[asterisk-commits] mnicholson: branch 1.8 r332021 - in /branches/1.8: ./ channels/ configs/

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Tue Aug 16 09:20:47 CDT 2011


Author: mnicholson
Date: Tue Aug 16 09:20:43 2011
New Revision: 332021

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=332021
Log:
Added the 'storesipcause' option to sip.conf to allow the user to disable the
setting of HASH(SIP_CAUSE,<chan name>) on the channel.

Having chan_sip set HASH(SIP_CAUSE,<chan name>) on the channel carries a
significant performance penalty because of the usage of the MASTER_CHANNEL()
dialplan function.

Modified:
    branches/1.8/CHANGES
    branches/1.8/channels/chan_sip.c
    branches/1.8/configs/sip.conf.sample

Modified: branches/1.8/CHANGES
URL: http://svnview.digium.com/svn/asterisk/branches/1.8/CHANGES?view=diff&rev=332021&r1=332020&r2=332021
==============================================================================
--- branches/1.8/CHANGES (original)
+++ branches/1.8/CHANGES Tue Aug 16 09:20:43 2011
@@ -43,7 +43,9 @@
    and enables symmetric RTP support.
  * Slave SIP channels now set HASH(SIP_CAUSE,<slave-channel-name>) on each
    response.  This permits the master channel to know how each channel dialled
-   in a multi-channel setup resolved in an individual way.
+   in a multi-channel setup resolved in an individual way. This carries a
+   performance penalty and can be disabled in sip.conf using the
+   'storesipcause' option.
  * Added 'externtcpport' and 'externtlsport' options to allow custom port
    configuration for the externip and externhost options when tcp or tls is used.
  * Added support for message body (stored in content variable) to SIP NOTIFY message

Modified: branches/1.8/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.8/channels/chan_sip.c?view=diff&rev=332021&r1=332020&r2=332021
==============================================================================
--- branches/1.8/channels/chan_sip.c (original)
+++ branches/1.8/channels/chan_sip.c Tue Aug 16 09:20:43 2011
@@ -745,6 +745,8 @@
 static int global_min_se;                     /*!< Lowest threshold for session refresh interval  */
 static int global_max_se;                     /*!< Highest threshold for session refresh interval */
 
+static int global_store_sip_cause;    /*!< Whether the MASTER_CHANNEL(HASH(SIP_CAUSE,[chan_name])) var should be set */
+
 static int global_dynamic_exclude_static = 0; /*!< Exclude static peers from contact registrations */
 /*@}*/
 
@@ -17495,6 +17497,7 @@
 		ast_cli(a->fd, "  SIP realtime:           Enabled\n" );
 	ast_cli(a->fd, "  Qualify Freq :          %d ms\n", global_qualifyfreq);
 	ast_cli(a->fd, "  Q.850 Reason header:    %s\n", AST_CLI_YESNO(ast_test_flag(&global_flags[1], SIP_PAGE2_Q850_REASON)));
+	ast_cli(a->fd, "  Store SIP_CAUSE:        %s\n", AST_CLI_YESNO(global_store_sip_cause));
 	ast_cli(a->fd, "\nNetwork QoS Settings:\n");
 	ast_cli(a->fd, "---------------------------\n");
 	ast_cli(a->fd, "  IP ToS SIP:             %s\n", ast_tos2str(global_tos_sip));
@@ -24422,7 +24425,7 @@
 
 			handle_response(p, respid, e + len, req, seqno);
 
-			if (p->owner) {
+			if (global_store_sip_cause && p->owner) {
 				struct ast_channel *owner = p->owner;
 
 				snprintf(causevar, sizeof(causevar), "MASTER_CHANNEL(HASH(SIP_CAUSE,%s))", owner->name);
@@ -27402,6 +27405,7 @@
 	global_shrinkcallerid = 1;
 	authlimit = DEFAULT_AUTHLIMIT;
 	authtimeout = DEFAULT_AUTHTIMEOUT;
+	global_store_sip_cause = TRUE;
 
 	sip_cfg.matchexternaddrlocally = DEFAULT_MATCHEXTERNADDRLOCALLY;
 
@@ -27871,6 +27875,8 @@
 			} else {
 				global_st_refresher = i;
 			}
+		} else if (!strcasecmp(v->name, "storesipcause")) {
+			global_store_sip_cause = ast_true(v->value);
 		} else if (!strcasecmp(v->name, "qualifygap")) {
 			if (sscanf(v->value, "%30d", &global_qualify_gap) != 1) {
 				ast_log(LOG_WARNING, "Invalid qualifygap '%s' at line %d of %s\n", v->value, v->lineno, config);

Modified: branches/1.8/configs/sip.conf.sample
URL: http://svnview.digium.com/svn/asterisk/branches/1.8/configs/sip.conf.sample?view=diff&rev=332021&r1=332020&r2=332021
==============================================================================
--- branches/1.8/configs/sip.conf.sample (original)
+++ branches/1.8/configs/sip.conf.sample Tue Aug 16 09:20:43 2011
@@ -1005,6 +1005,17 @@
                               ; but occasionally has spikes.
 
 ; jblog = no                  ; Enables jitterbuffer frame logging. Defaults to "no".
+
+;----------------------------- SIP_CAUSE reporting ---------------------------------
+; storesipcause = yes         ; This option causes chan_sip to set the
+			      ; HASH(SIP_CAUSE,<channel name>) channel variable
+			      ; to the value of the last sip response.
+			      ; WARNING: enabling this option carries a
+			      ; significant performance burden. It should only
+			      ; be used in low call volume situations. For
+			      ; historical reasons, this option defaults to
+			      ; "yes".
+
 ;-----------------------------------------------------------------------------------
 
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