diff options
| author | file <file@f38db490-d61c-443f-a65b-d21fe96a405b> | 2007-02-01 22:26:11 +0000 |
|---|---|---|
| committer | file <file@f38db490-d61c-443f-a65b-d21fe96a405b> | 2007-02-01 22:26:11 +0000 |
| commit | b638ef0a340dcbc851cacc052df4514254ce9628 (patch) | |
| tree | 0e2aba2b97ac02a5a73f5a6e0fa2e378bfaf3c47 /channels | |
| parent | 19a1456f1a20e33d91b05ef13945c53b3feee708 (diff) | |
Merged revisions 53104 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r53104 | file | 2007-02-01 16:24:32 -0600 (Thu, 01 Feb 2007) | 10 lines
Merged revisions 53103 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r53103 | file | 2007-02-01 16:21:56 -0600 (Thu, 01 Feb 2007) | 2 lines
Copy noncodeccapability over to the joint variable so that telephone-event will get transmitted in the sent INVITE.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@53105 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'channels')
| -rw-r--r-- | channels/chan_sip.c | 3 |
1 files changed, 2 insertions, 1 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c index da6092056..4767ac6a1 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -2961,7 +2961,8 @@ static int sip_call(struct ast_channel *ast, char *dest, int timeout) p->callingpres = ast->cid.cid_pres; p->jointcapability = ast_translate_available_formats(p->capability, p->prefcodec); - + p->jointnoncodeccapability = p->noncodeccapability; + /* If there are no audio formats left to offer, punt */ if (!(p->jointcapability & AST_FORMAT_AUDIO_MASK)) { ast_log(LOG_WARNING, "No audio format found to offer. Cancelling call to %s\n", p->username); |