James FitzGibbon wrote: > On 8/1/07, *Linux Lover* <[EMAIL PROTECTED] > <mailto:[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 the way to have * send dtmf is with the D option, w inserts a half second pause.
As an example I have a remote location that needs special 911, so they have a landline that connects to a linksys SPA, it doesnt like being passed the destination number through sip, so O do it this way: exten => 911,1,Dial(SIP/08CCB243-911,,D(w911)) works awesome, it connects, plays back the DTMF, and then passes the audio stream to the caller. Anthony _______________________________________________ --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
