diff options
| author | Mark Michelson <mmichelson@digium.com> | 2008-09-09 16:20:41 +0000 |
|---|---|---|
| committer | Mark Michelson <mmichelson@digium.com> | 2008-09-09 16:20:41 +0000 |
| commit | 01b2894d2eebfbdfdb61960272cbc0b1bfef1e4c (patch) | |
| tree | 63b10bf48c43a290d14dbe1b75a7761cb941fbb2 /channels/chan_sip.c | |
| parent | 4f077691bca39763cc0b6bf304de588068244c73 (diff) | |
Merged revisions 142079 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r142079 | mmichelson | 2008-09-09 11:19:17 -0500 (Tue, 09 Sep 2008) | 21 lines
When determining if codecs used by SIP peers allow
the media to be natively bridged, use the jointcapability
instead of the peercapability.
It seems that the intent of using the peercapability was to
expand the choice of codecs for the call to increase the
chances of being able to native bridge the channels. The
problem is that if a codec were settled on for the native
bridge and that wasn't a codec that was configured to be used
by Asterisk for that peer, then Asterisk would send a
REINVITE with no codecs in the SDP which is a bug no matter
how you slice it.
(closes issue #13076)
Reported by: ramonpeek
Patches:
13076.patch uploaded by putnopvut (license 60)
Tested by: tbelder
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@142080 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'channels/chan_sip.c')
| -rw-r--r-- | channels/chan_sip.c | 2 |
1 files changed, 1 insertions, 1 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c index bcecd54aa..40054a305 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -22402,7 +22402,7 @@ static int sip_sipredirect(struct sip_pvt *p, const char *dest) static int sip_get_codec(struct ast_channel *chan) { struct sip_pvt *p = chan->tech_pvt; - return p->peercapability ? p->peercapability : p->capability; + return p->jointcapability ? p->jointcapability : p->capability; } /*! \brief Send a poke to all known peers |