The purpose of enabling remote users is to pass the registration and traffic generated from those remote registrations through sipxrelay. By disabling remote users, the functions is closed and it is doing exactly what you told it to do.
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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