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.
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