No. Per their setup requirements, I don't register. I have the option unchecked under ITSP account. Correct. They just route to the IP we provided them.
On 2/1/2010 8:07 PM, Tony Graziano wrote: > Verizon isn't supporting registrations are they? I think they are just > sending the DID toa specific ip address. > ============================ > 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: Eric Varsanyi<[email protected]> > Cc: Scott Lawrence<[email protected]>; [email protected] > <[email protected]>; Tony Graziano > <[email protected]> > Sent: Mon Feb 01 20:23:10 2010 > Subject: Re: [sipx-users] "media release" support > > Thanks. Tools for my toolbox is what I need right now. > I guess I'm looking for that perfect mix of decentralized vs > centralized. I need to have as little as possible running at the remote > facility. This goes way beyond just telephony. With that being said I'm > going to take the points presented by you guys back to my boss. Thanks > for many of those points Tony. Given the price of a small FXO device, I > think I can make the sell for it. I can probably sell it on the point of > good PR (in case the circuit goes down) over safety. I know it shouldn't > be that way, but it is for me. > So, with trying to keep as much central as possible, if I configure > nothing but sipxbridge at the remote sight, can a I keep calls to and > from the PSTN from coming back through my central server? I would have > the phones registering with and using voicemail on the central server. I > would specify the appropriate sipxbridge instance in a dial plan for > each group of users. Does that work from a network flow point of view? > If not, skip the next paragraph. > If the above works (and I've sold my boss on POTS backup), I need to > configure the handsets (Polycom 450s) to use an FXO gateway if the MPLS > connection/sipxbridge gateway is down. I'm slightly partial to the > Audiocodes MP series. I got one working for inbound backup in about 20 > minutes, so that is a testament to Sipx's ability to manage it! Is it > possible to configure the Polycoms to use an Audiocodes FXO directly > with their primary line for outbound calls if they can't get out through > sipxbridge? If not, is their a minimal sipx setup I can use as a > secondary sipx server that the handsets could use if the primary isn't > available? > > Thanks as always, > Matthew > > On 2/1/2010 5:37 PM, Eric Varsanyi wrote: > >> In the polycoms (and I presume pretty much any other multi-line phone) you >> can register external lines that register direct from the handset to a >> voip provider. SipXecs manages the phone configuration in this case but >> otherwise doesn't get involved. The normal sipXecs lines continue to work >> normally. >> >> This wouldn't be seamless but it might be useful in your situation as the >> talk path would go direct from the branch office to Verizon. I use this >> sort of setup on the 650's as a backup in case the sipXecs server goes >> down and I want to dial 911 via a local Patton bridge. Unless verizon >> supports it you wouldn't get forward on no answer to voice mail, likely >> that would kill the idea right there. Verizon may also not support >> multiple registrations on a given DID number so each user would need a >> separate external phone #. >> >> Likely its not appealing at all, just a possible tool in the toolbox. >> >> -Eric >> >> >> >> _______________________________________________ sipx-users mailing list [email protected] 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/
