[OpenSIPS-Users] Handing BYE instead of CANCEL before Answer

[Russell Bierschbach]

That SIP message is actually OpenSIPS receiving the BYE from the customer, the problem is OpenSIPS is not forwarding/relaying that BYE message to the Asterisk server that is handling the call, so Asterisk doesn't know the call ended, and it's still letting the phone ring (and possibly be answered) on the other side of the call.  I will get a more complete SIP trace up.

My environment basically looks like this:

Customers -> OpenSIPS Load Balancing -> Asterisk Pool -> OpenSIPS -> Vendors

Russ

-----Original Message-----
From: users-bounces at lists.opensips.org [mailto:users-bounces at lists.opensips.org] On Behalf Of Anca Vamanu
Sent: Friday, December 10, 2010 5:08 AM
To: users at lists.opensips.org
Subject: Re: [OpenSIPS-Users] Handing BYE instead of CANCEL before Answer

Hi Russell,

I see from the trace that the BYE is forwarded by OpenSIPS to the other 
end here:

E..:.. at .@.12
.B.O}j......&   .BYE sip:11110702923002334053 at public.address SIP/2.0
Max-Forwards: 67
To: 11110702923002334053 
<sip:11110702923002334053 at customer.address>;tag=as6fbb0b2b
From: <sip:+963968716414 at customer.address>;tag=3500915879-587280
Contact: <sip:customer.address:5060;transport=udp>
Call-ID: 31586975-3500915879-587143 at customer.net
CSeq: 2 BYE
Via: SIP/2.0/UDP public.address;branch=z9hG4bK9188.2b1bd8b4.0
Via: SIP/2.0/UDP 
customer.address:5060;rport=5060;received=customer.address;branch=z9hG4bK15b3b0b1ea8e4d40f85f0cc6056f0968
Content-Length: 0

But I don't know of a way you could replace that Bye with a Cancel..

Regards,

-- 
Anca Vamanu
www.voice-system.ro




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