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... 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. On Wed, Aug 11, 2010 at 4:25 PM, Ujjval Karihaloo <[email protected]>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? > > > > _______________________________________________ > sipx-users mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > -- ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: [email protected] Fax: 434.984.8431 Email: [email protected] LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: [email protected] Fax: 434.984.8427 Helpdesk Contract Customers: http://www.myitdepartment.net/gethelp/ Why do mathematicians always confuse Halloween and Christmas? Because 31 Oct = 25 Dec.
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