Extensions.conf configuration

Hi,

I’m using asterisk 20.6.0 and a Grandstream HT813. I’ve been trying things out in extensions.conf but being completely new I’ve been stumped. I’d like to do the following:

1 call comes in on fxs, fxo does not ring yet
2 asterisk gets an invite and plays a message
3a if the user presses a dtmf key:
4 asterisk rings the fxs
5 if the fxs is answered before 4 rings
6 call proceeds normally until fxs is hung up
else
7 call goes to telco voicemail on the fxo side
else
8 asterisk hangs up the fxs

Sorry for being clueless. Is this do-able or am I trying to do the impossible? I’m not sure the HT813 has the proper configurations to do this. I can get as far as step 7, but I can never get the telco to go to voicemail

Stumped. :frowning:

Thanks for your time.

Dave

Playback. Note you have the option of using early media here, but the network may not allow it.

Read and conditional branching.

Dial

Not possible to do accurately. You have to use a timeout on Dial, which has been chosen to achieve this.

Wait until telco does its missed call thing. You can’t make an outgoing FXO call as it busy on the incoming one, and it is going to be difficult or impossible to get the gateway to do a register recall, although you could try Transfer, or SendDTMF, the latter using the F pseudo digit for recall.

Doing this after 7 is going to be difficult. Asterisk will hang it up when the Dail completes, and as it is still on hook, you won’t get a called party hangup handler run.

Hi david551,

Thanks for the reply. I’m still learning and I don’t know what you mean by “early media.”

I’ll investigate this just to learn. And I understand:

So there doesn’t appear to be a work-around at all. I’ll still try to get my head around everthing, and I do have a fall-back plan that will work, though not optimal.

Thanks again. Your comments are much appreciated.

Dave

I got things a bit confused on the last point. What I was trying to say is that I don’t think you can use the option to continue the dialplan on the called channel, after the calling one hangs up, because the called channel isn’t yet up. The bit about hangup handlers was talking about the wrong mechanism.

Hi david551,

I understand. Thanks. Just one question. I’m finding it difficult to wade through the documentation for extensions.conf commands. Is their a succinct document or user’s manual somewhere (other than the mutiple web pages at asterisk.org)?

Thanks again,

Dave

The Github sample is here: asterisk/configs/samples/extensions.conf.sample at master · asterisk/asterisk · GitHub