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@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@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....< .@.'. ...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@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