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/1.2@53103 65c4cc65-6c06-0410-ace0-fbb531ad65f3

channels/chan_sip.c

@@ -2083,6 +2083,7 @@ static int sip_call(struct ast_channel *ast, char *dest, int timeout)
 	if ( res != -1 ) {
 		p->callingpres = ast->cid.cid_pres;
 		p->jointcapability = p->capability;
+		p->jointnoncodeccapability = p->noncodeccapability;
 		transmit_invite(p, SIP_INVITE, 1, 2);
 		if (p->maxtime) {
 			/* Initialize auto-congest time */