Don't recreate stuff on authentication (so it works on reinvites)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@6517 65c4cc65-6c06-0410-ace0-fbb531ad65f3

channels/chan_sip.c

@@ -1978,7 +1978,7 @@ 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;
-		transmit_invite(p, SIP_INVITE, 1, 1);
+		transmit_invite(p, SIP_INVITE, 1, 2);
 		if (p->maxtime) {
 			/* Initialize auto-congest time */
 			p->initid = ast_sched_add(sched, p->maxtime * 4, auto_congest, p);
@@ -4529,7 +4529,10 @@ static int transmit_invite(struct sip_pvt *p, int sipmethod, int sdp, int init)
 		/* Bump branch even on initial requests */
 		p->branch ^= rand();
 		build_via(p, p->via, sizeof(p->via));
+		if (init > 1)
 			initreqprep(&req, p, sipmethod);
+		else
+			reqprep(&req, p, sipmethod, 0, 1);
 	} else
 		reqprep(&req, p, sipmethod, 0, 1);
 		
@@ -10654,9 +10657,9 @@ static int sip_poke_peer(struct sip_peer *peer)
 	ast_set_flag(p, SIP_OUTGOING);
 #ifdef VOCAL_DATA_HACK
 	ast_copy_string(p->username, "__VOCAL_DATA_SHOULD_READ_THE_SIP_SPEC__", sizeof(p->username));
-	transmit_invite(p, SIP_INVITE, 0, 1);
+	transmit_invite(p, SIP_INVITE, 0, 2);
 #else
-	transmit_invite(p, SIP_OPTIONS, 0, 1);
+	transmit_invite(p, SIP_OPTIONS, 0, 2);
 #endif
 	gettimeofday(&peer->ps, NULL);
 	peer->pokeexpire = ast_sched_add(sched, DEFAULT_MAXMS * 2, sip_poke_noanswer, peer);