[SR-Users] No BYE from Called party

Tue Sep 9 14:25:38 CEST 2014

Hello, this is me again)

I added record_route header, that was no result....

I did some tests with my problem and have some results than confused me
very hard...

I registed my trunk from asterisk to provider directly. Do some calls. No
errors- allpackets sends and recieved perfectly. Rgen I catch logs off
calls from kamailio ans asterisk to same trunk on same porviser. I eq
results and was  surprised - packets are the same (without sdp off course
and little things such as uac-agent and other)

Maby I missed something but now I cannot find any reason why call to trunk
not catches BYE from called party

I added my traces at attachement...

thanks for help

2014-09-05 16:45 GMT+04:00 Daniel-Constantin Mierla <miconda at gmail.com>:

>  Be sure you checked the two types of ack requests: hop-by-hop (for
> negative replies, where the contact is not important at all) and end-to-end
> (which is for a 200ok).
>
> Also, even not required by rfc, some UA implementations can be broken.
>
> Anyhow, if you tested and doesn't help, I would try to use record_route()
> for ACK. If that doesn't help, you will need the help of the provider to
> tell you why it doesn't send the BYE.
>
> Cheers,
> Daniel
>
>
> On 05/09/14 12:55, Yuriy Gorlichenko wrote:
>
> RFC not specified Contack header at ACK... So anyway I already tried it
> yesterday))  Unsuccessfull...
>
>
> 2014-09-05 12:54 GMT+04:00 Daniel-Constantin Mierla <miconda at gmail.com>:
>
>>  Hello,
>>
>> I noticed that the ACK is missing the Contact header -- not sure if specs
>> mention anything about being mandatory or not, but you can try to get the
>> contact there.
>>
>> Cheers,
>> Daniel
>>
>>
>> On 05/09/14 08:37, Yuriy Gorlichenko wrote:
>>
>>  Hello All. I have kamailio with provider connection (trunk)
>> When I call to external number through my provider call extablished Ok.
>> But when i try hangup call from external number no BYE sended to me. When I
>> hangup call from my kamailio (internal num) I send by to exteral number and
>> it respond me Ok so session if fully complete. I guess that BYE from
>> external number not recieves to me because I have wrong routing header
>> fields at my INVITe  or ACK messages, but can not find any information what
>> what header must recieve info to external number where send BYE at hangup
>> or thomething like this.
>>
>> This is my little dump for situation wherer I hangup from internal number
>> and BYE finished successfully:
>>
>>
>>
>>  IP my.kamailio.com.5068 > my.proider.com.5060: UDP, length 1599
>> E....< . at .'.
>> ...6........G.RINVITE sip:12345678900 at my.provider.com:5060 SIP/2.0
>> Record-Route: <sip:my.kamailio.com:5068;nat=yes;ftag=as5872f19e;lr=on>
>> Via: SIP/2.0/UDP my.kamailio.com:5068
>> ;branch=z9hG4bK5c63.57e9ba848fadd4e228caa4445baf9d76.2
>> Via: SIP/2.0/UDP
>> my.client.internal.ip:50600;branch=z9hG4bK4d90cebf;rport=50600
>> Max-Forwards: 70
>> From: <sip:TrunkNum at my.provider.com>;tag=as5872f19e
>> To: <sip:12345678900 at my.provider.com:5068>
>> Contact:<TrunkNum at my.kamailio.com:5068>
>> Call-ID: 42b819d45c08c9d304343bf976c5b405 at my.client.internal.ip:50600
>> CSeq: 102 INVITE
>> User-Agent: Asterisk PBX 12.5.0
>> Date: Thu, 04 Sep 2014 21:53:13 GMT
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
>> PUBLISH, MESSAGE
>> Supported: replaces, timer
>> Content-Type: application/sdp
>> Content-Length: 544
>> Proxy-Authorization: Digest username="TrunkNum", realm="my.provider.com",
>> nonce="VAjgfVQI31EsekTXtYjBzKa59XFSJB5P", uri="
>> sip:12345678900 at my.provider.com:5060", qop=auth, nc=00000001,
>> cnonce="3619116795", response="f5bc1d8125dd9e448d2e73764823adee",
>> algorithm=MD5
>>
>>  v=0
>> o=root 1022912010 1022912010 IN IP4 my.kamailio.com
>> s=Asterisk PBX 12.5.0
>> c=IN IP4 my.kamailio.com
>> t=0 0
>> a=ice-lite
>> m=audio 30032 RTP/AVP 0 3 8 101
>> a=rtpmap:0 PCMU/8000
>> 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=maxptime:150
>> a=sendrecv
>> a=rtcp:30033
>> a=ice-ufrag:3o8JrqkF
>> a=ice-pwd:m8q4khQT8NKSABqeUkKs7ed8ClR2
>> a=candidate:TgT1dfTnI3kBgWQ
>>
>>  IP my.proider.com.5060 > my.kamailio.com.5068: UDP, length 1882
>> E..... ./...6...
>> ........b..SIP/2.0 200 OK
>> Via: SIP/2.0/UDP my.kamailio.com:5068;rport=5068;received=my.kamailio.com
>> ;branch=z9hG4bK5c63.57e9ba848fadd4e228caa4445baf9d76.2
>> Via: SIP/2.0/UDP
>> my.client.internal.ip:50600;branch=z9hG4bK4d90cebf;rport=50600
>> Record-Route: <sip:my.proider.com;lr=on;ftag=as5872f19e>
>> Record-Route: <sip:my.kamailio.com:5068;nat=yes;ftag=as5872f19e;lr=on>
>> From: <sip:TrunkNum at my.provider.com>;tag=as5872f19e
>> To: <sip:12345678900 at my.provider.com:5068>;tag=5rF0FNamQ99gH
>> Call-ID: 42b819d45c08c9d304343bf976c5b405 at my.client.internal.ip:50600
>> CSeq: 102 INVITE
>> Contact: <sip:12345678900 at externail.number.end.ip:5060;transport=udp>
>> User-Agent: provider agent
>> 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: 746
>> X-provider agentOutboundGateway: sip:5574012345678900 at 62.93.147.149
>> X-provider agentOutboundCarrierID: 23705946361020
>> X-provider agentCarrierRate: 0.20180
>> X-provider agentCloudRate: 0.00300
>> Remote-Party-ID: "Outbound Call" <sip:5060 at my.provider.com
>> >;party=calling;privacy=off;screen=no
>>
>>  v=0
>> o=FreeSWITCH 1409844386 1409844387 IN IP4 externail.number.end.ip
>> s=FreeSWITCH
>> c=IN IP4 externail.number.end.ip
>> t=0 0
>> a=msid-semantic: WMS hQBNQPw0VjJInvOA0H6HPZyePNO0IIqP
>> m=audio 23216 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
>> a=ssrc:2990874569 cname:pRs5xP
>>
>>
>>
>>
>>
>>  IP my.kamailio.com.5068 > my.proider.com.5060: UDP, length 596
>> E..p. at ..@.K\
>> ...6........\`.ACK
>> sip:12345678900 at externail.number.end.ip:5060;transport=udp SIP/2.0
>> Via: SIP/2.0/UDP my.kamailio.com:5068
>> ;branch=z9hG4bK5c63.8c0bfc7de0669f1b8252b97b8431b2e1.0
>> Via: SIP/2.0/UDP
>> my.client.internal.ip:50600;branch=z9hG4bK5420b559;rport=50600
>> Route: <sip:my.proider.com;lr=on;ftag=as5872f19e>
>> Max-Forwards: 70
>> From: <sip:TrunkNum at sip.callsion.com>;tag=as5872f19e
>> To: <sip:12345678900 at my.provider.com:5068>;tag=5rF0FNamQ99gH
>> Call-ID: 42b819d45c08c9d304343bf976c5b405 at my.client.internal.ip:50600
>> CSeq: 102 ACK
>> User-Agent: Asterisk PBX 12.5.0
>> Content-Length: 0
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>  IP my.kamailio.com.5068 > my.proider.com.5060: UDP, length 617
>> E....D.. at .KC
>> ...6........q.ZBYE
>> sip:12345678900 at externail.number.end.ip:5060;transport=udp SIP/2.0
>> Via: SIP/2.0/UDP my.kamailio.com:5068
>> ;branch=z9hG4bK6c63.3deda3c1fee0ef86d7eae9549dd1b194.0
>> Via: SIP/2.0/UDP
>> my.client.internal.ip:50600;branch=z9hG4bK65e48b85;rport=50600
>> Route: <sip:my.proider.com;lr=on;ftag=as5872f19e>
>> Max-Forwards: 70
>> From: <sip:TrunkNum at sip.callsion.com>;tag=as5872f19e
>> To: <sip:12345678900 at my.provider.com:5068>;tag=5rF0FNamQ99gH
>> Call-ID: 42b819d45c08c9d304343bf976c5b405 at my.client.internal.ip:50600
>> CSeq: 103 BYE
>> User-Agent: Asterisk PBX 12.5.0
>> X-Asterisk-HangupCause: Normal Clearing
>> X-Asterisk-HangupCauseCode: 16
>> Content-Length: 0
>>
>>
>>  IP my.proider.com.5060 > my.kamailio.com.5068: UDP, length 568
>> E..T....-.676...
>> ........ at ..SIP/2.0 200 OK
>> Via: SIP/2.0/UDP my.kamailio.com:5068;received=my.kamailio.com
>> ;branch=z9hG4bK6c63.3deda3c1fee0ef86d7eae9549dd1b194.0
>> Via: SIP/2.0/UDP
>> my.client.internal.ip:50600;branch=z9hG4bK65e48b85;rport=50600
>> From: <sip:TrunkNum at sip.callsion.com>;tag=as5872f19e
>> To: <sip:12345678900 at my.provider.com:5068>;tag=5rF0FNamQ99gH
>> Call-ID: 42b819d45c08c9d304343bf976c5b405 at my.client.internal.ip:50600
>> CSeq: 103 BYE
>> User-Agent: provider agent
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER,
>> NOTIFY
>> Supported: timer, precondition, path, replaces
>> Content-Length: 0
>>
>>
>>
>>
>> Thanks for help.
>>
>>
>>  _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users at lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>> --
>> Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
>> Next Kamailio Advanced Trainings 2014 - http://www.asipto.com
>> Sep 22-25, Berlin, Germany
>>
>>
>> _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users at lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>
> --
> Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
> Next Kamailio Advanced Trainings 2014 - http://www.asipto.com
> Sep 22-25, Berlin, Germany
>
>
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Reliably Transmitting (NAT) tomy.provider.com:5060:
INVITE sip:To_num at my.provider.com SIP/2.0
Via: SIP/2.0/UDP my.server.com:5068;branch=z9hG4bK541a7830;rport
Max-Forwards: 70
From: "203" <sip:From_num at my.provider.com>;tag=as17dca408
To: <sip:To_num at my.provider.com>
Contact: <sip:From_num at my.server.com:5068>
Call-ID: 4458465f68af28ca0ccd8a073120c228 at my.provider.com
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.11.0
Date: Sun, 07 Sep 2014 00:15:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 283

v=0
o=root 1016367194 1016367194 IN IP4 my.server.com
s=Asterisk PBX 11.11.0
c=IN IP4 my.server.com
t=0 0
m=audio 18258 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<--- SIP read from UDP:my.provider.com:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP my.server.com:5068;branch=z9hG4bK541a7830;rport=5068
From: "203" <sip:From_num at my.provider.com>;tag=as17dca408
To: <sip:To_num at my.provider.com>;tag=7aca57471283d03f3321fdece75f8666.0554
Call-ID: 4458465f68af28ca0ccd8a073120c228 at my.provider.com
CSeq: 102 INVITE
Proxy-Authenticate: Digest realm="my.provider.com", nonce="VAukwFQLo5TCvnt9GWaCSWg8ULPgIzU9", qop="auth"
Server: kamailio (4.1.2 (x86_64/linux))
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Transmitting (NAT) tomy.provider.com:5060:
ACK sip:To_num at my.provider.com SIP/2.0
Via: SIP/2.0/UDP my.server.com:5068;branch=z9hG4bK541a7830;rport
Max-Forwards: 70
From: "203" <sip:From_num at my.provider.com>;tag=as17dca408
To: <sip:To_num at my.provider.com>;tag=7aca57471283d03f3321fdece75f8666.0554
Contact: <sip:From_num at my.server.com:5068>
Call-ID: 4458465f68af28ca0ccd8a073120c228 at my.provider.com
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.11.0
Content-Length: 0

Reliably Transmitting (NAT) tomy.provider.com:5060:
INVITE sip:To_num at my.provider.com SIP/2.0
Via: SIP/2.0/UDP my.server.com:5068;branch=z9hG4bK3a86505a;rport
Max-Forwards: 70
From: "203" <sip:From_num at my.provider.com>;tag=as17dca408
To: <sip:To_num at my.provider.com>
Contact: <sip:From_num at my.server.com:5068>
Call-ID: 4458465f68af28ca0ccd8a073120c228 at my.provider.com
CSeq: 103 INVITE
User-Agent: Asterisk PBX 11.11.0
Proxy-Authorization: Digest username="From_num", realm="my.provider.com", algorithm=MD5, uri="sip:To_num at my.provider.com", nonce="VAukwFQLo5TCvnt9GWaCSWg8ULPgIzU9", response="c21f09cb598343b7c4638de6f0aa5678", qop=auth, cnonce="7fd459c6", nc=00000001
Date: Sun, 07 Sep 2014 00:15:16 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 283

v=0
o=root 1016367194 1016367195 IN IP4 my.server.com
s=Asterisk PBX 11.11.0
c=IN IP4 my.server.com
t=0 0
m=audio 18258 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:my.provider.com:5060 --->
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP my.server.com:5068;branch=z9hG4bK3a86505a;rport=5068
From: "203" <sip:From_num at my.provider.com>;tag=as17dca408
To: <sip:To_num at my.provider.com>
Call-ID: 4458465f68af28ca0ccd8a073120c228 at my.provider.com
CSeq: 103 INVITE
Server: kamailio (4.1.2 (x86_64/linux))
Content-Length: 0

<--- SIP read from UDP:my.provider.com:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP my.server.com:5068;branch=z9hG4bK3a86505a;rport=5068
Record-Route: <sip:my.provider.com;lr=on;ftag=as17dca408;did=84c.788>
From: "203" <sip:From_num at my.provider.com>;tag=as17dca408
To: <sip:To_num at my.provider.com>;tag=ZZ7H26FHvvQDe
Call-ID: 4458465f68af28ca0ccd8a073120c228 at my.provider.com
CSeq: 103 INVITE
Contact: <sip:To_num at 212.100.254.141:5060;transport=udp>
User-Agent: Plivo
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: "To_num" <sip:To_num at my.provider.com>;party=calling;privacy=off;screen=no

<--- SIP read from UDP:my.provider.com:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP my.server.com:5068;branch=z9hG4bK3a86505a;rport=5068
Record-Route: <sip:my.provider.com;lr=on;ftag=as17dca408;did=84c.788>
From: "203" <sip:From_num at my.provider.com>;tag=as17dca408
To: <sip:To_num at my.provider.com>;tag=ZZ7H26FHvvQDe
Call-ID: 4458465f68af28ca0ccd8a073120c228 at my.provider.com
CSeq: 103 INVITE
Contact: <sip:To_num at 212.100.254.141:5060;transport=udp>
User-Agent: Plivo
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: 253
Remote-Party-ID: "To_num" <sip:To_num at my.provider.com>;party=calling;privacy=off;screen=no

v=0
o=FreeSWITCH 1410020988 1410020989 IN IP4 212.100.254.141
s=FreeSWITCH
c=IN IP4 212.100.254.141
t=0 0
m=audio 27930 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
<------------->

<--- SIP read from UDP:my.provider.com:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP my.server.com:5068;branch=z9hG4bK3a86505a;rport=5068
Record-Route: <sip:my.provider.com;lr=on;ftag=as17dca408;did=84c.788>
From: "203" <sip:From_num at my.provider.com>;tag=as17dca408
To: <sip:To_num at my.provider.com>;tag=ZZ7H26FHvvQDe
Call-ID: 4458465f68af28ca0ccd8a073120c228 at my.provider.com
CSeq: 103 INVITE
Contact: <sip:To_num at 212.100.254.141:5060;transport=udp>
User-Agent: Plivo
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: 253
X-PlivoOutboundGateway: sip:55740To_num at 62.93.147.149
X-PlivoOutboundCarrierID: 23705946361020
X-PlivoCarrierRate: 0.20180
X-PlivoCloudRate: 0.00300
Remote-Party-ID: "Outbound Call" <sip:5060 at my.provider.com>;party=calling;privacy=off;screen=no

v=0
o=FreeSWITCH 1410020988 1410020989 IN IP4 212.100.254.141
s=FreeSWITCH
c=IN IP4 212.100.254.141
t=0 0
m=audio 27930 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

ACK sip:To_num at 212.100.254.141:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP my.server.com:5068;branch=z9hG4bK09fc11af;rport
Route: <sip:my.provider.com;lr=on;ftag=as17dca408;did=84c.788>
Max-Forwards: 70
From: "203" <sip:From_num at my.provider.com>;tag=as17dca408
To: <sip:To_num at my.provider.com>;tag=ZZ7H26FHvvQDe
Contact: <sip:From_num at my.server.com:5068>
Call-ID: 4458465f68af28ca0ccd8a073120c228 at my.provider.com
CSeq: 103 ACK
User-Agent: Asterisk PBX 11.11.0
Content-Length: 0

<--- SIP read from UDP:my.provider.com:5060 --->
BYE sip:From_num at my.server.com:5068 SIP/2.0
Record-Route: <sip:my.provider.com;lr=on;ftag=ZZ7H26FHvvQDe>
Via: SIP/2.0/UDPmy.provider.com:5060;branch=z9hG4bK43a4.4e3a49dd8c75379912ac6ff1a9904244.0
Via: SIP/2.0/UDP 212.100.254.141;rport=5060;branch=z9hG4bKaH07HF0SavaDN
Max-Forwards: 16
From: <sip:To_num at my.provider.com>;tag=ZZ7H26FHvvQDe
To: "203" <sip:From_num at my.provider.com>;tag=as17dca408
Call-ID: 4458465f68af28ca0ccd8a073120c228 at my.provider.com
CSeq: 64688402 BYE
Contact: <sip:To_num at 212.100.254.141:5060;transport=udp>
User-Agent: Plivo
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY
Supported: timer, precondition, path, replaces
Reason: Q.850;cause=16
Content-Length: 0

<--- Transmitting (NAT) tomy.provider.com:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDPmy.provider.com:5060;branch=z9hG4bK43a4.4e3a49dd8c75379912ac6ff1a9904244.0;received=my.provider.com;rport=5060
Via: SIP/2.0/UDP 212.100.254.141;rport=5060;branch=z9hG4bKaH07HF0SavaDN
Record-Route: <sip:my.provider.com;lr=on;ftag=ZZ7H26FHvvQDe>
From: <sip:To_num at my.provider.com>;tag=ZZ7H26FHvvQDe
To: "203" <sip:From_num at my.provider.com>;tag=as17dca408
Call-ID: 4458465f68af28ca0ccd8a073120c228 at my.provider.com
CSeq: 64688402 BYE
Server: Asterisk PBX 11.11.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

-------------- next part --------------

IP 10.0.1.18.5068 > my.provider.com.5060: UDP, length 1667
E..... . at ...
...6..........  INVITE sip:to_num at my.porvider.com:5060 SIP/2.0
Record-Route: <sip:my.server.com:5068;nat=yes;ftag=as1e4f255b;lr=on>
Via: SIP/2.0/UDP my.server.com:5068;branch=z9hG4bK63d8.69a95d540b964f650a4e836fdf09f889.0
Via: SIP/2.0/UDP 10.0.1.6:50600;branch=z9hG4bK14d967b4;rport=50600
Max-Forwards: 70
From: <sip:From_num at my.porvider.com>;tag=as1e4f255b
To: <sip:to_num at my.porvider.com:5068>
Contact:<From_num at my.server.com:5068>
Call-ID: 3e69bacb64ccbea83f407f302c33485c at 10.0.1.6:50600
CSeq: 102 INVITE
User-Agent: Asterisk PBX 12.5.0
Date: Sun, 07 Sep 2014 00:16:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 884

v=0
o=root 753935994 753935994 IN IP4 my.server.com
s=Asterisk PBX 12.5.0
c=IN IP4 my.server.com
t=0 0
a=ice-lite
m=audio 31630 RTP/SAVPF 107 8 0 101
a=rtpmap:107 opus/48000/2
a=fmtp:107 maxplaybackrate=48000;sprop-maxcapturerate=48000;minptime=10;maxaveragebitrate=20000;stereo=0;sprop-stereo=0;cbr=0;useinbandfec=0;usedtx=0
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:60
a=sendrecv
a=rtcp:31631
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:4gd7wtcvq2iKds54tdCSUD/PWnO5iNTdRo8pcD2H
a=setup:actpass
a=fingerprint:sha-1 97:8B:D7:80:8B:07:EA:7E:1A:5C:19:5F:1E:5E:A8:3D:3E:0F:B4:B5
a=ice-ufrag:rVq84T0p

IP my.provider.com.5060 > 10.0.1.18.5068: UDP, length 621
E.......-.-.6...
........u.XSIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP my.server.com:5068;branch=z9hG4bK63d8.69a95d540b964f650a4e836fdf09f889.0;received=my.server.com
Via: SIP/2.0/UDP 10.0.1.6:50600;branch=z9hG4bK14d967b4;rport=50600
From: <sip:From_num at my.porvider.com>;tag=as1e4f255b
To: <sip:to_num at my.porvider.com:5068>;tag=7aca57471283d03f3321fdece75f8666.3802
Call-ID: 3e69bacb64ccbea83f407f302c33485c at 10.0.1.6:50600
CSeq: 102 INVITE
Proxy-Authenticate: Digest realm="my.porvider.com", nonce="VAulMFQLpAR2E9StClcRd8QzYT9h/rgs", qop="auth"
Server: kamailio (4.1.2 (x86_64/linux))
Content-Length: 0

IP 10.0.1.18.5068 > my.provider.com.5060: UDP, length 404
E....... at ...
...6...........ACK sip:to_num at my.porvider.com:5060 SIP/2.0
Via: SIP/2.0/UDP my.server.com:5068;branch=z9hG4bK63d8.69a95d540b964f650a4e836fdf09f889.0
Max-Forwards: 70
From: <sip:From_num at my.porvider.com>;tag=as1e4f255b
To: <sip:to_num at my.porvider.com:5068>;tag=7aca57471283d03f3321fdece75f8666.3802
Call-ID: 3e69bacb64ccbea83f407f302c33485c at 10.0.1.6:50600
CSeq: 102 ACK
Content-Length: 0

IP 10.0.1.18.5068 > my.provider.com.5060: UDP, length 1939
E..... . at ...
...6..........bINVITE sip:to_num at my.porvider.com:5060 SIP/2.0
Record-Route: <sip:my.server.com:5068;nat=yes;ftag=as1e4f255b;lr=on>
Via: SIP/2.0/UDP my.server.com:5068;branch=z9hG4bK63d8.69a95d540b964f650a4e836fdf09f889.1
Via: SIP/2.0/UDP 10.0.1.6:50600;branch=z9hG4bK14d967b4;rport=50600
Max-Forwards: 70
From: <sip:From_num at my.porvider.com>;tag=as1e4f255b
To: <sip:to_num at my.porvider.com:5068>
Contact:<From_num at my.server.com:5068>
Call-ID: 3e69bacb64ccbea83f407f302c33485c at 10.0.1.6:50600
CSeq: 102 INVITE
User-Agent: Asterisk PBX 12.5.0
Date: Sun, 07 Sep 2014 00:16:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 884
Proxy-Authorization: Digest username="From_num", realm="my.porvider.com", nonce="VAulMFQLpAR2E9StClcRd8QzYT9h/rgs", uri="sip:to_num at my.porvider.com:5060", qop=auth, nc=00000001, cnonce="2656360386", response="7747e0ee79de90627d43076ff41fbcbf", algorithm=MD5

v=0
o=root 753935994 753935994 IN IP4 my.server.com
s=Asterisk PBX 12.5.0
c=IN IP4 my.server.com
t=0 0
a=ice-lite
m=audio 31630 RTP/SAVPF 107 8 0 101
a=rtpmap:107 opus/48000/2
a=fmtp:107 maxplaybackrate=48000;sprop-maxcapturerate=48000;minptime=10;maxaveragebitrate=20000;stereo=0;sprop-stereo=0;cbr=0;useinbandfec=0;usedtx=0
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a
IP 10.0.1.18.5068 > my.provider.com.5060: UDP, length 1939
E..... . at ...
...6..........aINVITE sip:to_num at my.porvider.com:5060 SIP/2.0
Record-Route: <sip:my.server.com:5068;nat=yes;ftag=as1e4f255b;lr=on>
Via: SIP/2.0/UDP my.server.com:5068;branch=z9hG4bK63d8.69a95d540b964f650a4e836fdf09f889.2
Via: SIP/2.0/UDP 10.0.1.6:50600;branch=z9hG4bK14d967b4;rport=50600
Max-Forwards: 70
From: <sip:From_num at my.porvider.com>;tag=as1e4f255b
To: <sip:to_num at my.porvider.com:5068>
Contact:<From_num at my.server.com:5068>
Call-ID: 3e69bacb64ccbea83f407f302c33485c at 10.0.1.6:50600
CSeq: 102 INVITE
User-Agent: Asterisk PBX 12.5.0
Date: Sun, 07 Sep 2014 00:16:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 884
Proxy-Authorization: Digest username="From_num", realm="my.porvider.com", nonce="VAulMFQLpAR2E9StClcRd8QzYT9h/rgs", uri="sip:to_num at my.porvider.com:5060", qop=auth, nc=00000001, cnonce="2656360386", response="7747e0ee79de90627d43076ff41fbcbf", algorithm=MD5

v=0
o=root 753935994 753935994 IN IP4 my.server.com
s=Asterisk PBX 12.5.0
c=IN IP4 my.server.com
t=0 0
a=ice-lite
m=audio 31630 RTP/SAVPF 107 8 0 101
a=rtpmap:107 opus/48000/2
a=fmtp:107 maxplaybackrate=48000;sprop-maxcapturerate=48000;minptime=10;maxaveragebitrate=20000;stereo=0;sprop-stereo=0;cbr=0;useinbandfec=0;usedtx=0
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a
IP my.provider.com.5060 > 10.0.1.18.5068: UDP, length 493
E..     ....-..36...
...........SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP my.server.com:5068;branch=z9hG4bK63d8.69a95d540b964f650a4e836fdf09f889.1;rport=5068;received=my.server.com
Via: SIP/2.0/UDP 10.0.1.6:50600;branch=z9hG4bK14d967b4;rport=50600
From: <sip:From_num at my.porvider.com>;tag=as1e4f255b
To: <sip:to_num at my.porvider.com:5068>
Call-ID: 3e69bacb64ccbea83f407f302c33485c at 10.0.1.6:50600
CSeq: 102 INVITE
Server: kamailio (4.1.2 (x86_64/linux))
Content-Length: 0

IP my.provider.com.5060 > 10.0.1.18.5068: UDP, length 493
E..     ....-..26...
...........SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP my.server.com:5068;branch=z9hG4bK63d8.69a95d540b964f650a4e836fdf09f889.2;rport=5068;received=my.server.com
Via: SIP/2.0/UDP 10.0.1.6:50600;branch=z9hG4bK14d967b4;rport=50600
From: <sip:From_num at my.porvider.com>;tag=as1e4f255b
To: <sip:to_num at my.porvider.com:5068>
Call-ID: 3e69bacb64ccbea83f407f302c33485c at 10.0.1.6:50600
CSeq: 102 INVITE
Server: kamailio (4.1.2 (x86_64/linux))
Content-Length: 0

IP my.provider.com.5060 > 10.0.1.18.5068: UDP, length 962
E.......-.,\6...
..........ESIP/2.0 180 Ringing
Via: SIP/2.0/UDP my.server.com:5068;rport=5068;received=my.server.com;branch=z9hG4bK63d8.69a95d540b964f650a4e836fdf09f889.1
Via: SIP/2.0/UDP 10.0.1.6:50600;branch=z9hG4bK14d967b4;rport=50600
Record-Route: <sip:my.provider.com;lr=on;ftag=as1e4f255b;did=66e.0d32>
Record-Route: <sip:my.server.com:5068;nat=yes;ftag=as1e4f255b;lr=on>
From: <sip:From_num at my.porvider.com>;tag=as1e4f255b
To: <sip:to_num at my.porvider.com:5068>;tag=N3gvp42FtHX2Q
Call-ID: 3e69bacb64ccbea83f407f302c33485c at 10.0.1.6:50600
CSeq: 102 INVITE
Contact: <sip:to_num at 173.203.60.50:5060;transport=udp>
User-Agent: Plivo
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: "to_num" <sip:to_num at my.porvider.com>;party=calling;privacy=off;screen=no

IP my.provider.com.5060 > 10.0.1.18.5068: UDP, length 1852
E..... .-.
]6...
........D..SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP my.server.com:5068;rport=5068;received=my.server.com;branch=z9hG4bK63d8.69a95d540b964f650a4e836fdf09f889.1
Via: SIP/2.0/UDP 10.0.1.6:50600;branch=z9hG4bK14d967b4;rport=50600
Record-Route: <sip:my.provider.com;lr=on;ftag=as1e4f255b;did=66e.0d32>
Record-Route: <sip:my.server.com:5068;nat=yes;ftag=as1e4f255b;lr=on>
From: <sip:From_num at my.porvider.com>;tag=as1e4f255b
To: <sip:to_num at my.porvider.com:5068>;tag=N3gvp42FtHX2Q
Call-ID: 3e69bacb64ccbea83f407f302c33485c at 10.0.1.6:50600
CSeq: 102 INVITE
Contact: <sip:to_num at 173.203.60.50:5060;transport=udp>
User-Agent: Plivo
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: 843
Remote-Party-ID: "to_num" <sip:to_num at my.porvider.com>;party=calling;privacy=off;screen=no

v=0
o=FreeSWITCH 1410024665 1410024666 IN IP4 173.203.60.50
s=FreeSWITCH
c=IN IP4 173.203.60.50
t=0 0
a=msid-semantic: WMS uj5Wid9eUx5EVx5w0GQ8ee1jRO5oK8N4
m=audio 24364 RTP/SAVPF 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=silenceSupp:off - - - -
a=ptime:20
a=fingerprint:sha-1 B3:91:E8:A1:EC:76:7B:33:B9:15:51:85:33:4D:F2:F5:8F:56:E3:66
a=rtcp:24365 IN IP4 173.203.60.50
a=ssrc:4174795900 cname:hJQqNQvydq2ZNYmD
a=ssrc:417479590
IP 10.0.1.18.5068 > 10.0.1.6.50600: UDP, length 1416
E....4.. at ...
...
..........]SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.0.1.6:50600;branch=z9hG4bK14d967b4;rport=50600
Record-Route: <sip:my.provider.com;lr=on;ftag=as1e4f255b;did=66e.0d32>
Record-Route: <sip:my.server.com:5068;nat=yes;ftag=as1e4f255b;lr=on>
From: <sip:From_num at my.porvider.com>;tag=as1e4f255b
To: <sip:to_num at my.porvider.com:5068>;tag=N3gvp42FtHX2Q
Call-ID: 3e69bacb64ccbea83f407f302c33485c at 10.0.1.6:50600
CSeq: 102 INVITE
Contact: <sip:to_num at 173.203.60.50:5060;transport=udp>
User-Agent: Plivo
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: 537
Remote-Party-ID: "to_num" <sip:to_num at my.porvider.com>;party=calling;privacy=off;screen=no

v=0
o=FreeSWITCH 1410024665 1410024666 IN IP4 my.server.com
s=FreeSWITCH
c=IN IP4 my.server.com
t=0 0
a=msid-semantic: WMS uj5Wid9eUx5EVx5w0GQ8ee1jRO5oK8N4
m=audio 31644 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=silenceSupp:off - - - -
a=ptime:20
a=ssrc:4174795900 cname:hJQqNQvydq2ZNYmD
a=ssrc:4174795900 msid:uj5Wid9eUx5EVx5w0GQ8ee1jRO5oK8N4 a0
a=ssrc:4174795900 mslabel:uj5Wid9eUx5EVx5w0GQ8ee1jRO5oK8N4
a=ssrc:4174795900 label:uj5Wid9eUx5EVx5w0GQ8ee1jRO5oK8N4a0
a=sendrecv
a=rtcp:31645

IP 10.0.1.6.50600 > 10.0.1.18.5068: UDP, length 1189
E....... at .u.
...
..........aSIP/2.0 183 Session Progress
Via: SIP/2.0/UDP my.server.com:5068;branch=z9hG4bKab21.96d614380b8f81c32e9f7a109018eb9f.0;received=10.0.1.18;rport=5068
Via: SIP/2.0/WSS li77nhg5m00n.invalid;rport=55969;received=85.21.140.253;branch=z9hG4bK499016
Record-Route: <sip:my.server.com:5068;nat=yes;ftag=12c6uakhrr;lr=on>
From: <sip:callision.device-129 at my.server.com>;tag=12c6uakhrr
To: <sip:to_num at my.server.com>;tag=as067c9c96
Call-ID: ndfkjm4puad2qp15a4hs
CSeq: 8339 INVITE
Server: Asterisk PBX 12.5.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:to_num at 10.0.1.6:50600>
Content-Type: application/sdp
Content-Length: 446

v=0
o=root 222668154 222668154 IN IP4 10.0.1.6
s=Asterisk PBX 12.5.0
c=IN IP4 10.0.1.6
t=0 0
m=audio 18894 RTP/AVP 111 8 0 126
a=rtpmap:111 opus/48000/2
a=fmtp:111 maxplaybackrate=48000;sprop-maxcapturerate=48000;minptime=10;maxaveragebitrate=20000;stereo=0;sprop-stereo=0;cbr=0;useinbandfec=0;usedtx=0
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16
a=ptime:20
a=maxptime:60
a=sendrecv

IP my.provider.com.5060 > 10.0.1.18.5068: UDP, length 950
E.......-.,f6...
..........~SIP/2.0 180 Ringing
Via: SIP/2.0/UDP my.server.com:5068;rport=5068;received=my.server.com;branch=z9hG4bK63d8.69a95d540b964f650a4e836fdf09f889.2
Via: SIP/2.0/UDP 10.0.1.6:50600;branch=z9hG4bK14d967b4;rport=50600
Record-Route: <sip:my.provider.com;lr=on;ftag=as1e4f255b>
Record-Route: <sip:my.server.com:5068;nat=yes;ftag=as1e4f255b;lr=on>
From: <sip:From_num at my.porvider.com>;tag=as1e4f255b
To: <sip:to_num at my.porvider.com:5068>;tag=74gH2tQ7eXQUS
Call-ID: 3e69bacb64ccbea83f407f302c33485c at 10.0.1.6:50600
CSeq: 102 INVITE
Contact: <sip:to_num at 67.192.253.160:5060;transport=udp>
User-Agent: Plivo
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: "to_num" <sip:to_num at my.porvider.com>;party=calling;privacy=off;screen=no

IP my.provider.com.5060 > 10.0.1.18.5068: UDP, length 1845
E..... .-.
[6...
........=..SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP my.server.com:5068;rport=5068;received=my.server.com;branch=z9hG4bK63d8.69a95d540b964f650a4e836fdf09f889.2
Via: SIP/2.0/UDP 10.0.1.6:50600;branch=z9hG4bK14d967b4;rport=50600
Record-Route: <sip:my.provider.com;lr=on;ftag=as1e4f255b>
Record-Route: <sip:my.server.com:5068;nat=yes;ftag=as1e4f255b;lr=on>
From: <sip:From_num at my.porvider.com>;tag=as1e4f255b
To: <sip:to_num at my.porvider.com:5068>;tag=74gH2tQ7eXQUS
Call-ID: 3e69bacb64ccbea83f407f302c33485c at 10.0.1.6:50600
CSeq: 102 INVITE
Contact: <sip:to_num at 67.192.253.160:5060;transport=udp>
User-Agent: Plivo
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: 848
Remote-Party-ID: "to_num" <sip:to_num at my.porvider.com>;party=calling;privacy=off;screen=no

v=0
o=FreeSWITCH 1410017037 1410017038 IN IP4 67.192.253.160
s=FreeSWITCH
c=IN IP4 67.192.253.160
t=0 0
a=msid-semantic: WMS V88QhOyTjnVml04opHY2O8BvgVuG59FR
m=audio 31992 RTP/SAVPF 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=silenceSupp:off - - - -
a=ptime:20
a=fingerprint:sha-1 E0:94:0F:6B:7E:AA:26:4A:B0:73:79:73:EA:E2:7D:5C:89:25:FF:5B
a=rtcp:31993 IN IP4 67.192.253.160
a=ssrc:2388829308 cname:8ExgiWZsaKEmBzBR
a=ssrc:2388829308 msid:V8

IP my.provider.com.5060 > 10.0.1.18.5068: UDP, length 1991
E..... .-.
Y6...
.........N.SIP/2.0 200 OK
Via: SIP/2.0/UDP my.server.com:5068;rport=5068;received=my.server.com;branch=z9hG4bK63d8.69a95d540b964f650a4e836fdf09f889.1
Via: SIP/2.0/UDP 10.0.1.6:50600;branch=z9hG4bK14d967b4;rport=50600
Record-Route: <sip:my.provider.com;lr=on;ftag=as1e4f255b;did=66e.0d32>
Record-Route: <sip:my.server.com:5068;nat=yes;ftag=as1e4f255b;lr=on>
From: <sip:From_num at my.porvider.com>;tag=as1e4f255b
To: <sip:to_num at my.porvider.com:5068>;tag=N3gvp42FtHX2Q
Call-ID: 3e69bacb64ccbea83f407f302c33485c at 10.0.1.6:50600
CSeq: 102 INVITE
Contact: <sip:to_num at 173.203.60.50:5060;transport=udp>
User-Agent: Plivo
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: 843
X-PlivoOutboundGateway: sip:55740to_num at 62.93.147.149
X-PlivoOutboundCarrierID: 23705946361020
X-PlivoCarrierRate: 0.20180
X-PlivoCloudRate: 0.00300
Remote-Party-ID: "Outbound Call" <sip:5060 at my.porvider.com>;party=calling;privacy=off;screen=no

v=0
o=FreeSWITCH 1410024665 1410024666 IN IP4 173.203.60.50
s=FreeSWITCH
c=IN IP4 173.203.60.50
t=0 0
a=msid-semantic: WMS uj5Wid9eUx5EVx5w0GQ8ee1jRO5oK8N4
m=audio 24364 RTP/SAVPF 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=silenceSupp:off - - - -
a=ptime:20
a=fingerprint:sha-1 B3:91:E8:A1:EC:

IP 10.0.1.18.5068 > my.provider.com.5060: UDP, length 605
E..y.... at ...
...6........e at .ACK sip:to_num at 173.203.60.50:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP my.server.com:5068;branch=z9hG4bK63d8.68911b0bbc61d8d862bcbea2b9d7441c.0
Via: SIP/2.0/UDP 10.0.1.6:50600;branch=z9hG4bK79426f65;rport=50600
Route: <sip:my.provider.com;lr=on;ftag=as1e4f255b;did=66e.0d32>
Max-Forwards: 70
From: <sip:From_num at my.porvider.com>;tag=as1e4f255b
To: <sip:to_num at my.porvider.com:5068>;tag=N3gvp42FtHX2Q
Contact:<From_num at my.server.com:5068>
Call-ID: 3e69bacb64ccbea83f407f302c33485c at 10.0.1.6:50600
CSeq: 102 ACK
User-Agent: Asterisk PBX 12.5.0
Content-Length: 0