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: https://origsvn.digium.com/svn/asterisk/branches/1.4@53104 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 5ab633cb29c2fbbd67d11dd137bbe77fe0c288a9..22953bbcf049129109ff1558460e96db5d0a56f7 100644 (file)
--- 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)) {