diff options
| author | Joshua Colp <jcolp@digium.com> | 2009-06-19 15:41:24 +0000 |
|---|---|---|
| committer | Joshua Colp <jcolp@digium.com> | 2009-06-19 15:41:24 +0000 |
| commit | e85296e24493709535857ae10e70d1067b9691ea (patch) | |
| tree | ba2c013a77fabb40e732c6cda3198cdb67835304 /channels | |
| parent | ad0d1bfd9e531b8fd0a7c04238f69850c07824bd (diff) | |
Add support for allowing an RTP engine to decide on whether it is possible for specific formats to be transcoded for an RTP instance.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201902 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'channels')
| -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 97168e7ee..c61db0cbf 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -5338,7 +5338,7 @@ static int sip_call(struct ast_channel *ast, char *dest, int timeout) return res; } p->callingpres = ast->cid.cid_pres; - p->jointcapability = ast_translate_available_formats(p->capability, p->prefcodec); + p->jointcapability = ast_rtp_instance_available_formats(p->rtp, p->capability, p->prefcodec); p->jointnoncodeccapability = p->noncodeccapability; /* If there are no audio formats left to offer, punt */ |