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asterisk/res/res_pjsip_messaging.c

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/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 2013, Digium, Inc.
*
* Kevin Harwell <kharwell@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.
*/
/*** MODULEINFO
<depend>pjproject</depend>
<depend>res_pjsip</depend>
<depend>res_pjsip_session</depend>
<support_level>core</support_level>
***/
/*** DOCUMENTATION
<info name="MessageDestinationInfo" language="en_US" tech="PJSIP">
<para>The <literal>destination</literal> parameter is used to construct
the Request URI for an outgoing message. It can be in one of the following
formats, all prefixed with the <literal>pjsip:</literal> message tech.</para>
<para>
</para>
<enumlist>
<enum name="endpoint">
<para>Request URI comes from the endpoint's default aor and contact.</para>
</enum>
<enum name="endpoint/aor">
<para>Request URI comes from the specific aor/contact.</para>
</enum>
<enum name="endpoint@domain">
<para>Request URI from the endpoint's default aor and contact. The domain is discarded.</para>
</enum>
</enumlist>
<para>
</para>
<para>These all use the endpoint to send the message with the specified URI:</para>
<para>
</para>
<enumlist>
<enum name="endpoint/&lt;sip[s]:host&gt;>"/>
<enum name="endpoint/&lt;sip[s]:user@host&gt;"/>
<enum name="endpoint/&quot;display name&quot; &lt;sip[s]:host&gt;"/>
<enum name="endpoint/&quot;display name&quot; &lt;sip[s]:user@host&gt;"/>
<enum name="endpoint/sip[s]:host"/>
<enum name="endpoint/sip[s]:user@host"/>
<enum name="endpoint/host"/>
<enum name="endpoint/user@host"/>
</enumlist>
<para>
</para>
<para>These all use the default endpoint to send the message with the specified URI:</para>
<para>
</para>
<enumlist>
<enum name="&lt;sip[s]:host&gt;"/>
<enum name="&lt;sip[s]:user@host&gt;"/>
<enum name="&quot;display name&quot; &lt;sip[s]:host&gt;"/>
<enum name="&quot;display name&quot; &lt;sip[s]:user@host&gt;"/>
<enum name="sip[s]:host"/>
<enum name="sip[s]:user@host"/>
</enumlist>
<para>
</para>
<para>These use the default endpoint to send the message with the specified host:</para>
<para>
</para>
<enumlist>
<enum name="host"/>
<enum name="user@host"/>
</enumlist>
<para>
</para>
<para>This form is similar to a dialstring:</para>
<para>
</para>
<enumlist>
<enum name="PJSIP/user@endpoint"/>
</enumlist>
<para>
</para>
<para>You still need to prefix the destination with
the <literal>pjsip:</literal> message technology prefix. For example:
<literal>pjsip:PJSIP/8005551212@myprovider</literal>.
The endpoint contact's URI will have the <literal>user</literal> inserted
into it and will become the Request URI. If the contact URI already has
a user specified, it will be replaced.
</para>
<para>
</para>
</info>
<info name="MessageFromInfo" language="en_US" tech="PJSIP">
<para>The <literal>from</literal> parameter is used to specity the <literal>From:</literal>
header in the outgoing SIP MESSAGE. It will override the value specified in
MESSAGE(from) which itself will override any <literal>from</literal> value from
an incoming SIP MESSAGE.
</para>
<para>
</para>
</info>
<info name="MessageToInfo" language="en_US" tech="PJSIP">
<para>The <literal>to</literal> parameter is used to specity the <literal>To:</literal>
header in the outgoing SIP MESSAGE. It will override the value specified in
MESSAGE(to) which itself will override any <literal>to</literal> value from
an incoming SIP MESSAGE.
</para>
<para>
</para>
</info>
***/
#include "asterisk.h"
#include <pjsip.h>
#include <pjsip_ua.h>
#include "asterisk/message.h"
#include "asterisk/module.h"
#include "asterisk/pbx.h"
#include "asterisk/res_pjsip.h"
#include "asterisk/res_pjsip_session.h"
#include "asterisk/taskprocessor.h"
#include "asterisk/test.h"
#include "asterisk/uri.h"
const pjsip_method pjsip_message_method = {PJSIP_OTHER_METHOD, {"MESSAGE", 7} };
#define MAX_HDR_SIZE 512
#define MAX_BODY_SIZE 1024
#define MAX_USER_SIZE 128
static struct ast_taskprocessor *message_serializer;
/*!
* \internal
* \brief Checks to make sure the request has the correct content type.
*
* \details This module supports the following media types: "text/plain".
* Return unsupported otherwise.
*
* \param rdata The SIP request
*/
static enum pjsip_status_code check_content_type(const pjsip_rx_data *rdata)
{
int res;
if (rdata->msg_info.msg->body && rdata->msg_info.msg->body->len) {
res = ast_sip_is_content_type(
&rdata->msg_info.msg->body->content_type, "text", "plain");
} else {
res = rdata->msg_info.ctype &&
ast_sip_is_content_type(
&rdata->msg_info.ctype->media, "text", "plain");
}
return res ? PJSIP_SC_OK : PJSIP_SC_UNSUPPORTED_MEDIA_TYPE;
}
/*!
* \internal
* \brief Checks to make sure the request has the correct content type.
*
* \details This module supports the following media types: "text/\*", "application/\*".
* Return unsupported otherwise.
*
* \param rdata The SIP request
*/
static enum pjsip_status_code check_content_type_in_dialog(const pjsip_rx_data *rdata)
{
int res = PJSIP_SC_UNSUPPORTED_MEDIA_TYPE;
static const pj_str_t text = { "text", 4};
static const pj_str_t application = { "application", 11};
if (!(rdata->msg_info.msg->body && rdata->msg_info.msg->body->len > 0)) {
return res;
}
/* We'll accept any text/ or application/ content type */
if (pj_stricmp(&rdata->msg_info.msg->body->content_type.type, &text) == 0
|| pj_stricmp(&rdata->msg_info.msg->body->content_type.type, &application) == 0) {
res = PJSIP_SC_OK;
} else if (rdata->msg_info.ctype
&& (pj_stricmp(&rdata->msg_info.ctype->media.type, &text) == 0
|| pj_stricmp(&rdata->msg_info.ctype->media.type, &application) == 0)) {
res = PJSIP_SC_OK;
}
return res;
}
/*!
* \brief Find a contact and insert a "user@" into its URI.
*
* \param to Original destination (for error messages only)
* \param endpoint_name Endpoint name (for error messages only)
* \param aors Command separated list of AORs
* \param user The user to insert in the contact URI
* \param uri Pointer to buffer in which to return the URI
*
* \return 0 Success
* \return -1 Fail
*
* \note If the contact URI found for the endpoint already has a user in
* its URI, it will be replaced.
*/
static int insert_user_in_contact_uri(const char *to, const char *endpoint_name, const char *aors,
const char *user, char **uri)
{
char *scheme = NULL;
char *contact_uri = NULL;
char *after_scheme = NULL;
char *host;
struct ast_sip_contact *contact = NULL;
contact = ast_sip_location_retrieve_contact_from_aor_list(aors);
if (!contact) {
/*
* We're getting the contact using the same method as
* ast_sip_create_request() so if there's no contact
* we can never send this message.
*/
ast_log(LOG_WARNING, "Dest: '%s' MSG SEND FAIL: Couldn't find contact for endpoint '%s'\n",
to, endpoint_name);
return -1;
}
contact_uri = ast_strdupa(contact->uri);
ao2_cleanup(contact);
ast_debug(3, "Dest: '%s' User: '%s' Endpoint: '%s' ContactURI: '%s'\n", to, user, endpoint_name, contact_uri);
/*
* Contact URIs must have a scheme so we must insert the user between it and the host.
*/
scheme = contact_uri;
after_scheme = strchr(contact_uri, ':');
if (!after_scheme) {
/* A contact URI without a scheme? Something's wrong. Bail */
ast_log(LOG_WARNING, "Dest: '%s' MSG SEND FAIL: There was no scheme in the contact URI '%s'\n",
to, contact_uri);
return -1;
}
/*
* Terminate the scheme.
*/
*after_scheme = '\0';
after_scheme++;
/*
* If the contact_uri already has a user, the host starts after the '@', otherwise
* the host is at after_scheme.
*
* We're going to ignore the existing user.
*/
host = strchr(after_scheme, '@');
if (host) {
host++;
} else {
host = after_scheme;
}
*uri = ast_malloc(strlen(scheme) + strlen(user) + strlen(host) + 3 /* One for the ':', '@' and terminating NULL */);
sprintf(*uri, "%s:%s@%s", scheme, user, host); /* Safe */
return 0;
}
/*!
* \internal
* \brief Get endpoint and URI when the destination is only a single token
*
* "to" could be one of the following:
* \verbatim
endpoint_name
hostname
* \endverbatim
*
* \param to Destination specified in MessageSend
* \param destination
* \param uri Pointer to URI variable. Must be freed by caller
* \return endpoint
*/
static struct ast_sip_endpoint *handle_single_token(const char *to, char *destination, char **uri) {
char *endpoint_name = NULL;
struct ast_sip_endpoint *endpoint = NULL;
struct ast_sip_contact *contact = NULL;
/*
* If "to" is just one token, it could be an endpoint name
* or a hostname without a scheme.
*/
endpoint = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "endpoint", destination);
if (!endpoint) {
/*
* We can only assume it's a hostname.
*/
char *temp_uri = ast_malloc(strlen(destination) + strlen("sip:") + 1);
sprintf(temp_uri, "sip:%s", destination);
*uri = temp_uri;
endpoint = ast_sip_default_outbound_endpoint();
ast_debug(3, "Dest: '%s' Didn't find endpoint so adding scheme and using URI '%s' with default endpoint\n",
to, *uri);
return endpoint;
}
/*
* It's an endpoint
*/
endpoint_name = destination;
contact = ast_sip_location_retrieve_contact_from_aor_list(endpoint->aors);
if (!contact) {
/*
* We're getting the contact using the same method as
* ast_sip_create_request() so if there's no contact
* we can never send this message.
*/
ast_log(LOG_WARNING, "Dest: '%s' MSG SEND FAIL: Found endpoint '%s' but didn't find an aor/contact for it\n",
to, endpoint_name);
ao2_cleanup(endpoint);
*uri = NULL;
return NULL;
}
*uri = ast_strdup(contact->uri);
ast_debug(3, "Dest: '%s' Found endpoint '%s' and found contact with URI '%s'\n",
to, endpoint_name, *uri);
ao2_cleanup(contact);
return endpoint;
}
/*!
* \internal
* \brief Get endpoint and URI when the destination contained a '/'
*
* "to" could be one of the following:
* \verbatim
endpoint/aor
endpoint/<sip[s]:host>
endpoint/<sip[s]:user@host>
endpoint/"Bob" <sip[s]:host>
endpoint/"Bob" <sip[s]:user@host>
endpoint/sip[s]:host
endpoint/sip[s]:user@host
endpoint/host
endpoint/user@host
* \endverbatim
*
* \param to Destination specified in MessageSend
* \param uri Pointer to URI variable. Must be freed by caller
* \param destination, slash, atsign, scheme
* \return endpoint
*/
static struct ast_sip_endpoint *handle_slash(const char *to, char *destination, char **uri,
char *slash, char *atsign, char *scheme)
{
char *endpoint_name = NULL;
struct ast_sip_endpoint *endpoint = NULL;
struct ast_sip_contact *contact = NULL;
char *user = NULL;
char *afterslash = slash + 1;
struct ast_sip_aor *aor;
if (ast_begins_with(destination, "PJSIP/")) {
ast_debug(3, "Dest: '%s' Dialplan format'\n", to);
/*
* This has to be the form PJSIP/user@endpoint
*/
if (!atsign || strchr(afterslash, '/')) {
/*
* If there's no "user@" or there's a slash somewhere after
* "PJSIP/" then we go no further.
*/
*uri = NULL;
ast_log(LOG_WARNING,
"Dest: '%s' MSG SEND FAIL: Destinations beginning with 'PJSIP/' must be in the form of 'PJSIP/user@endpoint'\n",
to);
return NULL;
}
*atsign = '\0';
user = afterslash;
endpoint_name = atsign + 1;
ast_debug(3, "Dest: '%s' User: '%s' Endpoint: '%s'\n", to, user, endpoint_name);
} else {
/*
* Either...
* endpoint/aor
* endpoint/uri
*/
*slash = '\0';
endpoint_name = destination;
ast_debug(3, "Dest: '%s' Endpoint: '%s'\n", to, endpoint_name);
}
endpoint = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "endpoint", endpoint_name);
if (!endpoint) {
*uri = NULL;
ast_log(LOG_WARNING, "Dest: '%s' MSG SEND FAIL: Didn't find endpoint with name '%s'\n",
to, endpoint_name);
return NULL;
}
if (scheme) {
/*
* If we found a scheme, then everything after the slash MUST be a URI.
* We don't need to do any further modification.
*/
*uri = ast_strdup(afterslash);
ast_debug(3, "Dest: '%s' Found endpoint '%s' and found URI '%s' after '/'\n",
to, endpoint_name, *uri);
return endpoint;
}
if (user) {
/*
* This has to be the form PJSIP/user@endpoint
*/
int rc;
/*
* Set the return URI to be the endpoint's contact URI with the user
* portion set to the user that was specified before the endpoint name.
*/
rc = insert_user_in_contact_uri(to, endpoint_name, endpoint->aors, user, uri);
if (rc != 0) {
/*
* insert_user_in_contact_uri prints the warning message.
*/
ao2_cleanup(endpoint);
endpoint = NULL;
*uri = NULL;
}
ast_debug(3, "Dest: '%s' User: '%s' Endpoint: '%s' URI: '%s'\n", to, user,
endpoint_name, *uri);
return endpoint;
}
/*
* We're now left with two possibilities...
* endpoint/aor
* endpoint/uri-without-scheme
*/
aor = ast_sip_location_retrieve_aor(afterslash);
if (!aor) {
/*
* It's probably a URI without a scheme but we don't have a way to tell
* for sure. We're going to assume it is and prepend it with a scheme.
*/
*uri = ast_malloc(strlen(afterslash) + strlen("sip:") + 1);
sprintf(*uri, "sip:%s", afterslash);
ast_debug(3, "Dest: '%s' Found endpoint '%s' but didn't find aor after '/' so using URI '%s'\n",
to, endpoint_name, *uri);
return endpoint;
}
/*
* Only one possibility left... There was an aor name after the slash.
*/
ast_debug(3, "Dest: '%s' Found endpoint '%s' and found aor '%s' after '/'\n",
to, endpoint_name, ast_sorcery_object_get_id(aor));
contact = ast_sip_location_retrieve_first_aor_contact(aor);
if (!contact) {
/*
* An aor without a contact is useless and since
* ast_sip_create_message() won't be able to find one
* either, we just need to bail.
*/
ast_log(LOG_WARNING, "Dest: '%s' MSG SEND FAIL: Found endpoint '%s' but didn't find contact for aor '%s'\n",
to, endpoint_name, ast_sorcery_object_get_id(aor));
ao2_cleanup(aor);
ao2_cleanup(endpoint);
*uri = NULL;
return NULL;
}
*uri = ast_strdup(contact->uri);
ast_debug(3, "Dest: '%s' Found endpoint '%s' and found contact with URI '%s' for aor '%s'\n",
to, endpoint_name, *uri, ast_sorcery_object_get_id(aor));
ao2_cleanup(contact);
ao2_cleanup(aor);
return endpoint;
}
/*!
* \internal
* \brief Get endpoint and URI when the destination contained a '\@' but no '/' or scheme
*
* "to" could be one of the following:
* \verbatim
<sip[s]:user@host>
"Bob" <sip[s]:user@host>
sip[s]:user@host
user@host
* \endverbatim
*
* \param to Destination specified in MessageSend
* \param uri Pointer to URI variable. Must be freed by caller
* \param destination, slash, atsign, scheme
* \return endpoint
*/
static struct ast_sip_endpoint *handle_atsign(const char *to, char *destination, char **uri,
char *slash, char *atsign, char *scheme)
{
char *endpoint_name = NULL;
struct ast_sip_endpoint *endpoint = NULL;
struct ast_sip_contact *contact = NULL;
char *afterat = atsign + 1;
*atsign = '\0';
endpoint_name = destination;
/* Apparently there may be ';<user_options>' after the endpoint name ??? */
AST_SIP_USER_OPTIONS_TRUNCATE_CHECK(endpoint_name);
endpoint = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "endpoint", endpoint_name);
if (!endpoint) {
/*
* It's probably a uri with a user but without a scheme but we don't have a way to tell.
* We're going to assume it is and prepend it with a scheme.
*/
*uri = ast_malloc(strlen(to) + strlen("sip:") + 1);
sprintf(*uri, "sip:%s", to);
endpoint = ast_sip_default_outbound_endpoint();
ast_debug(3, "Dest: '%s' Didn't find endpoint before the '@' so using URI '%s' with default endpoint\n",
to, *uri);
return endpoint;
}
/*
* OK, it's an endpoint and a domain (which we ignore)
*/
contact = ast_sip_location_retrieve_contact_from_aor_list(endpoint->aors);
if (!contact) {
/*
* We're getting the contact using the same method as
* ast_sip_create_request() so if there's no contact
* we can never send this message.
*/
ao2_cleanup(endpoint);
endpoint = NULL;
*uri = NULL;
ast_log(LOG_WARNING, "Dest: '%s' MSG SEND FAIL: Found endpoint '%s' but didn't find contact\n",
to, endpoint_name);
return NULL;
}
*uri = ast_strdup(contact->uri);
ao2_cleanup(contact);
ast_debug(3, "Dest: '%s' Found endpoint '%s' and found contact with URI '%s' (discarding domain %s)\n",
to, endpoint_name, *uri, afterat);
return endpoint;
}
/*!
* \internal
* \brief Retrieves an endpoint and URI from the "to" string.
*
* This URI is used as the Request URI.
*
* Expects the given 'to' to be in one of the following formats:
* Why we allow so many is a mystery.
*
* Basic:
*
* endpoint : We'll get URI from the default aor/contact
* endpoint/aor : We'll get the URI from the specific aor/contact
* endpoint@domain : We toss the domain part and just use the endpoint
*
* These all use the endpoint and specified URI:
* \verbatim
endpoint/<sip[s]:host>
endpoint/<sip[s]:user@host>
endpoint/"Bob" <sip[s]:host>
endpoint/"Bob" <sip[s]:user@host>
endpoint/sip[s]:host
endpoint/sip[s]:user@host
endpoint/host
endpoint/user@host
\endverbatim
*
* These all use the default endpoint and specified URI:
* \verbatim
<sip[s]:host>
<sip[s]:user@host>
"Bob" <sip[s]:host>
"Bob" <sip[s]:user@host>
sip[s]:host
sip[s]:user@host
\endverbatim
*
* These use the default endpoint and specified host:
* \verbatim
host
user@host
\endverbatim
*
* This form is similar to a dialstring:
* \verbatim
PJSIP/user@endpoint
\endverbatim
*
* In this case, the user will be added to the endpoint contact's URI.
* If the contact URI already has a user, it will be replaced.
*
* The ones that have the sip[s] scheme are the easiest to parse.
* The rest all have some issue.
*
* endpoint vs host : We have to test for endpoint first
* endpoint/aor vs endpoint/host : We have to test for aor first
* What if there's an aor with the same
* name as the host?
* endpoint@domain vs user@host : We have to test for endpoint first.
* What if there's an endpoint with the
* same name as the user?
*
* \param to 'To' field with possible endpoint
* \param uri Pointer to a char* which will be set to the URI.
* Must be ast_free'd by the caller.
*
* \note The logic below could probably be condensed but then it wouldn't be
* as clear.
*/
static struct ast_sip_endpoint *get_outbound_endpoint(const char *to, char **uri)
{
char *destination;
char *slash = NULL;
char *atsign = NULL;
char *scheme = NULL;
struct ast_sip_endpoint *endpoint = NULL;
destination = ast_strdupa(to);
slash = strchr(destination, '/');
atsign = strchr(destination, '@');
scheme = S_OR(strstr(destination, "sip:"), strstr(destination, "sips:"));
if (!slash && !atsign && !scheme) {
/*
* If there's only a single token, it can be either...
* endpoint
* host
*/
return handle_single_token(to, destination, uri);
}
if (slash) {
/*
* If there's a '/', then the form must be one of the following...
* PJSIP/user@endpoint
* endpoint/aor
* endpoint/uri
*/
return handle_slash(to, destination, uri, slash, atsign, scheme);
}
if (!endpoint && atsign && !scheme) {
/*
* If there's an '@' but no scheme then it's either following an endpoint name
* and being followed by a domain name (which we discard).
* OR is's a user@host uri without a scheme. It's probably the latter but because
* endpoint@domain looks just like user@host, we'll test for endpoint first.
*/
return handle_atsign(to, destination, uri, slash, atsign, scheme);
}
/*
* If all else fails, we assume it's a URI or just a hostname.
*/
if (scheme) {
*uri = ast_strdup(destination);
ast_debug(3, "Dest: '%s' Didn't find an endpoint but did find a scheme so using URI '%s' with default endpoint\n",
to, *uri);
} else {
*uri = ast_malloc(strlen(destination) + strlen("sip:") + 1);
sprintf(*uri, "sip:%s", destination);
ast_debug(3, "Dest: '%s' Didn't find an endpoint and didn't find scheme so adding scheme and using URI '%s' with default endpoint\n",
to, *uri);
}
endpoint = ast_sip_default_outbound_endpoint();
return endpoint;
}
/*!
* \internal
* \brief Replace the To URI in the tdata with the supplied one
*
* \param tdata the outbound message data structure
* \param to URI to replace the To URI with
*
* \return 0: success, -1: failure
*/
static int update_to_uri(pjsip_tx_data *tdata, char *to)
{
pjsip_name_addr *parsed_name_addr;
pjsip_sip_uri *sip_uri;
pjsip_name_addr *tdata_name_addr;
pjsip_sip_uri *tdata_sip_uri;
char *buf = NULL;
#define DEBUG_BUF_SIZE 256
parsed_name_addr = (pjsip_name_addr *) pjsip_parse_uri(tdata->pool, to, strlen(to),
PJSIP_PARSE_URI_AS_NAMEADDR);
if (!parsed_name_addr || (!PJSIP_URI_SCHEME_IS_SIP(parsed_name_addr->uri)
&& !PJSIP_URI_SCHEME_IS_SIPS(parsed_name_addr->uri))) {
ast_log(LOG_WARNING, "To address '%s' is not a valid SIP/SIPS URI\n", to);
return -1;
}
sip_uri = pjsip_uri_get_uri(parsed_name_addr->uri);
if (DEBUG_ATLEAST(3)) {
buf = ast_alloca(DEBUG_BUF_SIZE);
pjsip_uri_print(PJSIP_URI_IN_FROMTO_HDR, sip_uri, buf, DEBUG_BUF_SIZE);
ast_debug(3, "Parsed To: %.*s %s\n", (int)parsed_name_addr->display.slen,
parsed_name_addr->display.ptr, buf);
}
tdata_name_addr = (pjsip_name_addr *) PJSIP_MSG_TO_HDR(tdata->msg)->uri;
if (!tdata_name_addr || (!PJSIP_URI_SCHEME_IS_SIP(tdata_name_addr->uri)
&& !PJSIP_URI_SCHEME_IS_SIPS(tdata_name_addr->uri))) {
/* Highly unlikely but we have to check */
ast_log(LOG_WARNING, "tdata To address '%s' is not a valid SIP/SIPS URI\n", to);
return -1;
}
tdata_sip_uri = pjsip_uri_get_uri(tdata_name_addr->uri);
if (DEBUG_ATLEAST(3)) {
buf[0] = '\0';
pjsip_uri_print(PJSIP_URI_IN_FROMTO_HDR, tdata_sip_uri, buf, DEBUG_BUF_SIZE);
ast_debug(3, "Original tdata To: %.*s %s\n", (int)tdata_name_addr->display.slen,
tdata_name_addr->display.ptr, buf);
}
/* Replace the uri */
pjsip_sip_uri_assign(tdata->pool, tdata_sip_uri, sip_uri);
/* The display name isn't part of the URI so we need to replace it separately */
pj_strdup(tdata->pool, &tdata_name_addr->display, &parsed_name_addr->display);
if (DEBUG_ATLEAST(3)) {
buf[0] = '\0';
pjsip_uri_print(PJSIP_URI_IN_FROMTO_HDR, tdata_sip_uri, buf, 256);
ast_debug(3, "New tdata To: %.*s %s\n", (int)tdata_name_addr->display.slen,
tdata_name_addr->display.ptr, buf);
}
return 0;
#undef DEBUG_BUF_SIZE
}
/*!
* \internal
* \brief Update the display name in the To uri in the tdata with the one from the supplied uri
*
* \param tdata the outbound message data structure
* \param to uri containing the display name to replace in the the To uri
*
* \return 0: success, -1: failure
*/
static int update_to_display_name(pjsip_tx_data *tdata, char *to)
{
pjsip_name_addr *parsed_name_addr;
parsed_name_addr = (pjsip_name_addr *) pjsip_parse_uri(tdata->pool, to, strlen(to),
PJSIP_PARSE_URI_AS_NAMEADDR);
if (parsed_name_addr) {
if (pj_strlen(&parsed_name_addr->display)) {
pjsip_name_addr *name_addr =
(pjsip_name_addr *) PJSIP_MSG_TO_HDR(tdata->msg)->uri;
pj_strdup(tdata->pool, &name_addr->display, &parsed_name_addr->display);
}
return 0;
}
return -1;
}
/*!
* \internal
* \brief Overwrite fields in the outbound 'From' header
*
* The outbound 'From' header is created/added in ast_sip_create_request with
* default data. If available that data may be info specified in the 'from_user'
* and 'from_domain' options found on the endpoint. That information will be
* overwritten with data in the given 'from' parameter.
*
* \param tdata the outbound message data structure
* \param from info to copy into the header
*
* \return 0: success, -1: failure
*/
static int update_from(pjsip_tx_data *tdata, char *from)
{
pjsip_name_addr *name_addr;
pjsip_sip_uri *uri;
pjsip_name_addr *parsed_name_addr;
if (ast_strlen_zero(from)) {
return 0;
}
name_addr = (pjsip_name_addr *) PJSIP_MSG_FROM_HDR(tdata->msg)->uri;
uri = pjsip_uri_get_uri(name_addr);
parsed_name_addr = (pjsip_name_addr *) pjsip_parse_uri(tdata->pool, from,
strlen(from), PJSIP_PARSE_URI_AS_NAMEADDR);
if (parsed_name_addr) {
pjsip_sip_uri *parsed_uri;
if (!PJSIP_URI_SCHEME_IS_SIP(parsed_name_addr->uri)
&& !PJSIP_URI_SCHEME_IS_SIPS(parsed_name_addr->uri)) {
ast_log(LOG_WARNING, "From address '%s' is not a valid SIP/SIPS URI\n", from);
return -1;
}
parsed_uri = pjsip_uri_get_uri(parsed_name_addr->uri);
if (pj_strlen(&parsed_name_addr->display)) {
pj_strdup(tdata->pool, &name_addr->display, &parsed_name_addr->display);
}
/* Unlike the To header, we only want to replace the user, host and port */
pj_strdup(tdata->pool, &uri->user, &parsed_uri->user);
pj_strdup(tdata->pool, &uri->host, &parsed_uri->host);
uri->port = parsed_uri->port;
return 0;
} else {
/* assume it is 'user[@domain]' format */
char *domain = strchr(from, '@');
if (domain) {
pj_str_t pj_from;
pj_strset3(&pj_from, from, domain);
pj_strdup(tdata->pool, &uri->user, &pj_from);
pj_strdup2(tdata->pool, &uri->host, domain + 1);
} else {
pj_strdup2(tdata->pool, &uri->user, from);
}
return 0;
}
return -1;
}
/*!
* \internal
* \brief Checks if the given msg var name should be blocked.
*
* \details Some headers are not allowed to be overriden by the user.
* Determine if the given var header name from the user is blocked for
* an outgoing MESSAGE.
*
* \param name name of header to see if it is blocked.
*
* \retval TRUE if the given header is blocked.
*/
static int is_msg_var_blocked(const char *name)
{
int i;
/* Don't block the Max-Forwards header because the user can override it */
static const char *hdr[] = {
"To",
"From",
"Via",
"Route",
"Contact",
"Call-ID",
"CSeq",
"Allow",
"Content-Length",
"Content-Type",
"Request-URI",
};
for (i = 0; i < ARRAY_LEN(hdr); ++i) {
if (!strcasecmp(name, hdr[i])) {
/* Block addition of this header. */
return 1;
}
}
return 0;
}
/*!
* \internal
* \brief Copies any other msg vars over to the request headers.
*
* \param msg The msg structure to copy headers from
* \param tdata The SIP transmission data
*/
static enum pjsip_status_code vars_to_headers(const struct ast_msg *msg, pjsip_tx_data *tdata)
{
const char *name;
const char *value;
int max_forwards;
struct ast_msg_var_iterator *iter;
for (iter = ast_msg_var_iterator_init(msg);
ast_msg_var_iterator_next(msg, iter, &name, &value);
ast_msg_var_unref_current(iter)) {
if (!strcasecmp(name, "Max-Forwards")) {
/* Decrement Max-Forwards for SIP loop prevention. */
if (sscanf(value, "%30d", &max_forwards) != 1 || --max_forwards == 0) {
ast_msg_var_iterator_destroy(iter);
ast_log(LOG_NOTICE, "MESSAGE(Max-Forwards) reached zero. MESSAGE not sent.\n");
return -1;
}
sprintf((char *) value, "%d", max_forwards);
ast_sip_add_header(tdata, name, value);
} else if (!is_msg_var_blocked(name)) {
ast_sip_add_header(tdata, name, value);
}
}
ast_msg_var_iterator_destroy(iter);
return PJSIP_SC_OK;
}
/*!
* \internal
* \brief Copies any other request header data over to ast_msg structure.
*
* \param rdata The SIP request
* \param msg The msg structure to copy headers into
*/
static int headers_to_vars(const pjsip_rx_data *rdata, struct ast_msg *msg)
{
char *c;
char name[MAX_HDR_SIZE];
char buf[MAX_HDR_SIZE];
int res = 0;
pjsip_hdr *h = rdata->msg_info.msg->hdr.next;
pjsip_hdr *end= &rdata->msg_info.msg->hdr;
while (h != end) {
if ((res = pjsip_hdr_print_on(h, buf, sizeof(buf)-1)) > 0) {
buf[res] = '\0';
if ((c = strchr(buf, ':'))) {
ast_copy_string(buf, ast_skip_blanks(c + 1), sizeof(buf));
}
ast_copy_pj_str(name, &h->name, sizeof(name));
if ((res = ast_msg_set_var(msg, name, buf)) != 0) {
break;
}
}
h = h->next;
}
return 0;
}
/*!
* \internal
* \brief Prints the message body into the given char buffer.
*
* \details Copies body content from the received data into the given
* character buffer removing any extra carriage return/line feeds.
*
* \param rdata The SIP request
* \param buf Buffer to fill
* \param len The length of the buffer
*/
static int print_body(pjsip_rx_data *rdata, char *buf, int len)
{
int res;
if (!rdata->msg_info.msg->body || !rdata->msg_info.msg->body->len) {
return 0;
}
if ((res = rdata->msg_info.msg->body->print_body(
rdata->msg_info.msg->body, buf, len)) < 0) {
return res;
}
/* remove any trailing carriage return/line feeds */
while (res > 0 && ((buf[--res] == '\r') || (buf[res] == '\n')));
buf[++res] = '\0';
return res;
}
/*!
* \internal
* \brief Converts a 'sip:' uri to a 'pjsip:' so it can be found by
* the message tech.
*
* \param buf uri to insert 'pjsip' into
* \param size length of the uri in buf
* \param capacity total size of buf
*/
static char *sip_to_pjsip(char *buf, int size, int capacity)
{
int count;
const char *scheme;
char *res = buf;
/* remove any wrapping brackets */
if (*buf == '<') {
++buf;
--size;
}
scheme = strncmp(buf, "sip", 3) ? "pjsip:" : "pj";
count = strlen(scheme);
if (count + size >= capacity) {
ast_log(LOG_WARNING, "Unable to handle MESSAGE- incoming uri "
"too large for given buffer\n");
return NULL;
}
memmove(res + count, buf, size);
memcpy(res, scheme, count);
buf += size - 1;
if (*buf == '>') {
*buf = '\0';
}
return res;
}
/*!
* \internal
* \brief Converts a pjsip_rx_data structure to an ast_msg structure.
*
* \details Attempts to fill in as much information as possible into the given
* msg structure copied from the given request data.
*
* \param rdata The SIP request
* \param msg The asterisk message structure to fill in.
*/
static enum pjsip_status_code rx_data_to_ast_msg(pjsip_rx_data *rdata, struct ast_msg *msg)
{
RAII_VAR(struct ast_sip_endpoint *, endpt, NULL, ao2_cleanup);
pjsip_uri *ruri = rdata->msg_info.msg->line.req.uri;
pjsip_sip_uri *sip_ruri;
pjsip_name_addr *name_addr;
char buf[MAX_BODY_SIZE];
const char *field;
const char *context;
char exten[AST_MAX_EXTENSION];
int res = 0;
int size;
if (!PJSIP_URI_SCHEME_IS_SIP(ruri) && !PJSIP_URI_SCHEME_IS_SIPS(ruri)) {
return PJSIP_SC_UNSUPPORTED_URI_SCHEME;
}
sip_ruri = pjsip_uri_get_uri(ruri);
ast_copy_pj_str(exten, &sip_ruri->user, AST_MAX_EXTENSION);
/*
* We may want to match in the dialplan without any user
* options getting in the way.
*/
AST_SIP_USER_OPTIONS_TRUNCATE_CHECK(exten);
endpt = ast_pjsip_rdata_get_endpoint(rdata);
ast_assert(endpt != NULL);
context = S_OR(endpt->message_context, endpt->context);
res |= ast_msg_set_context(msg, "%s", context);
res |= ast_msg_set_exten(msg, "%s", exten);
/* to header */
name_addr = (pjsip_name_addr *)rdata->msg_info.to->uri;
size = pjsip_uri_print(PJSIP_URI_IN_FROMTO_HDR, name_addr, buf, sizeof(buf) - 1);
if (size <= 0) {
return PJSIP_SC_INTERNAL_SERVER_ERROR;
}
buf[size] = '\0';
res |= ast_msg_set_to(msg, "%s", sip_to_pjsip(buf, ++size, sizeof(buf) - 1));
/* from header */
name_addr = (pjsip_name_addr *)rdata->msg_info.from->uri;
size = pjsip_uri_print(PJSIP_URI_IN_FROMTO_HDR, name_addr, buf, sizeof(buf) - 1);
if (size <= 0) {
return PJSIP_SC_INTERNAL_SERVER_ERROR;
}
buf[size] = '\0';
res |= ast_msg_set_from(msg, "%s", buf);
field = pj_sockaddr_print(&rdata->pkt_info.src_addr, buf, sizeof(buf) - 1, 3);
res |= ast_msg_set_var(msg, "PJSIP_RECVADDR", field);
switch (rdata->tp_info.transport->key.type) {
case PJSIP_TRANSPORT_UDP:
case PJSIP_TRANSPORT_UDP6:
field = "udp";
break;
case PJSIP_TRANSPORT_TCP:
case PJSIP_TRANSPORT_TCP6:
field = "tcp";
break;
case PJSIP_TRANSPORT_TLS:
case PJSIP_TRANSPORT_TLS6:
field = "tls";
break;
default:
field = rdata->tp_info.transport->type_name;
}
ast_msg_set_var(msg, "PJSIP_TRANSPORT", field);
if (print_body(rdata, buf, sizeof(buf) - 1) > 0) {
res |= ast_msg_set_body(msg, "%s", buf);
}
/* endpoint name */
res |= ast_msg_set_tech(msg, "%s", "PJSIP");
res |= ast_msg_set_endpoint(msg, "%s", ast_sorcery_object_get_id(endpt));
if (endpt->id.self.name.valid) {
res |= ast_msg_set_var(msg, "PJSIP_ENDPOINT", endpt->id.self.name.str);
}
res |= headers_to_vars(rdata, msg);
return !res ? PJSIP_SC_OK : PJSIP_SC_INTERNAL_SERVER_ERROR;
}
struct msg_data {
struct ast_msg *msg;
char *destination;
char *from;
};
static void msg_data_destroy(void *obj)
{
struct msg_data *mdata = obj;
ast_free(mdata->from);
ast_free(mdata->destination);
ast_msg_destroy(mdata->msg);
}
static struct msg_data *msg_data_create(const struct ast_msg *msg, const char *destination, const char *from)
{
char *uri_params;
struct msg_data *mdata = ao2_alloc(sizeof(*mdata), msg_data_destroy);
if (!mdata) {
return NULL;
}
/* typecast to suppress const warning */
mdata->msg = ast_msg_ref((struct ast_msg *) msg);
/* To starts with 'pjsip:' which needs to be removed. */
if (!(destination = strchr(destination, ':'))) {
ao2_ref(mdata, -1);
return NULL;
}
++destination;/* Now skip the ':' */
mdata->destination = ast_strdup(destination);
mdata->from = ast_strdup(from);
/*
* Sometimes from URI can contain URI parameters, so remove them.
*
* sip:user;user-options@domain;uri-parameters
*/
uri_params = strchr(mdata->from, '@');
if (uri_params && (uri_params = strchr(mdata->from, ';'))) {
*uri_params = '\0';
}
return mdata;
}
static void update_content_type(pjsip_tx_data *tdata, struct ast_msg *msg, struct ast_sip_body *body)
{
static const pj_str_t CONTENT_TYPE = { "Content-Type", sizeof("Content-Type") - 1 };
const char *content_type = ast_msg_get_var(msg, pj_strbuf(&CONTENT_TYPE));
if (content_type) {
pj_str_t type, subtype;
pjsip_ctype_hdr *parsed;
/* Let pjsip do the parsing for us */
parsed = pjsip_parse_hdr(tdata->pool, &CONTENT_TYPE,
ast_strdupa(content_type), strlen(content_type),
NULL);
if (!parsed) {
ast_log(LOG_WARNING, "Failed to parse '%s' as a content type. Using text/plain\n",
content_type);
return;
}
/* We need to turn type and subtype into zero-terminated strings */
pj_strdup_with_null(tdata->pool, &type, &parsed->media.type);
pj_strdup_with_null(tdata->pool, &subtype, &parsed->media.subtype);
body->type = pj_strbuf(&type);
body->subtype = pj_strbuf(&subtype);
}
}
/*!
* \internal
* \brief Send a MESSAGE
*
* \param data The outbound message data structure
*
* \return 0: success, -1: failure
*
* mdata contains the To and From specified in the call to the MessageSend
* dialplan app. It also contains the ast_msg object that contains the
* message body and may contain the To and From from the channel datastore,
* usually set with the MESSAGE or MESSAGE_DATA dialplan functions but
* could also come from an incoming sip MESSAGE.
*
* The mdata->to is always used as the basis for the Request URI
* while the mdata->msg->to is used for the To header. If
* mdata->msg->to isn't available, mdata->to is used for the To header.
*
*/
static int msg_send(void *data)
{
struct msg_data *mdata = data; /* The caller holds a reference */
struct ast_sip_body body = {
.type = "text",
.subtype = "plain",
.body_text = ast_msg_get_body(mdata->msg)
};
pjsip_tx_data *tdata;
RAII_VAR(char *, uri, NULL, ast_free);
RAII_VAR(struct ast_sip_endpoint *, endpoint, NULL, ao2_cleanup);
ast_debug(3, "mdata From: %s msg From: %s mdata Destination: %s msg To: %s\n",
mdata->from, ast_msg_get_from(mdata->msg), mdata->destination, ast_msg_get_to(mdata->msg));
endpoint = get_outbound_endpoint(mdata->destination, &uri);
if (!endpoint) {
ast_log(LOG_ERROR,
"PJSIP MESSAGE - Could not find endpoint '%s' and no default outbound endpoint configured\n",
mdata->destination);
ast_test_suite_event_notify("MSG_ENDPOINT_URI_FAIL",
"MdataFrom: %s\r\n"
"MsgFrom: %s\r\n"
"MdataDestination: %s\r\n"
"MsgTo: %s\r\n",
mdata->from,
ast_msg_get_from(mdata->msg),
mdata->destination,
ast_msg_get_to(mdata->msg));
return -1;
}
ast_debug(3, "Request URI: %s\n", uri);
if (ast_sip_create_request("MESSAGE", NULL, endpoint, uri, NULL, &tdata)) {
ast_log(LOG_WARNING, "PJSIP MESSAGE - Could not create request\n");
return -1;
}
/* If there was a To in the actual message, */
if (!ast_strlen_zero(ast_msg_get_to(mdata->msg))) {
char *msg_to = ast_strdupa(ast_msg_get_to(mdata->msg));
/*
* It's possible that the message To was copied from
* an incoming MESSAGE in which case it'll have the
* pjsip: tech prepended to it. We need to remove it.
*/
if (ast_begins_with(msg_to, "pjsip:")) {
msg_to += 6;
}
update_to_uri(tdata, msg_to);
} else {
/*
* If there was no To in the message, it's still possible
* that there is a display name in the mdata To. If so,
* we'll copy the URI display name to the tdata To.
*/
update_to_display_name(tdata, uri);
}
if (!ast_strlen_zero(mdata->from)) {
update_from(tdata, mdata->from);
} else if (!ast_strlen_zero(ast_msg_get_from(mdata->msg))) {
update_from(tdata, (char *)ast_msg_get_from(mdata->msg));
}
#ifdef TEST_FRAMEWORK
{
pjsip_name_addr *tdata_name_addr;
pjsip_sip_uri *tdata_sip_uri;
char touri[128];
char fromuri[128];
tdata_name_addr = (pjsip_name_addr *) PJSIP_MSG_TO_HDR(tdata->msg)->uri;
tdata_sip_uri = pjsip_uri_get_uri(tdata_name_addr->uri);
pjsip_uri_print(PJSIP_URI_IN_FROMTO_HDR, tdata_sip_uri, touri, sizeof(touri));
tdata_name_addr = (pjsip_name_addr *) PJSIP_MSG_FROM_HDR(tdata->msg)->uri;
tdata_sip_uri = pjsip_uri_get_uri(tdata_name_addr->uri);
pjsip_uri_print(PJSIP_URI_IN_FROMTO_HDR, tdata_sip_uri, fromuri, sizeof(fromuri));
ast_test_suite_event_notify("MSG_FROMTO_URI",
"MdataFrom: %s\r\n"
"MsgFrom: %s\r\n"
"MdataDestination: %s\r\n"
"MsgTo: %s\r\n"
"Endpoint: %s\r\n"
"RequestURI: %s\r\n"
"ToURI: %s\r\n"
"FromURI: %s\r\n",
mdata->from,
ast_msg_get_from(mdata->msg),
mdata->destination,
ast_msg_get_to(mdata->msg),
ast_sorcery_object_get_id(endpoint),
uri,
touri,
fromuri
);
}
#endif
update_content_type(tdata, mdata->msg, &body);
if (ast_sip_add_body(tdata, &body)) {
pjsip_tx_data_dec_ref(tdata);
ast_log(LOG_ERROR, "PJSIP MESSAGE - Could not add body to request\n");
return -1;
}
/*
* This copies any headers set with MESSAGE_DATA() to the
* tdata.
*/
vars_to_headers(mdata->msg, tdata);
ast_debug(1, "Sending message to '%s' (via endpoint %s) from '%s'\n",
uri, ast_sorcery_object_get_id(endpoint), mdata->from);
if (ast_sip_send_request(tdata, NULL, endpoint, NULL, NULL)) {
ast_log(LOG_ERROR, "PJSIP MESSAGE - Could not send request\n");
return -1;
}
return 0;
}
static int sip_msg_send(const struct ast_msg *msg, const char *destination, const char *from)
{
struct msg_data *mdata;
int res;
if (ast_strlen_zero(destination)) {
ast_log(LOG_ERROR, "SIP MESSAGE - a 'To' URI must be specified\n");
return -1;
}
mdata = msg_data_create(msg, destination, from);
if (!mdata) {
return -1;
}
res = ast_sip_push_task_wait_serializer(message_serializer, msg_send, mdata);
ao2_ref(mdata, -1);
return res;
}
static const struct ast_msg_tech msg_tech = {
.name = "pjsip",
.msg_send = sip_msg_send,
};
static pj_status_t send_response(pjsip_rx_data *rdata, enum pjsip_status_code code,
pjsip_dialog *dlg, pjsip_transaction *tsx)
{
pjsip_tx_data *tdata;
pj_status_t status;
status = ast_sip_create_response(rdata, code, NULL, &tdata);
if (status != PJ_SUCCESS) {
ast_log(LOG_ERROR, "Unable to create response (%d)\n", status);
return status;
}
if (dlg && tsx) {
status = pjsip_dlg_send_response(dlg, tsx, tdata);
} else {
struct ast_sip_endpoint *endpoint;
endpoint = ast_pjsip_rdata_get_endpoint(rdata);
status = ast_sip_send_stateful_response(rdata, tdata, endpoint);
ao2_cleanup(endpoint);
}
if (status != PJ_SUCCESS) {
ast_log(LOG_ERROR, "Unable to send response (%d)\n", status);
}
return status;
}
static pj_bool_t module_on_rx_request(pjsip_rx_data *rdata)
{
enum pjsip_status_code code;
struct ast_msg *msg;
/* if not a MESSAGE, don't handle */
if (pjsip_method_cmp(&rdata->msg_info.msg->line.req.method, &pjsip_message_method)) {
return PJ_FALSE;
}
code = check_content_type(rdata);
if (code != PJSIP_SC_OK) {
send_response(rdata, code, NULL, NULL);
return PJ_TRUE;
}
msg = ast_msg_alloc();
if (!msg) {
send_response(rdata, PJSIP_SC_INTERNAL_SERVER_ERROR, NULL, NULL);
return PJ_TRUE;
}
code = rx_data_to_ast_msg(rdata, msg);
if (code != PJSIP_SC_OK) {
send_response(rdata, code, NULL, NULL);
ast_msg_destroy(msg);
return PJ_TRUE;
}
if (!ast_msg_has_destination(msg)) {
ast_debug(1, "MESSAGE request received, but no handler wanted it\n");
send_response(rdata, PJSIP_SC_NOT_FOUND, NULL, NULL);
ast_msg_destroy(msg);
return PJ_TRUE;
}
/* Send it to the messaging core.
*
* If we are unable to send a response, the most likely reason is that we
* are handling a retransmission of an incoming MESSAGE and were unable to
* create a transaction due to a duplicate key. If we are unable to send
* a response, we should not queue the message to the dialplan
*/
if (!send_response(rdata, PJSIP_SC_ACCEPTED, NULL, NULL)) {
ast_msg_queue(msg);
}
return PJ_TRUE;
}
static int incoming_in_dialog_request(struct ast_sip_session *session, struct pjsip_rx_data *rdata)
{
enum pjsip_status_code code;
int rc;
pjsip_dialog *dlg = session->inv_session->dlg;
pjsip_transaction *tsx = pjsip_rdata_get_tsx(rdata);
struct ast_msg_data *msg;
struct ast_party_caller *caller;
pjsip_name_addr *name_addr;
size_t from_len;
size_t to_len;
struct ast_msg_data_attribute attrs[4];
int pos = 0;
int body_pos;
if (!session->channel) {
send_response(rdata, PJSIP_SC_NOT_FOUND, dlg, tsx);
return 0;
}
code = check_content_type_in_dialog(rdata);
if (code != PJSIP_SC_OK) {
send_response(rdata, code, dlg, tsx);
return 0;
}
caller = ast_channel_caller(session->channel);
name_addr = (pjsip_name_addr *) rdata->msg_info.from->uri;
from_len = pj_strlen(&name_addr->display);
if (from_len) {
attrs[pos].type = AST_MSG_DATA_ATTR_FROM;
from_len++;
attrs[pos].value = ast_alloca(from_len);
ast_copy_pj_str(attrs[pos].value, &name_addr->display, from_len);
pos++;
} else if (caller->id.name.valid && !ast_strlen_zero(caller->id.name.str)) {
attrs[pos].type = AST_MSG_DATA_ATTR_FROM;
attrs[pos].value = caller->id.name.str;
pos++;
}
name_addr = (pjsip_name_addr *) rdata->msg_info.to->uri;
to_len = pj_strlen(&name_addr->display);
if (to_len) {
attrs[pos].type = AST_MSG_DATA_ATTR_TO;
to_len++;
attrs[pos].value = ast_alloca(to_len);
ast_copy_pj_str(attrs[pos].value, &name_addr->display, to_len);
pos++;
}
attrs[pos].type = AST_MSG_DATA_ATTR_CONTENT_TYPE;
attrs[pos].value = ast_alloca(rdata->msg_info.msg->body->content_type.type.slen
+ rdata->msg_info.msg->body->content_type.subtype.slen + 2);
sprintf(attrs[pos].value, "%.*s/%.*s",
(int)rdata->msg_info.msg->body->content_type.type.slen,
rdata->msg_info.msg->body->content_type.type.ptr,
(int)rdata->msg_info.msg->body->content_type.subtype.slen,
rdata->msg_info.msg->body->content_type.subtype.ptr);
pos++;
body_pos = pos;
attrs[pos].type = AST_MSG_DATA_ATTR_BODY;
attrs[pos].value = ast_malloc(rdata->msg_info.msg->body->len + 1);
if (!attrs[pos].value) {
send_response(rdata, PJSIP_SC_INTERNAL_SERVER_ERROR, dlg, tsx);
return 0;
}
ast_copy_string(attrs[pos].value, rdata->msg_info.msg->body->data, rdata->msg_info.msg->body->len + 1);
pos++;
msg = ast_msg_data_alloc(AST_MSG_DATA_SOURCE_TYPE_IN_DIALOG, attrs, pos);
if (!msg) {
ast_free(attrs[body_pos].value);
send_response(rdata, PJSIP_SC_INTERNAL_SERVER_ERROR, dlg, tsx);
return 0;
}
ast_debug(1, "Received in-dialog MESSAGE from '%s:%s': %s %s\n",
ast_msg_data_get_attribute(msg, AST_MSG_DATA_ATTR_FROM),
ast_channel_name(session->channel),
ast_msg_data_get_attribute(msg, AST_MSG_DATA_ATTR_TO),
ast_msg_data_get_attribute(msg, AST_MSG_DATA_ATTR_BODY));
rc = ast_msg_data_queue_frame(session->channel, msg);
ast_free(attrs[body_pos].value);
ast_free(msg);
if (rc != 0) {
send_response(rdata, PJSIP_SC_INTERNAL_SERVER_ERROR, dlg, tsx);
} else {
send_response(rdata, PJSIP_SC_ACCEPTED, dlg, tsx);
}
return 0;
}
static struct ast_sip_session_supplement messaging_supplement = {
.method = "MESSAGE",
.incoming_request = incoming_in_dialog_request
};
static pjsip_module messaging_module = {
.name = {"Messaging Module", 16},
.id = -1,
.priority = PJSIP_MOD_PRIORITY_APPLICATION,
.on_rx_request = module_on_rx_request,
};
static int load_module(void)
{
if (ast_sip_register_service(&messaging_module) != PJ_SUCCESS) {
return AST_MODULE_LOAD_DECLINE;
}
if (pjsip_endpt_add_capability(ast_sip_get_pjsip_endpoint(),
NULL, PJSIP_H_ALLOW, NULL, 1,
&pjsip_message_method.name) != PJ_SUCCESS) {
ast_sip_unregister_service(&messaging_module);
return AST_MODULE_LOAD_DECLINE;
}
if (ast_msg_tech_register(&msg_tech)) {
ast_sip_unregister_service(&messaging_module);
return AST_MODULE_LOAD_DECLINE;
}
message_serializer = ast_sip_create_serializer("pjsip/messaging");
if (!message_serializer) {
ast_sip_unregister_service(&messaging_module);
ast_msg_tech_unregister(&msg_tech);
return AST_MODULE_LOAD_DECLINE;
}
ast_sip_session_register_supplement(&messaging_supplement);
return AST_MODULE_LOAD_SUCCESS;
}
static int unload_module(void)
{
ast_sip_session_unregister_supplement(&messaging_supplement);
ast_msg_tech_unregister(&msg_tech);
ast_sip_unregister_service(&messaging_module);
ast_taskprocessor_unreference(message_serializer);
return 0;
}
AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "PJSIP Messaging Support",
.support_level = AST_MODULE_SUPPORT_CORE,
.load = load_module,
.unload = unload_module,
.load_pri = AST_MODPRI_APP_DEPEND,
.requires = "res_pjsip,res_pjsip_session",
);
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