search for: app_jack

...w to Asterisk, so please bear with me. >> I have made a success installation from source of Asterisk 14.4.0 on >> Debian Jessie (8.7). And I am running the Asterisk server, with several >> extensions and dialplans, all working well. >> >> However I am struggling to get app_jack to run. >> >> In menuselect I can see that it is XXX due to dependencies on jack and >> resample, however both Debian packages: resample and jackd are installed >> in this machine. >> >> Is there something that I am missing? > > Most probably some -dev fil...

Hello, I am new to Asterisk, so please bear with me. I have made a success installation from source of Asterisk 14.4.0 on Debian Jessie (8.7). And I am running the Asterisk server, with several extensions and dialplans, all working well. However I am struggling to get app_jack to run. In menuselect I can see that it is XXX due to dependencies on jack and resample, however both Debian packages: resample and jackd are installed in this machine. Is there something that I am missing? What do I need to do in order to have the jack module to run? Cheers -- oooooooooo.io bi...

Hello everyone! Sorry, if the whole task is silly, I'm new to this. I have my newly installed asterisk (1.6.0-beta9) and my AVM Fritz a1 card. I have a simple German isdn line and I have a microphone, headphones and a running JACKd (JACK Aduio Connection Kit). The question: Can I (mis)use my asterisk CLI interface to make and recieve calls coming in/going out via the ISDN-card,

...even if they're only listed as active channels. But JACK just emmits white noise, with a highger frequency than 8kHz, in my believe. A call with app_record shows, that the signal is clear and very ear friendly though. So how to proceed from here? What can I do to help someone look into app_jack, as I can't beyond the simplest of levels. Kindly yours Julien -------- Music was my first love and it will be my last (John Miles) ======== FIND MY WEB-PROJECT AT: ======== http://ltsb.sourceforge.net the Linux TextBased Studio guide ======= AND MY PERSONAL PAGES AT: ======= htt...

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:...

...-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 2) Jackd work...

Hi, I want to use asterisk with jack audio. I tried the next configurations : 1 ) app_jack.so -> does not work. 2 ) chan_alsa.so without jackd started my .asoundrc : pcm.!default { type hw card 1 } pcm.jack1 { type jack playback_ports { 0 alsa_pcm:playback_1 1 alsa_pcm:playback_2 } capture_ports { 0 alsa_pcm:capture_1 1 alsa_pc...

...ve just scanned through the debug log, defined like this in logger.conf: full => notice,warning,error,debug,verbose I couldn't see any reason for the connection not working. I called my friend, he heard ringing, accepted the call and then it got hungup. I didn't see any output from app_jack though. Any idea, how I can get more output from app_jack? Kindest regards and thanks JUlien -------- Music was my first love and it will be my last (John Miles) ======== FIND MY WEB-PROJECT AT: ======== http://ltsb.sourceforge.net the Linux TextBased Studio guide ======= AND...

Hello everyone! So I hacked app_jack.c today, as best I could. Whic came mostly down to inserting ast_log() messages. I discovered the following with JACK: When it starts, it tries to read 512 bytes and only gets 0. That clears up after a while. Sometimes a good time later than the reading comes the writing. And there the...

....so.0 libjackserver.so /usr/local/lib64/libjackserver.so libjack.so.0 /usr/local/lib64/libjack.so.0 libjack.so /usr/local/lib64/libjack.so So... When I run in asterisk source dir: ./configure --disable-xmldoc, output has this line: checking for jack/jack.h... yes however, make menuselect shows XXX app_jack System information: motiejus at pbx3:/usr/src/asterisk-1.6.2.7$ uname -a Linux pbx3 2.6.18.8-xenU #2 SMP Thu Apr 29 15:55:34 EEST 2010 x86_64 GNU/Linux motiejus at pbx3:/usr/src/asterisk-1.6.2.7$ cat /etc/debian_version 5.0.4 Any help/suggestions how should I report Jack to asterisk really apprec...

Hi All, Is there an option in asterisk to manipulate the audio in a call? I would like to, for example change the voice of one caller but without manipulating the audio that comes from another caller. I have read about something called JACK but i don't know if i can use it for this (or how to use it). I am playing with audiohooks now, but I don't think I can change only my voice - whole

Hi, I'm running Asterisk 11.7.0 with app_jack and I'm finding a software echo. When I answer a call, I get two dahdi jack ports to connect the call. One for the phone mic and the other one for the phone headset. The audio that I send to the phone, comes back in the other jack port. For example, If I create a jack metro that beeps every sec...

> All patches need to go into JIRA with a license agreement to be > accepted. Understood, but I was using it as an illustration. Note, however, that, from a legal perspective, a patch such as this has no protectable IP (you can't copyright the only way of doing something) and the GNU projects have a formal rule that sufficiently-small patches need no assignments for that reason, which