search for: jack_hook

...16 --realtime --no-mlock -d alsa --playback hw:1,0 --capture hw:1,0 --rate 8000 --period 1024 --shorts --inchannels 2 --outchannels 2 --dither triangular & 1 ) I open asterisk with chan_alsa.so connected (with asoundrc) to the kernel dummy sound card (allow me dial command). I do a call with a JACK_HOOK from app_jack.so, sound is sent but no one is received. My extensions.conf : exten => _0.,1,Answer exten => _0.,n,Set(JACK_HOOK(manipulate,c(asterisk))i(from_voip:input)o(to_voip:outpu t)))=on) exten => _0.,n,Dial(SIP/freephonie-out/${EXTEN:1}) Asterisk command : console dial 0xxxxxxxx...

I use Jack for getting callee sound. Dial with option M(): JACK_HOOK(manipulate,i(rec_737219:input),o(rec_737219:output),c(rec_737219))=on This works fine, but I need to connect the sound channel to Jack *before* the actual answer. As you can see in the AMI log, between "Ringing" to JACK_HOOK there is a 6 second break. I don't want that. I need a way...

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

...y recommended, but I have a specific use-case, and my Asterisk server will only be handling a small number of concurrent callers. I can provide more detail on my python script process, my need for Jack, etc. My need for parallel running processes on the dialplan was spurned by issues I’ve had with JACK_HOOK, which are detailed here: https://community.asterisk.org/t/jack-hook-issue-and-finding-working-alternative/86039 Thanks in advance for any help or suggestions! -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attac...

Hello list, can a conversation be recorded without the caller or callee having to press some combination that is defined in features.conf ?? Like in queues.conf you have the ability to record a conversation with MixMonitor when the caller is connected to an agent/member of the queue. Can this auto-recording also be implied on normal Dial(something) ?? So that when the call is picked up (and

...gt; specific use-case, and my Asterisk server will only be handling a small > number of concurrent callers. > > I can provide more detail on my python script process, my need for Jack, > etc. My need for parallel running processes on the dialplan was spurned by > issues I’ve had with JACK_HOOK, which are detailed here: > https://community.asterisk.org/t/jack-hook-issue-and-finding-working-alternative/86039 > > Thanks in advance for any help or suggestions! > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided...

Hello asterisk users! I, like many people, have a cell phone. I also have some SIP phone devices (software and hardware). I'd like to have one number that rings all my phones and routes the call to wherever I pick up. However, my cell phone has its own call forwarding voicemail. I can't just turn that off, because then direct-to-cell calls wouldn't ever get to voicemail - that

Greetings! I now found someone to test gtalk with and found out, that app_jack has a problem here. My voice gets transmitted fine, but I only get white noise from the other party. I tried to set my JACK samplerate to 8000 to make sure it's no libresample problem, the results were the same. My setup is: Linux Debian Lenny Kernel: 2.6.30.4 PREEMPT (self-built) JACKd: jackd version

Hi, Thanks for your info, What is the impact of the following line in dialplan, Dial(SIP/19201/19202,300) On Thu, Sep 3, 2015 at 7:20 PM, Vinicius Fontes <vinicius at aittelecom.com.br> wrote: > You might want to use the Originate() application instead. Check its usage > by issuing the command 'core show application originate' on Asterisk CLI. > > 2015-09-03