[asterisk-commits] oej: trunk r93160 - in /trunk: CHANGES configs/sip.conf.sample

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Sun Dec 16 02:19:38 CST 2007


Author: oej
Date: Sun Dec 16 02:19:38 2007
New Revision: 93160

URL: http://svn.digium.com/view/asterisk?view=rev&rev=93160
Log:
Update documentation

Modified:
    trunk/CHANGES
    trunk/configs/sip.conf.sample

Modified: trunk/CHANGES
URL: http://svn.digium.com/view/asterisk/trunk/CHANGES?view=diff&rev=93160&r1=93159&r2=93160
==============================================================================
--- trunk/CHANGES (original)
+++ trunk/CHANGES Sun Dec 16 02:19:38 2007
@@ -101,6 +101,9 @@
   * A new option called "callcounter" (global/peer/user level) enables call counters needed
     for better status reports needed for queues and SIP subscriptions. (Call-Limit was previously
     used to enable this functionality).
+  * New settings for timer T1 and timer B on a global level or per device. This makes it 
+    possible to force timeout faster on non-responsive SIP servers. These settings are
+    considered advanced, so don't use them unless you have a problem.
 
 IAX2 changes
 ------------

Modified: trunk/configs/sip.conf.sample
URL: http://svn.digium.com/view/asterisk/trunk/configs/sip.conf.sample?view=diff&rev=93160&r1=93159&r2=93160
==============================================================================
--- trunk/configs/sip.conf.sample (original)
+++ trunk/configs/sip.conf.sample Sun Dec 16 02:19:38 2007
@@ -81,13 +81,6 @@
 				; and subscriptions (seconds)
 ;minexpiry=60			; Minimum length of registrations/subscriptions (default 60)
 ;defaultexpiry=120		; Default length of incoming/outgoing registration
-;t1min=100			; Minimum roundtrip time for messages to monitored hosts
-				; Defaults to 100 ms
-;timert1=500		; Default T1 timer
-				; Defaults to 500 ms
-;timerb=32000		; Call setup timer. If a provisional response is not received
-						; in this amount of time, the call will autocongest
-				; Defaults to 64*timert1
 ;notifymimetype=text/plain	; Allow overriding of mime type in MWI NOTIFY
 ;buggymwi=no			; Cisco SIP firmware doesn't support the MWI RFC
 				; fully. Enable this option to not get error messages
@@ -191,6 +184,19 @@
 				; this setting will enforce inactivation of the regexten
 				; extension for the peer
 ;
+;--------------------------- SIP timers ----------------------------------------------------
+; These timers are used primarily in INVITE transactions. 
+; The default for Timer T1 is 500 ms or the measured run-trip time between
+; Asterisk and the device if you have qualify=yes for the device.
+;
+;t1min=100			; Minimum roundtrip time for messages to monitored hosts
+				; Defaults to 100 ms
+;timert1=500		        ; Default T1 timer
+				; Defaults to 500 ms
+;timerb=32000		        ; Call setup timer. If a provisional response is not received
+				; in this amount of time, the call will autocongest
+				; Defaults to 64*timert1
+
 ;--------------------------- RTP timers ----------------------------------------------------
 ; These timers are currently used for both audio and video streams. The RTP timeouts
 ; are only applied to the audio channel.




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