I'd like to see a siptrace (not pcap) of the call myself, with the proxy set to debug.
On Mon, Aug 16, 2010 at 8:16 PM, Ujjval Karihaloo <ujj...@simplesignal.com>wrote: > Any other suggestions from group as to where to look for a solution to > this issue: > > > > Calling from PSTN and going out to PSTN(Hairpinned) . No UAs involved SIPx > not behind > NAT. SIP trunk is registered with ITSP > > PSTN--> ITSP (me)--> sipX (Sip Trunk) à Hits AA on SIPx and callers is > prompted to press 2 which thenroute the call back out > > > sipX--> SIP trunk --> ITSP-->PSTN > > > > > > > > *From:* Tony Graziano [mailto:tgrazi...@myitdepartment.net] > *Sent:* Friday, August 13, 2010 6:51 PM > > *To:* Ujjval Karihaloo > *Cc:* sipx-users@list.sipfoundry.org > *Subject:* Re: [sipx-users] SIP Trunk --> AA--> SIP trunk call flow > > > > Yes, this works for this site... > > > > ITSP1>AA>FWD through ITSP2 > > > > It just so happens they have two itsp's, one for inbound, the other for > outbound, but it does work. > > > > > > On Fri, Aug 13, 2010 at 8:42 PM, Tony Graziano < > tgrazi...@myitdepartment.net> wrote: > > 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. > > > > > -- > ====================== > Tony Graziano, Manager > Telephone: 434.984.8430 > sip: tgrazi...@voice.myitdepartment.net > Fax: 434.984.8431 > > Email: tgrazi...@myitdepartment.net > > LAN/Telephony/Security and Control Systems Helpdesk: > Telephone: 434.984.8426 > sip: 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 Fax: 434.984.8431 Email: tgrazi...@myitdepartment.net LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: 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.
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