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Update of docs 

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@45333 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Olle Johansson 2006-10-17 17:51:34 +00:00
parent 748de51a84
commit a8a26ad389
1 changed files with 5 additions and 0 deletions
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5
configs/sip.conf.sample
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@ -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