diff options
| author | oej <oej@f38db490-d61c-443f-a65b-d21fe96a405b> | 2006-11-16 15:03:49 +0000 |
|---|---|---|
| committer | oej <oej@f38db490-d61c-443f-a65b-d21fe96a405b> | 2006-11-16 15:03:49 +0000 |
| commit | afc4ff93833cc2f3790429eda0c581271f589885 (patch) | |
| tree | 51458603471c8e1cfca43f17050ed60a23431fb6 /configs/sip.conf.sample | |
| parent | f298b7dc446587c2706528a795caecbe0ca9a144 (diff) | |
- CANCEL is never authenticated (according to the RFC)
- Update docs on canreinvite. "nonat" is the recommended setting for most users with
phones behind a NAT.
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@47733 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'configs/sip.conf.sample')
| -rw-r--r-- | configs/sip.conf.sample | 6 |
1 files changed, 6 insertions, 0 deletions
diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample index eb5334693..6557cd551 100644 --- a/configs/sip.conf.sample +++ b/configs/sip.conf.sample @@ -248,6 +248,12 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; route = Assume NAT, don't send rport ; (work around more UNIDEN bugs) +;----------------------------------- MEDIA HANDLING -------------------------------- +; By default, Asterisk tries to re-invite the audio to an optimal path. If there's +; no reason for Asterisk to stay in the media path, the media will be redirected. +; This does not really work with in the case where Asterisk is outside and have +; clients on the inside of a NAT. In that case, you want to set canreinvite=nonat +; ;canreinvite=yes ; Asterisk by default tries to redirect the ; RTP media stream (audio) to go directly from ; the caller to the callee. Some devices do not |