similar to: chan_oss.c: Unable to register channel type 'OSS'

Zitat von Tzafrir Cohen <tzafrir.cohen at xorcom.com>: Hi, > Off-topic: any reason you don't use chan_alsa? This was the "Armbian installation", I didn't configured it extra... > Are you sure you quote the error message right? Copy+Paste... ;) But I searched a little bit and I really don't think, I need this module... As I undestand, I just need it, if I

So I have done through chan_console.c and searched for console_pct_lock() - every one - has a matching console_pvt_unlock() How is the deadlock occurring ? jerry -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20230908/dee530c8/attachment.html>

I am using usb-audio for Console/Dsp with asterisk. it is crashing 1.4.21 and also svn. During the brief times its working the audio is choppy but understandable. I have used aplay and arecord at the same time on the same wave file and they work fine every time and I have done it MANY times. Asterisk failes after 1 or 2 times. Any ideas on something I can try? Jerry

Hi again! I tried to attach two VoIP-phones to my new Asterisk 13.14.1 on a Banana PI with Armbian/Debian 9. First test was to call a test service that say the time. Works! Second test was to record my voice and play it again. Works! Third test was to call the other VoIP-phone. It does NOT work... :( Then I noticed that, by starting, Asterisk says the following messages: [Feb 15 18:42:54]

On 16-08-16 04:38, George Joseph wrote: > > > On Mon, Aug 15, 2016 at 1:24 PM, Jonas Kellens > <jonas.kellens at telenet.be <mailto:jonas.kellens at telenet.be>> wrote: > > Hello > > using pjproject 2.5.5 > using asterisk-certified-13.8-cert1 > > > IIRC there were API changes in pjproject 2.5 that aren't accounted for > in

I got this compiling the new cvs code ... any idea ? Tnx ! gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\"CVS-04/07/04-11:28:50\" -DINSTALL_PREFIX=\"\" -DASTETCDIR=\"/etc/asterisk\"

Hi list! My problem: I have three extensions in my Asterisk 1.8.30.0 and they have a voicemail. On two of these numbers the voicemail works without any problem, on the other it doesn't... I get this error: [Oct 17 17:01:29] WARNING[14700]: channel.c:5254 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x2 (gsm) [Oct 17 17:01:29] WARNING[14700]: file.c:957

Hi all, on my debian box i configured chan_oss to work with /dev/audio device. CLI console command and Dial(CONSOLE/dsp) work perfectly but i notice 2 problems: 1. audio is very low in volume, even if i set 100 the mixer volume (via cmd line setmixer utility) 2. the sound is very crappy: the voice is "vibrant", words sounds like 'ttthhhiiisss iiisss aaa ttteeessstt". Seems

Hi list! I'm using Asterisk 1.8.30.0 on a OpenWRT device and it works perfectly. I configured a voicemail and I receive an E-Mail with some information about the call. Again, wonderful! Now my wish: I'd like to have Asterisk to search the caller in a list file and send me the name corresponding to the number in the E-Mail of voicemail. Is it possible? I currently use ${VM_CALLERID} in

Daniel Heckl <daniel.heckl at gmail.com> schrieb: > You are searching for ?Call Pickup?. It is implemented in Asterisk by > default. > > https://wiki.asterisk.org/wiki/display/AST/Call+Pickup > <https://wiki.asterisk.org/wiki/display/AST/Call+Pickup> Take a look under > section ?Configuration Options?. Hi, Daniel! Thanks for your answer... I'm using Asterisk

Max Grobecker <max.grobecker at ml.grobecker.info> schrieb: Hello Max, > I'm also a customer of the DTAG. > Yesterday, the messed a bit with their DNS entries... Maybe they tried again to repair a working system... :) > If you are NOT using their DNS resolvers you got a "wrong" IP address back > that was not working. Besides that, you should disable SRV lookups

On 16-08-16 17:45, George Joseph wrote: > > > On Tue, Aug 16, 2016 at 3:21 AM, Jonas Kellens > <jonas.kellens at telenet.be <mailto:jonas.kellens at telenet.be>> wrote: > > On 16-08-16 04:38, George Joseph wrote: >> >> >> On Mon, Aug 15, 2016 at 1:24 PM, Jonas Kellens >> <jonas.kellens at telenet.be <mailto:jonas.kellens at

Can I dial directly from the asterisk console with the Dial() application? or, is channel originate preferred: channel originate SIP/thufir extension 18003569377 at outbound thanks, Thufir

Asterisk doesn't work with the latest CVS of ALSA (2003-06-24) . The module chan_alsa.so won't load even if the oss module, chan_oss.so, isn't loaded. There are no error messages. I've been chasing ALSA/Asterisk/client problems in one form or another for some time now. In previous versions of Asterisk and ALSA -- i.e., last week -- I could load either chan_oss.so or

Hi list! I'm using Asterisk 13.14.1 from Debian 9 repositories. The provider is Deutsche Telekom und Messagenet (just for receive). I can call and receive calls, but I have a little problem: there is a "delay" of about 1-1,5 seconds between the time the voice is sent and the time when the voice is received, so that it happens very often that the peer does not get my voice and try

Hi list! Right now I configured my Asterisk to forward the calls for the number X to both phones (mine and the phone of my wife). It works, of course, but I'm not enthusiast... I see what we have at office: if one phone rings, other phones in the same group can "catch the call", so that if a colleague is not present, another colleague can catch the call. I'd like to have the

Hi list! Yesterday Deutsche Telekom had a really big problem and Asterisk couldn't connect to the remote Server (by Telekom) until today about 7:30. Well, it could happen... What I find really annoying was that I needed to restart Asterisk as I checked with sipsak that the Telekom-Server works... I think, this should not be normal... Can someone explain me why it happens and what I have to

Joshua Asterisk 18.14.0 with chan_alsa and Console/dsp works. This does not work in 18.18.0 with chan_console enabled. I am on Ubuntu 20.04 LTS. Is there a howto for the new chan_console ? how can I get this working again ? I am trying to just play audio on pulse audio. Thanks, Jerry -------------- next part -------------- An HTML attachment was scrubbed... URL:

On Thu, Sep 7, 2023 at 3:15 PM Jerry Geis <jerry.geis at gmail.com> wrote: > Joshua > > Asterisk 18.14.0 with chan_alsa and Console/dsp works. > This does not work in 18.18.0 with chan_console enabled. > I am on Ubuntu 20.04 LTS. > > Is there a howto for the new chan_console ? > I'm not aware of one. The module itself has existed since at least Asterisk 1.8

> > > Just to verify that you did rerun configure after installing the libraries? > > Doug > Oh that is a good one - I thought I did - but apparently not. menuconfig now shows "*" So is chan_alsa going away ? What is it being replaced with? thank you! Jerry -------------- next part -------------- An HTML attachment was scrubbed... URL: