[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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