I have this same issue or very similar issue.

A - call comes from anywhere to my VoIP DID
B - Call is routed to a Phantom ext using SIP Alias of the DID in the Alias
section of the Phantom DID
C - Call is forwarded always to the Auto Attendant setup for this DID
D - User selects an option inside the AA and is routed to there destination
******* now from any point before this if the call hangs up the BYE is
accepted ********
E - User is routed from the option inside the AA to say a Call Queue
******* if the calling user hangs up while MoH is playing for the Call Queue
the call will not hangup when the BYE is sent from the ITSP, once the MoH is
finished playing i am assuming the call is dropped becuase it is not longer
available but the SipX system did not see this durring the playback of MoH
********







On Sat, Feb 6, 2010 at 8:22 AM, Sven Evensen <[email protected]>wrote:

>  I will first explain the scenario
>
>
>
> A Mobile 07795951717
>
> B Internal phantom user 8116
>
> C Internal phantom user 8216
>
> D Mobile 07791788997
>
>
>
> B calls A, A answers
>
> B transferred to C
>
> C transferred to D
>
> A hangs up before D answers
>
>
>
> This BYE coming from A is discarded by sipXBridge.
>
> If D answers, then A hangs up, the call disconnects properly
>
> This is sipX 4.0.4, using UK SIP providers. Exact same problem on two
> different servers using two different
>
> SIP providers
>
>
>
> In  sipXBridge.log I observe :
>
>
>
>
> "2010-02-05T16:22:27.283000Z":17845:JAVA:DEBUG:ec2-79-125-17-66.eu-west-1.compute.amazonaws.com:Thread-635:00000000:CallControlManager:"serverTransaction
> Not found -- stray request -- discarding "
>
>
>
> I have attached sipXBridge.log and the merged.xml. The snapshot is too
> big..
>
>
>
> Regards,
>
> Sven
>
> *Sven Evensen**, Technical Support Consultant*
>
> *OnRelay*
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-- 
Thank You,
Chris Rawlings
IT Consultant

phone - 610.741.3324

VCP RHCE RHCT MCSE
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