similar to: ALSA channel (console/dsp) problem

Hi folks, Its the forth day I'm sticking to a problem with chan_alsa, The sound played or captured from the device is choppy time to time. I mean when talking on a console/dsp microphone the other side hear my sound choppy and I'm hearing hers the same but not all the time during a call, sound sometimes are clear. Even when I'm putting the sip side on hold i hear the same choppy music

Dear folks, I couldnt receive faxes and get the following debug traces on the console, I appreciate any help or even hints. Using Spandsp-0.2 app_rxfax.c:76 span_message: FLOW Get at 9600bps, modem 1 app_rxfax.c:76 span_message: FLOW Changed from phase 3 to 5 app_rxfax.c:76 span_message: FLOW Non-ECM carrier up app_rxfax.c:76 span_message: FLOW Non-ECM carrier down app_rxfax.c:76 span_message:

Hi folks, I had a survey online but there i couldnt find a clean sample of CAS signalling on E1 interfaces. I defined a span with CAS framing and HDB3 line coding but dont know which signalling to use for channels. I'd use 3 bit CAS signalling and 20 incoming channels and 10 outgoing ones. Anyone could help me define the signalling for these channels. PS. Im using Sangoma cards. Any

Dear folks, I have problem in fax reception. The astrisk detects the fax tone and jusmps to the fax extension and rxfax application starts and the max machine starts the fax but saddenly stops and seems the rxfax have died. It doesnt returns, not files in the output dir and .......... Anyone have any idea or help how could i get whats wrong here. Regards. M. Shokuie Nia

Dear Folks, Anyone knows if Sangoma supports or going to provide support for battery removal detection on FXO lines?? As Tzafrir said earlier DAHDI supports it, which is a very nice feature but what about Sangoma? Regards. -- M. Shokuie Nia. -------------- next part -------------- An HTML attachment was scrubbed... URL:

I found this information elsewhere.... making it /root/.asoundrc helped. # This is .asoundrc # # this makes legacy OSS apps use alsa software mixing dmix pcm.dsp0 { type plug slave.pcm "dmix" } # mixer0 can stay unchanged, because it isn't used anyway, I guess ? ;) ctl.mixer0 { type hw card 0 } # this makes native ALSA apps default to using dmix pcm.!default {

I have setup asterisk alsa.conf to read the null device for ALSA and console/dsp input. asound.conf is pcm.nullpcm { type null } alsa.conf has input_device=plug:nullpcm Then when I call into the Console/dsp I get very choppy audio. I dont need and data from the microphone. I just want data out on console/dsp. Have I not setup the device properly? why would I be getting choppy audio

I have just installed the Alsa drivers for my 2.4.18-14 kernel (RH8). I have configured the sound card ok with alsaconf and tested with the aplay , works fine. But when I run asterisk it says.. ------------------------------- [chan_alsa.so] => (ALSA Console Channel Driver) Apr 20 18:28:34 ERROR[8192]: chan_alsa.c:339 alsa_card_init: snd_pcm_open failed: No such device or address Apr 20

I want to try ALSA driver since the sound quality of OSS driver is not so good (a little choppy but bearable when playing Diablo). After changing my .wine/config to [WinMM] ; Uncomment the "Drivers" line matching your sound setting. ;"Drivers" = "wineoss.drv" ; default for most common configurations ;"Drivers" = "winearts.drv" ; for KDE

Deaf folks, Actually this is my first post here, so sorry for any inconvenience. Im planning for a solution a bit larger in scale than ususal. I'm goin to use * as a PSTN gateway with E1 links and use two other 3rd party Gateways for FXO lines. I should be able to switch from every incoming channel to any outgoing one and also to some SIP softphones. I planned to use SER as a sip server but

Dear folks, I got a problem sending faxes using spandsp. Primerily, when the tiff file made using GIMP 2 with different compresions the fax app break downs whole *. Moreover when i made a tiff file using Microsoft mdi, everything works fine but on the other end of the call, the received fax is shrinked in size. Anyone has any idea whats the right file format and compression type for it? PS.

Dear folks, I would be very thankful if an experienced user can help me out here. I wanna use mfcr2 and unicall library on sangoma boards but so far impossible for me. As Im setting the framing type to CAS (TE_SIGMODE = CAS) on wanpipe I couldnt get the link alarm out (i looped a A102d links) but when setting it to CCS everything works fine and the green lights shine on the back. Can anyone

Dear folks, I'm using * realtime with no problems on most of the systems i've setup but rarely i confront this problem that the asterisk doesn't load from database when the systems rebooted and i have to reload it manually or restart it, but it would work fine afterward, no problem how many times you stop and start the *. It seems, there is a missequence of deamon loading at boot

I am trying to connect audio through HDMI on a config. aplay - l gives: **** List of PLAYBACK Hardware Devices **** card 0: NVidia [HDA NVidia], device 0: VT1708B Analog [VT1708B Analog] Subdevices: 2/2 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 card 0: NVidia [HDA NVidia], device 3: NVIDIA HDMI [NVIDIA HDMI] Subdevices: 1/1 Subdevice #0: subdevice #0 So I change my

I am running asterisk 1.4.26.1 and using ALSA not oss dahdi 2.2.0 and libpri-1.4.10 I am calling into console/dsp I hear the audio just fine then after the hangup I hear ringing on the console/dsp. Why would that be? I found this bug for OSS https://issues.asterisk.org/view.php?id=13686 Does the same thing exist in ALSA??? some traces below Jerry == Parsing

Hello, I just sent it with a wrong title... so once again: I just compiled and started Asterisk 1.0.2 following "Getting Started With Asterisk Version 0.1a" from http://www.automated.it/guidetoasterisk.htm I made only one change to default config files - I changed from using oss to alsa. I don't have any devices so far. I started asterisk from the command line: # asterisk -vc

What do I need to install for OSS or ALSA for console use only Sean -- +----------------------------------------------------+ |VOIP: FreeWorldDial 689482 VOIPBuster thecivvie | |GPG Key: http://thecivvie.fastmail.fm/thecivvie.asc | +----------------------------------------------------+ -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type:

Is there a way to have asterisk setup in ALSA mode (which is working) but also be able to "play file.wav" and hear that also. CONSOLE/dsp is working. If I stop asterisk and "play file.wav" that works. But if asterisk is running I cannot also play a file.wav I am not actually doing them at the same time. Just a couple files here and there to play. But as is I have to stop

Thanks for reading this. I've been pulling hair for days trying to resolve this, and any help someone can give me would be very much appreciated. I have an Asterisk box that is basically a P4-3GHz, a Digium-recommended SuperMicro X5SSE-GM motherboard, 2GB RAM, 250GB IDE hard-drive with UDMA, a SoundBlaster Live! 24 sound card, a Digium Wildcard TE110P, and a Digium Wildcard TDM400P with 4