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| author | file <file@f38db490-d61c-443f-a65b-d21fe96a405b> | 2007-02-01 22:24:32 +0000 |
|---|---|---|
| committer | file <file@f38db490-d61c-443f-a65b-d21fe96a405b> | 2007-02-01 22:24:32 +0000 |
| commit | 747f0c06670df4e7423be767e6919c91c5c9635a (patch) | |
| tree | 0d3238b44ff556a1b41fececd71983b4cb11d725 /channels | |
| parent | 7bacd30ba42fa80be69697f359132a22e82cf5d4 (diff) | |
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/branches/1.4@53104 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'channels')
| -rw-r--r-- | channels/chan_sip.c | 1 |
1 files changed, 1 insertions, 0 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c index 5ab633cb2..22953bbcf 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -2831,6 +2831,7 @@ static int sip_call(struct ast_channel *ast, char *dest, int timeout) if ( res != -1 ) { 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)) { |