From e446f4ca814d2832f66b6ef63a33859b3e024918 Mon Sep 17 00:00:00 2001 From: Mark Spencer Date: Thu, 27 May 2004 22:12:55 +0000 Subject: Merge OSS fixes for FreeBSD, implement rtptimeout and rtpholdtimeout git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@3097 65c4cc65-6c06-0410-ace0-fbb531ad65f3 --- configs/sip.conf.sample | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) (limited to 'configs') diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample index c314b7099..ddd335419 100755 --- a/configs/sip.conf.sample +++ b/configs/sip.conf.sample @@ -48,7 +48,10 @@ bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) ;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 @@ bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) ; port ; qualify ; defaultip - +; rtptimeout +; rtpholdtimeout ;[sip_proxy] ; For incoming calls only. Example: FWD (Free World Dialup) -- cgit v1.2.3