[OpenSIPS-Users] Opensips do not route ACK to Asterisk

[t0mmy]

Dear all, 

i'm having scenario where opensips 1.6.3 and asterisk is on the same host.
Opensips binds on port 5060 and asterisk on 5061. 

Opensips handles user registration, nat traversal  and "redirect" callls to
Asterisk PBX (configured as pstn gateway in address table and there is no
407 proxy auth req so they trust each other :))  . Asterisk then via sip
trunk send calls to our  provider. 

 Rtp proxy for nat traversal also running on same machine started and
opensips do not report any errors.   ( started like  rtpproxy -l
public_ip_opensips_and* -s udp:127.0.0.1:7890 -F -u rtpproxy). RTP proxy use
default udp range 10000-35000 and asterisk use 36000-65534 range. 


UA is X-lite and behind nat.
Calls are getting connected but they drop after 30 sec. X-lite recive 200 ok
from * and sends back ACK. But ACK not getting to Asterisk. Asterisk reports
retransmit timeout error.  



 I'm think that problem is like they say in sip-retransmit file 


- A SIP middlebox (SBC) that rewrites contact: headers 
  so that we can't reach the other side with our reply 
  or the ACK. 
- A badly configured SIP proxy that forgets to add 
  record-route headers to make sure that signalling works. 

When X-lite is in lan where is opensips error do not exists.  Calls are
working perfect!


Here is 200 ok that asterisk keep retransmiting:

^[[0KRetransmitting #2 (no NAT) to 192.168.1.42:5060: 
SIP/2.0 200 OK 
v: SIP/2.0/UDP
192.168.1.42;branch=z9hG4bKe31e.f8aa9df6.0;received=192.168.1.42 
v: SIP/2.0/UDP
PUBLIC_IP_OF_UA:59788;rport=59788;received=PUBLIC_IP_OF_UA;branch=z9hG4bK-d8754z-c754d54c1f044c74-1---d8754z- 
Record-Route: <sip:192.168.1.42;lr=on> 
f: "tommy2" <sip:6557181066 at PUBLIC_IP_OF_OPENSIPS_AND*>;tag=740b0d14 
t: "7890100" <sip:7890100 at 8PUBLIC_IP_OF_OPENSIPS_AND*>;tag=as57556733 
i: YjNjZThlNjljMjk2ODE5MmU1NDNiNTJhMTY5ZDg2MWQ. 
CSeq: 2 INVITE 
Server: Asterisk PBX 
^[[0Kllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH 
k: replaces, timer 
m: <sip:7890100 at 192.168.1.42:5061> 
c: application/sdp 
l: 261 

In opensips.cfg file i have this 2 section that involves ACK:

if ( is_method("ACK") ) {
				

					if ( t_check_trans() ) {
					# non loose-route, but stateful ACK; must be an ACK after 
					# a 487 or e.g. 404 from upstream server
					t_relay();
					exit;
				} else {
					# ACK without matching transaction ->
					# ignore and discard
					exit;
				}
			}
			sl_send_reply("404","Not here");
		}
		exit;
	}

and :


# preloaded route checking
	if (loose_route()) {
		xlog("L_ERR",
		"Attempt to route with preloaded Route's [$fu/$tu/$ru/$ci]");
		if (!is_method("ACK"))
			sl_send_reply("403","Preload Route denied");
		append_hf("P-hint: rr-enforced\r\n"); 
		route(1);
	}


i'm sending calls this way :

  
route[4] {
  #---- PSTN route ----#
  rewritehostport("192.168.1.42:5061");
	route(1);
	exit;
}



 I hope there is solution for this and what to thanks in advanced anyone who
try to help . 
Please help!!!!

Cheers 
  
T0mmy
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