I recommend looking at sipecs (which can include freeswitch as a module if you really want it). They have a complete install cd, web-gui, and it autodetects and configures a multitude of voip handsets--plug-n-play.
On Tue, Apr 8, 2014 at 9:37 AM, Tod Hansmann <[email protected]>wrote: > I second Chris' sentiments. I'll also suggest looking at a hosted solution > like getjive.com which is like 25/month per handset for as much calling as > needs be and a ton of features including IVR creation. If one does go the > route of self-hosting, please do FreeSwitch instead of Asterisk. It's far > more flexible and performant for when he wants to change it later, and if > this becomes a major point of his business it can be fleshed out quite a > bit to power the business without incurring a lot of cruft and code modules > all over the place that become hard to maintain. That's why FreeSwitch > exists, because Asterisk is architected pretty bad for long-term. > > Anyway, that's my two cents. > > Cheers, > -Tod Hansmann > > > > On Tue, Apr 8, 2014 at 8:48 AM, Chris Wood <[email protected]> wrote: > > > On Tue, Apr 8, 2014 at 5:14 AM, Dan Egli <[email protected]> wrote: > > > > > > Someone I talk to on a regular basis is getting ready to open his own > > > company. The expected size of the company would be about 10-15 people > > once > > > he finishes hiring, with more to follow in a year or two (he hopes). > What > > > he wants is to have all his phone calls and what not handled by > Asterisk > > > using IP based phones (obviously). I've talked with him a lot about > this, > > > but the points I could not remember dealt primarily with how the phone > > > company would handle things. > > > > > > Supposing that you wish to have the possibility of half of these people > > > (we'll say seven) placing calls or receiving calls at the same time. > > Under > > > standard phone setups (POTS) that would require seven incoming lines. > Is > > > that going to be the case if all he buys is a DID number and internet > > > service? I.e. does he have to tell the phone company to sell him > multiple > > > DID numbers in a roll down sequence? Or is simply having the phone call > > > come to the system sufficient and then Asterisk handles it from there? > I > > > admit I have not looked at Asterisk myself (I'd love to, but I've got a > > few > > > other things I really need to accomplish before I start studying a > > > VoIP/PBS setup in detail) so I don't know the answers. I tend to think > > that > > > it's likely that he'd need multiple DID numbers and the roll down, even > > > though it's all IP based, because it's likely the phone company that's > > > going to generate the busy signal, and if they only have one DID number > > on > > > file for you, then when a second person tries to call they're just > going > > to > > > get a busy signal (or voice mail if you have that service). But I > > honestly > > > don't know for sure. I'm sure he'd also need to purchase one extra DID > > > number, not in the roll down sequence, that he can configure to > > > specifically be for the fax machine whose duties I'm sure Asterisk > won't > > be > > > able to assume (forward to the fax, sure, but actually answer as a fax > > and > > > send/receive transmissions? Some how I doubt it.) > > > > Generally speaking, you purchase/subscribe to SIP trunks to handle > > calls. The number of concurrent calls you want to handle is the > > number of trunks you need. The DID is based on the number of phone > > numbers you want to have -- they will run across those trunks. You > > can have more DIDs than you have trunks. You can have one DID and > > multiple trunks (everybody has extensions instead of their own phone > > number). > > > > Companies typically charge you for DIDs and then charge you for > > trunks. The DID charges vary a lot and can be more expensive than > > they should be. > > > > For DID and trunks, at my last company we used: > > https://www.flowroute.com/ > > > > Fax can be obnoxious with VOIP. I would recommend getting a copper > > line from the local telco. Maybe somebody on the list has the magic > > sauce to make fax easy on voip. > > > > > > > So, what exactly do I tell him to ask the TeleCo for? He's going to be > in > > > southern Utah most likely (i.e. St. George area) so it's likely going > to > > be > > > Qwest or AT&T, unless you can get phone/DID service from someone like > > > McCloud USA, which I honestly don't know. He plans this to be a heavy > IP > > > business, so I don't think Comcast would be such a good idea, with the > > > monthly bandwidth caps they like to place on downloads and uploads (has > > > anyone heard of Comcast offering more than 1 mbit upload speeds?), not > to > > > mention their 250 GB/Mo download limit. > > > > Those limits on Comcast are for residential use. Your friend will > > want a business grade internet account. Which company supplies the > > SIP service can be independent of the company providing the internet > > service. I would get the most reliable ISP he can get in his area and > > then do the VOIP separate if the ISP isn't competitive. > > > > /* > > PLUG: http://plug.org, #utah on irc.freenode.net > > Unsubscribe: http://plug.org/mailman/options/plug > > Don't fear the penguin. > > */ > > > > /* > PLUG: http://plug.org, #utah on irc.freenode.net > Unsubscribe: http://plug.org/mailman/options/plug > Don't fear the penguin. > */ > /* PLUG: http://plug.org, #utah on irc.freenode.net Unsubscribe: http://plug.org/mailman/options/plug Don't fear the penguin. */
