Wow! Thank you so much, James - you have certainly clarified lots of things in my mind. You are correct about me overlooking the feedback issue (with the el-cheapo device). I see that I have to learn. This world of VoIP is new and mind boggling - to me.
Thanks, Lynn --- James FitzGibbon <[EMAIL PROTECTED]> wrote: > On 8/1/07, Linux Lover <[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 > > > > 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? > > > You can send arbitrary DTMF over any of Asterisk's > channels from the > dialplan. I just figured that this level of > integration was a bit deeper > than you were looking for as a first project. It > would be an interesting > experiment, to be sure. The biggest issue I'd think > would be feedback - you > can send the DTMF along the wire, but how do you > know that the SOHO box > interpreted it correctly? If the only feedback is > designed for a human (i.e. > auditory), then interpreting those cues with > Asterisk would be non-trivial. > > > > 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? > > > Yes, your Verizon POTS line would go into a FXO port > in your server (which > in Asterisk would be referenced as the channel > "Zap/1" - zaptel being > Asterisk's TDM driver) and your SIP phone would > connect via your standard > office network and be referenced as > "SIP/whateverusernameyouwant". > > A very simplistic example of bridging a call would > be: > > [from-verizon] > exten => s,1,Dial(SIP/whateverusername) > > Assuming that you'd configured zaptel to route calls > that come in on the FXO > port to the Asterisk context named "from-verizon", > then any such calls would > immediately cause Asterisk to ring your SIP phone, > and if answered to bridge > the two calls together. > > A more complex example that makes them press one to > call you and otherwise > lets them leave a message: > > [from-verizon] > exten => > s,1,Background(Press1ToTalkOr2ToLeaveAMessage) > exten => s,n,WaitExten(10) > > ; timeout > exten => t,1,Goto(vm,1) > > ; invalid > exten => i,1,Goto(vm,1) > > ; press 1 > exten => 1,1,Dial(SIP/101,20) > exten => 1,n,Goto(vm,1) > > ; press 2 > exten => 2,1,Goto(vm,1) > > ; all voicemail activity ends up here > exten => vm,1,VoiceMail(u101) > exten => vm,n,Hangup > > [from-officephone] > exten => *98,1,VoiceMailMain > extne => *98,n,Hangup > > Assuming you've now set up your SIP phone as > extension 101, this would play > a sound file saying "press 1 to talk to 2 to leave a > message". If they > press 1, your SIP phone rings. If they press 2, > they go to voicemail. If > they wait 10 seconds without pressing anything, or > press something other > than 1 or 2, they also go to voicemail. If they > press 1 to dial your phone > and you don't pick up after 20 seconds, they go to > voicemail. > > On your deskphone (could just as easily be a SIP > softphone if you prefer), > you can dial *98 to log in and pick up your new > voicemail messages. > > Hope that demystifies some of what you're trying to > do. > > -- > j. > > _______________________________________________ > --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 ____________________________________________________________________________________ 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
