See inline

From: Tony Graziano [mailto:tgrazi...@myitdepartment.net]
Sent: Wednesday, August 11, 2010 2:37 PM
To: Ujjval Karihaloo
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] SIP Trunk --> AA--> SIP trunk call flow

What you are describing is a hairpinned call. You should provide a siptrace of 
the call with the proxy at debug as a minimum.



You should also describe your environment...

what kind of phone/ua (firmware software version might be relevant), whether 
the UA or sipx is behind a nat or if the user is remote, and how you connect to 
the siptrunk...
Calling from PSTN and going out to PSTN. No UAs involved SIPx not behind NAT. 
SIP trunk is registered and normal outbound calls from a Sofphone registered to 
sipX and calling out the SIP trunk works.

PSTN--> ITSP (me)--> SIP Trunk to sipX--> sipX

sipX--> SIP trunk --> ITSP-->PSTN




I would also be curious to know if you created a phantom user (user with no 
phone) and set the account to forward all the time to one of the cell phones 
and changed the AA to point that option to the phantom user, whether or not you 
have audio.
No Phantom user...Just the Default AA in the System--> Dial Plan is using an AA 
with option 1 going to a PSTN number and option 2 going to a second PSTN number.
On Wed, Aug 11, 2010 at 4:25 PM, Ujjval Karihaloo 
<ujj...@simplesignal.com<mailto:ujj...@simplesignal.com>> wrote:
I have a call coming in via a Sip trunk to an extension assigned to an AA.

AA plays the prompts user to dial 1 or 2...

In either case I send the call back out over the SIP trunk to a Cell PSTN 
number. The call connects but I have no Audio either way.

Which Logs should I collect and provide to the group?


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