[asterisk-commits] mnicholson: branch 1.8 r333009 - in /branches/1.8: ./ channels/sip/include/ c...

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Tue Aug 23 13:11:54 CDT 2011


Author: mnicholson
Date: Tue Aug 23 13:11:50 2011
New Revision: 333009

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=333009
Log:
default 'sipstorecause' to no

AST-580

Modified:
    branches/1.8/CHANGES
    branches/1.8/UPGRADE.txt
    branches/1.8/channels/sip/include/sip.h
    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=333009&r1=333008&r2=333009
==============================================================================
--- branches/1.8/CHANGES (original)
+++ branches/1.8/CHANGES Tue Aug 23 13:11:50 2011
@@ -44,7 +44,7 @@
  * 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. This carries a
-   performance penalty and can be disabled in sip.conf using the
+   performance penalty and must be enabled 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.

Modified: branches/1.8/UPGRADE.txt
URL: http://svnview.digium.com/svn/asterisk/branches/1.8/UPGRADE.txt?view=diff&rev=333009&r1=333008&r2=333009
==============================================================================
--- branches/1.8/UPGRADE.txt (original)
+++ branches/1.8/UPGRADE.txt Tue Aug 23 13:11:50 2011
@@ -20,6 +20,10 @@
 
 From 1.6.2 to 1.8:
 
+* chan_sip no longer sets HASH(SIP_CAUSE,<chan name>) on channels by default.
+  This must now be enabled by setting 'sipstorecause' to 'yes' in sip.conf.
+  This carries a performance penalty.
+
 * Asterisk now requires libpri 1.4.11+ for PRI support.
 
 * A couple of CLI commands in res_ais were changed back to their original form:

Modified: branches/1.8/channels/sip/include/sip.h
URL: http://svnview.digium.com/svn/asterisk/branches/1.8/channels/sip/include/sip.h?view=diff&rev=333009&r1=333008&r2=333009
==============================================================================
--- branches/1.8/channels/sip/include/sip.h (original)
+++ branches/1.8/channels/sip/include/sip.h Tue Aug 23 13:11:50 2011
@@ -221,7 +221,7 @@
 #define DEFAULT_SDPOWNER   "root"          /*!< Default SDP username field in (o=) header unless re-defined in sip.conf */
 #define DEFAULT_ENGINE     "asterisk"      /*!< Default RTP engine to use for sessions */
 #define DEFAULT_CAPABILITY (AST_FORMAT_ULAW | AST_FORMAT_TESTLAW | AST_FORMAT_ALAW | AST_FORMAT_GSM | AST_FORMAT_H263);
-#define DEFAULT_STORE_SIP_CAUSE TRUE      /*!< Store HASH(SIP_CAUSE,<channel name>) for channels by default */
+#define DEFAULT_STORE_SIP_CAUSE FALSE      /*!< Don't store HASH(SIP_CAUSE,<channel name>) for channels by default */
 #endif
 /*@}*/
 

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=333009&r1=333008&r2=333009
==============================================================================
--- branches/1.8/configs/sip.conf.sample (original)
+++ branches/1.8/configs/sip.conf.sample Tue Aug 23 13:11:50 2011
@@ -1007,14 +1007,13 @@
 ; jblog = no                  ; Enables jitterbuffer frame logging. Defaults to "no".
 
 ;----------------------------- SIP_CAUSE reporting ---------------------------------
-; storesipcause = yes         ; This option causes chan_sip to set the
+; 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. For
-			      ; historical reasons, this option defaults to
-			      ; "yes".
+			      ; be used in low call volume situations. This
+                              ; option defaults to "no".
 
 ;-----------------------------------------------------------------------------------
 




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