similar to: codec_dahdi find_transcoders: Failed to open /dev/dahdi/transcode

I've got a question about codec_dahdi witrh a system running Asterisk 1.6.0.6 and DAHDI 2.1.0.4 with a TE410P card. The system is used primary to route calls between different PRI connections, so no transcoding between codecs is happening as far as I am aware. 1) How can I use codec_dahdi? Would it be useful when passing a call from one dahdi channel to another dahdi channel? 2) I'm

I have a CentOS release 4.7 box running asterisk-1.4.26.1 with dahdi-linux-2.2.0.2 and dahdi-tools-2.2.0 I regularly get these messages, is this something i should be worried about? [Aug 21 01:05:07] VERBOSE[4343] logger.c: codec_g726.so => (ITU G.726-32kbps G726 Transcoder) [Aug 21 01:05:07] ERROR[4343] codec_dahdi.c: Failed to open /dev/dahdi/transcode: No such file or directory [Aug

I am getting the following warnings on the CLI when loading Asterisk 1.6.1-rc1: [Feb 12 12:32:34] NOTICE[22261]: timing.c:59 ast_install_timing_functions: Multiple timing modules are loaded. You should only load one. [Feb 12 12:32:34] ERROR[22261]: codec_dahdi.c:398 find_transcoders: Failed to open /dev/dahdi/transcode: No such file or directory [Feb 12 12:32:33] WARNING[22261]:

I have a Linux-Vserver guest running CentOS 5.5 with Asterisk 1.6 installed from the asterisk.org and digium.com repositories. I have Asterisk starting (service asterisk start) but see errors about dahdi in /var/log/asterisk/messages. ... ERROR[25658] codec_dahdi.c: Failed to open /dev/dahdi/transcode: No such file or directory Linux-Vservers don't allow, under normal circumstances,

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 Thu, Jun 6, 2019 at 12:17 PM Malcolm Davenport <malcolmd at sangoma.com> wrote: > Howdy, > > There's a dahdi-linux-complete-3.1.0-rc1+3.1.0-rc1.tar.gz. > > Try that. > I noticed that was there, but I didn't try it originally because it's obviously a beta version. However, I did download it and try it. It does compile, but doesn't work correctly. For one

Hi when i use MeetMe, i have this errors: app_meetme.c: Unable to open pseudo device Where is the problems ? i have too warning and error into my logs: [Nov 1 07:26:17] WARNING[18544] res_musiconhold.c: Unable to open pseudo channel for timing... Sound may be choppy. [Nov 1 07:26:17] WARNING[18544] config.c: Realtime mapping for 'iaxpeers' found to engine 'mysql', but

Hi, I've got a brand new Asterisk 11 setup for which I would like to keep the number of loaded modules to a minimum. My goal is to this setup in a pure SIP environment, for switching incoming calls to outgoing tSIP trunks. When I leave autoload=yes in /etc/asterisk/modules.conf, I can handle an incoming SIP call with a Playback app. When I leave autoload=no in /etc/asterisk/modules.conf, it

I just upgraded to 1.6.2.rc-1 after running betas 2 and 3 with no problems and while everything seems fine i get these message at startup and than all is well. Should I be worried or do i need to let the team know about this? Also, is not finding "/dev/dahdi/transcode" a problem I should be worried about? And lastly conf2ael always segfaults when I try to run it. it did run once

I have built Asterisk 1.4.4 on my Solaris 10 x86 box: LDFLAGS='-R/usr/sfw/lib -R/opt/csw/lib -L/opt/csw/lib -L/usr/sfw/lib' CPPFLAGS=-I/opt/csw/include ./configure -with-curl=/opt/csw --without-oss --without-vpb --prefix=/opt/asterisk-1.4 The build and install go fine but the asterisk executable reproducibly dumps core with a segmentation violation. If I start it as: asterisk -gc and

> Message: 1 > Date: Tue, 15 May 2007 23:01:24 -0400 > From: Frank Tarczynski <ftarz at mindspring.com> > Subject: [asterisk-users] Asterisk 1.4.4 reproducibly dumps core on > Solaris 10 > To: asterisk-users at lists.digium.com > Message-ID: <464A7404.5000706 at mindspring.com> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > I have

Hello, I have transcoding card TC400P installed in server running Debian with Asterisk 1.4.23. Everything seams to be fine and after I boot up server I see in dmesg: 7.590966] Zapata Telephony Interface Registered on major 196 [ 7.590966] Zaptel Version: 1.4.12.1 [ 7.590966] Zaptel Echo Canceller: MG2 [ 7.610963] zttranscode: Loaded. [ 7.618969] wctc4xxp: tc400b0: Attached to

As best i could figure it out, I've installed dahdi and rc4. My TDM400P doesn't answer fxo or fxs. /etc/dahdi/system.conf: loadzone = us defaultzone=us fxoks=1,2 fxsks=4 /etc/asterisk/chan_dahdi.conf: [house-phones] context=internal ; Uses the [internal] context in extensions.conf signalling=fxo_ks ; fxo_ks Use FXO signalling for an FXS chanel dahdichan => 1 ;

Seems like I post about this about once a year, when it's time to upgrade Fedora. I first got this error trying to compile a patched version of dahdi-linux-2.11.1; I noticed that there is now a dahdi-linux-complete-3.0.0+3.0.0, so I tried that one with the same result. If I compile it while running kernel-4.16.8-300.fc28.x86_64, it compiles fine, but when I try to compile it while running

Hello list! I?m having an issue when reloading Asterisk, I?ve had this problem in Asterisk 1.6.1.6 so I upgrade to 1.6.2.1 version, but I still have the same error. For example, I send a "reload" in Asterisk CLI and this is the output: isb152*CLI> reload == Parsing '/etc/asterisk/extconfig.conf': == Found == Parsing '/etc/asterisk/manager.conf': == Found

Hi All, My research indicates ANI is not really supported with SIP Channels or passed between SIP servers, even with setting function CALLERID(ANI). So the only place this applies is on PRI interfaces, when sending calls out a ZAP PRI you can set the ANI to whatever and CID Number to a different whatever so on the other end of the PRI you will receive the two different values? Is this correct or

Using asterisk 1.6.2.1 and dahdi 2.2.0.2. dadhi-linux installed just fine. Using dahdi_dummy as there is no card in system. Did not install libpri, again, no card. When compiling asterisk, I include -with-dahdi and everything ./configure's fine but when I do 'make', everything goes fine but chan_dahdi doesn't get compiled. No errors. No warnings. It just skips it for some reason.

Hi All, I need to interface a codec-type device to Asterisk. The device uses a TI TLV320AIC1110 codec in 15 bit linear data mode with a 2.048 MHz clock supplied by the device. I am about to start on a custom hardware design to interface this device to the computer, but thought I'd ask here before I get started on it. Does anyone know of a hardware interface that is already being

Good afternoon, I'm trying to use the CLI command file convert on an Asterisk 1.4.26 server with a TC400B transcoding card. The transcoding card is working well for calls but I have some trouble converting sound files from alaw to g729. The command creates empty file as you can see below... CLI> file convert /var/lib/asterisk/sounds/fr/service_notactivated.alaw

Hello, SIP invites are accepted from imitator , but 'SIP 180' is not responded back to imitator. By inspecting the issue , we can *see* the response is generated and sent from asterisk (via asterisk logger ("sip debug" )) , but while sniffing the interface with tcpdump, we can't see 180 response on the interface. We don't have errors on the interface, firewall