[asterisk-users] Reading DTMF sent by callee during a SIP call

[Ishfaq Malik]

On 20 December 2013 16:13, Alex <ralienpp at gmail.com> wrote:

> Hi everyone,
>
> I am looking for advice about the design of a SIP-based intercom. I
> count on your help, as my current attempts are not fruitful (yet).
>
> This will be a pretty long message, so here's my fundamental question:
>
> Is there a way to interpret DTMF tones sent by the calee
> (not the caller) while a voice call is in progress?
>
>
>
>
>
>
> Here's the desired scenario:
>
> - there is a box with speakers and a mic
> - Asterisk is running on a computer inside that box
> - the box is embedded in a door
> - There are two user accounts, UserA and userB
> - UserA is a client that runs on the server*
> - UserA calls UserB and they are having a voice conversation
>
>
> Throughout the call, Asterisk must react to DTMF tones sent by userB;
> such that an action is executed when a specific key is pressed.
>
> The idea is to build an intercom that would enable me to open a door
> remotely, by relying entirely on SIP, so there would be no need to
> have some additional communication channel to send the "open door"
> signal.
>
>
>
>
> I have previously implemented IVRs using `Background` and jumped to
> specific extensions, when a button was pressed. But in that case, the
> extensions are dialed by the caller; whereas now the input must from
> the person who answered the call.
>
> If I use `Dial` and `Read` - the latter is only executed after `Dial`
> terminates - so this is not suitable.
>
>
> `Background` behaves like I need - but it plays back a predefined
> file, so it is not suitable for an interactive conversation.
>
>
>
> * Having a SIP client on the same machine as the Asterisk server
> itself is not possible, because both won't be able to bind to port
> 5060. My guess is that the solution is to originate a call from the
> CLI; but I haven't gotten to that part yet.
>
>
>
>
> Thank you for your patience, I am looking forward to your feedback,
> Alex
>
>
>
You could create your own feature in features.conf that executes a
Macro/Gosub defined in sip.conf...

Ish

-- 

Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
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