"Pulling" a call out of the VoiceMail system and &

[descore]

Hi,

I want to use Asterisk's voicemail system instead of our answering machine, but I have an issue with WAF (Wife Acceptance Factor) that I need to resolve before I can do it.

Basically when our answering machine takes a call you can hear the caller in the other end and if you pick up the handset, the answering machine stops.

This has 2 purposes:

1) If you are running around to find the cordless phone it gives you some extra time after the answering machine kicks in.

2) You can hear who is calling and if it's someone important you can pick it up even if you weren't bothered otherwise.

I would like to emulate this functionality with *.

So there are two separate issues here. I don't know if both can be solved and I think they have separate solutions anyway:

1) I want to be able to transfer a call that is currently in the VoiceMail system leaving a message, over to another extension by dialing some code.

2) If its possible, I want to be able to copy the sound of a call out of the * server's sound card.

Does anyone have any ideas for how to accomplish these?

TIA,
Descore

[PhoneBoy]

Asterisk stores the files as WAVs I believe, so it should be trivial to pump the voicemail sound out the speakers. Of course, that doesn't help when you're leaving a message. Not sure how to get the "live" call to play out the speakers.

[descore]

I guess it's something with Monitor. I wrote a post earlier about how I'd set up Asterisk to record calls, so I guess if I could pipe the output of Monitor into sox in realtime I could find a way get it out the speaker quite easily.

The other part of the problem is worse though - how do I get the call transferred back to my handset by entering a code or something? This is where I'm stuck, if I figure out how to do this I'm sure I can work out a way to get the sound out. I'm sure there are some * commands that will let me do it but I need some sort of example to work with since I'm pretty new with the dialplan syntax.

[PhoneBoy]

I believe once the call hits the voicemail system, it's basically a "black box" you can't get it out of.

[muppetmaster]

You could hack the code, but beyond that not aware of a way to do it. But this vmail is no different than those provided by your local RBOC. It's time to ditch the old answering machine concept anyway...

[descore]

Ok I guess I'll have to stick with the "old fashioned" answering machine until I have some time to mess around with the code.

What I envision to do is something like this:

; My master extension
exten => 100,1,Dial(SIP/101&SIP/102,120)
exten => 100,2,SetGlobalVar(CallInVoicemail=${CALLID}) ; Is there something like this
exten => 100,3,MonitorCustom("/usr/bin/mymixerscript")
exten => 100,4,VoiceMail(u100)

; Code for grabbing call in voicemail
exten => 199,1,GrabCall(${CALLINVOICEMAIL})
exten => 199,102,Playback(custom/no-call-in-voicemail)

So I'd need two "new" commands:

MonitorCustom - Like Monitor, but will pass to the program two file descriptors or unix sockets with the raw audio of the two sides of the call. A small standalone app would then mix the channels and play them via /dev/dsp or something.

GrabCall - Would behave a bit like Dial, but instead connect the caller to an existing call in the system by it's ID. Whatever application was currently using the call would get signalled that the user has hung up.

This is just brainstorming but I'm probably going to subscribe to the dev list to get some feedback.

Anyway thanks Phoneboy and Muppetmaster for the inspiration, I hope I can get this to work as soon as I have a bit of time to do some coding

Cheers,

Descore

[muppetmaster]

If you do end up doing some hacking, would be more than happy to be an alpha tester.

[descore]

Sure thing muppetmaster, I've already posted on the dev list and I'll let you know about any progress.