[asterisk-users] Asterisk Management API

[Peter Childs]

On 11 March 2010 21:09, Matt Riddell <lists at venturevoip.com> wrote:
> On 9/03/10 9:13 PM, Peter Childs wrote:
>> Also is there some way to get the starting end to auto pickup, (or at
>> least hit for this to happen (I'm using SIP if that helps))
>
> When you make an originate request it works like this:
>
> 1. Call is made to the "Channel" parameter.
> 2. When the "Channel" answers it connects the other end to the
> application/context/extension.
>
> So, send the channel to the SIP device and then the other end won't
> start till the SIP device picks up.
>

Yes I got that, and it seams to work quite well, It does mean that its
more difficult to actually have a call going to a dead phone when it
gets sent from the wrong channel in error.

>>> 2. Send DTMF to the far end, PlayDTMF looks like it should work but it
>>> seams to send the Play the DTMF to my end not the far end.
>>
>> I seam to be able to send it to the far end by finding far end
>> channel's name and using that instead, but this does not work if the
>> far end is not a channel, (eg the Answer phone) but I hope that will
>> not really be a problem...
>
> Again, looks like you have the order of the channels round the wrong way.
>
> If you originated to a SIP device and sent the other end to the
> application PlayDTMF, then it would be sent to the SIP device (if that's
> what you want).
>

I figured that out. It means that if you want to control your calls
when in you own menus, you can't do it by send DTMF but need to use
the underlining application/dial-plan. which makes things more complex
than they should be.

Peter.