diff options
| author | oej <oej@f38db490-d61c-443f-a65b-d21fe96a405b> | 2006-11-16 15:12:30 +0000 |
|---|---|---|
| committer | oej <oej@f38db490-d61c-443f-a65b-d21fe96a405b> | 2006-11-16 15:12:30 +0000 |
| commit | 735aaa7959782d2fb1f86009ad7b245053458985 (patch) | |
| tree | 809d31ec081e66c1d2bdb31b89b2eebfe6febaed /configs | |
| parent | e48a9fd51b820160cce8e505d7e87a8ce2ed150c (diff) | |
- CANCEL never uses authentication
- Add docs on canreinvite
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@47734 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'configs')
| -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 169941e08..eeb29a18e 100644 --- a/configs/sip.conf.sample +++ b/configs/sip.conf.sample @@ -266,6 +266,12 @@ 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 |