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: https://origsvn.digium.com/svn/asterisk/trunk@53105 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index da609205616557736536b1cbca7e0b4e114133c1..4767ac6a16021ece0f112212845c4ff48f47d01e 100644 (file)
--- 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);