similar to: Echo cancellation, when software doesn't cut it. Whats next?

I notice the zaptel Makefile option the mark2 option & KFLAGS+=-DAGGRESSIVE_SUPPRESSOR is now gone. Does simply adding these options still compile in a certain echo can - or is there an other method of activating a particular can. I have not had to update my machine that is connected to pstn for a while & I don't want to jump into echotraining without a way to quickly enable what has

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

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 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,

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.

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

Hello all, I have a WildCard TE410P setup and working with a full PRI with the latest CVS. SIP and IAX2 gateways are accessing the PRI without issue, however; echo is very prominent in some calls and is only heard by the IAX2/SIP client. The echo Is not present in calls to cell phones because they are digital, centrex land Lines have a barely noticable echo, but analog lines aint so pretty.

> -----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: >

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

Hi, i am from Germany, so excuse my School English. I use Asteriks 1.2.12.1, zaptel 1.2.9 and mISDN Rc25 - since my update of Asterisk 2 wooks ago, Echos accure in my SIP Calls. I use SNOM 360, sometimes there is no echo (for example if i call myself via SIP->Asterisk->SIPProvider->TELEKOM->ISDN) but if i call other people there occures Echo many times. The Routing is always the

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

I have been reading with great interest the posts on trouble shooting echo cancellation with *. Is it just coincidence that all of this discussion has been with analog lines. Are PRI's susceptible to echo problem like POTS lines. Thanks for clearing this up. Alan

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;

I'm using bristuffed Asterisk with ISDN/ZAPHFC I have VERY ugly (outgoing) sound through ISDN/HFC if echocancel=yes in zapata.conf, but without echocancel I have bad (incoming) echo Through PSTN/FXO sound is ok with or without echocancel. I tried other echo cancellers (in zconfig.h) two times: ECHO_CAN_KB1 (this was default) ECHO_CAN_MARK2 ECHO_CAN_MG2 after any change I compiled (make

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

hi! In your experience what is the best choice for echo canceller ? Which one should work better: STEVE, STEVE2, MARK, MARK2, MARK3, KB1 ? Regards, Claudio -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050929/f556bf91/attachment.htm

I have 2 Digium 4 port FXO cards in my system. The system is a P4 2.4Ghz, 512MB RAM, Promise FastTrax 100 TX2 Pro Raid, 80GB RAID1 for storage - whitebox - running RedHat 9. With pretty much any CVS HEAD version we are getting, what I will call, "phantom" calls on some lines. What I mean by a phantom call is that the line will ring, Asterisk will log that the Zap channel has been

I have an x100p which suffers from echo (no surprise there apparantly :), and a few of the things I've read about tuning out echo say the first thing to get right is the tx and rx gain, and for that you need a few different types of test lines from the telco (Telstra, Optus, whatever). Does anyone have any numbers in Australia for such lines? I think they just need to send out a tone of a