[asterisk-users] Getting calee audio in Asterisk (real time)

[Motiejus Jakštys]

Just FYI how I solved this:

I figured out that JACK_HOOK`ing for open channel does not connect
input and output ports. So instead of
 *CLI> core set chanvar SIP/poly1-ab23jadf234 JACK_HOOK(manipulate) on

you shoud use:
 *CLI> core set chanvar SIP/poly1-ab23jadf234
JACK_HOOK(manipulate,i(SIP/poly1-ab23jadf234:input),o(SIP/poly1-ab23jadf234:output))
on

Then all works fine and you get leg B's channel.

---------- Forwarded message ----------
From: Motiejus Jakštys <desired.mta at gmail.com>
Date: 2010/5/5
Subject: Re: Getting calee audio in Asterisk (real time)
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>


Update:

I thought this may be the solution:
 *CLI> core set chanvar SIP/poly1-ab23jadf234 JACK_HOOK(manipulate) on
(For 1.6.2 it's dialplan set chanvar SIP/poly1-ab23jadf234
JACK_HOOK(manipulate) on )
Source: voip-info.org

The command opens two jack ports: Channel:input and channel:output. At
once command is executed, sound on the caller is gone.
Question: what should this CLI command do in reality? Is it a bug or
expected behaviour?

Then I connect those two ports hoping it will return the sound to the caller:
jack_connect SIP/PBX2-0000000d:output SIP/PBX2-0000000d:input
Then the calee hears garbled sound. Sample of all process is here. It
is recorded by MixMonitor on the machine where jack takes process.

Asterisk 1.6.2.6 (upgrading/downgrading/patching is not a problem).

Waiting for your suggestions... Maybe I can do this in totally
different approach?

Regards
Motiejus Jakštys
http://m.jakstys.lt/

2010/5/5 Motiejus Jakštys <desired.mta at gmail.com>
>
> Hello,
> I need to capture calee's audio in real-time in order to capture operator messages (I've written sound recognition software that works with Jack: http://github.com/Motiejus/SoundPatty/).
> Jack does the following:
> Incoming call audio -> audio in to jack, audio out from jack -> current Asterisk application
> Outgoing call audio <- current Asterisk application
>
> However, I need vica-versa:
> Incoming call audio -> current Asterisk application
> Outgoing call audio <- Audio from jack, Audio into Jack <- current Asterisk application
> or at least
> Incoming call audio -> current Asterisk application
> Audio to jack       <- current Asterisk application
> Outgoing call audio <- current Asterisk application
>
> Any idea how I could accomplish this?
>