similar to: Early injecting Jack between call parties

Hello, My configuration is : Card 0 - kernel dummy sound card Card 1 - my soundcard I have a jackd running in background. My jackd launch command is : jackd --port-max 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

Hi, I am trying to write a dialplan that will use Dial() to call two local extensions. One extension will run an AGI script (a continuous background process, running until hangup), the other will connect the active channel to Jack() (also running as continuous process, until hangup). This is my current dialplan attempt: --------------------- [from-internal] exten = 514316XXXX,1,Answer() same

Asterisk will try calling both at once. As soon as one is answered it cancels the call to the other. What you can do is for extension 101 to put it in it's own context and then call the agi from the h extension. So something like this: [from-internal] exten = 514316XXXX,1,Answer() same => n,Playback(hello) same => n,Dial(LOCAL/100 at extensions&LOCAL/101 at extensions) [extensions]

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

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

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

I have been trying to use Illustrator 8 with wine, and it works fine except often it doesn't recognize the release of a mouse click, so tools/selections continue even after you've released the mouse button. I have no idea as to how to solve this, but anyone with any info it would be much appreciated. Thanks On Tue, 18 Dec 2001 14:39:10 -0500, Joel8bit wrote: > Hello, > Does anyone

Hi! well the subject is telling everything - I''m just curious to know how other railers handle this (for me the answer is yes, clearly!) cheers Thibaut -------------- next part -------------- An HTML attachment was scrubbed... URL: http://wrath.rubyonrails.org/pipermail/rails/attachments/20060512/0b59b57b/attachment-0001.html

Hi, I am in a predicament and any help/pointers would be appreciated. We are using chanspy to listen in on conversations. We are doing this via a web interface. The web interface lists all the ongoing calls. We click on a call and then my local phone rings allowing me to spy on the session I clicked on. But "most" of the time, when I start listening in, the two parties that are in

Hello, is it possible to disable the CDR record creation for all dialed parties? From my limited testing it looks like CDR_PROP(disable) is effective only for the first party (the one specified before the first ampersand in the Dial application argument) and I can't find any way to disable it for the other ones (I think the CDR in question is written after the Dial completes). Is it by design?

Hello, list. My name is Max, I'm a web developer. I found Markdown convention very useful, I use it everywhere nowadays. Michel Fortin's markdown-php seems to be the only available PHP implementation. And, judging by number of people using it, it works well :) But there is many thing that's keep it from being perfect: * One large file with functions not being sorted in a logical