Re: [SR-Users] No BYE from Called party

9 Sep 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 &lt;miconda(a)gmail.com&gt;om>:

...
   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 &lt;miconda(a)gmail.com&gt;om>:

   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....< .@.'.
 ...6........G.RINVITE sip:12345678900@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@my.provider.com>;tag=as5872f19e
 To: <sip:12345678900@my.provider.com:5068>
 Contact:<TrunkNum@my.kamailio.com:5068>
 Call-ID: 42b819d45c08c9d304343bf976c5b405@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@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@my.provider.com>;tag=as5872f19e
 To: <sip:12345678900@my.provider.com:5068>;tag=5rF0FNamQ99gH
 Call-ID: 42b819d45c08c9d304343bf976c5b405@my.client.internal.ip:50600
 CSeq: 102 INVITE
 Contact: <sip:12345678900@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@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@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.@..@.K\
 ...6........\`.ACK
 sip:12345678900@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@sip.callsion.com>;tag=as5872f19e
 To: <sip:12345678900@my.provider.com:5068>;tag=5rF0FNamQ99gH
 Call-ID: 42b819d45c08c9d304343bf976c5b405@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..@.KC
 ...6........q.ZBYE
 sip:12345678900@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@sip.callsion.com>;tag=as5872f19e
 To: <sip:12345678900@my.provider.com:5068>;tag=5rF0FNamQ99gH
 Call-ID: 42b819d45c08c9d304343bf976c5b405@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...
 ........@..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@sip.callsion.com>;tag=as5872f19e
 To: <sip:12345678900@my.provider.com:5068>;tag=5rF0FNamQ99gH
 Call-ID: 42b819d45c08c9d304343bf976c5b405@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@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(a)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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Re: [SR-Users] No BYE from Called party