From 57fe6882acaeb4c392bc1516e1dc898423665071 Mon Sep 17 00:00:00 2001
From: Joshua Colp <jcolp@digium.com>
Date: Thu, 1 Feb 2007 22:24:32 +0000
Subject: [PATCH] Merged revisions 53103 via svnmerge from
 https://origsvn.digium.com/svn/asterisk/branches/1.2

........
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.

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53104 65c4cc65-6c06-0410-ace0-fbb531ad65f3
---
 channels/chan_sip.c | 1 +
 1 file changed, 1 insertion(+)

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 5ab633cb29..22953bbcf0 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)) {