[asterisk-commits] oej: trunk r48112 - in /trunk: ./ configs/sip.conf.sample

[asterisk-commits at lists.digium.com]

Author: oej
Date: Wed Nov 29 13:47:45 2006
New Revision: 48112

URL: http://svn.digium.com/view/asterisk?view=rev&rev=48112
Log:
Explain RTP timeouts

Modified:
    trunk/   (props changed)
    trunk/configs/sip.conf.sample

Propchange: trunk/
------------------------------------------------------------------------------
Binary property 'branch-1.4-merged' - no diff available.

Modified: trunk/configs/sip.conf.sample
URL: http://svn.digium.com/view/asterisk/trunk/configs/sip.conf.sample?view=diff&rev=48112&r1=48111&r2=48112
==============================================================================
--- trunk/configs/sip.conf.sample (original)
+++ trunk/configs/sip.conf.sample Wed Nov 29 13:47:45 2006
@@ -94,9 +94,11 @@
 ;language=en			; Default language setting for all users/peers
 				; This may also be set for individual users/peers
 ;relaxdtmf=yes			; Relax dtmf handling
-;rtptimeout=60			; Terminate call if 60 seconds of no RTP activity
-				; when we're not on hold
-;rtpholdtimeout=300		; Terminate call if 300 seconds of no RTP activity
+;rtptimeout=60			; Terminate call if 60 seconds of no RTP or RTCP activity
+				; when we're not on hold. This is to be able to hangup
+				; a call in the case of a phone disappearing from the net,
+				; like a powerloss or grandma tripping over a cable.
+;rtpholdtimeout=300		; Terminate call if 300 seconds of no RTP or RTCP activity
 				; when we're on hold (must be > rtptimeout)
 ;trustrpid = no			; If Remote-Party-ID should be trusted
 ;sendrpid = yes			; If Remote-Party-ID should be sent