[SR-Users] AT&T and UAC cancel after 200 OK

Thu Nov 6 12:44:12 CET 2014

Hello. I use UAC module for trunks and have trouble when call to mobile
endpoint of AT&T provider.

When endpoint pickup call ans 200 OK reply come into kamailio, kamailio
send CANCEL. so at the endpoint I see than call is dropped and at kamailio
client I hear voicemail speech form AT&T endpoint.

Provider say that his trace is OK...

My sip trace bellow:

IP 10.0.2.4.5068 > 34.43.4.23.5060: UDP, length 1129
E....... at .l.
...6........q..INVITE sip:98765432100 at phone.myprovider.com SIP/2.0
Record-Route: <sip:sip.myservice.info:5068;nat=yes;ftag=as288f6857;lr=on>
Via: SIP/2.0/UDP sip.myservice.info:5068
;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.0
Via: SIP/2.0/UDP 10.0.1.41:50600
;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600
Max-Forwards: 70
From: "John Connor" <sip:12345678 at phone.myprovider.com>;tag=as288f6857
To: <sip:98765432100 at phone.myprovider.com>
Contact:<sip:12345678 at sip.myservice.info:5068>
Call-ID: 6717947d5e5f445314ba762f154be72f at 10.0.1.41:50600
CSeq: 102 INVITE
User-Agent: Asterisk PBX 12.5.0
Date: Mon, 03 Nov 2014 20:21:03 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 314

v=0
o=root 1363870848 1363870848 IN IP4 68.34.22.11
s=Asterisk PBX 12.5.0
c=IN IP4 68.34.22.11
t=0 0
m=audio 30030 RTP/AVP 8 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
a=rtcp:30031

IP 34.43.4.23.5060 > 10.0.2.4.5068: UDP, length 651
E.......*.Z.6...
.........}.SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP sip.myservice.info:5068
;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.0;received=68.34.22.11
Via: SIP/2.0/UDP 10.0.1.41:50600
;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600
From: "John Connor" <sip:12345678 at phone.myprovider.com>;tag=as288f6857
To: <sip:98765432100 at phone.myprovider.com
>;tag=04e2a294d0728b89107e081a19babab4.43a7
Call-ID: 6717947d5e5f445314ba762f154be72f at 10.0.1.41:50600
CSeq: 102 INVITE
Proxy-Authenticate: Digest realm="phone.myprovider.com",
nonce="VFfk3VRX47HuWpmSTOv4DPvdulTv2Eeq", qop="auth"
Server: kamailio (4.1.2 (x86_64/linux))
Content-Length: 0

IP 10.0.2.4.5068 > 34.43.4.23.5060: UDP, length 410
E....... at .o.
...6.........j%ACK sip:98765432100 at phone.myprovider.com SIP/2.0
Via: SIP/2.0/UDP sip.myservice.info:5068
;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.0
Max-Forwards: 70
From: "John Connor" <sip:12345678 at phone.myprovider.com>;tag=as288f6857
To: <sip:98765432100 at phone.myprovider.com
>;tag=04e2a294d0728b89107e081a19babab4.43a7
Call-ID: 6717947d5e5f445314ba762f154be72f at 10.0.1.41:50600
CSeq: 102 ACK
Content-Length: 0

IP 10.0.2.4.5068 > 34.43.4.23.5060: UDP, length 1402
E....... at .k.
...6..........`INVITE sip:98765432100 at phone.myprovider.com SIP/2.0
Record-Route: <sip:sip.myservice.info:5068;nat=yes;ftag=as288f6857;lr=on>
Via: SIP/2.0/UDP sip.myservice.info:5068
;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.1.cs102
Via: SIP/2.0/UDP 10.0.1.41:50600
;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600
Max-Forwards: 70
From: "John Connor" <sip:12345678 at phone.myprovider.com>;tag=as288f6857
To: <sip:98765432100 at phone.myprovider.com>
Contact:<sip:12345678 at sip.myservice.info:5068>
Call-ID: 6717947d5e5f445314ba762f154be72f at 10.0.1.41:50600
CSeq: 103 INVITE
User-Agent: Asterisk PBX 12.5.0
Date: Mon, 03 Nov 2014 20:21:03 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 314
Proxy-Authorization: Digest username="12345678", realm="phone.myprovider.com",
nonce="VFfk3VRX47HuWpmSTOv4DPvdulTv2Eeq", uri="
sip:98765432100 at phone.myprovider.com", qop=auth, nc=00000001,
cnonce="2107612791", response="51db231dedb4a74396447729a192a8d3",
algorithm=MD5

v=0
o=root 1363870848 1363870848 IN IP4 68.34.22.11
s=Asterisk PBX 12.5.0
c=IN IP4 68.34.22.11
t=0 0
m=audio 30030 RTP/AVP 8 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
a=rtcp:30031

IP 10.0.2.4.5068 > 34.43.4.23.5060: UDP, length 1402
E....... at .k.
...6.........._INVITE sip:98765432100 at phone.myprovider.com SIP/2.0
Record-Route: <sip:sip.myservice.info:5068;nat=yes;ftag=as288f6857;lr=on>
Via: SIP/2.0/UDP sip.myservice.info:5068
;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.2.cs102
Via: SIP/2.0/UDP 10.0.1.41:50600
;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600
Max-Forwards: 70
From: "John Connor" <sip:12345678 at phone.myprovider.com>;tag=as288f6857
To: <sip:98765432100 at phone.myprovider.com>
Contact:<sip:12345678 at sip.myservice.info:5068>
Call-ID: 6717947d5e5f445314ba762f154be72f at 10.0.1.41:50600
CSeq: 104 INVITE
User-Agent: Asterisk PBX 12.5.0
Date: Mon, 03 Nov 2014 20:21:03 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 314
Proxy-Authorization: Digest username="12345678", realm="phone.myprovider.com",
nonce="VFfk3VRX47HuWpmSTOv4DPvdulTv2Eeq", uri="
sip:98765432100 at phone.myprovider.com", qop=auth, nc=00000001,
cnonce="2107612791", response="51db231dedb4a74396447729a192a8d3",
algorithm=MD5

v=0
o=root 1363870848 1363870848 IN IP4 68.34.22.11
s=Asterisk PBX 12.5.0
c=IN IP4 68.34.22.11
t=0 0
m=audio 30030 RTP/AVP 8 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
a=rtcp:30031

IP 34.43.4.23.5060 > 10.0.2.4.5068: UDP, length 529
E..-....*.[.6...
.........xiSIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP sip.myservice.info:5068
;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.2.cs102;rport=1024;received=68.34.22.11
Via: SIP/2.0/UDP 10.0.1.41:50600
;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600
From: "John Connor" <sip:12345678 at phone.myprovider.com>;tag=as288f6857
To: <sip:98765432100 at phone.myprovider.com>
Call-ID: 6717947d5e5f445314ba762f154be72f at 10.0.1.41:50600
CSeq: 104 INVITE
Server: kamailio (4.1.2 (x86_64/linux))
Content-Length: 0

IP 34.43.4.23.5060 > 10.0.2.4.5068: UDP, length 529
E..-....*.[.6...
.........xkSIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP sip.myservice.info:5068
;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.1.cs102;rport=1024;received=68.34.22.11
Via: SIP/2.0/UDP 10.0.1.41:50600
;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600
From: "John Connor" <sip:12345678 at phone.myprovider.com>;tag=as288f6857
To: <sip:98765432100 at phone.myprovider.com>
Call-ID: 6717947d5e5f445314ba762f154be72f at 10.0.1.41:50600
CSeq: 103 INVITE
Server: kamailio (4.1.2 (x86_64/linux))
Content-Length: 0

IP 34.43.4.23.5060 > 10.0.2.4.5068: UDP, length 998
E.......*.Y16...
.........^.SIP/2.0 180 Ringing
Via: SIP/2.0/UDP sip.myservice.info:5068
;rport=1024;received=68.34.22.11;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.2.cs102
Via: SIP/2.0/UDP 10.0.1.41:50600
;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600
Record-Route: <sip:34.43.4.23;lr=on;ftag=as288f6857;did=229.d4f>
Record-Route: <sip:sip.myservice.info:5068;nat=yes;ftag=as288f6857;lr=on>
From: "John Connor" <sip:12345678 at phone.myprovider.com>;tag=as288f6857
To: <sip:98765432100 at phone.myprovider.com>;tag=etQmF8eNZ7NKm
Call-ID: 6717947d5e5f445314ba762f154be72f at 10.0.1.41:50600
CSeq: 104 INVITE
Contact: <sip:98765432100 at 98.129.251.83:5060;transport=udp>
User-Agent: myprovider
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER,
NOTIFY
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Length: 0
Remote-Party-ID: "98765432100" <sip:98765432100 at phone.myprovider.com
>;party=calling;privacy=off;screen=no

IP 34.43.4.23.5060 > 10.0.2.4.5068: UDP, length 986
E.......*.Y<6...
...........SIP/2.0 180 Ringing
Via: SIP/2.0/UDP sip.myservice.info:5068
;rport=1024;received=68.34.22.11;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.1.cs102
Via: SIP/2.0/UDP 10.0.1.41:50600
;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600
Record-Route: <sip:34.43.4.23;lr=on;ftag=as288f6857>
Record-Route: <sip:sip.myservice.info:5068;nat=yes;ftag=as288f6857;lr=on>
From: "John Connor" <sip:12345678 at phone.myprovider.com>;tag=as288f6857
To: <sip:98765432100 at phone.myprovider.com>;tag=UrBQrZp9vSm3p
Call-ID: 6717947d5e5f445314ba762f154be72f at 10.0.1.41:50600
CSeq: 103 INVITE
Contact: <sip:98765432100 at 173.203.60.50:5060;transport=udp>
User-Agent: myprovider
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER,
NOTIFY
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Length: 0
Remote-Party-ID: "98765432100" <sip:98765432100 at phone.myprovider.com
>;party=calling;privacy=off;screen=no

IP 34.43.4.23.5060 > 10.0.2.4.5068: UDP, length 1294
E..*....*.X.6...
.........:.SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP sip.myservice.info:5068
;rport=1024;received=68.34.22.11;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.2.cs102
Via: SIP/2.0/UDP 10.0.1.41:50600
;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600
Record-Route: <sip:34.43.4.23;lr=on;ftag=as288f6857;did=229.d4f>
Record-Route: <sip:sip.myservice.info:5068;nat=yes;ftag=as288f6857;lr=on>
From: "John Connor" <sip:12345678 at phone.myprovider.com>;tag=as288f6857
To: <sip:98765432100 at phone.myprovider.com>;tag=etQmF8eNZ7NKm
Call-ID: 6717947d5e5f445314ba762f154be72f at 10.0.1.41:50600
CSeq: 104 INVITE
Contact: <sip:98765432100 at 98.129.251.83:5060;transport=udp>
User-Agent: myprovider
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER,
NOTIFY
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 249
Remote-Party-ID: "98765432100" <sip:98765432100 at phone.myprovider.com
>;party=calling;privacy=off;screen=no

v=0
o=FreeSWITCH 1415023959 1415023960 IN IP4 98.129.251.83
s=FreeSWITCH
c=IN IP4 98.129.251.83
t=0 0
m=audio 22108 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20

IP 34.43.4.23.5060 > 10.0.2.4.5068: UDP, length 1282
E.......*.X.6...
........
..SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP sip.myservice.info:5068
;rport=1024;received=68.34.22.11;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.1.cs102
Via: SIP/2.0/UDP 10.0.1.41:50600
;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600
Record-Route: <sip:34.43.4.23;lr=on;ftag=as288f6857>
Record-Route: <sip:sip.myservice.info:5068;nat=yes;ftag=as288f6857;lr=on>
From: "John Connor" <sip:12345678 at phone.myprovider.com>;tag=as288f6857
To: <sip:98765432100 at phone.myprovider.com>;tag=UrBQrZp9vSm3p
Call-ID: 6717947d5e5f445314ba762f154be72f at 10.0.1.41:50600
CSeq: 103 INVITE
Contact: <sip:98765432100 at 173.203.60.50:5060;transport=udp>
User-Agent: myprovider
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER,
NOTIFY
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 249
Remote-Party-ID: "98765432100" <sip:98765432100 at phone.myprovider.com
>;party=calling;privacy=off;screen=no

v=0
o=FreeSWITCH 1415020637 1415020638 IN IP4 173.203.60.50
s=FreeSWITCH
c=IN IP4 173.203.60.50
t=0 0
m=audio 25430 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20

IP 34.43.4.23.5060 > 10.0.2.4.5068: UDP, length 1428
E.......*.W.6...
.........H.SIP/2.0 200 OK
Via: SIP/2.0/UDP sip.myservice.info:5068
;rport=1024;received=68.34.22.11;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.2.cs102
Via: SIP/2.0/UDP 10.0.1.41:50600
;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600
Record-Route: <sip:34.43.4.23;lr=on;ftag=as288f6857;did=229.d4f>
Record-Route: <sip:sip.myservice.info:5068;nat=yes;ftag=as288f6857;lr=on>
From: "John Connor" <sip:12345678 at phone.myprovider.com>;tag=as288f6857
To: <sip:98765432100 at phone.myprovider.com>;tag=etQmF8eNZ7NKm
Call-ID: 6717947d5e5f445314ba762f154be72f at 10.0.1.41:50600
CSeq: 104 INVITE
Contact: <sip:98765432100 at 98.129.251.83:5060;transport=udp>
User-Agent: myprovider
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER,
NOTIFY
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 249
X-myproviderOutboundGateway: sip:+98765432100 at 67.231.8.195
X-myproviderOutboundCarrierID: 29927192767592
X-myproviderCarrierRate: 0.00900
X-myproviderCloudRate: 0.00300
Remote-Party-ID: "Outbound Call" <sip:5060 at phone.myprovider.com
>;party=calling;privacy=off;screen=no

v=0
o=FreeSWITCH 1415023959 1415023960 IN IP4 98.129.251.83
s=FreeSWITCH
c=IN IP4 98.129.251.83
t=0 0
m=audio 22108 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20

IP 10.0.2.4.5068 > 34.43.4.23.5060: UDP, length 412
E....... at .o.
...6.........j.CANCEL sip:98765432100 at phone.myprovider.com SIP/2.0
Via: SIP/2.0/UDP sip.myservice.info:5068
;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.1.cs102
Max-Forwards: 70
From: "John Connor" <sip:12345678 at phone.myprovider.com>;tag=as288f6857
To: <sip:98765432100 at ints.myservice.info:5068>
Call-ID: 6717947d5e5f445314ba762f154be72f at 10.0.1.41:50600
CSeq: 105 CANCEL
Content-Length: 0
Reason: SIP;cause=200

IP 10.0.2.4.5068 > 34.43.4.23.5060: UDP, length 647
E....... at .n.
...6...........ACK sip:98765432100 at 98.129.251.83:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP sip.myservice.info:5068
;branch=z9hG4bK3f75.0ec383b6dc8fbb1861dfe644f92ee9ca.0.cs102
Via: SIP/2.0/UDP 10.0.1.41:50600
;received=10.0.1.41;branch=z9hG4bK010bb317;rport=50600
Route: <sip:34.43.4.23;lr=on;ftag=as288f6857;did=229.d4f>
Max-Forwards: 70
From: "John Connor" <sip:skynet.device-3 at 10.0.1.41:50600>;tag=as288f6857
To: <sip:98765432100 at ints.myservice.info:5068>;tag=etQmF8eNZ7NKm
Contact: <sip:skynet.device-3 at 10.0.1.41:50600>
Call-ID: 6717947d5e5f445314ba762f154be72f at 10.0.1.41:50600
CSeq: 105 ACK
User-Agent: Asterisk PBX 12.5.0
Content-Length: 0

IP 34.43.4.23.5060 > 10.0.2.4.5068: UDP, length 1428
E.......*.W~6...
.........H.SIP/2.0 200 OK
Via: SIP/2.0/UDP sip.myservice.info:5068
;rport=1024;received=68.34.22.11;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.2.cs102
Via: SIP/2.0/UDP 10.0.1.41:50600
;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600
Record-Route: <sip:34.43.4.23;lr=on;ftag=as288f6857;did=229.d4f>
Record-Route: <sip:sip.myservice.info:5068;nat=yes;ftag=as288f6857;lr=on>
From: "John Connor" <sip:12345678 at phone.myprovider.com>;tag=as288f6857
To: <sip:98765432100 at phone.myprovider.com>;tag=etQmF8eNZ7NKm
Call-ID: 6717947d5e5f445314ba762f154be72f at 10.0.1.41:50600
CSeq: 104 INVITE
Contact: <sip:98765432100 at 98.129.251.83:5060;transport=udp>
User-Agent: myprovider
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER,
NOTIFY
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 249
X-myproviderOutboundGateway: sip:+98765432100 at 67.231.8.195
X-myproviderOutboundCarrierID: 29927192767592
X-myproviderCarrierRate: 0.00900
X-myproviderCloudRate: 0.00300
Remote-Party-ID: "Outbound Call" <sip:5060 at phone.myprovider.com
>;party=calling;privacy=off;screen=no

v=0
o=FreeSWITCH 1415023959 1415023960 IN IP4 98.129.251.83
s=FreeSWITCH
c=IN IP4 98.129.251.83
t=0 0
m=audio 22108 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20

IP 10.0.2.4.5068 > 34.43.4.23.5060: UDP, length 647
E....... at .n.
...6..........jACK sip:98765432100 at 98.129.251.83:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP sip.myservice.info:5068
;branch=z9hG4bK3f75.6a1e71bf5fcd88f137733279fa31770c.0.cs102
Via: SIP/2.0/UDP 10.0.1.41:50600
;received=10.0.1.41;branch=z9hG4bK1f95579c;rport=50600
Route: <sip:34.43.4.23;lr=on;ftag=as288f6857;did=229.d4f>
Max-Forwards: 70
From: "John Connor" <sip:skynet.device-3 at 10.0.1.41:50600>;tag=as288f6857
To: <sip:98765432100 at ints.myservice.info:5068>;tag=etQmF8eNZ7NKm
Contact: <sip:skynet.device-3 at 10.0.1.41:50600>
Call-ID: 6717947d5e5f445314ba762f154be72f at 10.0.1.41:50600
CSeq: 105 ACK
User-Agent: Asterisk PBX 12.5.0
Content-Length: 0

IP 10.0.2.4.5068 > 34.43.4.23.5060: UDP, length 406
E....... at .o.
...6.........?xCANCEL sip:98765432100 at phone.myprovider.com SIP/2.0
Via: SIP/2.0/UDP sip.myservice.info:5068
;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.1
Max-Forwards: 70
From: "John Connor" <sip:12345678 at phone.myprovider.com>;tag=as288f6857
To: <sip:98765432100 at ints.myservice.info:5068>
Call-ID: 6717947d5e5f445314ba762f154be72f at 10.0.1.41:50600
CSeq: 102 CANCEL
Content-Length: 0
Reason: SIP;cause=200

IP 34.43.4.23.5060 > 10.0.2.4.5068: UDP, length 246
E.......*.\.6...
...........OPTIONS sip:68.34.22.11:1024 SIP/2.0
Via: SIP/2.0/UDP 10.202.129.254:5060;branch=0
From: sip:keepalive at myprovider.com;tag=29ebf5d
To: sip:68.34.22.11:1024
Call-ID: 6360a4bb-53bebf7-5686ca6 at 10.202.129.254
CSeq: 1 OPTIONS
Content-Length: 0

IP 10.0.2.4.5068 > 34.43.4.23.5060: UDP, length 330
E..f.... at .p.
...6........R.<SIP/2.0 405 Method not allowed
Via: SIP/2.0/UDP 10.202.129.254:5060;branch=0;rport=5060;received=34.43.4.23
From: sip:keepalive at myprovider.com;tag=29ebf5d
To: sip:68.34.22.11:1024;tag=c6894543322e2a4942d921e4298ce904.1ec7
Call-ID: 6360a4bb-53bebf7-5686ca6 at 10.202.129.254
CSeq: 1 OPTIONS
Server: MS Lync
Content-Length: 0

IP 34.43.4.23.5060 > 10.0.2.4.5068: UDP, length 1428
E.......*.W|6...
.........H.SIP/2.0 200 OK
Via: SIP/2.0/UDP sip.myservice.info:5068
;rport=1024;received=68.34.22.11;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.2.cs102
Via: SIP/2.0/UDP 10.0.1.41:50600
;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600
Record-Route: <sip:34.43.4.23;lr=on;ftag=as288f6857;did=229.d4f>
Record-Route: <sip:sip.myservice.info:5068;nat=yes;ftag=as288f6857;lr=on>
From: "John Connor" <sip:12345678 at phone.myprovider.com>;tag=as288f6857
To: <sip:98765432100 at phone.myprovider.com>;tag=etQmF8eNZ7NKm
Call-ID: 6717947d5e5f445314ba762f154be72f at 10.0.1.41:50600
CSeq: 104 INVITE
Contact: <sip:98765432100 at 98.129.251.83:5060;transport=udp>
User-Agent: myprovider
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER,
NOTIFY
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 249
X-myproviderOutboundGateway: sip:+98765432100 at 67.231.8.195
X-myproviderOutboundCarrierID: 29927192767592
X-myproviderCarrierRate: 0.00900
X-myproviderCloudRate: 0.00300
Remote-Party-ID: "Outbound Call" <sip:5060 at phone.myprovider.com
>;party=calling;privacy=off;screen=no

v=0
o=FreeSWITCH 1415023959 1415023960 IN IP4 98.129.251.83
s=FreeSWITCH
c=IN IP4 98.129.251.83
t=0 0
m=audio 22108 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20

IP 10.0.2.4.5068 > 34.43.4.23.5060: UDP, length 647
E....... at .n.
...6..........eACK sip:98765432100 at 98.129.251.83:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP sip.myservice.info:5068
;branch=z9hG4bK3f75.47488c128d3518b837410e363acc963e.0.cs102
Via: SIP/2.0/UDP 10.0.1.41:50600
;received=10.0.1.41;branch=z9hG4bK5e2d222b;rport=50600
Route: <sip:34.43.4.23;lr=on;ftag=as288f6857;did=229.d4f>
Max-Forwards: 70
From: "John Connor" <sip:skynet.device-3 at 10.0.1.41:50600>;tag=as288f6857
To: <sip:98765432100 at ints.myservice.info:5068>;tag=etQmF8eNZ7NKm
Contact: <sip:skynet.device-3 at 10.0.1.41:50600>
Call-ID: 6717947d5e5f445314ba762f154be72f at 10.0.1.41:50600
CSeq: 105 ACK
User-Agent: Asterisk PBX 12.5.0
Content-Length: 0

IP 10.0.2.4.5068 > 34.43.4.23.5060: UDP, length 406
E....... at .o.
...6.........?xCANCEL sip:98765432100 at phone.myprovider.com SIP/2.0
Via: SIP/2.0/UDP sip.myservice.info:5068
;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.1
Max-Forwards: 70
From: "John Connor" <sip:12345678 at phone.myprovider.com>;tag=as288f6857
To: <sip:98765432100 at ints.myservice.info:5068>
Call-ID: 6717947d5e5f445314ba762f154be72f at 10.0.1.41:50600
CSeq: 102 CANCEL
Content-Length: 0
Reason: SIP;cause=200

IP 34.43.4.23.5060 > 10.0.2.4.5068: UDP, length 1428
E.......*.W{6...
.........H.SIP/2.0 200 OK
Via: SIP/2.0/UDP sip.myservice.info:5068
;rport=1024;received=68.34.22.11;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.2.cs102
Via: SIP/2.0/UDP 10.0.1.41:50600
;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600
Record-Route: <sip:34.43.4.23;lr=on;ftag=as288f6857;did=229.d4f>
Record-Route: <sip:sip.myservice.info:5068;nat=yes;ftag=as288f6857;lr=on>
From: "John Connor" <sip:12345678 at phone.myprovider.com>;tag=as288f6857
To: <sip:98765432100 at phone.myprovider.com>;tag=etQmF8eNZ7NKm
Call-ID: 6717947d5e5f445314ba762f154be72f at 10.0.1.41:50600
CSeq: 104 INVITE
Contact: <sip:98765432100 at 98.129.251.83:5060;transport=udp>
User-Agent: myprovider
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER,
NOTIFY
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 249
X-myproviderOutboundGateway: sip:+98765432100 at 67.231.8.195
X-myproviderOutboundCarrierID: 29927192767592
X-myproviderCarrierRate: 0.00900
X-myproviderCloudRate: 0.00300
Remote-Party-ID: "Outbound Call" <sip:5060 at phone.myprovider.com
>;party=calling;privacy=off;screen=no

v=0
o=FreeSWITCH 1415023959 1415023960 IN IP4 98.129.251.83
s=FreeSWITCH
c=IN IP4 98.129.251.83
t=0 0
m=audio 22108 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20

IP 10.0.2.4.5068 > 34.43.4.23.5060: UDP, length 647
E....... at .n.
...6..........;ACK sip:98765432100 at 98.129.251.83:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP sip.myservice.info:5068
;branch=z9hG4bK3f75.00885667802b30315ce7f8b5db31e2bd.0.cs102
Via: SIP/2.0/UDP 10.0.1.41:50600
;received=10.0.1.41;branch=z9hG4bK6081ac89;rport=50600
Route: <sip:34.43.4.23;lr=on;ftag=as288f6857;did=229.d4f>
Max-Forwards: 70
From: "John Connor" <sip:skynet.device-3 at 10.0.1.41:50600>;tag=as288f6857
To: <sip:98765432100 at ints.myservice.info:5068>;tag=etQmF8eNZ7NKm
Contact: <sip:skynet.device-3 at 10.0.1.41:50600>
Call-ID: 6717947d5e5f445314ba762f154be72f at 10.0.1.41:50600
CSeq: 105 ACK
User-Agent: Asterisk PBX 12.5.0
Content-Length: 0

IP 10.0.2.4.5068 > 34.43.4.23.5060: UDP, length 406
E....... at .o.
...6.........?xCANCEL sip:98765432100 at phone.myprovider.com SIP/2.0
Via: SIP/2.0/UDP sip.myservice.info:5068
;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.1
Max-Forwards: 70
From: "John Connor" <sip:12345678 at phone.myprovider.com>;tag=as288f6857
To: <sip:98765432100 at ints.myservice.info:5068>
Call-ID: 6717947d5e5f445314ba762f154be72f at 10.0.1.41:50600
CSeq: 102 CANCEL
Content-Length: 0
Reason: SIP;cause=200
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