Yes. I have a site that does this. Let me check how it is configured and post back. ============================ Tony Graziano, Manager Telephone: 434.984.8430 Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 Fax: 434.984.8427 Helpdesk Contract Customers: http://www.myitdepartment.net/gethelp/ ----- Original Message ----- From: Ujjval Karihaloo <ujj...@simplesignal.com> To: Ujjval Karihaloo <ujj...@simplesignal.com>; Tony Graziano <tgrazi...@myitdepartment.net> Cc: sipx-users@list.sipfoundry.org <sipx-users@list.sipfoundry.org> Sent: Fri Aug 13 20:36:14 2010 Subject: RE: [sipx-users] SIP Trunk --> AA--> SIP trunk call flow Looks my call forward off of an Extension on SIPX to my cell phone is not working either... Any ideas befor eI start digging into traces....with sipX using so many internal ports (5090, 5080, 5090..15060)...the trace is complicated to debug From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Ujjval Karihaloo Sent: Friday, August 13, 2010 6:27 PM To: Tony Graziano Cc: sipx-users@list.sipfoundry.org Subject: Re: [sipx-users] SIP Trunk --> AA--> SIP trunk call flow Hi All: Does anyone have this call flow working. Call from ITSP and Autoattendant hairpins the cal lback to ITSP..I get no audio either way... Seems like a common scenario.... Call Forward always should work the same way..I will verify that too... From: Tony Graziano [mailto:tgrazi...@myitdepartment.net] Sent: Wednesday, August 11, 2010 3:07 PM To: Ujjval Karihaloo Cc: sipx-users@list.sipfoundry.org Subject: Re: [sipx-users] SIP Trunk --> AA--> SIP trunk call flow I think it might matter what the softphone is, and the version... I would "try" the AA using the phantom user just to see if the audio works. its not a solution, just a troubleshooting step to determine if the proxy cares or not. On Wed, Aug 11, 2010 at 4:42 PM, Ujjval Karihaloo <ujj...@simplesignal.com<mailto:ujj...@simplesignal.com>> wrote: See inline From: Tony Graziano [mailto:tgrazi...@myitdepartment.net<mailto:tgrazi...@myitdepartment.net>] Sent: Wednesday, August 11, 2010 2:37 PM To: Ujjval Karihaloo Cc: sipx-users@list.sipfoundry.org<mailto: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? _______________________________________________ sipx-users mailing list sipx-users@list.sipfoundry.org<mailto:sipx-users@list.sipfoundry.org> List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net<mailto:tgrazi...@voice.myitdepartment.net> Fax: 434.984.8431 Email: tgrazi...@myitdepartment.net<mailto:tgrazi...@myitdepartment.net> LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net<mailto:helpd...@voice.myitdepartment.net> 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. -- ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net<mailto:tgrazi...@voice.myitdepartment.net> Fax: 434.984.8431 Email: tgrazi...@myitdepartment.net<mailto:tgrazi...@myitdepartment.net> LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net<mailto:helpd...@voice.myitdepartment.net> 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. _______________________________________________ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
