search for: compandent

Hi: I?Bought TDM2400P ,with 24 FXO ports , I installed? the asterisk 1.4.28 and?dahdi 2.2.0?and then i compiled them and configured all, i pluged into 8 pstn line into the Rj connector and then i got messages on asterisk console that alarm cleared on channels 1->8??, at this step everything should go fine ,but when i try to make a call?through the TDM?there is silence and no audio,and then i

Hello folks, Although Jack graciously permitted me to keep tinkering on the Postfish alongside OggFile work ;-) allowing it to sit there in a partially tinkered state continued to be a distraction. So yesterday and today I cleaned up the source repository and put together a prerelease. With this prerelease, I'm putting the project out of mind until I put out a beta of OggFile. Actually,

Hello folks, Although Jack graciously permitted me to keep tinkering on the Postfish alongside OggFile work ;-) allowing it to sit there in a partially tinkered state continued to be a distraction. So yesterday and today I cleaned up the source repository and put together a prerelease. With this prerelease, I'm putting the project out of mind until I put out a beta of OggFile. Actually,

On Tue, 9 Feb 2016, Larry Turnbull wrote: > One of the streams is an old time radio stream and I use ezstream to run the > prerecorded shows. > > Is there a package or some sort of way I can apply dynamic compression to > the stream? Ezstream really isn't designed for this. It's primarily meant for streaming files as they are, which is a very light-weight operation, with

Hi all: I am managing a radio station that has 7 streams. I am using icecast, ices and ezstream on the streams and overall it is working pretty well. One of the streams is an old time radio stream and I use ezstream to run the prerecorded shows. The streams are 128K mp3 streams. Also this station is running on a VPS using Ubuntu 14.04. My question: Is there a package or some sort of

Has anyone needed a coded that Asterisk does not natively support, such as MELPe or CVSD? If so, did you find a pure software solution and provided that as an addition to Asterisk? Was that solution successful? Has using an I/F card with a DSP proved to be the better solutions? We are beginners with Asterisk so any help/advice on how to best implement non-native Codecs into Asterisk will be

The Asterisk Development Team is pleased to announce the release of DAHDI-Linux and DAHDI-Tools version 2.4.0. DAHDI-Linux 2.4.0, DAHDI-Tools 2.4.0, and DAHDI-Linux-Complete are available for immediate download at: http://downloads.asterisk.org/pub/telephony/dahdi-linux http://downloads.asterisk.org/pub/telephony/dahdi-tools http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete In

The Asterisk Development Team is pleased to announce the release of DAHDI-Linux and DAHDI-Tools version 2.4.0. DAHDI-Linux 2.4.0, DAHDI-Tools 2.4.0, and DAHDI-Linux-Complete are available for immediate download at: http://downloads.asterisk.org/pub/telephony/dahdi-linux http://downloads.asterisk.org/pub/telephony/dahdi-tools http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete In

Thanks Geoff, I will give it a try. I am reencoding all files to a 128K mp3, 44100 sample rate stereo stream for consistency. So yes I have the decode and encode lines in the xml file as you outlined. I will check out sox and see how it does. I am also exploring mp3gain to run across the entire library and see if getting all files to the same level will help. Larry -----Original

Dear Stefan, In the paper "Improved Forward-Adaptive Prediction for MPEG-4 Audio Lossless Coding", a non-linear compander is applied to the parcor coefficients prior to quantization. This compander is designed in order to minimize quantization error, especially for magnitudes close to unity. If you determine the typical distribution of magnitudes of the LPC coefficients, you could

Hi, I had previously posted about connecting an AS5300 to * via SIP/H323. I got it to work via SIP, but only 1 call at a time would work, and if a user from the * side hung up, the cisco would'nt catch the hangup. I an now trying to hook up to the cisco via E1, with a Sangoma A101 card in my * box. I would like it such that I call from * via E1/PRI to the cisco, and call out via R2 to

Hi list! The stability of * with fax (or the lack of it) is causing me headaches. To solve it, I was thinking to put a TE205P card in the * box, connect the E1 pri on one port and a channelbank (I was thinking of the Rhino) on the other port in T1 mode. (Has anyone tried this??) The TE205P supports channel bridging which sounded like the ideal (but not cheap) solution when combined with a

Hi everyone, We have been working with Asterisk for a while now and would really like to expand its capabilities by fully integrating it with our Definity g3si and are wondering about other peoples experiences with similar setups. Thus far we have only been able to achieve a partial 1 way integration, but ultimately would love to route inbound voip calls to the asterisk out of the lucent.

Hi. I'm wondering if anyone out there can give me a little help trying to figure something out: I presently am running CentOS4.1 with GNOME as my desktop (It runs quite well, I must say!). I just have a little curiosity: I have a few audio files (.wav, .mp3 and .ogg) and I like to "preview" them. When run my mouse over the .wav files (and mp3 files) the

Dear all, I have an Asterisk SIP server in a LAN environment and I want your opinion in order to decide the use of an audio codec: What audio codec is better, G.711a or G.711u ??? Which suites to my LAN voip calls ??? Thank you !!! Alejandro -------------- next part -------------- An HTML attachment was scrubbed... URL:

> 1) AGC: This seems to work pretty well in all cases. I had previously > hacked-in the "compander" filter from sox for a similar effect. What > I've noticed is that speex_preprocess's AGC has no "knobs", and it > seems to use an attack/decay that is a lot faster than what I had > chosen from the sox compander, but it works pretty well nonetheless.

Hi, Can someone please comment about the below issue [root at host0040 kaushal]# file obd-demo.mp3 obd-demo.mp3: MPEG ADTS, layer III, v1, 256 kBits, 44.1 kHz, Monaural [root at host0040 kaushal]# sox obd-demo.mp3 -e stat sox: Failed reading obd-demo.mp3: Do not understand format type: mp3 [root at host0040 kaushal]# sox -V obd-demo.mp3 -r 8000 -c 1 -t ul -w vm-intro.ulaw sox: Failed reading

First, I'd just like to thank the Speex community, and Jean-Marc especially, for their great work. I'm developing a VoIP library (which uses IAX, the asterisk protocol) as the network protocol. I've been putting off integrating Speex for a while, as things have been working pretty well so far with GSM. (for those interested, the code is at iaxclient.sourceforge.net). However,

Xiph.Org announces the release of libao-1.0.0, libvorbis-1.3.1 and vorbis-tools-1.4.0. This is a coordinated update of the audio libraries and tools to deploy improved surround-sound support across the libraries and toolchain. libao improvements: - AO returned to active development - Added surround channel mapping API and capability - Updated all drivers on modern installs - New config file

Xiph.Org announces the release of libao-1.0.0, libvorbis-1.3.1 and vorbis-tools-1.4.0. This is a coordinated update of the audio libraries and tools to deploy improved surround-sound support across the libraries and toolchain. libao improvements: - AO returned to active development - Added surround channel mapping API and capability - Updated all drivers on modern installs - New config file