similar to: Bad recorded audio quality (upgrade).

Hi, I'm trying to compile asterisk 1.2.14 on a Debian Sarge amd64 with kernel 2.6.8-12-amd64-k8 make[2]: Entering directory `/usr/src/asterisk-1.2.14/codecs/gsm' gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -fomit-frame-pointer -fPIC -c -DNeedFunctionPrototypes=1 -funroll-loops

Hi All, I'm stumped on this and I looking for some clues to fix this. This is a new install of Slackware 12.1 onto an IBM x330 Server. Asterisk 1.4.20.1 plays the wav files and the Cepstral_Allison Swift just fine, but when I play the gsm files the audio quite choppy. And, the files produced from the MixMonitor don't even record any audio other than noise. I have a hard drive from

Hello Out of curiosity, has someone managed to run Asterisk on a Beagleboard for home-use? www.beagleboard.org As an alternative to a PC, it can be powered from a USB hub, so that would make for a compact, fanless Asterisk server. Thank you.

I was successful in compiling asterisk in raspbien except for the following error If I enable the gsm codec. It appears there is something in the Makefile n this directory that needs to be changed. Probably involving optimization. Not sure why it does not recognize the processor since it is one that is mentioned in the Makefile.? Any help would be appreciated. make[2]: Entering directory

Hello, I am trying to install a Billion ISDN on Asterisk I have Debian Etch and I installed theese packages: apt-get install linux-headers-`uname -r` apt-get install make apt-get install ncurses-base ncurses-bin ncurses-term apt-get install libncurses5 libncurses5-dev apt-get install bison apt-get install openssl apt-get install libssl0.9.8 apt-get install libssl-dev apt-get install

Hello Im trying to install an old version of Asterisk. But it isnt working: when I run "make install": gcc -o gentone gentone.c -lm ./gentone busy 480 620 Wavelength 1 (in samples): 16.66667 Minimum samples (1): 50 (3.000000.3 wavelengths) Wavelength 1 (in samples): 12.90323 Minimum samples (1): 400 (31.000000.3 wavelengths) Need 400 samples Wrote busy.h ./gentone ringtone 440 480

Trying to build asterisk 1.6.0-rc3, it fails with this message: make[2]: Entering directory `/home/fhimpe/rpm/BUILD/asterisk-1.6.1.0-rc3/ main/editline' /bin/sh makelist -h common.c > common.h /bin/sh makelist -h emacs.c> emacs.h /bin/sh makelist -h vi.c > vi.h /bin/sh makelist -fh common.h emacs.h vi.h > fcns.h /bin/sh makelist -fc common.h emacs.h vi.h > fcns.c if uname -s |

Hi, Does anyone know who did the prompts for French and Russian for Asterisk? I need some custom prompts. Regards, Dovid -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160511/ae5eea65/attachment.html>

Hi! I have configured the language support in asterisk to reproduce spanish prompts. I have lines for it in sip.conf, iax.conf, zapata.conf and voicemail.conf as shown: [general] ... language=es ... In zaptel.conf loadzone = es defaultzone = es When I check my voicemail I get in the CLI: -- Playing 'digits/4' (language 'en') -- Playing 'vm-Old' (language

I'm running asterisk/zaptel 1.4.5. If I load the ztdummy module, the dialplan hangs when it tries to play audio (i.e. Playback) -- and I just hear static on the line. I'm running this on a debian system. I actually have it working on a different debian system but have yet to discover the important difference between the two installations. Any advice on where to look? Thanks John

Hi all, I am trying to implement native format (ulaw) voice prompts and music on hold. Different documentation has different file extensions. Does Asterisk recognise them all? So far I have .ulaw .ul .pcm . Which should I use so Asterisk recognises them as native uLaw files.... -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi List, I was thinking the other day that even with T.38 there are still some issues with faxing. I was thinking of a protocol that instead of just sending down the fax tones an ATA or "VOIP fax machine" would get the entire fax convert it into some sort of image and pass it down the line to the receiving end. I got the idea from RFC2833. Yes I know that fax machines send bit by bit and

Hi, everyone I'm investigating if I could give asterisk CLI access to one of our clients. If I add that user to asterisk group and set his shell to /usr/sbin/rasterisk, is there a possibility for a user to brake our of asterisk CLI to normal shell? Thanks in advance

All too common and largely undocumented. I had this same problem. Installing ztdummy changes Asterisk to use it for timing of playback, apparently. Removing ztdummy "fixed" the problem. To get it all to work, I had to upgrade to to at least kernel 2.6.23.11 (previous versions are either missing options are just broken.) After doing this, I recompiled ztdummy and it worked. Note

Hi users, I have Asterisk 1.6.2.20 in Ubuntu 10.04. I am trying to convert a gsm file to G729 using file convert, but I am facing error as follows, file convert /tmp/welcome.gsm /tmp/welcome.g729 Failed to convert /tmp/welcome.gsm to /tmp/welcome.g729! Command 'file convert /tmp/welcome.gsm /tmp/welcome.g729' failed. [Dec 20 17:24:18] WARNING[2221]: translate.c:256

> Hi Daniel, > > I'm intrigued by this and wanted to try it out - but I'm wondering how > you get Asterisk to call sox at all during Voicemail()? Our server > doesn't even have sox installed, so I'm not sure how to go about > tricking Asterisk into running a different one. To do anything useful you would have to get sox installed on your server. But to get

Does anybody have (or know of) a command line application that would: ) Eliminate pops and other random loud noises. ) Trim leading and trailing silence. ) Trim pauses exceeding x milliseconds to y milliseconds. ) Normalize what's left. I know about normalize and have figured out how to trim leading and trailing silence in sox, but I'm looking for more :) Thanks in advance,

We just completed a new implementation of voicemail for Asterisk. It's much cleaner than Comedian mail and can emulate several voicemail user interfaces, including Audix. It's a great replacement for Audix. All of the sounds/prompts are presently being re-recorded by a professional female voice. If you are interest in the app, let us know at nt_jnewman at yahoo.com. Justin

Hi, all. I've finally made the jump from 1.4 to 1.8. I've installed everything (I think), my Sangoma card initializes right... but there's no "dahdi" command -- not from the base, nor as a subset of the "core" commands. I've got my channels configured in my chan_dahdi.conf file. What am I missing, here? Thanks... -Ken