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
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
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
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>
From: <sip:TrunkNum@my.provider.com>;tag=as5872f19e
To: <sip:12345678900@my.provider.com:5068>;tag=5rF0FNamQ99gH
CSeq: 102 INVITE
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
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
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
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
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
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 list
sr-users@lists.sip-router.org
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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

_______________________________________________
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 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