[asterisk-users] Hardware that can ring my phone?

[Anthony Francis]

James FitzGibbon wrote:
> On 8/1/07, *Linux Lover* <linuxlover992000 at yahoo.com 
> <mailto:linuxlover992000 at yahoo.com>> 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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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