similar to: sip + zap problem

Hi all, I was excited to see the announcement on the list regarding the fix for the echo problems on Digium FXO cards! I have 2 X101P's, TDM400P with 4 FXS modules and couple of XLite softphones. A few months back,I had gone thru the recommendation on the list to remove echo from the SIP phones(I never did have any echo on the TDM400P FXS phones), and had removed about 90% of the echo.

I have a bad echo problem on my TDM400P with one FXO module installed. I have tried a few things, such as: * setting rxgain and txgain to 0 * setting echocancelwhenbridged to no / yes * settting echocancel to 64 / no / yes * setting echocanceltraining to 800 / no / yes * MG2 echo cancellation * MARK2 echo cancellation * KB1 echo cancellation * AGGRESSIVE_SUPPRESSOR option of MARK2 Each time

Hello, We have an asterisk setup at our client's site using a TE205P. The line to telco is a 23 channels T1 PRI, however the line has random echo problems (about 5-10% of the calls)! Can anybody tell me if echo cancellation is really needed even at a T1 PRI to the telco? Because people keep saying when they deploy voip solution in Hong Kong using T1 PRI, there is no need of echo cancellation.

After speaking with several people, and even participating in a forum of several other people with echo issues, I thought I'd share what we've done (well actually what our chief R&D engineer, Brett Bourn has done...) First let me say that normal cheapy PC hardware couldn't be made to function with out echo. We tried on both the single port Digium T1 card and the 4 port Digium T1

I have an adit 600 with one fxo card connected to a Digium single span T1 card. CallerID, disconnect supervision work perfect, however the users complain that they have some sound quality issues, after testing it I realized that whenever one is in a phone call they get like silence between the sounds coming from the other party, almost like a cell phone, in other words if there is no sound coming

Hello, I have an Adit 600 (3 FXS cards) hooked up to a digium T1 card in my asterisk box. I 'connected' the slots to the a:1 T1 interfaces via the command line. The slots (3 fxs) are configured with 'ls' signaling. I configured the T1 card with the same line settings as the T1 interfaces on the adit and I get green lights on both the T1 card and the T1 interface on the adit (so

I've spent all day today trying to fix an echo problem and I've made no ground whatsoever. I have Digium TDM400 with 3 FXO & 1 FXS. I've tried this computer at two completely different sites with different phone providers. I've tried compiling & installing different versions of Zaptel (currently running 1.4.0, started at 1.2.9 and worked my way up). I've tried

I'm running Asterisk CVS-HEAD-05/24/04-17:37:48 on kernel 2.4.25-gentoo-r3. I have a Digium TDM-400P card with 4 FXO ports. Here are the pertinent files: zaptel.conf: fxsks=1-4 loadzone = us defaultzone=us zapata.conf: [channels] context=north_in_pots_vip group=1 signalling=fxs_ks usecallerid=no hidecallerid=no callwaiting=no restrictcid=no threewaycalling=no echocancel=1

Hello all, I have a new Carrier Access Adit 600 and am having a problem dialing out and receiving incoming calls. After the initial ring, the calling party will here nothing. There is a silence, but the SIP phones that is supposed to ring will ring. The calling party will not hear the sound of the ring that * sends out on the line. Additionally, if someone picks up the receiver there is no

I'm having issues with echo. My setup is Polycom IP501 phones connected to an Adit 600 via T100P. 13 Lines going to the Adit. All echo so far is on the local side (Employees hears own voice, but only on some calls). Watching the channel with ztmonitor, I notice that TXGAIN is pegged out most of the time. RXGAIN is anywhere between 45 and 60%. Is it safe to assume that playing around

I know a few ppl have those CAC Adit 600's with t400 I can't seem to get my second span up on the T400 connected to the second spand on the adit (A:2) A:1 seems ok Can someone post they zaptel.conf span defintions And maybe a "print config" from the adit 600 cli I think my issue is timing srcs the coding, framing. bld out are all matched thx -------------- next part

Hello, I've got very annoying behaviour from our asterisk PBX. We have 12 channels T1 e&m wink start for TDM and using iax softphones internally (iaxcomm, but tried firefly-thirdparty and discarded for bad sound quality). Slackware 9.1 w/ kernel 2.4.26+ digium TE110P card. In some cases when call is placed from softphone to TDM, system does not detect call answered on Zap channel and

Folks, I've been following recent discussions regarding echo and echo training with much interest, since it's a problem we've never been able to eliminate here. We're facing two challenges presently, and they may be related (or not): a) Cisco 7960s in the office here echo back to our staff, but the customer hears decent sound. b) We've yet to be able to pass faxes through

Hi list, would like to know if some of you have tested asterisk connected to an EADS 6550 analogique PBX (also know as Nexpan50). Our set up is a Dell Optiplex with 1 TDM400 4 FXO, 1 TDM400 4 FXS, (no other card, each of them have their own IRQ) all ports connected to the EADS. We have GS ATA286 as EP. Calling from ATA to ATA give a perfect sound. Calling to PSTN numbers or reverse side,

This may be commonly known but I haven't come across it so here goes, maybe it'll help someone: I have terrible echo with asterisk 1.2, zaptel 1.2.12, and a TDM400P with 1 FXS and two FXO modules. The Mark2 echo canceller with Aggressive turned on was the only setting that would make it acceptable. I found fxotune with this zaptel version to be broken. I pulled the latest fxotune.c

Hey guys, I've been struggling with hangup detection on a Centrex system for a bit now, I was on site Saturday and I took my DMM with me. It would appear that this Centrex service provider uses polarity reversal at the beginning of a call and several times after hangup. I've been reading the archives for the last year and it looks like there was some work done that matches my

Hello, We're in the process of setting up an Asterisk server, and are having echo problems. We have a Digium TE110P, and have tried the MG and MARK2 AGGRESSIVE echo cancellers, with a variety of gain levels and training times, and with both trunk and 1.2 branch versions of Zaptel, Libpre, and Asterisk. In all cases, callers from the PSTN hear their own voice echoed back after 1.5-2 seconds;

Asterisk apparently has five echo cancellation algorithms: STEVE, STEVE2, MARK, MARK2 and MARK3. The current default appears to be MARK2. My question is, has anyone had any experience with any of the others (other than MARK2), and is there some conventional wisdom as to when to use one over another? TIA Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815

I've spent the last week or so troubleshooting echo problems at my Wife's business, and I've been able to clear up about 99% of the echo, but there is still a little residual echo that I can't seem to "tweak out". The users describe it as "buzzing or crackling", but what it sounds like to me is a slight echo, but just of one syllable of the word.

> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Andrew > Kohlsmith > Sent: Thursday, September 29, 2005 2:23 PM > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] Best echo canceller? > > > On Thursday 29 September 2005 17:04, Claudio Canseco wrote: >