strchr(app, ','))) {
*args++ = '\0';
answer_exec->args = ast_strdup(args);
}
/* Copy application name */
ast_copy_string(answer_exec->app, app, sizeof(answer_exec->app));
return answer_exec;
}
/*! \brief Disable function for 'ANSWER_EXEC' option */
static int answer_exec_disable(void *data)
{
struct answer_exec_struct *answer_exec = data;
/* Make sure we have a value */
if (!answer_exec)
return -1;
/* If arguments are present, free them too */
if (answer_exec->args)
ast_free(answer_exec->args);
/* This is simple - just free the structure */
ast_free(answer_exec);
return 0;
}
static void *music_enable(void *data)
{
return ast_strdup(data);
}
static int music_disable(void *data)
{
if (!data)
return -1;
ast_free(data);
return 0;
}
static void *predial_enable(void *data)
{
return ast_strdup(data);
}
static int predial_disable(void *data)
{
if (!data) {
return -1;
}
ast_free(data);
return 0;
}
/*! \brief Application execution function for 'ANSWER_EXEC' option */
static void answer_exec_run(struct ast_dial *dial, struct ast_dial_channel *dial_channel, char *app, char *args)
{
struct ast_channel *chan = dial_channel->owner;
/* Execute the application, if available */
if (ast_pbx_exec_application(chan, app, args)) {
/* If the application was not found, return immediately */
return;
}
/* If another thread is not taking over hang up the channel */
ast_mutex_lock(&dial->lock);
if (dial->thread != AST_PTHREADT_STOP) {
ast_hangup(chan);
dial_channel->owner = NULL;
}
ast_mutex_unlock(&dial->lock);
return;
}
struct ast_option_types {
enum ast_dial_option option;
ast_dial_option_cb_enable enable;
ast_dial_option_cb_disable disable;
};
/*!
* \brief Map options to respective handlers (enable/disable).
*
* \note This list MUST be perfectly kept in order with enum
* ast_dial_option, or else madness will happen.
*/
static const struct ast_option_types option_types[] = {
{ AST_DIAL_OPTION_RINGING, NULL, NULL }, /*!< Always indicate ringing to caller */
{ AST_DIAL_OPTION_ANSWER_EXEC, answer_exec_enable, answer_exec_disable }, /*!< Execute application upon answer in async mode */
{ AST_DIAL_OPTION_MUSIC, music_enable, music_disable }, /*!< Play music to the caller instead of ringing */
{ AST_DIAL_OPTION_DISABLE_CALL_FORWARDING, NULL, NULL }, /*!< Disable call forwarding on channels */
{ AST_DIAL_OPTION_PREDIAL, predial_enable, predial_disable }, /*!< Execute a subroutine on the outbound channels prior to dialing */
{ AST_DIAL_OPTION_DIAL_REPLACES_SELF, NULL, NULL }, /*!< The dial operation is a replacement for the requester */
2015-02-26 18:53:36 +00:00
{ AST_DIAL_OPTION_SELF_DESTROY, NULL, NULL}, /*!< Destroy self at end of ast_dial_run */
{ AST_DIAL_OPTION_MAX, NULL, NULL }, /*!< Terminator of list */
};
/*! \brief Maximum number of channels we can watch at a time */
#define AST_MAX_WATCHERS 256
/*! \brief Macro for finding the option structure to use on a dialed channel */
#define FIND_RELATIVE_OPTION(dial, dial_channel, ast_dial_option) (dial_channel->options[ast_dial_option] ? dial_channel->options[ast_dial_option] : dial->options[ast_dial_option])
/*! \brief Macro that determines whether a channel is the caller or not */
#define IS_CALLER(chan, owner) (chan == owner ? 1 : 0)
/*! \brief New dialing structure
* \note Create a dialing structure
* \return Returns a calloc'd ast_dial structure, NULL on failure
*/
struct ast_dial *ast_dial_create(void)
{
struct ast_dial *dial = NULL;
/* Allocate new memory for structure */
if (!(dial = ast_calloc(1, sizeof(*dial))))
return NULL;
/* Initialize list of channels */
AST_LIST_HEAD_INIT(&dial->channels);
/* Initialize thread to NULL */
dial->thread = AST_PTHREADT_NULL;
/* No timeout exists... yet */
dial->timeout = -1;
dial->actual_timeout = -1;
/* Can't forget about the lock */
ast_mutex_init(&dial->lock);
return dial;
}
static int dial_append_common(struct ast_dial *dial, struct ast_dial_channel *channel,
const char *tech, const char *device, const struct ast_assigned_ids *assignedids)
{
/* Record technology and device for when we actually dial */
channel->tech = ast_strdup(tech);
channel->device = ast_strdup(device);
/* Store the assigned id */
if (assignedids && !ast_strlen_zero(assignedids->uniqueid)) {
channel->assignedid1 = ast_strdup(assignedids->uniqueid);
if (!ast_strlen_zero(assignedids->uniqueid2)) {
channel->assignedid2 = ast_strdup(assignedids->uniqueid2);
}
}
/* Grab reference number from dial structure */
channel->num = ast_atomic_fetchadd_int(&dial->num, +1);
/* No timeout exists... yet */
channel->timeout = -1;
/* Insert into channels list */
AST_LIST_INSERT_TAIL(&dial->channels, channel, list);
return channel->num;
}
/*! \brief Append a channel
* \note Appends a channel to a dialing structure
* \return Returns channel reference number on success, -1 on failure
*/
int ast_dial_append(struct ast_dial *dial, const char *tech, const char *device, const struct ast_assigned_ids *assignedids)
{
struct ast_dial_channel *channel = NULL;
/* Make sure we have required arguments */
if (!dial || !tech || !device)
return -1;
/* Allocate new memory for dialed channel structure */
if (!(channel = ast_calloc(1, sizeof(*channel))))
return -1;
return dial_append_common(dial, channel, tech, device, assignedids);
}
int ast_dial_append_channel(struct ast_dial *dial, struct ast_channel *chan)
{
struct ast_dial_channel *channel;
char *tech;
char *device;
char *dash;
if (!dial || !chan) {
return -1;
}
channel = ast_calloc(1, sizeof(*channel));
if (!channel) {
return -1;
}
channel->owner = chan;
tech = ast_strdupa(ast_channel_name(chan));
device = strchr(tech, '/');
if (!device) {
ast_free(channel);
return -1;
}
*device++ = '\0';
dash = strrchr(device, '-');
if (dash) {
*dash = '\0';
}
return dial_append_common(dial, channel, tech, device, NULL);
}
/*! \brief Helper function that requests all channels */
static int begin_dial_prerun(struct ast_dial_channel *channel, struct ast_channel *chan, struct ast_format_cap *cap, const char *predial_string)
{
struct ast_format_cap *cap_all_audio = NULL;
struct ast_format_cap *cap_request;
struct ast_format_cap *requester_cap = NULL;
struct ast_assigned_ids assignedids = {
.uniqueid = channel->assignedid1,
.uniqueid2 = channel->assignedid2,
};
Detect potential forwarding loops based on count. A potential problem that can arise is the following: * Bob's phone is programmed to automatically forward to Carol. * Carol's phone is programmed to automatically forward to Bob. * Alice calls Bob. If left unchecked, this results in an endless loops of call forwards that would eventually result in some sort of fiery crash. Asterisk's method of solving this issue was to track which interfaces had been dialed. If a destination were dialed a second time, then the attempt to call that destination would fail since a loop was detected. The problem with this method is that call forwarding has evolved. Some SIP phones allow for a user to manually forward an incoming call to an ad-hoc destination. This can mean that: * There are legitimate use cases where a device may be dialed multiple times, or * There can be human error when forwarding calls. This change removes the old method of detecting forwarding loops in favor of keeping a count of the number of destinations a channel has dialed on a particular branch of a call. If the number exceeds the set number of max forwards, then the call fails. This approach has the following advantages over the old: * It is much simpler. * It can detect loops involving local channels. * It is user configurable. The only disadvantage it has is that in the case where there is a legitimate forwarding loop present, it takes longer to detect it. However, the forwarding loop is still properly detected and the call is cleaned up as it should be. Address review feedback on gerrit. * Correct "mfgium" to "Digium" * Decrement max forwards by one in the case where allocation of the max forwards datastore is required. * Remove irrelevant code change from pjsip_global_headers.c ASTERISK-24958 #close Change-Id: Ia7e4b7cd3bccfbd34d9a859838356931bba56c23
2015-04-15 15:38:02 +00:00
if (chan) {
int max_forwards;
ast_channel_lock(chan);
max_forwards = ast_max_forwards_get(chan);
requester_cap = ao2_bump(ast_channel_nativeformats(chan));
Detect potential forwarding loops based on count. A potential problem that can arise is the following: * Bob's phone is programmed to automatically forward to Carol. * Carol's phone is programmed to automatically forward to Bob. * Alice calls Bob. If left unchecked, this results in an endless loops of call forwards that would eventually result in some sort of fiery crash. Asterisk's method of solving this issue was to track which interfaces had been dialed. If a destination were dialed a second time, then the attempt to call that destination would fail since a loop was detected. The problem with this method is that call forwarding has evolved. Some SIP phones allow for a user to manually forward an incoming call to an ad-hoc destination. This can mean that: * There are legitimate use cases where a device may be dialed multiple times, or * There can be human error when forwarding calls. This change removes the old method of detecting forwarding loops in favor of keeping a count of the number of destinations a channel has dialed on a particular branch of a call. If the number exceeds the set number of max forwards, then the call fails. This approach has the following advantages over the old: * It is much simpler. * It can detect loops involving local channels. * It is user configurable. The only disadvantage it has is that in the case where there is a legitimate forwarding loop present, it takes longer to detect it. However, the forwarding loop is still properly detected and the call is cleaned up as it should be. Address review feedback on gerrit. * Correct "mfgium" to "Digium" * Decrement max forwards by one in the case where allocation of the max forwards datastore is required. * Remove irrelevant code change from pjsip_global_headers.c ASTERISK-24958 #close Change-Id: Ia7e4b7cd3bccfbd34d9a859838356931bba56c23
2015-04-15 15:38:02 +00:00
ast_channel_unlock(chan);
if (max_forwards <= 0) {
ast_log(LOG_WARNING, "Cannot dial from channel '%s'. Max forwards exceeded\n",
ast_channel_name(chan));
}
}
if (!channel->owner) {
if (cap && ast_format_cap_count(cap)) {
cap_request = cap;
} else if (requester_cap) {
cap_request = requester_cap;
} else {
cap_all_audio = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
ast_format_cap_append_by_type(cap_all_audio, AST_MEDIA_TYPE_AUDIO);
cap_request = cap_all_audio;
}
/* If we fail to create our owner channel bail out */
if (!(channel->owner = ast_request(channel->tech, cap_request, &assignedids, chan, channel->device, &channel->cause))) {
ao2_cleanup(cap_all_audio);
return -1;
}
cap_request = NULL;
ao2_cleanup(requester_cap);
media formats: re-architect handling of media for performance improvements In the old times media formats were represented using a bit field. This was fast but had a few limitations. 1. Asterisk was limited in how many formats it could handle. 2. Formats, being a bit field, could not include any attribute information. A format was strictly its type, e.g., "this is ulaw". This was changed in Asterisk 10 (see https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal for notes on that work) which led to the creation of the ast_format structure. This structure allowed Asterisk to handle attributes and bundle information with a format. Additionally, ast_format_cap was created to act as a container for multiple formats that, together, formed the capability of some entity. Another mechanism was added to allow logic to be registered which performed format attribute negotiation. Everywhere throughout the codebase Asterisk was changed to use this strategy. Unfortunately, in software, there is no free lunch. These new capabilities came at a cost. Performance analysis and profiling showed that we spend an inordinate amount of time comparing, copying, and generally manipulating formats and their related structures. Basic prototyping has shown that a reasonably large performance improvement could be made in this area. This patch is the result of that project, which overhauled the media format architecture and its usage in Asterisk to improve performance. Generally, the new philosophy for handling formats is as follows: * The ast_format structure is reference counted. This removed a large amount of the memory allocations and copying that was done in prior versions. * In order to prevent race conditions while keeping things performant, the ast_format structure is immutable by convention and lock-free. Violate this tenet at your peril! * Because formats are reference counted, codecs are also reference counted. The Asterisk core generally provides built-in codecs and caches the ast_format structures created to represent them. Generally, to prevent inordinate amounts of module reference bumping, codecs and formats can be added at run-time but cannot be removed. * All compatibility with the bit field representation of codecs/formats has been moved to a compatibility API. The primary user of this representation is chan_iax2, which must continue to maintain its bit-field usage of formats for interoperability concerns. * When a format is negotiated with attributes, or when a format cannot be represented by one of the cached formats, a new format object is created or cloned from an existing format. That format may have the same codec underlying it, but is a different format than a version of the format with different attributes or without attributes. * While formats are reference counted objects, the reference count maintained on the format should be manipulated with care. Formats are generally cached and will persist for the lifetime of Asterisk and do not explicitly need to have their lifetime modified. An exception to this is when the user of a format does not know where the format came from *and* the user may outlive the provider of the format. This occurs, for example, when a format is read from a channel: the channel may have a format with attributes (hence, non-cached) and the user of the format may last longer than the channel (if the reference to the channel is released prior to the format's reference). For more information on this work, see the API design notes: https://wiki.asterisk.org/wiki/display/AST/Media+Format+Rewrite Finally, this work was the culmination of a large number of developer's efforts. Extra thanks goes to Corey Farrell, who took on a large amount of the work in the Asterisk core, chan_sip, and was an invaluable resource in peer reviews throughout this project. There were a substantial number of patches contributed during this work; the following issues/patch names simply reflect some of the work (and will cause the release scripts to give attribution to the individuals who work on them). Reviews: https://reviewboard.asterisk.org/r/3814 https://reviewboard.asterisk.org/r/3808 https://reviewboard.asterisk.org/r/3805 https://reviewboard.asterisk.org/r/3803 https://reviewboard.asterisk.org/r/3801 https://reviewboard.asterisk.org/r/3798 https://reviewboard.asterisk.org/r/3800 https://reviewboard.asterisk.org/r/3794 https://reviewboard.asterisk.org/r/3793 https://reviewboard.asterisk.org/r/3792 https://reviewboard.asterisk.org/r/3791 https://reviewboard.asterisk.org/r/3790 https://reviewboard.asterisk.org/r/3789 https://reviewboard.asterisk.org/r/3788 https://reviewboard.asterisk.org/r/3787 https://reviewboard.asterisk.org/r/3786 https://reviewboard.asterisk.org/r/3784 https://reviewboard.asterisk.org/r/3783 https://reviewboard.asterisk.org/r/3778 https://reviewboard.asterisk.org/r/3774 https://reviewboard.asterisk.org/r/3775 https://reviewboard.asterisk.org/r/3772 https://reviewboard.asterisk.org/r/3761 https://reviewboard.asterisk.org/r/3754 https://reviewboard.asterisk.org/r/3753 https://reviewboard.asterisk.org/r/3751 https://reviewboard.asterisk.org/r/3750 https://reviewboard.asterisk.org/r/3748 https://reviewboard.asterisk.org/r/3747 https://reviewboard.asterisk.org/r/3746 https://reviewboard.asterisk.org/r/3742 https://reviewboard.asterisk.org/r/3740 https://reviewboard.asterisk.org/r/3739 https://reviewboard.asterisk.org/r/3738 https://reviewboard.asterisk.org/r/3737 https://reviewboard.asterisk.org/r/3736 https://reviewboard.asterisk.org/r/3734 https://reviewboard.asterisk.org/r/3722 https://reviewboard.asterisk.org/r/3713 https://reviewboard.asterisk.org/r/3703 https://reviewboard.asterisk.org/r/3689 https://reviewboard.asterisk.org/r/3687 https://reviewboard.asterisk.org/r/3674 https://reviewboard.asterisk.org/r/3671 https://reviewboard.asterisk.org/r/3667 https://reviewboard.asterisk.org/r/3665 https://reviewboard.asterisk.org/r/3625 https://reviewboard.asterisk.org/r/3602 https://reviewboard.asterisk.org/r/3519 https://reviewboard.asterisk.org/r/3518 https://reviewboard.asterisk.org/r/3516 https://reviewboard.asterisk.org/r/3515 https://reviewboard.asterisk.org/r/3512 https://reviewboard.asterisk.org/r/3506 https://reviewboard.asterisk.org/r/3413 https://reviewboard.asterisk.org/r/3410 https://reviewboard.asterisk.org/r/3387 https://reviewboard.asterisk.org/r/3388 https://reviewboard.asterisk.org/r/3389 https://reviewboard.asterisk.org/r/3390 https://reviewboard.asterisk.org/r/3321 https://reviewboard.asterisk.org/r/3320 https://reviewboard.asterisk.org/r/3319 https://reviewboard.asterisk.org/r/3318 https://reviewboard.asterisk.org/r/3266 https://reviewboard.asterisk.org/r/3265 https://reviewboard.asterisk.org/r/3234 https://reviewboard.asterisk.org/r/3178 ASTERISK-23114 #close Reported by: mjordan media_formats_translation_core.diff uploaded by kharwell (License 6464) rb3506.diff uploaded by mjordan (License 6283) media_format_app_file.diff uploaded by kharwell (License 6464) misc-2.diff uploaded by file (License 5000) chan_mild-3.diff uploaded by file (License 5000) chan_obscure.diff uploaded by file (License 5000) jingle.diff uploaded by file (License 5000) funcs.diff uploaded by file (License 5000) formats.diff uploaded by file (License 5000) core.diff uploaded by file (License 5000) bridges.diff uploaded by file (License 5000) mf-codecs-2.diff uploaded by file (License 5000) mf-app_fax.diff uploaded by file (License 5000) mf-apps-3.diff uploaded by file (License 5000) media-formats-3.diff uploaded by file (License 5000) ASTERISK-23715 rb3713.patch uploaded by coreyfarrell (License 5909) rb3689.patch uploaded by mjordan (License 6283) ASTERISK-23957 rb3722.patch uploaded by mjordan (License 6283) mf-attributes-3.diff uploaded by file (License 5000) ASTERISK-23958 Tested by: jrose rb3822.patch uploaded by coreyfarrell (License 5909) rb3800.patch uploaded by jrose (License 6182) chan_sip.diff uploaded by mjordan (License 6283) rb3747.patch uploaded by jrose (License 6182) ASTERISK-23959 #close Tested by: sgriepentrog, mjordan, coreyfarrell sip_cleanup.diff uploaded by opticron (License 6273) chan_sip_caps.diff uploaded by mjordan (License 6283) rb3751.patch uploaded by coreyfarrell (License 5909) chan_sip-3.diff uploaded by file (License 5000) ASTERISK-23960 #close Tested by: opticron direct_media.diff uploaded by opticron (License 6273) pjsip-direct-media.diff uploaded by file (License 5000) format_cap_remove.diff uploaded by opticron (License 6273) media_format_fixes.diff uploaded by opticron (License 6273) chan_pjsip-2.diff uploaded by file (License 5000) ASTERISK-23966 #close Tested by: rmudgett rb3803.patch uploaded by rmudgetti (License 5621) chan_dahdi.diff uploaded by file (License 5000) ASTERISK-24064 #close Tested by: coreyfarrell, mjordan, opticron, file, rmudgett, sgriepentrog, jrose rb3814.patch uploaded by rmudgett (License 5621) moh_cleanup.diff uploaded by opticron (License 6273) bridge_leak.diff uploaded by opticron (License 6273) translate.diff uploaded by file (License 5000) rb3795.patch uploaded by rmudgett (License 5621) tls_fix.diff uploaded by mjordan (License 6283) fax-mf-fix-2.diff uploaded by file (License 5000) rtp_transfer_stuff uploaded by mjordan (License 6283) rb3787.patch uploaded by rmudgett (License 5621) media-formats-explicit-translate-format-3.diff uploaded by file (License 5000) format_cache_case_fix.diff uploaded by opticron (License 6273) rb3774.patch uploaded by rmudgett (License 5621) rb3775.patch uploaded by rmudgett (License 5621) rtp_engine_fix.diff uploaded by opticron (License 6273) rtp_crash_fix.diff uploaded by opticron (License 6273) rb3753.patch uploaded by mjordan (License 6283) rb3750.patch uploaded by mjordan (License 6283) rb3748.patch uploaded by rmudgett (License 5621) media_format_fixes.diff uploaded by opticron (License 6273) rb3740.patch uploaded by mjordan (License 6283) rb3739.patch uploaded by mjordan (License 6283) rb3734.patch uploaded by mjordan (License 6283) rb3689.patch uploaded by mjordan (License 6283) rb3674.patch uploaded by coreyfarrell (License 5909) rb3671.patch uploaded by coreyfarrell (License 5909) rb3667.patch uploaded by coreyfarrell (License 5909) rb3665.patch uploaded by mjordan (License 6283) rb3625.patch uploaded by coreyfarrell (License 5909) rb3602.patch uploaded by coreyfarrell (License 5909) format_compatibility-2.diff uploaded by file (License 5000) core.diff uploaded by file (License 5000) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419044 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-20 22:06:33 +00:00
ao2_cleanup(cap_all_audio);
}
accountcode: Slightly change accountcode propagation. The previous behavior was to simply set the accountcode of an outgoing channel to the accountcode of the channel initiating the call. It was done this way a long time ago to allow the accountcode set on the SIP/100 channel to be propagated to a local channel so the dialplan execution on the Local;2 channel would have the SIP/100 accountcode available. SIP/100 -> Local;1/Local;2 -> SIP/200 Propagating the SIP/100 accountcode to the local channels is very useful. Without any dialplan manipulation, all channels in this call would have the same accountcode. Using dialplan, you can set a different accountcode on the SIP/200 channel either by setting the accountcode on the Local;2 channel or by the Dial application's b(pre-dial), M(macro) or U(gosub) options, or by the FollowMe application's b(pre-dial) option, or by the Queue application's macro or gosub options. Before Asterisk v12, the altered accountcode on SIP/200 will remain until the local channels optimize out and the accountcode would change to the SIP/100 accountcode. Asterisk v1.8 attempted to add peeraccount support but ultimately had to punt on the support. The peeraccount support was rendered useless because of how the CDR code needed to unconditionally force the caller's accountcode onto the peer channel's accountcode. The CEL events were thus intentionally made to always use the channel's accountcode as the peeraccount value. With the arrival of Asterisk v12, the situation has improved somewhat so peeraccount support can be made to work. Using the indicated example, the the accountcode values become as follows when the peeraccount is set on SIP/100 before calling SIP/200: SIP/100 ---> Local;1 ---- Local;2 ---> SIP/200 acct: 100 \/ acct: 200 \/ acct: 100 \/ acct: 200 peer: 200 /\ peer: 100 /\ peer: 200 /\ peer: 100 If a channel already has an accountcode it can only change by the following explicit user actions: 1) A channel originate method that can specify an accountcode to use. 2) The calling channel propagating its non-empty peeraccount or its non-empty accountcode if the peeraccount was empty to the outgoing channel's accountcode before initiating the dial. e.g., Dial and FollowMe. The exception to this propagation method is Queue. Queue will only propagate peeraccounts this way only if the outgoing channel does not have an accountcode. 3) Dialplan using CHANNEL(accountcode). 4) Dialplan using CHANNEL(peeraccount) on the other end of a local channel pair. If a channel does not have an accountcode it can get one from the following places: 1) The channel driver's configuration at channel creation. 2) Explicit user action as already indicated. 3) Entering a basic or stasis-mixing bridge from a peer channel's peeraccount value. You can specify the accountcode for an outgoing channel by setting the CHANNEL(peeraccount) before using the Dial, FollowMe, and Queue applications. Queue adds the wrinkle that it will not overwrite an existing accountcode on the outgoing channel with the calling channels values. Accountcode and peeraccount values propagate to an outgoing channel before dialing. Accountcodes also propagate when channels enter or leave a basic or stasis-mixing bridge. The peeraccount value only makes sense for mixing bridges with two channels; it is meaningless otherwise. * Made peeraccount functional by changing accountcode propagation as described above. * Fixed CEL extracting the wrong ie value for the peeraccount. This was done intentionally in Asterisk v1.8 when that version had to punt on peeraccount. * Fixed a few places dealing with accountcodes that were reading from channels without the lock held. AFS-65 #close Review: https://reviewboard.asterisk.org/r/3601/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419520 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-24 22:48:38 +00:00
if (chan) {
ast_channel_lock_both(chan, channel->owner);
} else {
ast_channel_lock(channel->owner);
}
ast_channel_stage_snapshot(channel->owner);
ast_channel_appl_set(channel->owner, "AppDial2");
ast_channel_data_set(channel->owner, "(Outgoing Line)");
stasis: Reduce creation of channel snapshots to improve performance During some performance testing of Asterisk with AGI, ARI, and lots of Local channels, we noticed that there's quite a hit in performance during channel creation and releasing to the dialplan (ARI continue). After investigating the performance spike that occurs during channel creation, we discovered that we create a lot of channel snapshots that are technically unnecessary. This includes creating snapshots during: * AGI execution * Returning objects for ARI commands * During some Local channel operations * During some dialling operations * During variable setting * During some bridging operations And more. This patch does the following: - It removes a number of fields from channel snapshots. These fields were rarely used, were expensive to have on the snapshot, and hurt performance. This included formats, translation paths, Log Call ID, callgroup, pickup group, and all channel variables. As a result, AMI Status, "core show channel", "core show channelvar", and "pjsip show channel" were modified to either hit the live channel or not show certain pieces of data. While this is unfortunate, the performance gain from this patch is worth the loss in behaviour. - It adds a mechanism to publish a cached snapshot + blob. A large number of publications were changed to use this, including: - During Dial begin - During Variable assignment (if no AMI variables are emitted - if AMI variables are set, we have to make snapshots when a variable is changed) - During channel pickup - When a channel is put on hold/unhold - When a DTMF digit is begun/ended - When creating a bridge snapshot - When an AOC event is raised - During Local channel optimization/Local bridging - When endpoint snapshots are generated - All AGI events - All ARI responses that return a channel - Events in the AgentPool, MeetMe, and some in Queue - Additionally, some extraneous channel snapshots were being made that were unnecessary. These were removed. - The result of ast_hashtab_hash_string is now cached in stasis_cache. This reduces a large number of calls to ast_hashtab_hash_string, which reduced the amount of time spent in this function in gprof by around 50%. #ASTERISK-23811 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3568/ ........ Merged revisions 416211 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416216 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-13 18:24:49 +00:00
memset(ast_channel_whentohangup(channel->owner), 0, sizeof(*ast_channel_whentohangup(channel->owner)));
/* Inherit everything from he who spawned this dial */
if (chan) {
ast_channel_inherit_variables(chan, channel->owner);
ast_channel_datastore_inherit(chan, channel->owner);
Detect potential forwarding loops based on count. A potential problem that can arise is the following: * Bob's phone is programmed to automatically forward to Carol. * Carol's phone is programmed to automatically forward to Bob. * Alice calls Bob. If left unchecked, this results in an endless loops of call forwards that would eventually result in some sort of fiery crash. Asterisk's method of solving this issue was to track which interfaces had been dialed. If a destination were dialed a second time, then the attempt to call that destination would fail since a loop was detected. The problem with this method is that call forwarding has evolved. Some SIP phones allow for a user to manually forward an incoming call to an ad-hoc destination. This can mean that: * There are legitimate use cases where a device may be dialed multiple times, or * There can be human error when forwarding calls. This change removes the old method of detecting forwarding loops in favor of keeping a count of the number of destinations a channel has dialed on a particular branch of a call. If the number exceeds the set number of max forwards, then the call fails. This approach has the following advantages over the old: * It is much simpler. * It can detect loops involving local channels. * It is user configurable. The only disadvantage it has is that in the case where there is a legitimate forwarding loop present, it takes longer to detect it. However, the forwarding loop is still properly detected and the call is cleaned up as it should be. Address review feedback on gerrit. * Correct "mfgium" to "Digium" * Decrement max forwards by one in the case where allocation of the max forwards datastore is required. * Remove irrelevant code change from pjsip_global_headers.c ASTERISK-24958 #close Change-Id: Ia7e4b7cd3bccfbd34d9a859838356931bba56c23
2015-04-15 15:38:02 +00:00
ast_max_forwards_decrement(channel->owner);
/* Copy over callerid information */
ast_party_redirecting_copy(ast_channel_redirecting(channel->owner), ast_channel_redirecting(chan));
ast_channel_dialed(channel->owner)->transit_network_select = ast_channel_dialed(chan)->transit_network_select;
ast_connected_line_copy_from_caller(ast_channel_connected(channel->owner), ast_channel_caller(chan));
ast_channel_language_set(channel->owner, ast_channel_language(chan));
if (channel->options[AST_DIAL_OPTION_DIAL_REPLACES_SELF]) {
ast_channel_req_accountcodes(channel->owner, chan, AST_CHANNEL_REQUESTOR_REPLACEMENT);
} else {
ast_channel_req_accountcodes(channel->owner, chan, AST_CHANNEL_REQUESTOR_BRIDGE_PEER);
}
if (ast_strlen_zero(ast_channel_musicclass(channel->owner)))
ast_channel_musicclass_set(channel->owner, ast_channel_musicclass(chan));
ast_channel_adsicpe_set(channel->owner, ast_channel_adsicpe(chan));
ast_channel_transfercapability_set(channel->owner, ast_channel_transfercapability(chan));
accountcode: Slightly change accountcode propagation. The previous behavior was to simply set the accountcode of an outgoing channel to the accountcode of the channel initiating the call. It was done this way a long time ago to allow the accountcode set on the SIP/100 channel to be propagated to a local channel so the dialplan execution on the Local;2 channel would have the SIP/100 accountcode available. SIP/100 -> Local;1/Local;2 -> SIP/200 Propagating the SIP/100 accountcode to the local channels is very useful. Without any dialplan manipulation, all channels in this call would have the same accountcode. Using dialplan, you can set a different accountcode on the SIP/200 channel either by setting the accountcode on the Local;2 channel or by the Dial application's b(pre-dial), M(macro) or U(gosub) options, or by the FollowMe application's b(pre-dial) option, or by the Queue application's macro or gosub options. Before Asterisk v12, the altered accountcode on SIP/200 will remain until the local channels optimize out and the accountcode would change to the SIP/100 accountcode. Asterisk v1.8 attempted to add peeraccount support but ultimately had to punt on the support. The peeraccount support was rendered useless because of how the CDR code needed to unconditionally force the caller's accountcode onto the peer channel's accountcode. The CEL events were thus intentionally made to always use the channel's accountcode as the peeraccount value. With the arrival of Asterisk v12, the situation has improved somewhat so peeraccount support can be made to work. Using the indicated example, the the accountcode values become as follows when the peeraccount is set on SIP/100 before calling SIP/200: SIP/100 ---> Local;1 ---- Local;2 ---> SIP/200 acct: 100 \/ acct: 200 \/ acct: 100 \/ acct: 200 peer: 200 /\ peer: 100 /\ peer: 200 /\ peer: 100 If a channel already has an accountcode it can only change by the following explicit user actions: 1) A channel originate method that can specify an accountcode to use. 2) The calling channel propagating its non-empty peeraccount or its non-empty accountcode if the peeraccount was empty to the outgoing channel's accountcode before initiating the dial. e.g., Dial and FollowMe. The exception to this propagation method is Queue. Queue will only propagate peeraccounts this way only if the outgoing channel does not have an accountcode. 3) Dialplan using CHANNEL(accountcode). 4) Dialplan using CHANNEL(peeraccount) on the other end of a local channel pair. If a channel does not have an accountcode it can get one from the following places: 1) The channel driver's configuration at channel creation. 2) Explicit user action as already indicated. 3) Entering a basic or stasis-mixing bridge from a peer channel's peeraccount value. You can specify the accountcode for an outgoing channel by setting the CHANNEL(peeraccount) before using the Dial, FollowMe, and Queue applications. Queue adds the wrinkle that it will not overwrite an existing accountcode on the outgoing channel with the calling channels values. Accountcode and peeraccount values propagate to an outgoing channel before dialing. Accountcodes also propagate when channels enter or leave a basic or stasis-mixing bridge. The peeraccount value only makes sense for mixing bridges with two channels; it is meaningless otherwise. * Made peeraccount functional by changing accountcode propagation as described above. * Fixed CEL extracting the wrong ie value for the peeraccount. This was done intentionally in Asterisk v1.8 when that version had to punt on peeraccount. * Fixed a few places dealing with accountcodes that were reading from channels without the lock held. AFS-65 #close Review: https://reviewboard.asterisk.org/r/3601/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419520 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-24 22:48:38 +00:00
ast_channel_unlock(chan);
}
ast_channel_stage_snapshot_done(channel->owner);
ast_channel_unlock(channel->owner);
if (!ast_strlen_zero(predial_string)) {
if (chan) {
ast_autoservice_start(chan);
}
ast_pre_call(channel->owner, predial_string);
if (chan) {
ast_autoservice_stop(chan);
}
}
return 0;
}
int ast_dial_prerun(struct ast_dial *dial, struct ast_channel *chan, struct ast_format_cap *cap)
{
struct ast_dial_channel *channel;
int res = -1;
char *predial_string = dial->options[AST_DIAL_OPTION_PREDIAL];
AST_LIST_LOCK(&dial->channels);
AST_LIST_TRAVERSE(&dial->channels, channel, list) {
if ((res = begin_dial_prerun(channel, chan, cap, predial_string))) {
break;
}
}
AST_LIST_UNLOCK(&dial->channels);
return res;
}
/*! \brief Helper function that does the beginning dialing per-appended channel */
static int begin_dial_channel(struct ast_dial_channel *channel, struct ast_channel *chan, int async, const char *predial_string, struct ast_channel *forwarder_chan)
{
int res = 1;
char forwarder[AST_CHANNEL_NAME];
/* If no owner channel exists yet execute pre-run */
if (!channel->owner && begin_dial_prerun(channel, chan, NULL, predial_string)) {
return 0;
}
if (forwarder_chan) {
ast_copy_string(forwarder, ast_channel_name(forwarder_chan), sizeof(forwarder));
ast_channel_lock(channel->owner);
pbx_builtin_setvar_helper(channel->owner, "FORWARDERNAME", forwarder);
ast_channel_unlock(channel->owner);
}
/* Attempt to actually call this device */
if ((res = ast_call(channel->owner, channel->device, 0))) {
res = 0;
ast_hangup(channel->owner);
channel->owner = NULL;
} else {
Update Asterisk's CDRs for the new bridging framework This patch is the initial push to update Asterisk's CDR engine for the new bridging framework. This patch guts the existing CDR engine and builds the new on top of messages coming across Stasis. As changes in channel state and bridge state are detected, CDRs are built and dispatched accordingly. This fundamentally changes CDRs in a few ways. (1) CDRs are now *very* reflective of the actual state of channels and bridges. This means CDRs track well with what an actual channel is doing - which is useful in transfer scenarios (which were previously difficult to pin down). It does, however, mean that CDRs cannot be 'fooled'. Previous behavior in Asterisk allowed for CDR applications, channels, and other properties to be spoofed in parts of the code - this no longer works. (2) CDRs have defined behavior in multi-party scenarios. This behavior will not be what everyone wants, but it is a defined behavior and as such, it is predictable. (3) The CDR manipulation functions and applications have been overhauled. Major changes have been made to ResetCDR and ForkCDR in particular. Many of the options for these two applications no longer made any sense with the new framework and the (slightly) more immutable nature of CDRs. There are a plethora of other changes. For a full description of CDR behavior, see the CDR specification on the Asterisk wiki. (closes issue ASTERISK-21196) Review: https://reviewboard.asterisk.org/r/2486/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@391947 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-06-17 03:00:38 +00:00
ast_channel_publish_dial(async ? NULL : chan, channel->owner, channel->device, NULL);
res = 1;
ast_verb(3, "Called %s\n", channel->device);
}
return res;
}
/*! \brief Helper function that does the beginning dialing per dial structure */
Update Asterisk's CDRs for the new bridging framework This patch is the initial push to update Asterisk's CDR engine for the new bridging framework. This patch guts the existing CDR engine and builds the new on top of messages coming across Stasis. As changes in channel state and bridge state are detected, CDRs are built and dispatched accordingly. This fundamentally changes CDRs in a few ways. (1) CDRs are now *very* reflective of the actual state of channels and bridges. This means CDRs track well with what an actual channel is doing - which is useful in transfer scenarios (which were previously difficult to pin down). It does, however, mean that CDRs cannot be 'fooled'. Previous behavior in Asterisk allowed for CDR applications, channels, and other properties to be spoofed in parts of the code - this no longer works. (2) CDRs have defined behavior in multi-party scenarios. This behavior will not be what everyone wants, but it is a defined behavior and as such, it is predictable. (3) The CDR manipulation functions and applications have been overhauled. Major changes have been made to ResetCDR and ForkCDR in particular. Many of the options for these two applications no longer made any sense with the new framework and the (slightly) more immutable nature of CDRs. There are a plethora of other changes. For a full description of CDR behavior, see the CDR specification on the Asterisk wiki. (closes issue ASTERISK-21196) Review: https://reviewboard.asterisk.org/r/2486/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@391947 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-06-17 03:00:38 +00:00
static int begin_dial(struct ast_dial *dial, struct ast_channel *chan, int async)
{
struct ast_dial_channel *channel = NULL;
int success = 0;
char *predial_string = dial->options[AST_DIAL_OPTION_PREDIAL];
/* Iterate through channel list, requesting and calling each one */
AST_LIST_LOCK(&dial->channels);
AST_LIST_TRAVERSE(&dial->channels, channel, list) {
success += begin_dial_channel(channel, chan, async, predial_string, NULL);
}
AST_LIST_UNLOCK(&dial->channels);
/* If number of failures matches the number of channels, then this truly failed */
return success;
}
/*! \brief Helper function to handle channels that have been call forwarded */
static int handle_call_forward(struct ast_dial *dial, struct ast_dial_channel *channel, struct ast_channel *chan)
{
struct ast_channel *original = channel->owner;
char *tmp = ast_strdupa(ast_channel_call_forward(channel->owner));
char *tech = "Local", *device = tmp, *stuff;
char *predial_string = dial->options[AST_DIAL_OPTION_PREDIAL];
/* If call forwarding is disabled just drop the original channel and don't attempt to dial the new one */
if (FIND_RELATIVE_OPTION(dial, channel, AST_DIAL_OPTION_DISABLE_CALL_FORWARDING)) {
ast_hangup(original);
channel->owner = NULL;
return 0;
}
/* Figure out the new destination */
if ((stuff = strchr(tmp, '/'))) {
*stuff++ = '\0';
tech = tmp;
device = stuff;
} else {
const char *forward_context;
char destination[AST_MAX_CONTEXT + AST_MAX_EXTENSION + 1];
ast_channel_lock(original);
forward_context = pbx_builtin_getvar_helper(original, "FORWARD_CONTEXT");
snprintf(destination, sizeof(destination), "%s@%s", tmp, S_OR(forward_context, ast_channel_context(original)));
ast_channel_unlock(original);
device = ast_strdupa(destination);
}
/* Drop old destination information */
ast_free(channel->tech);
ast_free(channel->device);
ast_free(channel->assignedid1);
channel->assignedid1 = NULL;
ast_free(channel->assignedid2);
channel->assignedid2 = NULL;
/* Update the dial channel with the new destination information */
channel->tech = ast_strdup(tech);
channel->device = ast_strdup(device);
AST_LIST_UNLOCK(&dial->channels);
/* Drop the original channel */
channel->owner = NULL;
/* Finally give it a go... send it out into the world */
begin_dial_channel(channel, chan, chan ? 0 : 1, predial_string, original);
ast_channel_publish_dial_forward(chan, original, channel->owner, NULL, "CANCEL",
ast_channel_call_forward(original));
ast_hangup(original);
return 0;
}
/*! \brief Helper function that finds the dialed channel based on owner */
static struct ast_dial_channel *find_relative_dial_channel(struct ast_dial *dial, struct ast_channel *owner)
{
struct ast_dial_channel *channel = NULL;
AST_LIST_LOCK(&dial->channels);
AST_LIST_TRAVERSE(&dial->channels, channel, list) {
if (channel->owner == owner)
break;
}
AST_LIST_UNLOCK(&dial->channels);
return channel;
}
static void set_state(struct ast_dial *dial, enum ast_dial_result state)
{
dial->state = state;
if (dial->state_callback)
dial->state_callback(dial);
}
/*! \brief Helper function that handles frames */
static void handle_frame(struct ast_dial *dial, struct ast_dial_channel *channel, struct ast_frame *fr, struct ast_channel *chan)
{
if (fr->frametype == AST_FRAME_CONTROL) {
switch (fr->subclass.integer) {
case AST_CONTROL_ANSWER:
if (chan) {
ast_verb(3, "%s answered %s\n", ast_channel_name(channel->owner), ast_channel_name(chan));
} else {
ast_verb(3, "%s answered\n", ast_channel_name(channel->owner));
}
AST_LIST_LOCK(&dial->channels);
AST_LIST_REMOVE(&dial->channels, channel, list);
AST_LIST_INSERT_HEAD(&dial->channels, channel, list);
AST_LIST_UNLOCK(&dial->channels);
ast_channel_publish_dial(chan, channel->owner, channel->device, "ANSWER");
set_state(dial, AST_DIAL_RESULT_ANSWERED);
break;
case AST_CONTROL_BUSY:
Replace direct access to channel name with accessor functions There are many benefits to making the ast_channel an opaque handle, from increasing maintainability to presenting ways to kill masquerades. This patch kicks things off by taking things a field at a time, renaming the field to '__do_not_use_${fieldname}' and then writing setters/getters and converting the existing code to using them. When all fields are done, we can move ast_channel to a C file from channel.h and lop off the '__do_not_use_'. This patch sets up main/channel_interal_api.c to be the only file that actually accesses the ast_channel's fields directly. The intent would be for any API functions in channel.c to use the accessor functions. No more monkeying around with channel internals. We should use our own APIs. The interesting changes in this patch are the addition of channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to use accessor functions when ast_channel is really opaque), and some re-working of the way channel iterators/callbacks are handled so as to avoid creating fake ast_channels on the stack to pass in matching data by directly accessing fields (since "name" is a stringfield and the fake channel doesn't init the stringfields, you can't use the ast_channel_name_set() function). I went with ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a setter. The majority of the grunt-work for this change was done by writing a semantic patch using Coccinelle ( http://coccinelle.lip6.fr/ ). Review: https://reviewboard.asterisk.org/r/1655/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-09 22:15:50 +00:00
ast_verb(3, "%s is busy\n", ast_channel_name(channel->owner));
ast_channel_publish_dial(chan, channel->owner, channel->device, "BUSY");
ast_hangup(channel->owner);
channel->cause = AST_CAUSE_USER_BUSY;
channel->owner = NULL;
break;
case AST_CONTROL_CONGESTION:
Replace direct access to channel name with accessor functions There are many benefits to making the ast_channel an opaque handle, from increasing maintainability to presenting ways to kill masquerades. This patch kicks things off by taking things a field at a time, renaming the field to '__do_not_use_${fieldname}' and then writing setters/getters and converting the existing code to using them. When all fields are done, we can move ast_channel to a C file from channel.h and lop off the '__do_not_use_'. This patch sets up main/channel_interal_api.c to be the only file that actually accesses the ast_channel's fields directly. The intent would be for any API functions in channel.c to use the accessor functions. No more monkeying around with channel internals. We should use our own APIs. The interesting changes in this patch are the addition of channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to use accessor functions when ast_channel is really opaque), and some re-working of the way channel iterators/callbacks are handled so as to avoid creating fake ast_channels on the stack to pass in matching data by directly accessing fields (since "name" is a stringfield and the fake channel doesn't init the stringfields, you can't use the ast_channel_name_set() function). I went with ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a setter. The majority of the grunt-work for this change was done by writing a semantic patch using Coccinelle ( http://coccinelle.lip6.fr/ ). Review: https://reviewboard.asterisk.org/r/1655/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-09 22:15:50 +00:00
ast_verb(3, "%s is circuit-busy\n", ast_channel_name(channel->owner));
ast_channel_publish_dial(chan, channel->owner, channel->device, "CONGESTION");
ast_hangup(channel->owner);
channel->cause = AST_CAUSE_NORMAL_CIRCUIT_CONGESTION;
channel->owner = NULL;
break;
Merged revisions 335078 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r335078 | mjordan | 2011-09-09 11:27:01 -0500 (Fri, 09 Sep 2011) | 29 lines Merged revisions 335064 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335064 | mjordan | 2011-09-09 11:09:09 -0500 (Fri, 09 Sep 2011) | 23 lines Updated SIP 484 handling; added Incomplete control frame When a SIP phone uses the dial application and receives a 484 Address Incomplete response, if overlapped dialing is enabled for SIP, then the 484 Address Incomplete is forwarded back to the SIP phone and the HANGUPCAUSE channel variable is set to 28. Previously, the Incomplete application dialplan logic was automatically triggered; now, explicit dialplan usage of the application is required. Additionally, this patch adds a new AST_CONTOL_FRAME type called AST_CONTROL_INCOMPLETE. If a channel driver receives this control frame, it is an indication that the dialplan expects more digits back from the device. If the device supports overlap dialing it should attempt to notify the device that the dialplan is waiting for more digits; otherwise, it can handle the frame in a manner appropriate to the channel driver. (closes issue ASTERISK-17288) Reported by: Mikael Carlsson Tested by: Matthew Jordan Review: https://reviewboard.asterisk.org/r/1416/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335079 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-09 16:28:23 +00:00
case AST_CONTROL_INCOMPLETE:
ast_verb(3, "%s dialed Incomplete extension %s\n", ast_channel_name(channel->owner), ast_channel_exten(channel->owner));
if (chan) {
ast_indicate(chan, AST_CONTROL_INCOMPLETE);
} else {
ast_hangup(channel->owner);
channel->cause = AST_CAUSE_UNALLOCATED;
channel->owner = NULL;
}
Merged revisions 335078 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r335078 | mjordan | 2011-09-09 11:27:01 -0500 (Fri, 09 Sep 2011) | 29 lines Merged revisions 335064 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335064 | mjordan | 2011-09-09 11:09:09 -0500 (Fri, 09 Sep 2011) | 23 lines Updated SIP 484 handling; added Incomplete control frame When a SIP phone uses the dial application and receives a 484 Address Incomplete response, if overlapped dialing is enabled for SIP, then the 484 Address Incomplete is forwarded back to the SIP phone and the HANGUPCAUSE channel variable is set to 28. Previously, the Incomplete application dialplan logic was automatically triggered; now, explicit dialplan usage of the application is required. Additionally, this patch adds a new AST_CONTOL_FRAME type called AST_CONTROL_INCOMPLETE. If a channel driver receives this control frame, it is an indication that the dialplan expects more digits back from the device. If the device supports overlap dialing it should attempt to notify the device that the dialplan is waiting for more digits; otherwise, it can handle the frame in a manner appropriate to the channel driver. (closes issue ASTERISK-17288) Reported by: Mikael Carlsson Tested by: Matthew Jordan Review: https://reviewboard.asterisk.org/r/1416/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335079 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-09 16:28:23 +00:00
break;
case AST_CONTROL_RINGING:
Replace direct access to channel name with accessor functions There are many benefits to making the ast_channel an opaque handle, from increasing maintainability to presenting ways to kill masquerades. This patch kicks things off by taking things a field at a time, renaming the field to '__do_not_use_${fieldname}' and then writing setters/getters and converting the existing code to using them. When all fields are done, we can move ast_channel to a C file from channel.h and lop off the '__do_not_use_'. This patch sets up main/channel_interal_api.c to be the only file that actually accesses the ast_channel's fields directly. The intent would be for any API functions in channel.c to use the accessor functions. No more monkeying around with channel internals. We should use our own APIs. The interesting changes in this patch are the addition of channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to use accessor functions when ast_channel is really opaque), and some re-working of the way channel iterators/callbacks are handled so as to avoid creating fake ast_channels on the stack to pass in matching data by directly accessing fields (since "name" is a stringfield and the fake channel doesn't init the stringfields, you can't use the ast_channel_name_set() function). I went with ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a setter. The majority of the grunt-work for this change was done by writing a semantic patch using Coccinelle ( http://coccinelle.lip6.fr/ ). Review: https://reviewboard.asterisk.org/r/1655/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-09 22:15:50 +00:00
ast_verb(3, "%s is ringing\n", ast_channel_name(channel->owner));
ast_channel_publish_dial(chan, channel->owner, channel->device, "RINGING");
if (chan && !dial->options[AST_DIAL_OPTION_MUSIC])
ast_indicate(chan, AST_CONTROL_RINGING);
set_state(dial, AST_DIAL_RESULT_RINGING);
break;
case AST_CONTROL_PROGRESS:
ast_channel_publish_dial(chan, channel->owner, channel->device, "PROGRESS");
if (chan) {
ast_verb(3, "%s is making progress, passing it to %s\n", ast_channel_name(channel->owner), ast_channel_name(chan));
ast_indicate(chan, AST_CONTROL_PROGRESS);
} else {
ast_verb(3, "%s is making progress\n", ast_channel_name(channel->owner));
}
set_state(dial, AST_DIAL_RESULT_PROGRESS);
break;
case AST_CONTROL_VIDUPDATE:
if (!chan) {
break;
}
Replace direct access to channel name with accessor functions There are many benefits to making the ast_channel an opaque handle, from increasing maintainability to presenting ways to kill masquerades. This patch kicks things off by taking things a field at a time, renaming the field to '__do_not_use_${fieldname}' and then writing setters/getters and converting the existing code to using them. When all fields are done, we can move ast_channel to a C file from channel.h and lop off the '__do_not_use_'. This patch sets up main/channel_interal_api.c to be the only file that actually accesses the ast_channel's fields directly. The intent would be for any API functions in channel.c to use the accessor functions. No more monkeying around with channel internals. We should use our own APIs. The interesting changes in this patch are the addition of channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to use accessor functions when ast_channel is really opaque), and some re-working of the way channel iterators/callbacks are handled so as to avoid creating fake ast_channels on the stack to pass in matching data by directly accessing fields (since "name" is a stringfield and the fake channel doesn't init the stringfields, you can't use the ast_channel_name_set() function). I went with ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a setter. The majority of the grunt-work for this change was done by writing a semantic patch using Coccinelle ( http://coccinelle.lip6.fr/ ). Review: https://reviewboard.asterisk.org/r/1655/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-09 22:15:50 +00:00
ast_verb(3, "%s requested a video update, passing it to %s\n", ast_channel_name(channel->owner), ast_channel_name(chan));
ast_indicate(chan, AST_CONTROL_VIDUPDATE);
break;
case AST_CONTROL_SRCUPDATE:
if (!chan) {
break;
}
Replace direct access to channel name with accessor functions There are many benefits to making the ast_channel an opaque handle, from increasing maintainability to presenting ways to kill masquerades. This patch kicks things off by taking things a field at a time, renaming the field to '__do_not_use_${fieldname}' and then writing setters/getters and converting the existing code to using them. When all fields are done, we can move ast_channel to a C file from channel.h and lop off the '__do_not_use_'. This patch sets up main/channel_interal_api.c to be the only file that actually accesses the ast_channel's fields directly. The intent would be for any API functions in channel.c to use the accessor functions. No more monkeying around with channel internals. We should use our own APIs. The interesting changes in this patch are the addition of channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to use accessor functions when ast_channel is really opaque), and some re-working of the way channel iterators/callbacks are handled so as to avoid creating fake ast_channels on the stack to pass in matching data by directly accessing fields (since "name" is a stringfield and the fake channel doesn't init the stringfields, you can't use the ast_channel_name_set() function). I went with ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a setter. The majority of the grunt-work for this change was done by writing a semantic patch using Coccinelle ( http://coccinelle.lip6.fr/ ). Review: https://reviewboard.asterisk.org/r/1655/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-09 22:15:50 +00:00
ast_verb(3, "%s requested a source update, passing it to %s\n", ast_channel_name(channel->owner), ast_channel_name(chan));
ast_indicate(chan, AST_CONTROL_SRCUPDATE);
break;
case AST_CONTROL_CONNECTED_LINE:
if (!chan) {
break;
}
Replace direct access to channel name with accessor functions There are many benefits to making the ast_channel an opaque handle, from increasing maintainability to presenting ways to kill masquerades. This patch kicks things off by taking things a field at a time, renaming the field to '__do_not_use_${fieldname}' and then writing setters/getters and converting the existing code to using them. When all fields are done, we can move ast_channel to a C file from channel.h and lop off the '__do_not_use_'. This patch sets up main/channel_interal_api.c to be the only file that actually accesses the ast_channel's fields directly. The intent would be for any API functions in channel.c to use the accessor functions. No more monkeying around with channel internals. We should use our own APIs. The interesting changes in this patch are the addition of channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to use accessor functions when ast_channel is really opaque), and some re-working of the way channel iterators/callbacks are handled so as to avoid creating fake ast_channels on the stack to pass in matching data by directly accessing fields (since "name" is a stringfield and the fake channel doesn't init the stringfields, you can't use the ast_channel_name_set() function). I went with ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a setter. The majority of the grunt-work for this change was done by writing a semantic patch using Coccinelle ( http://coccinelle.lip6.fr/ ). Review: https://reviewboard.asterisk.org/r/1655/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-09 22:15:50 +00:00
ast_verb(3, "%s connected line has changed, passing it to %s\n", ast_channel_name(channel->owner), ast_channel_name(chan));
if (ast_channel_connected_line_sub(channel->owner, chan, fr, 1)) {
ast_indicate_data(chan, AST_CONTROL_CONNECTED_LINE, fr->data.ptr, fr->datalen);
}
break;
case AST_CONTROL_REDIRECTING:
if (!chan) {
break;
}
Replace direct access to channel name with accessor functions There are many benefits to making the ast_channel an opaque handle, from increasing maintainability to presenting ways to kill masquerades. This patch kicks things off by taking things a field at a time, renaming the field to '__do_not_use_${fieldname}' and then writing setters/getters and converting the existing code to using them. When all fields are done, we can move ast_channel to a C file from channel.h and lop off the '__do_not_use_'. This patch sets up main/channel_interal_api.c to be the only file that actually accesses the ast_channel's fields directly. The intent would be for any API functions in channel.c to use the accessor functions. No more monkeying around with channel internals. We should use our own APIs. The interesting changes in this patch are the addition of channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to use accessor functions when ast_channel is really opaque), and some re-working of the way channel iterators/callbacks are handled so as to avoid creating fake ast_channels on the stack to pass in matching data by directly accessing fields (since "name" is a stringfield and the fake channel doesn't init the stringfields, you can't use the ast_channel_name_set() function). I went with ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a setter. The majority of the grunt-work for this change was done by writing a semantic patch using Coccinelle ( http://coccinelle.lip6.fr/ ). Review: https://reviewboard.asterisk.org/r/1655/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-09 22:15:50 +00:00
ast_verb(3, "%s redirecting info has changed, passing it to %s\n", ast_channel_name(channel->owner), ast_channel_name(chan));
if (ast_channel_redirecting_sub(channel->owner, chan, fr, 1)) {
Enhancements to connected line and redirecting work. From reviewboard: Digium has a commercial customer who has made extensive use of the connected party and redirecting information present in later versions of Asterisk Business Edition and which is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions have come about. This patch adds several enhancements to maximize usage of the connected party and redirecting information functionality. First, Asterisk trunk already had connected line interception macros. These macros allow you to manipulate connected line information before it was sent out to its target. This patch adds the same feature except for redirecting information instead. Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI, mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is that it can be set to whatever value the administrator likes. Later, when running connected line and redirecting macros, the admin can read the tag off the appropriate structure to determine what action to take. You can think of this sort of like a channel variable, except that instead of having the variable associated with a channel, the variable is associated with a specific identity within Asterisk. Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force a specific calling presentation value on the outgoing channel. Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party being transferred would not have the opportunity to run a connected line interception macro to possibly alter the transfer target's connected line information. The issue here was that during a blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line update. The way this was corrected was to add this new control frame subclass. Now, we queue an AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should be run. When ast_read is called to read the frame, ast_read responds by calling a callback function associated with the specific read action the control frame describes. In this case, the action taken is to run the connected line interception macro on the transferee's channel. Review: https://reviewboard.asterisk.org/r/652/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-17 15:36:31 +00:00
ast_indicate_data(chan, AST_CONTROL_REDIRECTING, fr->data.ptr, fr->datalen);
}
break;
case AST_CONTROL_PROCEEDING:
ast_channel_publish_dial(chan, channel->owner, channel->device, "PROCEEDING");
if (chan) {
ast_verb(3, "%s is proceeding, passing it to %s\n", ast_channel_name(channel->owner), ast_channel_name(chan));
ast_indicate(chan, AST_CONTROL_PROCEEDING);
} else {
ast_verb(3, "%s is proceeding\n", ast_channel_name(channel->owner));
}
set_state(dial, AST_DIAL_RESULT_PROCEEDING);
break;
case AST_CONTROL_HOLD:
if (!chan) {
break;
}
Replace direct access to channel name with accessor functions There are many benefits to making the ast_channel an opaque handle, from increasing maintainability to presenting ways to kill masquerades. This patch kicks things off by taking things a field at a time, renaming the field to '__do_not_use_${fieldname}' and then writing setters/getters and converting the existing code to using them. When all fields are done, we can move ast_channel to a C file from channel.h and lop off the '__do_not_use_'. This patch sets up main/channel_interal_api.c to be the only file that actually accesses the ast_channel's fields directly. The intent would be for any API functions in channel.c to use the accessor functions. No more monkeying around with channel internals. We should use our own APIs. The interesting changes in this patch are the addition of channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to use accessor functions when ast_channel is really opaque), and some re-working of the way channel iterators/callbacks are handled so as to avoid creating fake ast_channels on the stack to pass in matching data by directly accessing fields (since "name" is a stringfield and the fake channel doesn't init the stringfields, you can't use the ast_channel_name_set() function). I went with ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a setter. The majority of the grunt-work for this change was done by writing a semantic patch using Coccinelle ( http://coccinelle.lip6.fr/ ). Review: https://reviewboard.asterisk.org/r/1655/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-09 22:15:50 +00:00
ast_verb(3, "Call on %s placed on hold\n", ast_channel_name(chan));
ast_indicate_data(chan, AST_CONTROL_HOLD, fr->data.ptr, fr->datalen);
break;
case AST_CONTROL_UNHOLD:
if (!chan) {
break;
}
Replace direct access to channel name with accessor functions There are many benefits to making the ast_channel an opaque handle, from increasing maintainability to presenting ways to kill masquerades. This patch kicks things off by taking things a field at a time, renaming the field to '__do_not_use_${fieldname}' and then writing setters/getters and converting the existing code to using them. When all fields are done, we can move ast_channel to a C file from channel.h and lop off the '__do_not_use_'. This patch sets up main/channel_interal_api.c to be the only file that actually accesses the ast_channel's fields directly. The intent would be for any API functions in channel.c to use the accessor functions. No more monkeying around with channel internals. We should use our own APIs. The interesting changes in this patch are the addition of channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to use accessor functions when ast_channel is really opaque), and some re-working of the way channel iterators/callbacks are handled so as to avoid creating fake ast_channels on the stack to pass in matching data by directly accessing fields (since "name" is a stringfield and the fake channel doesn't init the stringfields, you can't use the ast_channel_name_set() function). I went with ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a setter. The majority of the grunt-work for this change was done by writing a semantic patch using Coccinelle ( http://coccinelle.lip6.fr/ ). Review: https://reviewboard.asterisk.org/r/1655/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-09 22:15:50 +00:00
ast_verb(3, "Call on %s left from hold\n", ast_channel_name(chan));
ast_indicate(chan, AST_CONTROL_UNHOLD);
break;
case AST_CONTROL_OFFHOOK:
case AST_CONTROL_FLASH:
break;
case AST_CONTROL_PVT_CAUSE_CODE:
if (chan) {
ast_indicate_data(chan, AST_CONTROL_PVT_CAUSE_CODE, fr->data.ptr, fr->datalen);
}
break;
case -1:
if (chan) {
/* Prod the channel */
ast_indicate(chan, -1);
}
break;
default:
break;
}
}
}
/*! \brief Helper function to handle when a timeout occurs on dialing attempt */
static int handle_timeout_trip(struct ast_dial *dial, struct timeval start)
{
struct ast_dial_channel *channel = NULL;
int diff = ast_tvdiff_ms(ast_tvnow(), start), lowest_timeout = -1, new_timeout = -1;
/* If there is no difference yet return the dial timeout so we can go again, we were likely interrupted */
if (!diff) {
return dial->timeout;
}
/* If the global dial timeout tripped switch the state to timeout so our channel loop will drop every channel */
if (diff >= dial->timeout) {
set_state(dial, AST_DIAL_RESULT_TIMEOUT);
new_timeout = 0;
}
/* Go through dropping out channels that have met their timeout */
AST_LIST_TRAVERSE(&dial->channels, channel, list) {
if (dial->state == AST_DIAL_RESULT_TIMEOUT || diff >= channel->timeout) {
ast_hangup(channel->owner);
channel->cause = AST_CAUSE_NO_ANSWER;
channel->owner = NULL;
} else if ((lowest_timeout == -1) || (lowest_timeout > channel->timeout)) {
lowest_timeout = channel->timeout;
}
}
/* Calculate the new timeout using the lowest timeout found */
if (lowest_timeout >= 0)
new_timeout = lowest_timeout - diff;
return new_timeout;
}
const char *ast_hangup_cause_to_dial_status(int hangup_cause)
{
switch(hangup_cause) {
case AST_CAUSE_BUSY:
return "BUSY";
case AST_CAUSE_CONGESTION:
return "CONGESTION";
case AST_CAUSE_NO_ROUTE_DESTINATION:
case AST_CAUSE_UNREGISTERED:
return "CHANUNAVAIL";
case AST_CAUSE_NO_ANSWER:
default:
return "NOANSWER";
}
}
/*! \brief Helper function that basically keeps tabs on dialing attempts */
static enum ast_dial_result monitor_dial(struct ast_dial *dial, struct ast_channel *chan)
{
int timeout = -1;
struct ast_channel *cs[AST_MAX_WATCHERS], *who = NULL;
struct ast_dial_channel *channel = NULL;
struct answer_exec_struct *answer_exec = NULL;
struct timeval start;
set_state(dial, AST_DIAL_RESULT_TRYING);
/* If the "always indicate ringing" option is set, change state to ringing and indicate to the owner if present */
if (dial->options[AST_DIAL_OPTION_RINGING]) {
set_state(dial, AST_DIAL_RESULT_RINGING);
if (chan)
ast_indicate(chan, AST_CONTROL_RINGING);
} else if (chan && dial->options[AST_DIAL_OPTION_MUSIC] &&
!ast_strlen_zero(dial->options[AST_DIAL_OPTION_MUSIC])) {
char *original_moh = ast_strdupa(ast_channel_musicclass(chan));
ast_indicate(chan, -1);
ast_channel_musicclass_set(chan, dial->options[AST_DIAL_OPTION_MUSIC]);
ast_moh_start(chan, dial->options[AST_DIAL_OPTION_MUSIC], NULL);
ast_channel_musicclass_set(chan, original_moh);
}
/* Record start time for timeout purposes */
start = ast_tvnow();
/* We actually figured out the maximum timeout we can do as they were added, so we can directly access the info */
timeout = dial->actual_timeout;
/* Go into an infinite loop while we are trying */
while ((dial->state != AST_DIAL_RESULT_UNANSWERED) && (dial->state != AST_DIAL_RESULT_ANSWERED) && (dial->state != AST_DIAL_RESULT_HANGUP) && (dial->state != AST_DIAL_RESULT_TIMEOUT)) {
int pos = 0, count = 0;
struct ast_frame *fr = NULL;
/* Set up channel structure array */
pos = count = 0;
if (chan)
cs[pos++] = chan;
/* Add channels we are attempting to dial */
AST_LIST_LOCK(&dial->channels);
AST_LIST_TRAVERSE(&dial->channels, channel, list) {
if (channel->owner) {
cs[pos++] = channel->owner;
count++;
}
}
AST_LIST_UNLOCK(&dial->channels);
/* If we have no outbound channels in progress, switch state to unanswered and stop */
if (!count) {
set_state(dial, AST_DIAL_RESULT_UNANSWERED);
break;
}
/* Just to be safe... */
if (dial->thread == AST_PTHREADT_STOP)
break;
/* Wait for frames from channels */
who = ast_waitfor_n(cs, pos, &timeout);
/* Check to see if our thread is being canceled */
if (dial->thread == AST_PTHREADT_STOP)
break;
/* If the timeout no longer exists OR if we got no channel it basically means the timeout was tripped, so handle it */
if (!timeout || !who) {
timeout = handle_timeout_trip(dial, start);
continue;
}
/* Find relative dial channel */
if (!chan || !IS_CALLER(chan, who))
channel = find_relative_dial_channel(dial, who);
/* See if this channel has been forwarded elsewhere */
if (!ast_strlen_zero(ast_channel_call_forward(who))) {
handle_call_forward(dial, channel, chan);
continue;
}
/* Attempt to read in a frame */
if (!(fr = ast_read(who))) {
/* If this is the caller then we switch state to hangup and stop */
if (chan && IS_CALLER(chan, who)) {
set_state(dial, AST_DIAL_RESULT_HANGUP);
break;
}
ast_channel_publish_dial(chan, who, channel->device, ast_hangup_cause_to_dial_status(ast_channel_hangupcause(who)));
ast_hangup(who);
channel->owner = NULL;
continue;
}
/* Process the frame */
handle_frame(dial, channel, fr, chan);
/* Free the received frame and start all over */
ast_frfree(fr);
}
/* Do post-processing from loop */
if (dial->state == AST_DIAL_RESULT_ANSWERED) {
/* Hangup everything except that which answered */
AST_LIST_LOCK(&dial->channels);
AST_LIST_TRAVERSE(&dial->channels, channel, list) {
if (!channel->owner || channel->owner == who)
continue;
ast_channel_publish_dial(chan, channel->owner, channel->device, "CANCEL");
ast_hangup(channel->owner);
channel->cause = AST_CAUSE_ANSWERED_ELSEWHERE;
channel->owner = NULL;
}
AST_LIST_UNLOCK(&dial->channels);
/* If ANSWER_EXEC is enabled as an option, execute application on answered channel */
if ((channel = find_relative_dial_channel(dial, who)) && (answer_exec = FIND_RELATIVE_OPTION(dial, channel, AST_DIAL_OPTION_ANSWER_EXEC))) {
channel->is_running_app = 1;
answer_exec_run(dial, channel, answer_exec->app, answer_exec->args);
channel->is_running_app = 0;
}
if (chan && dial->options[AST_DIAL_OPTION_MUSIC] &&
!ast_strlen_zero(dial->options[AST_DIAL_OPTION_MUSIC])) {
ast_moh_stop(chan);
}
} else if (dial->state == AST_DIAL_RESULT_HANGUP) {
/* Hangup everything */
AST_LIST_LOCK(&dial->channels);
AST_LIST_TRAVERSE(&dial->channels, channel, list) {
if (!channel->owner)
continue;
ast_channel_publish_dial(chan, channel->owner, channel->device, "CANCEL");
ast_hangup(channel->owner);
channel->cause = AST_CAUSE_NORMAL_CLEARING;
channel->owner = NULL;
}
AST_LIST_UNLOCK(&dial->channels);
}
2015-02-26 18:53:36 +00:00
if (dial->options[AST_DIAL_OPTION_SELF_DESTROY]) {
enum ast_dial_result state = dial->state;
ast_dial_destroy(dial);
return state;
}
return dial->state;
}
/*! \brief Dial async thread function */
static void *async_dial(void *data)
{
struct ast_dial *dial = data;
if (dial->callid) {
ast_callid_threadassoc_add(dial->callid);
}
/* This is really really simple... we basically pass monitor_dial a NULL owner and it changes it's behavior */
monitor_dial(dial, NULL);
return NULL;
}
/*! \brief Execute dialing synchronously or asynchronously
* \note Dials channels in a dial structure.
* \return Returns dial result code. (TRYING/INVALID/FAILED/ANSWERED/TIMEOUT/UNANSWERED).
*/
enum ast_dial_result ast_dial_run(struct ast_dial *dial, struct ast_channel *chan, int async)
{
enum ast_dial_result res = AST_DIAL_RESULT_TRYING;
/* Ensure required arguments are passed */
if (!dial) {
ast_debug(1, "invalid #1\n");
return AST_DIAL_RESULT_INVALID;
}
/* If there are no channels to dial we can't very well try to dial them */
if (AST_LIST_EMPTY(&dial->channels)) {
ast_debug(1, "invalid #2\n");
return AST_DIAL_RESULT_INVALID;
}
/* Dial each requested channel */
Update Asterisk's CDRs for the new bridging framework This patch is the initial push to update Asterisk's CDR engine for the new bridging framework. This patch guts the existing CDR engine and builds the new on top of messages coming across Stasis. As changes in channel state and bridge state are detected, CDRs are built and dispatched accordingly. This fundamentally changes CDRs in a few ways. (1) CDRs are now *very* reflective of the actual state of channels and bridges. This means CDRs track well with what an actual channel is doing - which is useful in transfer scenarios (which were previously difficult to pin down). It does, however, mean that CDRs cannot be 'fooled'. Previous behavior in Asterisk allowed for CDR applications, channels, and other properties to be spoofed in parts of the code - this no longer works. (2) CDRs have defined behavior in multi-party scenarios. This behavior will not be what everyone wants, but it is a defined behavior and as such, it is predictable. (3) The CDR manipulation functions and applications have been overhauled. Major changes have been made to ResetCDR and ForkCDR in particular. Many of the options for these two applications no longer made any sense with the new framework and the (slightly) more immutable nature of CDRs. There are a plethora of other changes. For a full description of CDR behavior, see the CDR specification on the Asterisk wiki. (closes issue ASTERISK-21196) Review: https://reviewboard.asterisk.org/r/2486/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@391947 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-06-17 03:00:38 +00:00
if (!begin_dial(dial, chan, async))
return AST_DIAL_RESULT_FAILED;
/* If we are running async spawn a thread and send it away... otherwise block here */
if (async) {
/* reference be released at dial destruction if it isn't NULL */
dial->callid = ast_read_threadstorage_callid();
dial->state = AST_DIAL_RESULT_TRYING;
/* Try to create a thread */
if (ast_pthread_create(&dial->thread, NULL, async_dial, dial)) {
/* Failed to create the thread - hangup all dialed channels and return failed */
ast_dial_hangup(dial);
res = AST_DIAL_RESULT_FAILED;
}
} else {
res = monitor_dial(dial, chan);
}
return res;
}
/*! \brief Return channel that answered
* \note Returns the Asterisk channel that answered
* \param dial Dialing structure
*/
struct ast_channel *ast_dial_answered(struct ast_dial *dial)
{
if (!dial)
return NULL;
return ((dial->state == AST_DIAL_RESULT_ANSWERED) ? AST_LIST_FIRST(&dial->channels)->owner : NULL);
}
/*! \brief Steal the channel that answered
* \note Returns the Asterisk channel that answered and removes it from the dialing structure
* \param dial Dialing structure
*/
struct ast_channel *ast_dial_answered_steal(struct ast_dial *dial)
{
struct ast_channel *chan = NULL;
if (!dial)
return NULL;
if (dial->state == AST_DIAL_RESULT_ANSWERED) {
chan = AST_LIST_FIRST(&dial->channels)->owner;
AST_LIST_FIRST(&dial->channels)->owner = NULL;
}
return chan;
}
/*! \brief Return state of dial
* \note Returns the state of the dial attempt
* \param dial Dialing structure
*/
enum ast_dial_result ast_dial_state(struct ast_dial *dial)
{
return dial->state;
}
/*! \brief Cancel async thread
* \note Cancel a running async thread
* \param dial Dialing structure
*/
enum ast_dial_result ast_dial_join(struct ast_dial *dial)
{
pthread_t thread;
/* If the dial structure is not running in async, return failed */
if (dial->thread == AST_PTHREADT_NULL)
return AST_DIAL_RESULT_FAILED;
/* Record thread */
thread = dial->thread;
/* Boom, commence locking */
ast_mutex_lock(&dial->lock);
/* Stop the thread */
dial->thread = AST_PTHREADT_STOP;
/* If the answered channel is running an application we have to soft hangup it, can't just poke the thread */
AST_LIST_LOCK(&dial->channels);
if (AST_LIST_FIRST(&dial->channels)->is_running_app) {
struct ast_channel *chan = AST_LIST_FIRST(&dial->channels)->owner;
if (chan) {
ast_channel_lock(chan);
ast_softhangup(chan, AST_SOFTHANGUP_EXPLICIT);
ast_channel_unlock(chan);
}
} else {
struct ast_dial_channel *channel = NULL;
/* Now we signal it with SIGURG so it will break out of it's waitfor */
pthread_kill(thread, SIGURG);
/* pthread_kill may not be enough, if outgoing channel has already got an answer (no more in waitfor) but is not yet running an application. Force soft hangup. */
AST_LIST_TRAVERSE(&dial->channels, channel, list) {
if (channel->owner) {
ast_softhangup(channel->owner, AST_SOFTHANGUP_EXPLICIT);
}
}
}
AST_LIST_UNLOCK(&dial->channels);
/* Yay done with it */
ast_mutex_unlock(&dial->lock);
/* Finally wait for the thread to exit */
pthread_join(thread, NULL);
/* Yay thread is all gone */
dial->thread = AST_PTHREADT_NULL;
return dial->state;
}
/*! \brief Hangup channels
* \note Hangup all active channels
* \param dial Dialing structure
*/
void ast_dial_hangup(struct ast_dial *dial)
{
struct ast_dial_channel *channel = NULL;
if (!dial)
return;
AST_LIST_LOCK(&dial->channels);
AST_LIST_TRAVERSE(&dial->channels, channel, list) {
ast_hangup(channel->owner);
channel->owner = NULL;
}
AST_LIST_UNLOCK(&dial->channels);
return;
}
int ast_dial_destroy(struct ast_dial *dial)
{
int i = 0;
struct ast_dial_channel *channel = NULL;
if (!dial)
return -1;
/* Hangup and deallocate all the dialed channels */
AST_LIST_LOCK(&dial->channels);
AST_LIST_TRAVERSE_SAFE_BEGIN(&dial->channels, channel, list) {
/* Disable any enabled options */
for (i = 0; i < AST_DIAL_OPTION_MAX; i++) {
if (!channel->options[i])
continue;
if (option_types[i].disable)
option_types[i].disable(channel->options[i]);
channel->options[i] = NULL;
}
/* Hang up channel if need be */
ast_hangup(channel->owner);
channel->owner = NULL;
/* Free structure */
ast_free(channel->tech);
ast_free(channel->device);
ast_free(channel->assignedid1);
ast_free(channel->assignedid2);
AST_LIST_REMOVE_CURRENT(list);
ast_free(channel);
}
AST_LIST_TRAVERSE_SAFE_END;
AST_LIST_UNLOCK(&dial->channels);
/* Disable any enabled options globally */
for (i = 0; i < AST_DIAL_OPTION_MAX; i++) {
if (!dial->options[i])
continue;
if (option_types[i].disable)
option_types[i].disable(dial->options[i]);
dial->options[i] = NULL;
}
/* Lock be gone! */
ast_mutex_destroy(&dial->lock);
/* Free structure */
ast_free(dial);
return 0;
}
int ast_dial_option_global_enable(struct ast_dial *dial, enum ast_dial_option option, void *data)
{
/* If the option is already enabled, return failure */
if (dial->options[option])
return -1;
/* Execute enable callback if it exists, if not simply make sure the value is set */
if (option_types[option].enable)
dial->options[option] = option_types[option].enable(data);
else
dial->options[option] = (void*)1;
return 0;
}
/*! \brief Helper function for finding a channel in a dial structure based on number
*/
static struct ast_dial_channel *find_dial_channel(struct ast_dial *dial, int num)
{
struct ast_dial_channel *channel = AST_LIST_LAST(&dial->channels);
/* We can try to predict programmer behavior, the last channel they added is probably the one they wanted to modify */
if (channel->num == num)
return channel;
/* Hrm not at the end... looking through the list it is! */
AST_LIST_LOCK(&dial->channels);
AST_LIST_TRAVERSE(&dial->channels, channel, list) {
if (channel->num == num)
break;
}
AST_LIST_UNLOCK(&dial->channels);
return channel;
}
int ast_dial_option_enable(struct ast_dial *dial, int num, enum ast_dial_option option, void *data)
{
struct ast_dial_channel *channel = NULL;
/* Ensure we have required arguments */
if (!dial || AST_LIST_EMPTY(&dial->channels))
return -1;
if (!(channel = find_dial_channel(dial, num)))
return -1;
/* If the option is already enabled, return failure */
if (channel->options[option])
return -1;
/* Execute enable callback if it exists, if not simply make sure the value is set */
if (option_types[option].enable)
channel->options[option] = option_types[option].enable(data);
else
channel->options[option] = (void*)1;
return 0;
}
int ast_dial_option_global_disable(struct ast_dial *dial, enum ast_dial_option option)
{
/* If the option is not enabled, return failure */
if (!dial->options[option]) {
return -1;
}
/* Execute callback of option to disable if it exists */
if (option_types[option].disable)
option_types[option].disable(dial->options[option]);
/* Finally disable option on the structure */
dial->options[option] = NULL;
return 0;
}
int ast_dial_option_disable(struct ast_dial *dial, int num, enum ast_dial_option option)
{
struct ast_dial_channel *channel = NULL;
/* Ensure we have required arguments */
if (!dial || AST_LIST_EMPTY(&dial->channels))
return -1;
if (!(channel = find_dial_channel(dial, num)))
return -1;
/* If the option is not enabled, return failure */
if (!channel->options[option])
return -1;
/* Execute callback of option to disable it if it exists */
if (option_types[option].disable)
option_types[option].disable(channel->options[option]);
/* Finally disable the option on the structure */
channel->options[option] = NULL;
return 0;
}
int ast_dial_reason(struct ast_dial *dial, int num)
{
struct ast_dial_channel *channel;
if (!dial || AST_LIST_EMPTY(&dial->channels) || !(channel = find_dial_channel(dial, num))) {
return -1;
}
return channel->cause;
}
struct ast_channel *ast_dial_get_channel(struct ast_dial *dial, int num)
{
struct ast_dial_channel *channel;
if (!dial || AST_LIST_EMPTY(&dial->channels) || !(channel = find_dial_channel(dial, num))) {
return NULL;
}
return channel->owner;
}
void ast_dial_set_state_callback(struct ast_dial *dial, ast_dial_state_callback callback)
{
dial->state_callback = callback;
}
void ast_dial_set_user_data(struct ast_dial *dial, void *user_data)
{
dial->user_data = user_data;
}
void *ast_dial_get_user_data(struct ast_dial *dial)
{
return dial->user_data;
}
void ast_dial_set_global_timeout(struct ast_dial *dial, int timeout)
{
dial->timeout = timeout;
if (dial->timeout > 0 && (dial->actual_timeout > dial->timeout || dial->actual_timeout == -1))
dial->actual_timeout = dial->timeout;
return;
}
void ast_dial_set_timeout(struct ast_dial *dial, int num, int timeout)
{
struct ast_dial_channel *channel = NULL;
if (!(channel = find_dial_channel(dial, num)))
return;
channel->timeout = timeout;
if (channel->timeout > 0 && (dial->actual_timeout > channel->timeout || dial->actual_timeout == -1))
dial->actual_timeout = channel->timeout;
return;
}