[SR-Users] No BYE from Called party
Daniel-Constantin Mierla
miconda at gmail.com
Fri Sep 5 14:45:20 CEST 2014
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
> <mailto: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
>> <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
>> <mailto: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
>> <mailto: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\ <mailto:E..p. at ..@.K%5C>
>> ...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
>> <mailto: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 <mailto: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
>> <mailto: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
>> <mailto: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 list
>> sr-users at lists.sip-router.org <mailto:sr-users at lists.sip-router.org>
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
> --
> Daniel-Constantin Mierla
> http://twitter.com/#!/miconda <http://twitter.com/#%21/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 <mailto:sr-users at 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
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