[asterisk-commits] mnicholson: trunk r332023 - in /trunk: ./ channels/ configs/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Tue Aug 16 09:41:27 CDT 2011
Author: mnicholson
Date: Tue Aug 16 09:41:23 2011
New Revision: 332023
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=332023
Log:
Merged revisions 332022 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................
r332022 | mnicholson | 2011-08-16 09:40:37 -0500 (Tue, 16 Aug 2011) | 16 lines
In 10 and trunk this option is disabled by default.
Merged revisions 332021 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r332021 | mnicholson | 2011-08-16 09:20:43 -0500 (Tue, 16 Aug 2011) | 7 lines
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:
trunk/ (props changed)
trunk/CHANGES
trunk/channels/chan_sip.c
trunk/configs/sip.conf.sample
Propchange: trunk/
------------------------------------------------------------------------------
Binary property 'branch-10-merged' - no diff available.
Modified: trunk/CHANGES
URL: http://svnview.digium.com/svn/asterisk/trunk/CHANGES?view=diff&rev=332023&r1=332022&r2=332023
==============================================================================
--- trunk/CHANGES (original)
+++ trunk/CHANGES Tue Aug 16 09:41:23 2011
@@ -190,6 +190,9 @@
SIP Changes
-----------
* Add T38 support for REJECTED state where T.38 Negotiation is explicitly rejected.
+ * Setting of HASH(SIP_CAUSE,<slave-channel-name>) on channels is now disabled
+ by default. It can be enabled using the 'storesipcause' option. This feature
+ has a significant performance penalty.
Queue changes
-------------
@@ -256,7 +259,9 @@
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: trunk/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/chan_sip.c?view=diff&rev=332023&r1=332022&r2=332023
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Tue Aug 16 09:41:23 2011
@@ -746,6 +746,8 @@
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 */
/*@}*/
@@ -17979,6 +17981,7 @@
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));
@@ -25072,7 +25075,7 @@
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);
@@ -28108,6 +28111,7 @@
global_shrinkcallerid = 1;
authlimit = DEFAULT_AUTHLIMIT;
authtimeout = DEFAULT_AUTHTIMEOUT;
+ global_store_sip_cause = FALSE;
sip_cfg.matchexternaddrlocally = DEFAULT_MATCHEXTERNADDRLOCALLY;
@@ -28585,6 +28589,8 @@
} 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: trunk/configs/sip.conf.sample
URL: http://svnview.digium.com/svn/asterisk/trunk/configs/sip.conf.sample?view=diff&rev=332023&r1=332022&r2=332023
==============================================================================
--- trunk/configs/sip.conf.sample (original)
+++ trunk/configs/sip.conf.sample Tue Aug 16 09:41:23 2011
@@ -1016,6 +1016,16 @@
; but occasionally has spikes.
; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
+
+;----------------------------- SIP_CAUSE reporting ---------------------------------
+; storesipcause = no ; 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. This
+ ; option defaults to "no".
+
;-----------------------------------------------------------------------------------
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