[SR-Users] No BYE from Called party

Fri Sep 5 10:54:56 CEST 2014

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 
> <http://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 at my.provider.com 
> <mailto:sip%3ATrunkNum at my.provider.com>>;tag=as5872f19e
> To: <sip:12345678900 at my.provider.com:5068 
> <http://sip:12345678900@my.provider.com:5068>>
> Contact:<TrunkNum at my.kamailio.com:5068 
> <http://TrunkNum@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 <http://my.provider.com>", 
> nonce="VAjgfVQI31EsekTXtYjBzKa59XFSJB5P", 
> uri="sip:12345678900 at my.provider.com:5060 
> <http://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 
> <http://my.kamailio.com>
> s=Asterisk PBX 12.5.0
> c=IN IP4 my.kamailio.com <http://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 
> <http://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 
> <http://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 
> <mailto:sip%3ATrunkNum at my.provider.com>>;tag=as5872f19e
> To: <sip:12345678900 at my.provider.com:5068 
> <http://sip:12345678900@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 
> <mailto:sip%3A5574012345678900 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 
> <mailto:sip%3A5060 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 <http://my.proider.com>;lr=on;ftag=as5872f19e>
> Max-Forwards: 70
> From: <sip:TrunkNum at sip.callsion.com 
> <mailto:sip%3ATrunkNum at sip.callsion.com>>;tag=as5872f19e
> To: <sip:12345678900 at my.provider.com:5068 
> <http://sip:12345678900@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 <http://my.proider.com>;lr=on;ftag=as5872f19e>
> Max-Forwards: 70
> From: <sip:TrunkNum at sip.callsion.com 
> <mailto:sip%3ATrunkNum at sip.callsion.com>>;tag=as5872f19e
> To: <sip:12345678900 at my.provider.com:5068 
> <http://sip:12345678900@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 
> <http://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 
> <mailto:sip%3ATrunkNum at sip.callsion.com>>;tag=as5872f19e
> To: <sip:12345678900 at my.provider.com:5068 
> <http://sip:12345678900@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.
>
>
> _______________________________________________
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-- 
Daniel-Constantin Mierla
http://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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