I've got a couple options. Option 1 - Just buy IP phones and use those. Have all of them talk to the SIP provider and be done with it.
Option 2 - Buy a VoIP appliance (there are lots of different ones) with enough analog jacks for your phones (or maybe a mix of analog and digital phones) and use their management gui (probably a local or remote web server) to configure the system. Option 3 - Roll your own Asterisk system or use Trixbox (which is easier than building Asterisk from scratch) and an analog card or two. I happen to have a Digium 8 port FXS card that we just took out of service, if you're interested. Chris On Mon, May 18, 2009 at 5:13 PM, Tim Jackson <jackson....@gmail.com> wrote: > > 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 -~----------~----~----~----~------~----~------~--~---
