similar to: HPEC audio clipping

I am running 64 bit linux on my Asterisk box and would like to get the new HPEC software running on it. However, while there are 32 bit modules available, there are no 64 bit modules on the ftp site: http://ftp.digium.com/pub/telephony/hpec/64-bit/ In some places on the digium website it states 32 bit only and other places including the documentation it states 32 & 64 bit are available.

Hi, everyone: I am developing a system using Asterisk, TDM-400 analog cards, analog lines, and Polycom SIP phones for internal extensions. Initially there was bad echo but after a series of efforts, I've managed to reduce it to a negligible level (it only happens when both parties speak simultaneously, and even there, only for a few hundred milliseconds). From an echo standpoint, things are

I recently read about the following new technologies from Digium. Has anyone tried the new HPEC or knows when it will be available? TDM800P and HPEC The TDM800P is an 8-port analog telephony interface card, so it fills the gap between Digium's 4-port and 24-port cards. Analog phones and POTS lines are going to be with us for some time, and demand for support for them remains high. The

Hello, 1. A patch allowing "capture audio streams in a way that will allow [us] to debug (and presumably fix) the problem" was mentioned by Kevin. Anything new about it ? I couldn't find it in Zaptel 1.2.17.1 nor 1.4.2.1 changelog. 2. Is there any Mantis bug report allowing someone to track this "HPEC audio clipping" feature ? It seems to me this hasn't been reported

Hi: I am working on a new system with a TDM-400P card with three FXO modules and one FXS module. The system has been in place for a week. Users are complaining of echo problems. I have noticed this echo myself. It varies in severity. It is sometimes bad enough to make it difficult to converse, but the users find it generally unacceptable. They miss their old phones, which just worked. As you can

Hi all, Just want to check from the list experienced personal about the Digium HPEC, where I had purchased the HPEC and wish to run with TDM card Sangoma A200. I can't install HPEC to run with Sangama A200 card, even I had changed my hpec file from i686 to i386. The error that I had as bellow; > Found key 'HPEC-XXXXXXXXXX' for 2 channels. > Found valid HPEC licenses for 2

My SM56 (Motorola X100P clone) has echo as hight as 38%, according to fxotune -d. But when trying to take action, it fxotune simply says it can't. ./fxotune -i3 -vvvv Running with parameters: doset=0 docalibrate=1 dodump=0 startdev=1 stopdev=252 calibtype=2 waveformtype=-1 delaytosilence=0 silencegoodfor=18

Just wondering how your experience is with HPEC, Is it just for analog interfaces or we can use it on TE122 as well? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080214/3f0580f1/attachment.htm

Hello, Has anyone tried the combination of asterisk 1.2.24, zaptel 1.4.5 and HPEC 9.00.003? In particular, with a hardware configuration similar to: Module 0: Installed -- AUTO FXO (FCC mode) Module 1: Installed -- AUTO FXO (FCC mode) Module 2: Installed -- AUTO FXO (FCC mode) Module 3: Not installed Found a Wildcard TDM: Wildcard TDM400P REV I (3 modules) I have two fully independent systems

Hi - I had a client recently move their asterisk system (asterisk 1.4.26, dahdi 2.2.0.1, aex800 w/vpm module) to a "new" location, a building that's nearly 150 years old. I was not personally able to go there, but the person who did the move said the building's demarc room was "scary"-- water leaks, jumbled and frayed wiring, and all sorts of other fun. The echo on

Hi All - I've been trying to compile Zaptel w/ HPEC, but I've been unsuccessful. The system is CentOS 4.4, zaptel version 1.2.15. I believe I've got all the requisite files, and they're in the right locations in the zaptel tree. When I compile, I get the following warning from make: Warning: could not find /usr/src/zaptel-1.2.15/hpec/.hpec_x86_32.o.cmd for

I was previously using an old computer running Asterisk 1.2 with zaptel. Once the CPU fried I switch to a new computer and I chose AsteriskNow 1.5 running in 64bits to simplify the installation process. I manage to find my way with configuring dahdi instead of zaptel and to switch all my previous config to the new computer. Now everything is fine except that even if I use the md2 echo

Does this make a significant improvement? The box in question I was going to try this with has a 4 port TDM card w/ plenty of horsepower, but I do intend to later migrate to a Soekris unit running Astlinux and therefore might not have the power to run it after. If the difference is significant, I may move to an ITX board so I could use a bigger CPU, but only if the hassle is worthwhile. Any

Hi I have a server with the last version of asterisk branches, zaptel branches, 2 Digium Card with TDM800P 16 HPEC channels (Echo Cancelation), 40 Grandstream BT200 and 10 Grandstream GXP2000. zapata.conf echocancel=64 rxgain=0 txgain=0 when i place a call o receive a call, I finish a sentence i hear a ssssssss, AND when the both side talks at the same time i have choppy audio. Any

Hi, folks: Yesterday I added a second TDM400P card to a working, echo-free server running HPEC. Today, I'm getting these messages: > Apr 20 09:12:12 WARNING[5679]: chan_zap.c:1551 zt_enable_ec: Unable to enable echo cancellation on channel 3 along with complaints of severe echo. The channels have all been tuned using 'fxotune -s' and the echo numbers are all under 2%. I guess

Hi, Could anyone tell me how to read the values in the "zonedata.c" file? I am looking at the "zt_tone_ringtone" field mainly. Thank you. Jad Wauthier -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070512/4c0387be/attachment.htm

Any idea why I can't build HPEC for zaptel 1.2.18? It builds fine with 1.4.3... PaulH

Hello, Are HPEC (ie zaptel 1.2.X) and bristuff compatible ? And the results ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070413/b5adde2a/attachment.htm

Hi List... I have an asterisk box with one TDM2400, it has 8 FXS's and 12 FXO's, and it also has the echo canceller... I'm running Asterisk 1.2.13 and Zaptel 1.2.12 with the Kernel 2.6.9-34.0.2.EL I'm using Polycom's 501 with the SIP 1.6.2.0041 The problem is when someone dials to or from the PSTN through the TDM2400, the voice quality is crappy...Instead of hearing:

I noticed that clipping occurs in lossy audio compression even if there is no clipping in the original file (though the original file has peaks that are just below the maximum). I know that this happens due to all the filtering involved during compression, but I'm wondering just how audible this clipping is because I don't hear anything wrong. I mean, as things are now, is it safe to say