Most ITSPs and their SBCs will release audio if the 2 endpoints are from behind the same firewall/IP... So audio wouldn't actually go out and come back in, just be hairpinned @ the ATA...
Might be a good thing to check with the ITSP though... -- Tim On Mon, May 18, 2009 at 5:08 PM, Don McMorris <don.mcmor...@gmail.com> wrote: > > On 5/18/09, Steven S. Critchfield <cri...@basesys.com> wrote: >> >> ----- "Kevin Hart" <bowl...@gmail.com> wrote: >> > Is there any decent, not bank breaking, SIP ATA's out there that can >> > handle >> > more than one phone line over voip? >> > >> > Say we have a small office and want to just use basic phones. Have a >> > SIP >> > server already out there that we can connect to on the net that will >> > provide >> > dial tone to us. Rather not build an Asterisk box for the >> > issue...just need >> > some adapters to plug those phones into. Like say at least 4 phones, >> > to say >> > 2-8 phone lines. >> > >> > Is this just a pipe dream? Am I going to be forced to build out a >> > full >> > phone system for this office just for this issue? >> >> Yes your idea is a bit of a pipe dream. Here is why. >> >> Without putting in asterisk or similar software, how do you handle the >> presentation of calls. Specifically, if a call comes in, which SIP adapter >> gets the call and which phone gets rung? Do you want more than one to be >> rung? > If I understand "Have a SIP server already out there that we can > connect to on the net that will provide dial tone to us" correctly, > you want to use a third-party SIP host with a 4-line ATA. That is, > all your calls (even intra-office) would be through the ATA, to the > ITSP, and back to the ATA... right? > >> >> You wouldn't be all that bad off using a asterisk box to be a funnel on >> both sides for your calls and presentation. Plus you have a few options >> for your hardware then. You could go and look ahead to being bigger and >> buy a channel bank and a T1 card to interface it in. This would let you >> use some cheap 2 line phones at your desktop and go from the 2-8 to 4-16 >> lines needed. This allows you to put some one on "hold" while you select >> the second line to confer with someone. Or you could turn off callwaiting >> and just see the blinking second line as an indicator instead of interupting >> calls. Of course you could do the same with some SIP UAs and reduce your >> wiring, but again, having asterisk or similar in your own control allows >> you to dictate behavior on your own terms instead of waiting for your >> provider set things up for you. >> >> -- >> Steven Critchfield cri...@basesys.com >> >> > >> > > > > --~--~---------~--~----~------------~-------~--~----~ You received this message because you are subscribed to the Google Groups "NLUG" group. To post to this group, send email to nlug-talk@googlegroups.com To unsubscribe from this group, send email to nlug-talk+unsubscr...@googlegroups.com For more options, visit this group at http://groups.google.com/group/nlug-talk?hl=en -~----------~----~----~----~------~----~------~--~---
