[SR-Users] Help With kamailio not responding ack propperly.

23 Oct 2013

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 properly, 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 appreciated .

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.
To: <sip:23276341079@2.0.0.1>.
Contact: <sip:+19812457865@1.1.1.1:5060>.
Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060.
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.
To: <sip:23276341079@2.0.0.1>.
Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060.
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
INVITE sip:76890723276341079@3.1.1.1:5060 SIP/2.0.
Record-Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>.
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.
To: <sip:76890723276341079@3.1.1.1>.
Contact: <sip:+19812457865@1.1.1.1:5060>.
Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060.
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.
To: <sip:76890723276341079@3.1.1.1>.
Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060.
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.
Record-Route: <sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>.
Record-Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>.
To: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967.
From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9.
Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060.
CSeq: 102 INVITE.
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER,
SUBSCRIBE, PRACK, UPDATE.
Contact: <sip:76890723276341079@3.1.1.2:5060>.
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.
Record-Route: <sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>.
Record-Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>.
To: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967.
From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9.
Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060.
CSeq: 102 INVITE.
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER,
SUBSCRIBE, PRACK, UPDATE.
Contact: <sip:76890723276341079@3.1.1.2:5060>.
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.
Record-Route: <sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>.
Record-Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>.
To: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967.
From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9.
Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060.
CSeq: 102 INVITE.
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER,
SUBSCRIBE, PRACK, UPDATE.
Contact: <sip:76890723276341079@3.1.1.2:5060>.
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.
Record-Route: <sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>.
Record-Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>.
To: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967.
From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9.
Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060.
CSeq: 102 INVITE.
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER,
SUBSCRIBE, PRACK, UPDATE.
Contact: <sip:76890723276341079@3.1.1.2:5060>.
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
ACK sip:76890723276341079@3.1.1.2:5060 SIP/2.0.
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK05b1c5df;rport.
Route:
<sip:2.0.0.1;lr=on;ftag=as4bc322e9>,<sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>.
Max-Forwards: 70.
From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9.
To: <sip:23276341079@2.0.0.1>;tag=3591552407-393967.
Contact: <sip:+19812457865@1.1.1.1:5060>.
Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060.
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
ACK sip:76890723276341079@3.1.1.2:5060 SIP/2.0.
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.
Route:
<sip:2.0.0.1;lr=on;ftag=as4bc322e9>,<sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>.
Max-Forwards: 16.
From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9.
To: <sip:76890723276341079@3.1.1.2>;tag=3591552407-393967.
Contact: <sip:+19812457865@1.1.1.1:5060>.
Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060.
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.
Record-Route: <sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>.
Record-Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>.
To: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967.
From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9.
Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060.
CSeq: 102 INVITE.
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER,
SUBSCRIBE, PRACK, UPDATE.
Contact: <sip:76890723276341079@3.1.1.2:5060>.
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.
Record-Route: <sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>.
Record-Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>.
To: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967.
From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9.
Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060.
CSeq: 102 INVITE.
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER,
SUBSCRIBE, PRACK, UPDATE.
Contact: <sip:76890723276341079@3.1.1.2:5060>.
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
ACK sip:76890723276341079@3.1.1.2:5060 SIP/2.0.
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK675a9141;rport.
Route:
<sip:2.0.0.1;lr=on;ftag=as4bc322e9>,<sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>.
Max-Forwards: 70.
From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9.
To: <sip:23276341079@2.0.0.1>;tag=3591552407-393967.
Contact: <sip:+19812457865@1.1.1.1:5060>.
Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060.
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
ACK sip:76890723276341079@3.1.1.2:5060 SIP/2.0.
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.
Route:
<sip:2.0.0.1;lr=on;ftag=as4bc322e9>,<sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>.
Max-Forwards: 16.
From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9.
To: <sip:76890723276341079@3.1.1.2>;tag=3591552407-393967.
Contact: <sip:+19812457865@1.1.1.1:5060>.
Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060.
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.
Record-Route: <sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>.
Record-Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>.
To: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967.
From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9.
Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060.
CSeq: 102 INVITE.
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER,
SUBSCRIBE, PRACK, UPDATE.
Contact: <sip:76890723276341079@3.1.1.2:5060>.
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.
Record-Route: <sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>.
Record-Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>.
To: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967.
From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9.
Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060.
CSeq: 102 INVITE.
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER,
SUBSCRIBE, PRACK, UPDATE.
Contact: <sip:76890723276341079@3.1.1.2:5060>.
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
ACK sip:76890723276341079@3.1.1.2:5060 SIP/2.0.
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK3fc3c548;rport.
Route:
<sip:2.0.0.1;lr=on;ftag=as4bc322e9>,<sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>.
Max-Forwards: 70.
From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9.
To: <sip:23276341079@2.0.0.1>;tag=3591552407-393967.
Contact: <sip:+19812457865@1.1.1.1:5060>.
Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060.
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
ACK sip:76890723276341079@3.1.1.2:5060 SIP/2.0.
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.
Route:
<sip:2.0.0.1;lr=on;ftag=as4bc322e9>,<sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>.
Max-Forwards: 16.
From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9.
To: <sip:76890723276341079@3.1.1.2>;tag=3591552407-393967.
Contact: <sip:+19812457865@1.1.1.1:5060>.
Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060.
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.
Record-Route: <sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>.
Record-Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>.
To: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967.
From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9.
Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060.
CSeq: 102 INVITE.
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER,
SUBSCRIBE, PRACK, UPDATE.
Contact: <sip:76890723276341079@3.1.1.2:5060>.
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.
Record-Route: <sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>.
Record-Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>.
To: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967.
From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9.
Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060.
CSeq: 102 INVITE.
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER,
SUBSCRIBE, PRACK, UPDATE.
Contact: <sip:76890723276341079@3.1.1.2:5060>.
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
ACK sip:76890723276341079@3.1.1.2:5060 SIP/2.0.
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK280c4e9c;rport.
Route:
<sip:2.0.0.1;lr=on;ftag=as4bc322e9>,<sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>.
Max-Forwards: 70.
From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9.
To: <sip:23276341079@2.0.0.1>;tag=3591552407-393967.
Contact: <sip:+19812457865@1.1.1.1:5060>.
Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060.
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
ACK sip:76890723276341079@3.1.1.2:5060 SIP/2.0.
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.
Route:
<sip:2.0.0.1;lr=on;ftag=as4bc322e9>,<sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>.
Max-Forwards: 16.
From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9.
To: <sip:76890723276341079@3.1.1.2>;tag=3591552407-393967.
Contact: <sip:+19812457865@1.1.1.1:5060>.
Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060.
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
BYE sip:+19812457865@1.1.1.1:5060 SIP/2.0.
Max-Forwards: 69.
Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>.
To: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9.
From: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967.
Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060.
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.
Contact: <sip:76890723276341079@3.1.1.2:5060>.
Content-Length: 0.
.

U 2013/10/23 17:26:36.355995 1.1.1.2:5060 -> 1.1.1.1:5060
BYE sip:+19812457865@1.1.1.1:5060 SIP/2.0.
Max-Forwards: 16.
Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>.
To: "+19812457865" <sip:176822213@1.1.1.1>;tag=as4bc322e9.
From: 0 <sip:079@3.1.1.1>;tag=3591552407-393967.
Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060.
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.
Contact: <sip:76890723276341079@3.1.1.2:5060>.
Content-Length: 0.
.

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[SR-Users] Help With kamailio not responding ack propperly.