02.04.2010 18:25:21
oldDude
Posts: 16
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Me again. This one seems strange. It appears that the TekSIP proxy getting the correct BYE (Proxy at 192.168.0.1:5060) but is copying it incorrectly before sending it downstream to the client at 192.168.0.121. I tried the same setup wih another Proxy and didn't have any problems. Any ideas?
SIP req. from : 192.168.0.2:5160 (UDP) - 4/2/2010 4:43:49 PM
BYE sip:8101@192.168.0.121 SIP/2.0
[Via] = SIP/2.0/UDP 192.168.0.1:5060;rport;branch=z9hG4bKPj829b93bb4f724a5fa8fbcefcd37d2e26
[Route] = <sip:192.168.0.1;lr>
[Max-Forwards] = 70
[From] = <sip:234@192.168.0.2;user=phone>;tag=39023f31bb6e43b7b8ed22e2f6730874
[To] = "8101" <sip:8101@192.168.0.2>;tag=C0EE058F-DF901570
[Call-ID] = c00d808b-eefe1a5-9ad97be@192.168.0.121
[CSeq] = 20697 BYE
[User-Agent] = SIPLink v2.1
[Content-Length] = 0
SIP req. to : 192.168.0.121:5060 (UDP) - 4/2/2010 4:43:49 PM
BYE sip:8101@192.168.0.121 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK-P899b4745j2b3bf2a
Via: SIP/2.0/UDP 192.168.0.1:5060;received=192.168.0.2;rport=5160;branch=z9hG4bKPj829b93bb4f724a5fa8fbcefcd37d2e26
Max-Forwards: 69
From: <sip:234@192.168.0.2;user=phone>;tag=39023f31bb6e43b7b8ed22e2f6730874
To: "8101" <sip:8101@192.168.0.2>;tag=C0EE058F-DF901570
Call-ID: c00d808b-eefe1a5-9ad97be@192.168.0.221
CSeq: 20697 BYE
Record-Route: <sip:192.168.0.1;lr>
User-Agent: SIPLink v2.1
Content-Length: 0
SIP req. from : 192.168.0.121:5060 (UDP) - 4/2/2010 4:43:49 PM
SIP/2.0 481 Call Leg/Transaction Does Not Exist
[Via] = SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK-P899b4745j2b3bf2a
[Via] = SIP/2.0/UDP 192.168.0.1:5060;received=192.168.0.2;rport=5160;branch=z9hG4bKPj829b93bb4f724a5fa8fbcefcd37d2e26
[From] = <sip:234@192.168.0.2;user=phone>;tag=39023f31bb6e43b7b8ed22e2f6730874
[To] = "8101" <sip:8101@192.168.0.2>;tag=C0EE058F-DF901570
[CSeq] = 20697 BYE
[Call-ID] = c00d808b-eefe1a5-9ad97be@192.168.0.221
[Record-Route] = <sip:192.168.0.1;lr>
[User-Agent] = PolycomSoundPointIP-SPIP_300-UA/1.6.6.0036
[Content-Length] = 0
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02.04.2010 21:26:49
Admin
Administrator
Posts: 254
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Hi,
Can you send me full sesison log which includes INVITE dialog?
Best regards,
Yasin KAPLAN
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04.04.2010 22:36:56
oldDude
Posts: 16
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It looks like I already overwrote the old log file. I'll set it up again tomorrow, and get back to you ASAP.
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05.04.2010 09:13:06
Admin
Administrator
Posts: 254
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OK, I'll be waiting for the log.
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05.04.2010 13:43:03
oldDude
Posts: 16
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I have reproduced the problem. In the trace, I initiate the call from the Polycom phone both times. In the first call, I end the call from the Polycom (at 121) and everything works fine. In the second call, I end the call from the other side and I see the same problem in the call-id. If the file attach didn't work, let me know...
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05.04.2010 14:27:39
oldDude
Posts: 16
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I just noticed another problem, but it appears to be mine. If you notice in the trace, the From and To: header contents have been switched in the BadBye. I'll check that from my side. But still, doesn't explain change in Call-ID.
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05.04.2010 16:40:12
Admin
Administrator
Posts: 254
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Hi,
Via header seems faulty in BYE request in second dialog;
[Via] = SIP/2.0/UDP 192.168.0.1:5060;rport;branch=z9hG4bKPj5ad507b9ecd74b5e8454178e59ed3651
it should be something like;
[Via] = SIP/2.0/UDP 192.168.0.2:5160;rport;branch=.....
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05.04.2010 17:33:00
oldDude
Posts: 16
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Yes, I just found the same thing. When I fix it at my end, I'll try again and let you know. I'm sure it will work better if I don't send you junk!
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05.04.2010 17:43:18
oldDude
Posts: 16
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Problem resolved. I was causing it by loading up bad Via info as you noticed. Thanks again. Everything works fines now...
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06.04.2010 08:37:17
Admin
Administrator
Posts: 254
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You welcome
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