So I am not sure this is "exactly" supported at this time... http://track.sipfoundry.org/browse/XX-7362
<http://track.sipfoundry.org/browse/XX-7362>as a result you might consider trying this: http://track.sipfoundry.org/browse/XX-7461 On Fri, Sep 24, 2010 at 4:04 PM, Ujjval Karihaloo <[email protected]>wrote: > I found out if I disable NAT traversal (for Endpoints) this call flows > works and I get 2 way a audio. > This would mean I can forward call off of an AA and get 2 way audio, but > endpoints behind NAT wont be able to register to SIPx... > > Any ideas? > > > > -----Original Message----- > From: Tony Graziano [mailto:[email protected]] > Sent: Sunday, August 22, 2010 7:53 AM > To: [email protected]; [email protected]; Ujjval Karihaloo > Cc: [email protected] > Subject: Re: [sipx-users] SIP Trunk --> AA--> SIP trunk call flow > > If it is available, I assume one could edit the xml file directly and > restart the service (no config changes from sipxconfig,so projection would > not happen) and try it? Or edit the vm "to on" by default? > ============================ > Tony Graziano, Manager > Telephone: 434.984.8430 > Fax: 434.984.8431 > > Email: [email protected] > > 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: [email protected] > <[email protected]> > To: 'Douglas Hubler' <[email protected]>; 'Ujjval Karihaloo' > <[email protected]> > Cc: [email protected] <[email protected]> > Sent: Sun Aug 22 09:37:36 2010 > Subject: Re: [sipx-users] SIP Trunk --> AA--> SIP trunk call flow > > > > > On Fri, Aug 20, 2010 at 3:52 PM, Ujjval Karihaloo > > <[email protected]> wrote: > > > Guys: > > > > > > > > > Looking for some help on this....has anyone tried thisL I get no Audio > > in > > > each direction. Tony tried it with 2 different ITSPs and it works...but > > I > > > cannot get it to work with only one ITSP that I have to test. > > > > > > Any other suggestions from group as to where to look for a solution > > to this > > > issue: > > > > Sounds like this issue that i'm still waiting for a toplink test > > account to test with > > http://track.sipfoundry.org/browse/XX-8663 > > > > Because the call seems to be successful, it's just that the audio is > > missing, I'd look closer at the RTP message in both the SDP portion of > > the SIP messages the wireshark of RTP source and destination. > > > > In a successful call, i'm not sure if the final RTP path should flow > > thru sipxbridge at all, in theory in shouldn't have to AFAIK. > > I think this question relates to: > http://track.sipfoundry.org/browse/XX-7362. While this was implemented, > the > setting in sipXconfig has not: http://track.sipfoundry.org/browse/XX-7461 > > The call flows through sipXrelay, which is part of the proxy, and not > sipXbridge. > > --martin > > > _______________________________________________ > 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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