Displaying 20 results from an estimated 200 matches similar to: "codec_dahdi find_transcoders: Failed to open /dev/dahdi/transcode"
2009 Feb 26
1
codec_dahdi and Asterisk 1.6.0.6
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
2009 Aug 21
2
codec_dahdi.c: Failed to open /dev/dahdi/transcode: No such file or directory
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
2009 Feb 12
1
1.6.1-rc1 errors
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]:
2010 Aug 05
1
Asterisk 1.6 without DAHDI
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,
2018 Feb 15
2
Problem with DAHDI
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]
2019 Jun 06
3
error compiling dahdi for recent kernels
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
2009 Nov 01
1
Error in MeetMe modules ?
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
2013 Mar 04
2
Asterisk 11 - How to trim the number of modules to minimum ?
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
2009 Sep 06
1
1.6.2-RC1 question
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
2007 May 15
1
Asterisk 1.4.4 reproducibly dumps core on Solaris 10
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
2007 Jul 26
0
Asterisk 1.4.9 reproducibly dumps core on Solaris 10
> 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
2010 Feb 19
1
transcoding with TC400P
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
2008 Sep 05
1
dahdi & tdm400p: no luck
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 ;
2019 Jun 06
2
error compiling dahdi for recent kernels
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
2010 Feb 02
0
Issue when reloading
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
2010 Oct 11
4
SIP and ANI
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
2009 Aug 14
1
chan_dahdi refuses to build
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.
2009 Nov 12
1
Codec interface
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
2009 Sep 09
1
CLI file convert from alaw to g729 with TC400B transcoding card results to an empty file
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
2009 Apr 19
1
issue with sip 180 responses
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