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....@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@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....@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? > -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users