Don't know if it can help you, but please take a look at this: http://sipx-wiki.calivia.com/index.php/SipXbridge_Overview_and_Configuration
That says: "Typically ITSPs do not handle certain types of SIP requests such as *REFER*which is used in Call Transfer operations. To implement call transfer, SipXbridge does signaling translation, converting a *REFER* request to an * INVITE* request to the call transfer target. Consequently, a ringing tone will not be heard at the calling phone during call transfers when the call is routed through SipXbridge" On Thu, Dec 3, 2009 at 8:04 AM, Tony Graziano <tgrazi...@myitdepartment.net>wrote: > When you login to sipxconfig it will display at the bottom. > > ============================ > 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: Dan White <dwh...@citadelitg.com> > To: Tony Graziano <tgrazi...@myitdepartment.net>; sipx-users > <sipx-users@list.sipfoundry.org> > Sent: Wed Dec 02 16:03:13 2009 > Subject: RE: [sipx-users] Problem with call forwarding > > Sorry, I am using Vitality.net as my sip trunk. > > I am not sure which version of the software I am using, how do I tell? > > > > Dan White > > From: Tony Graziano [mailto:tgrazi...@myitdepartment.net] > Sent: Wednesday, December 02, 2009 3:55 PM > To: Dan White > Subject: Re: [sipx-users] Problem with call forwarding > > > > A little better explanation would be helpful. > > > > What kind of gaateway is the call coming in on, out on? > > > > Is your forwarding a user forwarding (at the same time or if no answer? > If it goes out a PSTN (analog pots) gateway, you might want to confirm > the gateway works and has a more gracious timeout period since the call > setup is a w-e-e-e-e bit longer than a digital one. Are you using 4.0.4? > > > > Thanks. > > On Wed, Dec 2, 2009 at 3:46 PM, Dan White <dwh...@citadelitg.com> wrote: > > I setup a user when a call comes in it gets routed to an internal ext > say 204, if it no one picks up, it suppose to forward to an external > number, but this just doesn't seem to work. Sometimes it rings on the > other side, but you can't hear anything. Is this a known issue? > > > > Dan White > > > _______________________________________________ > sipx-users mailing list sipx-users@list.sipfoundry.org > List Archive: http://list.sipfoundry.org/archive/sipx-users > Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users > sipXecs IP PBX -- http://www.sipfoundry.org/ > _______________________________________________ > sipx-users mailing list sipx-users@list.sipfoundry.org > List Archive: http://list.sipfoundry.org/archive/sipx-users > Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users > sipXecs IP PBX -- http://www.sipfoundry.org/ >
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