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

[James FitzGibbon]

On 8/1/07, Linux Lover <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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