similar to: Questions regarding new echo cancellation features...

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

For those that have had X100P echo problems, it seems that somewhere just prior to Asterisk CVS-12/04/03-14:24:40 it has been fixed to where its hardly detectable for just a couple of seconds at the start of a call. (I use to have it rather bad for the first 15 seconds or so.) I'm using DECHO_CAN_MARK2 with KFLAGS+=-DAGGRESSIVE_SUPPRESSOR and echotraining=yes echocancel=yes

Over the last couple weeks I've tried everything I could get my hands on in an attempt to get rid of my echo problems. Using a CVS checkout of just yesterday, I've tried every echo cancellation routine in zconfig.h (including Mark2 w/Aggressive) , as well as the echotraining=800 mentioned on this list just last week. While some things worked better then others, I would consider none

Thank you for the suggestions Andrew. I have not come across some of them before and will give them a shot. Based on my reading, changing the motherboard should have minimal impact unless that motherboard and the TDM400P don't get along (aka. IRQ sharing). I have disabled everything that is not needed and I do not believe I have any IRQ problems and I am NEVER wrong ;). Calls are crisp and

Has anyone has issues with echo using a Wildcard with a PRI from a major Canadian Telco? (Bell, Telus, AllStream, Sprint, Group Telecom). We are using a T1 from GT that is giving use annoying echos whenever a SIP/IAX2 client calls a local analog line. Calling cells phones is no issue since its digital. Regardless, there should be no issue with echo on a PRI at all. NOC at GT is telling us

I'm testing out my SNOM 200 phone by trying to call out through NuFone. When I do so, I don't hear an echo at all (in fact I can't hear myself through the phone) but the callee can hear an echo when she speaks. NuFone tells me their network is totally digital and so can't be involved in an echo. This is all well and good, but the echo is still there. Any suggestions? As a

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,

Just updated from CVS 12-23-03 to tarbal 0.7.1. Identical settings in zconfig.h for echo cancellation (MARK2, aggressive OFF). The echo got worse, much worse. It takes longer to train and overspeak now disables cancellation, which it did not before. In fact, I now have echo on VoIP to VoIP on our local network, which I never had before. Was something changed with the echo supression which

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 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, I'm new to Asterisk and would like to know SIP support status. Are there any testing been done with some widely deployed client (Cisco SIP IP phone, ...)? I was using Vocal but I'm now interested in Asterisk as it seems to offer more features...if it supports SIP. Thanks for your help. Francois.

I've got echo problems. *** I'm looking for paid support. *** I'll accept free support, but don't mind paying if someone really knows what they are doing. I've read the wiki, etc. Played with the settings in zapata.conf Using V400P PSTN->_T1->_ASTERISK->_BROADVOICE->_PSTN ECHO ON CALLED PHONE PSTN->_T1->_ASTERISK->_T1->_PSTN NO ECHO

Hi, I have just begun working with TDMoE running between 2 fiber nics the dynamic span works great. In my main asterisk box's startup file I just 'modprobe tor2', then start asterisk. The zaptel, ztdynamic & ztd-eth modules all load by themselves when tor2 is loaded. If I stop asterisk then 'modprobe -r tor2' the tor2 module is removed but the other three remain. If I

Good Day all, I have a Fractional PRI connected to my Asterisk Box via a T100P card. When I initiate a call out to phone number 123-8888 the call sounds great no echo what so ever. If the person at 123-8888 hangs up and calls me right back (same handset on both sides) same trunk line The call always has echo on it. The Asterisk sip extension hears them selves echoing. The remote party

We have been pretty much able to solve our echo problems, except for the primary mode in which we desire to operate our system. See system diagram at bottom. Prior to making adjustments to cancel echos (all echocancel=no): Call Type Result (Before) --------- --------------- CP <- LEC PRI * TDMoE * FXO -> AP

Hello, Can the wildcard x100P be bought of the shelf in stores - say in Singapore. Thanks Varun

I came into this with my eyes wide open. I have read ABSOLUTELY EVERYTHING there is to be found on the net about avoiding echo problems BEFORE I even attempted to create a production system. Since lots of people are apparently using this in production environments now I just assumed that echo IS avoidable. As others have recommended, I created a test system with the proposed production

The echo canceller algorithms aren't doing anything. We get extreme echo during the conversation, it appears even before the call connects, the echo is there... This only happens with SIP to/from WCFXO (analog POTS). Looking at the Zaptel configuration: /etc/asterisk/zapata.conf: echocancel=yes echocancelwhenbridged=yes rxgain=0.8 txgain=0.8 (although none of the above options

New complile causing trouble... How do you enable MARK2 echo can with DAGGRESSIVE_ SUPPRESSION? looks like zconfig.h ..but I have quite a bit of echo on a bridgeded call and echocancelwhenbridged=no John ------------------------------------------------------ This message was sent using Monroe-Woodbury's WebMail. http://webmail.mw.k12.ny.us/ This e-mail was scanned and found clean by

I used to get bad echo and also voice breaking up during a conversation, so I upgraded the machine (from a PII 300 to a Athlon 1.8GHz) and have resolved the voice breaking up issues but still seem to have a fairly bad echo on *both* sides of the conversation. Currently I am only using the system for outbound calls, so I am always calling from the digium USB port phone and out to the telco using a