you would still need an fxo port of some sort for asterisk for it to "pretend" to be a phone.
Quoting Linux Lover <[EMAIL PROTECTED]>: > James, thank you for your educating answer. > > --- James FitzGibbon <[EMAIL PROTECTED]> > wrote: > >> >> This SOHO PBX box won't interop with Asterisk >> because it doesn't speak any >> of the protocols that Asterisk does. This box >> appears to be a solid-state >> (and I'd assume very feature restricted) alternative >> to Asterisk. That it >> happens to have both FXO (to the Telco) and FXS (to >> the analog phone) ports >> doesn't mean that it is usable as an analog >> interface for Asterisk. >> > > I tend agree with your evaluation. Still, I was > thinking that since all these el-cheapo SOHO PBX boxes > support manual attendant call transfer, what's to > prevent Asterisk from mimicking an attendant by > sending proper DTMF signals and make this box > "transfer" the call to the single analog phone in the > business? That is, Asterisk will connect (via RJ-11) > to the unit as the "attendant's phone", and my real > phone (only one in the system) will connect via a > second RJ-11 (there could be 4 of them). > > Or is Asterisk not capable of sending DTMF signals > over an RJ-11 connection? > > Not that I am rushing to buy this cheap box right now, > but I am curious whether this is possible at all - > perhaps to get a better feel of how flexible Asterisk > is. > >> >> The original single-FXO-port card from Digium was >> the X100P. These aren't >> sold anymore (the TDM400B modular card replaced it), >> but they can be found >> on eBay for $10-$30. If you can get your hands on >> one, you might consider >> going with a cheap SIP phone instead of a analog >> phone for your business. >> There isn't (as far as I know) a readily available >> cheap single-FXS-port >> card. If you go with an analog phone behind >> Asterisk, you'll need an FXS >> port. If you go with a SIP phone, you just need to >> have a network >> connection from the phone to the server, which might >> be cheaper. A quick >> search on eBay shows a few Grandstream Budgetone 101 >> phones (certainly not >> the best available, but they'll do the job) in the >> sub-$50 range. >> > > Do I undestand correctly that with this solution, I > will still be able to connect to my analog Verizon > phone line with the SIP phone? That is, the outside > world will see my phone as an ordinary phone, when in > fact I am using a SIP phone? If so, that means that > Asterisk does all the magic behind the scene, right? > > Thanks, > Lynn > > > > ____________________________________________________________________________________ > Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now (it's > updated for today's economy) at Yahoo! Games. > http://get.games.yahoo.com/proddesc?gamekey=monopolyherenow > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > Jon Pounder _/_/_/ _/ _/ _/ _/_/_/ _/ _/ _/_/_/_/ _/ _/_/ _/ _/ _/ _/_/ _/ _/_/ _/ _/ _/_/ _/ _/ _/ _/_/ _/ _/_/_/ _/ _/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com ---------------------------------------------------------------- This message was sent using IMP, the Internet Messaging Program. _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
