[Asterisk-cvs] asterisk/configs sip.conf.sample,1.28,1.29

[markster at lists.digium.com]

Update of /usr/cvsroot/asterisk/configs
In directory mongoose.digium.com:/tmp/cvs-serv4664/configs

Modified Files:
	sip.conf.sample 
Log Message:
Merge OSS fixes for FreeBSD, implement rtptimeout and rtpholdtimeout


Index: sip.conf.sample
===================================================================
RCS file: /usr/cvsroot/asterisk/configs/sip.conf.sample,v
retrieving revision 1.28
retrieving revision 1.29
diff -u -d -r1.28 -r1.29
--- sip.conf.sample	24 May 2004 15:09:34 -0000	1.28
+++ sip.conf.sample	27 May 2004 22:12:55 -0000	1.29
@@ -48,7 +48,10 @@
 ;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
+				; when we're on hold (must be > rtptimeout)
 
 ; Asterisk can register as a SIP user agent to a SIP proxy (provider)
 ; Format for the register statement is:
@@ -128,7 +131,8 @@
 ;                             port
 ;                             qualify
 ;                             defaultip
-
+;                             rtptimeout
+;                             rtpholdtimeout
 
 ;[sip_proxy]
 ; For incoming calls only. Example: FWD (Free World Dialup)