similar to: HPEC audio clipping

I have recently moved an asterisk system to a new location. This location is experiencing terrible echo. I installed the HPEC from Digium but that has caused a new problem. When HPEC is enabled and more that 16 taps are used, the audio from the outside caller gets clipped. Instead of hearing: Hello, my name is Mike one hears He o, m ame ike If the taps are set to less than 16, there

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

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

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 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.

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

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

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

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

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

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

I have just placed HPEC version 9.00.003 onto the Digium FTP site; this release cures the recently discovered bug where incoming audio was choppy or lost completely under specific circumstances (did not affect all users, but if a user's system had the problem it would have it on all calls). -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - "The Genuine Asterisk

Hi I've bought the Digium g729 codec and have installed it correctly (I think) >> voip*CLI> show g729 >> 0/0 encoders/decoders of 3 licensed channels are currently in use if I do an echo test either from a sip Cisco 7960 or another hard phone (unbranded) using g729 it sometimes works and sometimes the announcement about the test (echo-test.gsm) fails part way though but the

Hello all, I've been lobbying for some time at the #asterisk IRC channel. Until now, I still can't find a solution to my one way audio problem. I rebuilt the Asterisk-1.4.21.2 from the Debian Testing repository on my Debian Etch. I got a Digium TDM400P with 1 FXO (channel 4) and 1 FXS (channel 1). My SIP extension phone located inside the LAN is a SNOM 300 IP phone. This one way audio

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

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 TDM400 and 4 channels of HPEC. I don't use the POTs lines much so I didn't realize it wasn't working. This morning I was watching the console and noticed that the echo canceller didn't load when a call came in. /etc/dahdi/system.conf showed mg2 for all 4 channels. I changed them all to hpec and then restarted dahdi and Asterisk and suddenly HPEC was working. Is

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

Using a SIP connection with a CLEC, the downstream (received) audio is perfect when the mute button is activated on the phone. However, when there is upstream audio (i.e., talking or even breathing into the microphone), the downstream audio is cut off. It's kinda like having a half-duplex audio connection. When I divert outgoing calls to another provider, these calls are fine.