[asterisk-commits] file: trunk r403208 - in /trunk: ./ include/asterisk/ res/ res/res_pjsip/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Wed Nov 27 18:38:38 CST 2013
Author: file
Date: Wed Nov 27 18:38:36 2013
New Revision: 403208
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=403208
Log:
res_pjsip_session: Add configurable behavior for redirects.
The action taken when a redirect occurs is now configurable on a
per-endpoint basis. The redirect can either be treated as a redirect
to a local extension, to a URI that is dialed through the Asterisk
core, or to a URI that is dialed within PJSIP itself.
(closes issue ASTERISK-21710)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2963/
........
Merged revisions 403207 from http://svn.asterisk.org/svn/asterisk/branches/12
Modified:
trunk/ (props changed)
trunk/include/asterisk/res_pjsip.h
trunk/res/res_pjsip.c
trunk/res/res_pjsip/pjsip_configuration.c
trunk/res/res_pjsip_session.c
Propchange: trunk/
------------------------------------------------------------------------------
--- branch-12-merged (original)
+++ branch-12-merged Wed Nov 27 18:38:36 2013
@@ -1,1 +1,1 @@
-/branches/12:1-398558,398560-398577,398579-399305,399307-401390,401392-403175,403179
+/branches/12:1-398558,398560-398577,398579-399305,399307-401390,401392-403175,403179,403207
Modified: trunk/include/asterisk/res_pjsip.h
URL: http://svnview.digium.com/svn/asterisk/trunk/include/asterisk/res_pjsip.h?view=diff&rev=403208&r1=403207&r2=403208
==============================================================================
--- trunk/include/asterisk/res_pjsip.h (original)
+++ trunk/include/asterisk/res_pjsip.h Wed Nov 27 18:38:36 2013
@@ -314,6 +314,15 @@
AST_SIP_MEDIA_ENCRYPT_SDES,
/*! Offer encrypted session media with datagram TLS key exchange */
AST_SIP_MEDIA_ENCRYPT_DTLS,
+};
+
+enum ast_sip_session_redirect {
+ /*! User portion of the target URI should be used as the target in the dialplan */
+ AST_SIP_REDIRECT_USER = 0,
+ /*! Target URI should be used as the target in the dialplan */
+ AST_SIP_REDIRECT_URI_CORE,
+ /*! Target URI should be used as the target within chan_pjsip itself */
+ AST_SIP_REDIRECT_URI_PJSIP,
};
/*!
@@ -574,6 +583,8 @@
unsigned int faxdetect;
/*! Determines if transfers (using REFER) are allowed by this endpoint */
unsigned int allowtransfer;
+ /*! Method used when handling redirects */
+ enum ast_sip_session_redirect redirect_method;
};
/*!
Modified: trunk/res/res_pjsip.c
URL: http://svnview.digium.com/svn/asterisk/trunk/res/res_pjsip.c?view=diff&rev=403208&r1=403207&r2=403208
==============================================================================
--- trunk/res/res_pjsip.c (original)
+++ trunk/res/res_pjsip.c Wed Nov 27 18:38:36 2013
@@ -241,6 +241,27 @@
the endpoint can be identified in multiple ways.</para></note>
<enumlist>
<enum name="username" />
+ </enumlist>
+ </description>
+ </configOption>
+ <configOption name="redirect_method">
+ <synopsis>How redirects received from an endpoint are handled</synopsis>
+ <description><para>
+ When a redirect is received from an endpoint there are multiple ways it can be handled.
+ If this option is set to <literal>user</literal> the user portion of the redirect target
+ is treated as an extension within the dialplan and dialed using a Local channel. If this option
+ is set to <literal>uri_core</literal> the target URI is returned to the dialing application
+ which dials it using the PJSIP channel driver and endpoint originally used. If this option is
+ set to <literal>uri_pjsip</literal> the redirect occurs within chan_pjsip itself and is not exposed
+ to the core at all. The <literal>uri_pjsip</literal> option has the benefit of being more efficient
+ and also supporting multiple potential redirect targets. The con is that since redirection occurs
+ within chan_pjsip redirecting information is not forwarded and redirection can not be
+ prevented.
+ </para>
+ <enumlist>
+ <enum name="user" />
+ <enum name="uri_core" />
+ <enum name="uri_pjsip" />
</enumlist>
</description>
</configOption>
Modified: trunk/res/res_pjsip/pjsip_configuration.c
URL: http://svnview.digium.com/svn/asterisk/trunk/res/res_pjsip/pjsip_configuration.c?view=diff&rev=403208&r1=403207&r2=403208
==============================================================================
--- trunk/res/res_pjsip/pjsip_configuration.c (original)
+++ trunk/res/res_pjsip/pjsip_configuration.c Wed Nov 27 18:38:36 2013
@@ -452,6 +452,25 @@
}
*buf = ast_strdup(*buf);
+ return 0;
+}
+
+static int redirect_handler(const struct aco_option *opt, struct ast_variable *var, void *obj)
+{
+ struct ast_sip_endpoint *endpoint = obj;
+
+ if (!strcasecmp(var->value, "user")) {
+ endpoint->redirect_method = AST_SIP_REDIRECT_USER;
+ } else if (!strcasecmp(var->value, "uri_core")) {
+ endpoint->redirect_method = AST_SIP_REDIRECT_URI_CORE;
+ } else if (!strcasecmp(var->value, "uri_pjsip")) {
+ endpoint->redirect_method = AST_SIP_REDIRECT_URI_PJSIP;
+ } else {
+ ast_log(LOG_ERROR, "Unrecognized redirect method %s specified for endpoint %s\n",
+ var->value, ast_sorcery_object_get_id(endpoint));
+ return -1;
+ }
+
return 0;
}
@@ -1353,6 +1372,7 @@
ast_sorcery_object_field_register_custom(sip_sorcery, "endpoint", "dtls_ca_path", "", dtls_handler, dtlscapath_to_str, 0, 0);
ast_sorcery_object_field_register_custom(sip_sorcery, "endpoint", "dtls_setup", "", dtls_handler, dtlssetup_to_str, 0, 0);
ast_sorcery_object_field_register(sip_sorcery, "endpoint", "srtp_tag_32", "no", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, media.rtp.srtp_tag_32));
+ ast_sorcery_object_field_register_custom(sip_sorcery, "endpoint", "redirect_method", "user", redirect_handler, NULL, 0, 0);
if (ast_sip_initialize_sorcery_transport(sip_sorcery)) {
ast_log(LOG_ERROR, "Failed to register SIP transport support with sorcery\n");
Modified: trunk/res/res_pjsip_session.c
URL: http://svnview.digium.com/svn/asterisk/trunk/res/res_pjsip_session.c?view=diff&rev=403208&r1=403207&r2=403208
==============================================================================
--- trunk/res/res_pjsip_session.c (original)
+++ trunk/res/res_pjsip_session.c Wed Nov 27 18:38:36 2013
@@ -2065,13 +2065,31 @@
static pjsip_redirect_op session_inv_on_redirected(pjsip_inv_session *inv, const pjsip_uri *target, const pjsip_event *e)
{
struct ast_sip_session *session = inv->mod_data[session_module.id];
-
- if (PJSIP_URI_SCHEME_IS_SIP(target) || PJSIP_URI_SCHEME_IS_SIPS(target)) {
- const pjsip_sip_uri *uri = pjsip_uri_get_uri(target);
+ const pjsip_sip_uri *uri;
+
+ if (session->endpoint->redirect_method == AST_SIP_REDIRECT_URI_PJSIP) {
+ return PJSIP_REDIRECT_ACCEPT;
+ }
+
+ if (!PJSIP_URI_SCHEME_IS_SIP(target) && !PJSIP_URI_SCHEME_IS_SIPS(target)) {
+ return PJSIP_REDIRECT_STOP;
+ }
+
+ uri = pjsip_uri_get_uri(target);
+
+ if (session->endpoint->redirect_method == AST_SIP_REDIRECT_USER) {
char exten[AST_MAX_EXTENSION];
ast_copy_pj_str(exten, &uri->user, sizeof(exten));
ast_channel_call_forward_set(session->channel, exten);
+ } else if (session->endpoint->redirect_method == AST_SIP_REDIRECT_URI_CORE) {
+ char target_uri[PJSIP_MAX_URL_SIZE];
+ /* PJSIP/ + endpoint length + / + max URL size */
+ char forward[8 + strlen(ast_sorcery_object_get_id(session->endpoint)) + PJSIP_MAX_URL_SIZE];
+
+ pjsip_uri_print(PJSIP_URI_IN_REQ_URI, uri, target_uri, sizeof(target_uri));
+ sprintf(forward, "PJSIP/%s/%s", ast_sorcery_object_get_id(session->endpoint), target_uri);
+ ast_channel_call_forward_set(session->channel, forward);
}
return PJSIP_REDIRECT_STOP;
More information about the asterisk-commits
mailing list