diff options
| author | Mark Spencer <markster@digium.com> | 2004-05-27 22:12:55 +0000 |
|---|---|---|
| committer | Mark Spencer <markster@digium.com> | 2004-05-27 22:12:55 +0000 |
| commit | e446f4ca814d2832f66b6ef63a33859b3e024918 (patch) | |
| tree | ff3e36b608b7a399e59482f7cc4ab5f5d1ee65d1 /configs/sip.conf.sample | |
| parent | 4d1706d05fb1f76eade7ecdb0c303a6edbd3da5c (diff) | |
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
Diffstat (limited to 'configs/sip.conf.sample')
| -rwxr-xr-x | configs/sip.conf.sample | 8 |
1 files changed, 6 insertions, 2 deletions
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) |