Thank you , yes it is behind a private IP, and I am using advertise option,

i was able to solve my issue using 
modparam("rr", "enable_double_rr", 0)

and then resplying ack o a loose_route() now it seems to work fine.

thanks 
Andres Collazos,



On Tue, Oct 29, 2013 at 5:34 AM, Daniel-Constantin Mierla <miconda@gmail.com> wrote:
Hello,

the config file is not complete, you don't pass the parameters. The ngrep trace that kamailio is forwarding the invite from the private IP but the route header is having the public IP.

You have to set debug=3 in your config and send the log messages here to see what is executed.

As a guess hit - if kamailio is listening on a private ip, being behind a port forwarding nat firewall, you may consider:

listen=udp:privateip:5060 advertise publicip:5060

See Core Cookbook from the kamailio.org wiki for more details about the above parameter.

Cheers,
Daniel

On 10/24/13 12:01 AM, anfecora wrote:
Hi all, can anyone help me to find out what is wrong with my setup, i have an asterisk behind a kamailio, kamailio is proxying all packages  to the outside.

when the call is bridge it gets disconnected after a few seconds, it seems that our voip carrier is sending a bye because we didn't answer to their 200 ok propperly, but as the trace shows we did only that kamailio is answering to the contact header ip not the ip that is sending the ok.

any help is apreciated .

thanks.

my setup 

request_route {

        if (!mf_process_maxfwd_header("10")) {
                sl_send_reply("483","Too Many Hops");
                exit;
        }


        if(is_method("OPTIONS")) {
            # send reply for each options request
            sl_send_reply("200", "ok");
            exit();
         }

   if(method=="BYE") {
   #Account BYE transactions

};


if (method=="CANCEL") {
if (t_check_trans()) t_relay();

exit;
};



 if (loose_route()) {


t_relay();
                exit;
       }


 if (is_method("INVITE")) { 


                record_route();

        }
f (!t_relay_to_udp("3.1.1.1", "5060")) {
sl_reply_error();
exit;
};
exit
};

here is a trace to a call made to a hotel.
i had changed the real ips for obvious reasons.
thanks.


asterisk ip 1.1.1.1
kamailio internal 1.1.1.2
kamailio external 2.0.0.1
Voip Carrier 3.1.1.1
voip contact ip 3.1.1.2



U 2013/10/23 17:26:03.920163 1.1.1.1:5060 -> 1.1.1.2:5060
INVITE sip:23276341079@2.0.0.1 SIP/2.0.
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport.
Max-Forwards: 70.
From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9.
CSeq: 102 INVITE.
User-Agent: Asterisk PBX 1.8.15-cert2.
Date: Wed, 23 Oct 2013 21:26:46 GMT.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Privacy: off.
P-Asserted-Identity: sip:+19812457865@1.1.1.1.
Cisco-Guid: 25655507-3591552378-379709
Content-Type: application/sdp.
Content-Length: 333.
.
v=0.
o=root 519803789 519803789 IN IP4 1.1.1.1.
s=Asterisk PBX 1.8.15-cert2.
c=IN IP4 1.1.1.1.
t=0 0.
m=audio 49926 RTP/AVP 0 18 3 8 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=rtpmap:3 GSM/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
a=sendrecv.


U 2013/10/23 17:26:03.921355 1.1.1.2:5060 -> 1.1.1.1:5060
SIP/2.0 100 trying -- your call is important to us.
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060.
From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9.
CSeq: 102 INVITE.
Server: kamailio (4.0.4 (x86_64/linux)).
Content-Length: 0.
.


U 2013/10/23 17:26:03.921544 1.1.1.2:5060 -> 3.1.1.1:5060
Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bK887c.94fdcd27.0.
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060.
Max-Forwards: 16.
From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9.
CSeq: 102 INVITE.
User-Agent: Asterisk PBX 1.8.15-cert2.
Date: Wed, 23 Oct 2013 21:26:46 GMT.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Privacy: off.
P-Asserted-Identity: sip:+19812457865@1.1.1.1.
Cisco-Guid: 25655507-3591552378-379709
Content-Type: application/sdp.
Content-Length: 333.
.
v=0.
o=root 519803789 519803789 IN IP4 1.1.1.1.
s=Asterisk PBX 1.8.15-cert2.
c=IN IP4 1.1.1.1.
t=0 0.
m=audio 49926 RTP/AVP 0 18 3 8 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=rtpmap:3 GSM/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
a=sendrecv.


U 2013/10/23 17:26:03.955394 3.1.1.1:5060 -> 1.1.1.2:5060
SIP/2.0 100 Giving a try.
Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bK887c.94fdcd27.0.
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060.
From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9.
CSeq: 102 INVITE.
Server: gProxy (1.8.3 (i386/Linux)).
Content-Length: 0.
.


U 2013/10/23 17:26:04.424330 3.1.1.1:5060 -> 1.1.1.2:5060
SIP/2.0 183 Session Progress.
Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bK887c.94fdcd27.0.
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060.
To: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967.
From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9.
CSeq: 102 INVITE.
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE.
Call-Info: <sip:3.1.1.2>;method="NOTIFY;Event=telephone-event;Duration=1000".
Allow-Events: telephone-event.
Content-Type: application/sdp.
Content-Length: 202.
.
v=0.
o=MSXB 4755 8544 IN IP4 3.1.1.2.
s=sip call.
c=IN IP4 204.15.40.111.
t=0 0.
m=audio 33408 RTP/AVP 0 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.


U 2013/10/23 17:26:04.424521 1.1.1.2:5060 -> 1.1.1.1:5060
SIP/2.0 183 Session Progress.
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060.
To: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967.
From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9.
CSeq: 102 INVITE.
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE.
Call-Info: <sip:3.1.1.2>;method="NOTIFY;Event=telephone-event;Duration=1000".
Allow-Events: telephone-event.
Content-Type: application/sdp.
Content-Length: 202.
.
v=0.
o=MSXB 4755 8544 IN IP4 3.1.1.2.
s=sip call.
c=IN IP4 204.15.40.111.
t=0 0.
m=audio 33408 RTP/AVP 0 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.


U 2013/10/23 17:26:16.846067 3.1.1.1:5060 -> 1.1.1.2:5060
SIP/2.0 200 OK.
Session-Expires: 3600;refresher=uas.
Require: timer.
Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bK887c.94fdcd27.0.
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060.
To: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967.
From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9.
CSeq: 102 INVITE.
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE.
Call-Info: <sip:3.1.1.2>;method="NOTIFY;Event=telephone-event;Duration=1000".
Allow-Events: telephone-event.
Content-Type: application/sdp.
Content-Length: 202.
.
v=0.
o=MSXB 4755 8544 IN IP4 3.1.1.2.
s=sip call.
c=IN IP4 204.15.40.111.
t=0 0.
m=audio 33408 RTP/AVP 0 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.


U 2013/10/23 17:26:16.846201 1.1.1.2:5060 -> 1.1.1.1:5060
SIP/2.0 200 OK.
Session-Expires: 3600;refresher=uas.
Require: timer.
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060.
To: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967.
From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9.
CSeq: 102 INVITE.
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE.
Call-Info: <sip:3.1.1.2>;method="NOTIFY;Event=telephone-event;Duration=1000".
Allow-Events: telephone-event.
Content-Type: application/sdp.
Content-Length: 202.
.
v=0.
o=MSXB 4755 8544 IN IP4 3.1.1.2.
s=sip call.
c=IN IP4 204.15.40.111.
t=0 0.
m=audio 33408 RTP/AVP 0 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.


T 2013/10/23 17:26:16.846287 1.1.1.2:55305 -> 10.0.3.54:3306 [AP]
.....insert into acc (method,from_tag,to_tag,callid,sip_code,sip_reason,time,from_uri,to_uri,kekuintid,type_call,dst_ip,carriercode,callmode ) values ('INVITE','as4bc322e9','3591552407-393967','7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060','200','OK','2013-10-23 17:26:16','sip:+19812457865@1.1.1.1','sip:23276341079@2.0.0.1','+19812457865','1.1.1.1','sip:76890723276341079@3.1.1.1:5060','sip:23276341079@2.0.0.1','OUT')

U 2013/10/23 17:26:16.847421 1.1.1.1:5060 -> 1.1.1.2:5060
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK05b1c5df;rport.
Max-Forwards: 70.
From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9.
To: <sip:23276341079@2.0.0.1>;tag=3591552407-393967.
CSeq: 102 ACK.
User-Agent: Asterisk PBX 1.8.15-cert2.
Content-Length: 0.
.


U 2013/10/23 17:26:16.847651 1.1.1.2:5060 -> 3.1.1.2:5060
Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bKcydzigwkX.
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK05b1c5df;rport=5060.
Max-Forwards: 16.
From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9.
To: <sip:76890723276341079@3.1.1.2>;tag=3591552407-393967.
CSeq: 102 ACK.
User-Agent: Asterisk PBX 1.8.15-cert2.
Content-Length: 0.
.


U 2013/10/23 17:26:17.346094 3.1.1.1:5060 -> 1.1.1.2:5060
SIP/2.0 200 OK.
Session-Expires: 3600;refresher=uas.
Require: timer.
Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bK887c.94fdcd27.0.
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060.
To: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967.
From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9.
CSeq: 102 INVITE.
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE.
Call-Info: <sip:3.1.1.2>;method="NOTIFY;Event=telephone-event;Duration=1000".
Allow-Events: telephone-event.
Content-Type: application/sdp.
Content-Length: 202.
.
v=0.
o=MSXB 4755 8544 IN IP4 3.1.1.2.
s=sip call.
c=IN IP4 204.15.40.111.
t=0 0.
m=audio 33408 RTP/AVP 0 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.


U 2013/10/23 17:26:17.346262 1.1.1.2:5060 -> 1.1.1.1:5060
SIP/2.0 200 OK.
Session-Expires: 3600;refresher=uas.
Require: timer.
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060.
To: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967.
From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9.
CSeq: 102 INVITE.
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE.
Call-Info: <sip:3.1.1.2>;method="NOTIFY;Event=telephone-event;Duration=1000".
Allow-Events: telephone-event.
Content-Type: application/sdp.
Content-Length: 202.
.
v=0.
o=MSXB 4755 8544 IN IP4 3.1.1.2.
s=sip call.
c=IN IP4 204.15.40.111.
t=0 0.
m=audio 33408 RTP/AVP 0 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.


U 2013/10/23 17:26:17.349001 1.1.1.1:5060 -> 1.1.1.2:5060
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK675a9141;rport.
Max-Forwards: 70.
From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9.
To: <sip:23276341079@2.0.0.1>;tag=3591552407-393967.
CSeq: 102 ACK.
User-Agent: Asterisk PBX 1.8.15-cert2.
Content-Length: 0.
.


U 2013/10/23 17:26:17.349223 1.1.1.2:5060 -> 3.1.1.2:5060
Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bKcydzigwkX.
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK675a9141;rport=5060.
Max-Forwards: 16.
From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9.
To: <sip:76890723276341079@3.1.1.2>;tag=3591552407-393967.
CSeq: 102 ACK.
User-Agent: Asterisk PBX 1.8.15-cert2.
Content-Length: 0.
.


U 2013/10/23 17:26:18.347584 3.1.1.1:5060 -> 1.1.1.2:5060
SIP/2.0 200 OK.
Session-Expires: 3600;refresher=uas.
Require: timer.
Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bK887c.94fdcd27.0.
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060.
To: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967.
From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9.
CSeq: 102 INVITE.
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE.
Call-Info: <sip:3.1.1.2>;method="NOTIFY;Event=telephone-event;Duration=1000".
Allow-Events: telephone-event.
Content-Type: application/sdp.
Content-Length: 202.
.
v=0.
o=MSXB 4755 8544 IN IP4 3.1.1.2.
s=sip call.
c=IN IP4 204.15.40.111.
t=0 0.
m=audio 33408 RTP/AVP 0 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.


U 2013/10/23 17:26:18.347767 1.1.1.2:5060 -> 1.1.1.1:5060
SIP/2.0 200 OK.
Session-Expires: 3600;refresher=uas.
Require: timer.
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060.
To: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967.
From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9.
CSeq: 102 INVITE.
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE.
Call-Info: <sip:3.1.1.2>;method="NOTIFY;Event=telephone-event;Duration=1000".
Allow-Events: telephone-event.
Content-Type: application/sdp.
Content-Length: 202.
.
v=0.
o=MSXB 4755 8544 IN IP4 3.1.1.2.
s=sip call.
c=IN IP4 204.15.40.111.
t=0 0.
m=audio 33408 RTP/AVP 0 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.


U 2013/10/23 17:26:18.348867 1.1.1.1:5060 -> 1.1.1.2:5060
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK3fc3c548;rport.
Max-Forwards: 70.
From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9.
To: <sip:23276341079@2.0.0.1>;tag=3591552407-393967.
CSeq: 102 ACK.
User-Agent: Asterisk PBX 1.8.15-cert2.
Content-Length: 0.
.


U 2013/10/23 17:26:18.349133 1.1.1.2:5060 -> 3.1.1.2:5060
Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bKcydzigwkX.
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK3fc3c548;rport=5060.
Max-Forwards: 16.
From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9.
To: <sip:76890723276341079@3.1.1.2>;tag=3591552407-393967.
CSeq: 102 ACK.
User-Agent: Asterisk PBX 1.8.15-cert2.
Content-Length: 0.
.


U 2013/10/23 17:26:20.352624 3.1.1.1:5060 -> 1.1.1.2:5060
SIP/2.0 200 OK.
Session-Expires: 3600;refresher=uas.
Require: timer.
Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bK887c.94fdcd27.0.
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060.
To: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967.
From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9.
CSeq: 102 INVITE.
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE.
Call-Info: <sip:3.1.1.2>;method="NOTIFY;Event=telephone-event;Duration=1000".
Allow-Events: telephone-event.
Content-Type: application/sdp.
Content-Length: 202.
.
v=0.
o=MSXB 4755 8544 IN IP4 3.1.1.2.
s=sip call.
c=IN IP4 204.15.40.111.
t=0 0.
m=audio 33408 RTP/AVP 0 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.


U 2013/10/23 17:26:20.353056 1.1.1.2:5060 -> 1.1.1.1:5060
SIP/2.0 200 OK.
Session-Expires: 3600;refresher=uas.
Require: timer.
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060.
To: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967.
From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9.
CSeq: 102 INVITE.
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE.
Call-Info: <sip:3.1.1.2>;method="NOTIFY;Event=telephone-event;Duration=1000".
Allow-Events: telephone-event.
Content-Type: application/sdp.
Content-Length: 202.
.
v=0.
o=MSXB 4755 8544 IN IP4 3.1.1.2.
s=sip call.
c=IN IP4 204.15.40.111.
t=0 0.
m=audio 33408 RTP/AVP 0 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.


U 2013/10/23 17:26:20.354026 1.1.1.1:5060 -> 1.1.1.2:5060
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK280c4e9c;rport.
Max-Forwards: 70.
From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9.
To: <sip:23276341079@2.0.0.1>;tag=3591552407-393967.
CSeq: 102 ACK.
User-Agent: Asterisk PBX 1.8.15-cert2.
Content-Length: 0.
.


U 2013/10/23 17:26:20.354248 1.1.1.2:5060 -> 3.1.1.2:5060
Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bKcydzigwkX.
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK280c4e9c;rport=5060.
Max-Forwards: 16.
From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9.
To: <sip:76890723276341079@3.1.1.2>;tag=3591552407-393967.
CSeq: 102 ACK.
User-Agent: Asterisk PBX 1.8.15-cert2.
Content-Length: 0.
.


U 2013/10/23 17:26:36.355580 3.1.1.1:5060 -> 1.1.1.2:5060
Max-Forwards: 69.
To: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9.
From: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967.
CSeq: 2 BYE.
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE.
Via: SIP/2.0/UDP 3.1.1.1:5060;branch=z9hG4bKce8a.db93afa3.0.
Via: SIP/2.0/UDP 3.1.1.2:5060;branch=z9hG4bK96003ecbb11f5deaf6014235140e6952.
Content-Length: 0.
.


U 2013/10/23 17:26:36.355995 1.1.1.2:5060 -> 1.1.1.1:5060
Max-Forwards: 16.
To: "+19812457865" <sip:176822213@1.1.1.1>;tag=as4bc322e9.
From: 0 <sip:079@3.1.1.1>;tag=3591552407-393967.
CSeq: 2 BYE.
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE.
Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bKce8a.52d22d63.0.
Via: SIP/2.0/UDP 3.1.1.1:5060;branch=z9hG4bKce8a.db93afa3.0.
Via: SIP/2.0/UDP 3.1.1.2:5060;branch=z9hG4bK96003ecbb11f5deaf6014235140e6952.
Content-Length: 0.
.


_______________________________________________
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sr-users@lists.sip-router.org
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-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Trainings - Berlin, Nov 25-28
  - more details about Kamailio trainings at http://www.asipto.com -

_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users