On 8/1/07, <b class="gmail_sendername">Linux Lover</b> <<a href="mailto:linuxlover992000@yahoo.com">linuxlover992000@yahoo.com</a>> wrote:<div><span class="gmail_quote"></span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
> This SOHO PBX box won't interop with Asterisk<br>> because it doesn't speak any<br>> of the protocols that Asterisk does. This box<br><br>I tend agree with your evaluation. Still, I was<br>thinking that since all these el-cheapo SOHO PBX boxes
<br>support manual attendant call transfer, what's to<br>prevent Asterisk from mimicking an attendant by<br>sending proper DTMF signals and make this box<br>"transfer" the call to the single analog phone in the
<br>business? That is, Asterisk will connect (via RJ-11)<br>to the unit as the "attendant's phone", and my real<br>phone (only one in the system) will connect via a<br>second RJ-11 (there could be 4 of them).
<br><br>Or is Asterisk not capable of sending DTMF signals<br>over an RJ-11 connection?</blockquote><div><br>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.<br></div><br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<br>Do I undestand correctly that with this solution, I<br>will still be able to connect to my analog Verizon<br>phone line with the SIP phone? That is, the outside<br>world will see my phone as an ordinary phone, when in
<br>fact I am using a SIP phone? If so, that means that<br>Asterisk does all the magic behind the scene, right?</blockquote><div><br>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".
<br><br>A very simplistic example of bridging a call would be:<br><br>[from-verizon]<br>exten => s,1,Dial(SIP/whateverusername)<br><br>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.
<br><br>A more complex example that makes them press one to call you and otherwise lets them leave a message:<br><br>[from-verizon]<br>exten => s,1,Background(Press1ToTalkOr2ToLeaveAMessage)<br>exten => s,n,WaitExten(10)
<br><br>; timeout<br>exten => t,1,Goto(vm,1)<br><br>; invalid<br>exten => i,1,Goto(vm,1)<br><br>; press 1<br>exten => 1,1,Dial(SIP/101,20)<br>exten => 1,n,Goto(vm,1)<br><br>; press 2<br>exten => 2,1,Goto(vm,1)
<br><br>; all voicemail activity ends up here<br>exten => vm,1,VoiceMail(u101)<br>exten => vm,n,Hangup<br><br>[from-officephone]<br>exten => *98,1,VoiceMailMain<br>extne => *98,n,Hangup<br><br>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.
<br><br>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.<br><br>Hope that demystifies some of what you're trying to do.<br>
</div></div><br>-- <br>j.