From: Joshua Colp Date: Wed, 8 Aug 2007 13:52:13 +0000 (+0000) Subject: Merged revisions 78569 via svnmerge from X-Git-Tag: 1.6.0-beta1~3^2~1778 X-Git-Url: http://git.ipfire.org/gitweb.cgi?a=commitdiff_plain;h=afceb3e4aa15c8763935e4a4e89075ccacd55060;p=thirdparty%2Fasterisk.git Merged revisions 78569 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r78569 | file | 2007-08-08 10:51:01 -0300 (Wed, 08 Aug 2007) | 4 lines (closes issue #10335) Reported by: adamgundy Update sip.conf to include another scenario where directrtpsetup will fail. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@78570 65c4cc65-6c06-0410-ace0-fbb531ad65f3 --- diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample index 842c7b513a..c80ca87ccf 100644 --- a/configs/sip.conf.sample +++ b/configs/sip.conf.sample @@ -403,7 +403,8 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; the call directly with media peer-2-peer without re-invites. ; Will not work for video and cases where the callee sends ; RTP payloads and fmtp headers in the 200 OK that does not match the - ; callers INVITE. + ; callers INVITE. This will also fail if canreinvite is enabled when + ; the device is actually behind NAT. ;canreinvite=nonat ; An additional option is to allow media path redirection ; (reinvite) but only when the peer where the media is being