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| author | Olle Johansson <oej@edvina.net> | 2006-10-17 17:51:34 +0000 |
|---|---|---|
| committer | Olle Johansson <oej@edvina.net> | 2006-10-17 17:51:34 +0000 |
| commit | a8a26ad38999d88613248835b4b88e80622f007a (patch) | |
| tree | 4b7245ba07f95ab59f72c17d30e03ed6f6a31ca3 /configs/sip.conf.sample | |
| parent | 748de51a842136206887a9c0a1b7c8cec64c4fc3 (diff) | |
Update of docs
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@45333 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'configs/sip.conf.sample')
| -rw-r--r-- | configs/sip.conf.sample | 5 |
1 files changed, 5 insertions, 0 deletions
diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample index b6e84fa91..504b7c7c9 100644 --- a/configs/sip.conf.sample +++ b/configs/sip.conf.sample @@ -264,6 +264,11 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; behind a NAT, or for some other reason wants Asterisk to ; stay in the audio path, you may want to turn this off. + ; This setting also affect direct RTP + ; at call setup (a new feature in 1.4 - setting up the + ; call directly between the endpoints instead of sending + ; a re-INVITE). + ;canreinvite=nonat ; An additional option is to allow media path redirection ; (reinvite) but only when the peer where the media is being ; sent is known to not be behind a NAT (as the RTP core can |