similar to: Persuing the gtalk issue - not only jack-related

Hello everyone! So I'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?

Thanks J. It didn't work. On 05/10/2017 01:57 PM, J Montoya or A J Stiles wrote: > On Wednesday 10 May 2017, andre castro wrote: >> 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

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:

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

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, I have been headbanging with asterisk and Jack for a while, decited to ask other linuxists for an advice. The problem is that Jack is compiled from source (0.118) in /usr/local/, but menuselect says "XXX" for it (cannot enable it). I need jack... Otherwise I will inotify Monitor WAVs, what is bad :`( After installing jack from sources: Add system-wide

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 second and I connect it to the jack port that goes to

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

Karl Heyes <karl@xiph.org> writes: > On Mon, 2004-10-18 at 18:27, Jack O'Quin wrote: > > Karl Heyes <karl@xiph.org> writes: > > > > > Are you running with realtime privileges, for this you need to start as > > > root if you want that. Even with realtime privileges there may be odd > > > cases where the scheduling latency is a bit too

Hello everyone! I'm just trying to set up my new asterisk (version 1.6.2.8-rc1). I'd be very grateful, if someone could help me here. I'd be very glad, if one of you could test googletalk with me. Last time I tried (in 1.6.0.x times) it wouldn't work in the end. But here are my gtalk and jabber.conf files. Could you please take a look and tell me, if the settings same sane?

Hello everyone! So now I'm testing with chan_sip and I discovered, that I can make calls, 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

Karl Heyes <karl@xiph.org> writes: > Are you running with realtime privileges, for this you need to start as > root if you want that. Even with realtime privileges there may be odd > cases where the scheduling latency is a bit too high, it all depends on > the drivers and kernel version but the current state is not that bad and > getting better. There are several ways to

>I can't really comment on jackd itself, as I haven't looked that much >into it. If running jackd as root requires client apps to also run as >root then that is a pretty bad issue. I would of thought that running >jackd as a realtime process (whether root or not) would be an important >consideration. jackd (realtime) requires various resources which it shares with its

Hello, Disclaimer: I have no plan to break 3.x bitcode compatibility in any way. I’m experimenting with ideas to improve the current bitcode serialization and deserialization in general. I’m interested in improving the speed and the flexibility of the reader/writer, as well as making the code easier to deal with. My current griefs against our bitcode format are mainly the lack of built-in

On Mon, 18 Oct 2004 16:15:44 +0100, EvilOverlord wrote: >> The setup I'm working with is an ecasound session relaying audio between >> its input and output ports, with its output connected to all the ices >> clients for the different streams we're running. Then I'm >> switching the ecasound input between different stream sources (line in, >> recorded

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I am new to this list and just wanted to ask a question regarding Wine and JACK: I'm running JACK as my sound daemon and today I installed Wine (from the APT-Repository). I setup my .wine directory with the help of winetools. So far everything works fine, except sound output (tested with wmplayer and a simple wav file). So I selected

> On Sep 28, 2016, at 2:03 PM, Xinliang David Li <xinliangli at gmail.com> wrote: > > > > On Tue, Sep 27, 2016 at 12:56 PM, Mehdi Amini via llvm-dev <llvm-dev at lists.llvm.org <mailto:llvm-dev at lists.llvm.org>> wrote: > Hello, > > Disclaimer: I have no plan to break 3.x bitcode compatibility in any way. > > I’m experimenting with ideas to

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,

hello, I've just got ices-kh60 running with jack. It's working OK apart from a problem where the latency on the stream creeps up from about 15-20 secs to around 7 minutes. The stream plays OK, but the audio coming out is 7 minutes behind what goes in. I'm not sure how this can be happening, seeing as I don't think there are any buffers that long in the system, but it is.

I've been using JackWASIO and 10.5 Leopard fine, but since I switched to Snow Leo 10.6 I can't find a way to use wineasio low latency driver at all. Have tried to install some version of Wine (that comes with WineBottler) and a new CrossOver Games 8.0 but no luck: wineasio doesn't appear in the Audio preferences of winecfg. Is there a clear way (for someone who aren't in developing