similar to: codec_dahdi and Asterisk 1.6.0.6

I have asterisk 1.8.2.3 + A102D Sangoma card 2 port T1. when i am starting asterisk i am getting this error on console. func_callerid.so => (Party ID related dialplan functions (Caller-ID, Connected-line, Redirecting)) == Registered application 'PrivacyManager' app_privacy.so => (Require phone number to be entered, if no CallerID sent) == Registered custom function

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]:

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

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 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

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

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 ;

Hi, experts for example, the below case: You can see a lot of back and forth MTU probe packets been exchanged between tinc nodes, but it’s weird that, from the debug log, one line shows "No response to MTU probes from node1”, but it indeed received a lot of MTU probe response, and finally it get the conclusion of "Packet for node1 (1.1.1.1 port 443) larger than minimum MTU”.

On Sat, 2003-05-31 at 10:29, Stefan Neufeind wrote: > I have mp3s as input and can't convert them all. And since my > listeners will ONLY use mp3-streaming I don't want the overhead and > quality-loss of converting mp3 to ogg and later stream-transcode them > from ogg back to mp3. That's not worth it. > So why can't support for mp3 be added to ices2? From a

Tengo en una maquia ubuntu 8.10 el kernel es Linux 2.6.27-14-generic Quiero configurar una tarjeta x100p i usarla con asterisk, asi que descague compile e instale lo siguiente: asterisk-1.4.24 dahdi-linux-2.1.0.4 dahdi-tools-2.1.0.2 libpri-1.4.9 Sin embargo no logro configurar la tarjeta con exito, ya probe casi? todo. Esto aparece si ejecuto lspci: 04:06.0 Communication controller: Motorola

Yesterday, a low-duty production server that I maintain core-dumped. At the time there were only around 2 calls going through it. The strace on the screen made it look like it was caused by Dahdi. The machine is running asterisk-1.6.0.3 dahdi-linux-2.1.0.3 dahdi-tools-2.1.0.2 asterisk-addons-1.6.0 Kernel version 2.6.28 There is a genuine TDM400P (populated witrh 2xFXO cards and 2xFXS cards.

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi rgl crashes my R session, when resizing the rgl graphic window. I am using Ubuntu Maversick, with dual monitor setup. If I disconnect one monitor, I can resize it "a little bit", but it still craches if I enlarge it to much. I assume that the problem has to do with allocated graphic memory in the kernel, but why is R crashing

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi rgl crashes my R session, when resizing the rgl graphic window. I am using Ubuntu Maversick, with dual monitor setup. If I disconnect one monitor, I can resize it "a little bit", but it still craches if I enlarge it to much. I assume that the problem has to do with allocated graphic memory in the kernel, but why is R crashing

Hi, I would like to improve my understanding of T.38. 1. What T38FAX_VERSION_0 or T38FAX_VERSION_1 in chan_sip.c means ? voip-info.org implies one has to change values in chan_sip.c to make it work. Shall I set T38FAX_VERSION_1 or leave T38FAX_VERSION_0 in global_t38_capability ? Source code says "This is default: NO MMR and JBIG trancoding, NO fill bit removal, transferredTCF TCF, UDP FEC,

Hello! I use asterisk with TE420 as PRI switch for two channels : ;panasonic uplink group=3 context=panasuplink ; relaxdtmf=yes ; immediate=yes rxgain=0.0 txgain=0.0 mohsuggest=default jbenable = no ; jbenable = yes ; jbmaxsize = 200 ; display_send=name_initial display_send=name

On 31 May 2003 at 12:49, Karl Heyes wrote: > On Sat, 2003-05-31 at 10:29, Stefan Neufeind wrote: > > I have mp3s as input and can't convert them all. And since my > > listeners will ONLY use mp3-streaming I don't want the overhead and > > quality-loss of converting mp3 to ogg and later stream-transcode > > them from ogg back to mp3. That's not worth it.

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