similar to: Echo cancellation

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

Hi all, 1. Just bought a new TE411P and about to install it replacing the existing TE410P. I am assuming I need to set echocancel=no and echocancelwhenbridged=no now that it will be done in hardware, correct? 2. Is there any way to monitor hardware echo cancellation to ensure it is working (apart from being on a call)? 3. The Digium webite says "By supporting 16ms with 128 channels or

I have a 12 channel PRI with SNOM 190's and asterisk CVS from January. Most calls are fine, all incoming calls are fine, but I am getting echo on a significant number of outgoing calls. The person on the other side hears a perfect call, but the SIPphone side gets to hear themselves. It happens 100% of the time to some numbers (outgoing only), and only sporadically to others. Has anyone

Hi Everybody I've got a weird problem. I am running one Asterisk system on a dual processor box. This box mostly do VoIP only but it has a Fritz PCI ISDN card installed with latest drivers. Dialing out through the ISDN cards from an internal Snom phone works fine and so does dialing in. Except - if I load the ztdummy module (for IAX channels) the capi drivers starts acting up. It is hard

Ok I know I'm not the only one having echo problem with asterisk but the weird thing is that when I receive a call from a PSTN line on my TDM04B card I don't have any echo problem at the beginning of the call then after a few minutes I start having echo on my side only (the person calling from a regular phone doesn't have any echo), then it stop and come back all the way until the

Hello, I have a crossover PRI(Asterisk server to PBX) and a regular telco PRI T1 line and currently have echocancel=yes and echocancelwhenbridged=yes on those spans in zapata.conf. I was discussing CPU load with another Asterisk user and he mentioned that PRIs don't need echo cancelation and that turning it off will reduced CPU load on the server. I checked many sample configs and the

I have echocancel=yes in zapata.conf but when I do a zap show channel 1, I notice echo cancellation is turned off. I followed the article that talks about the order in which the statements need to be in zapata.conf to get echo canceling to work: http://lists.digium.com/pipermail/asterisk-users/2005-June/110615.html But it is still not working. Does anyone know how to get echo

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

I'm experience echo on outgoing calls: Snom 200 ----> Asterisk ----> T100P ----> PRI ----> called party I am getting echo on the Snom 200 phone. The called party does not hear the echo. Since the PRI is digital, I don't really understand where the echo is coming from. I turned on echotraining, echocancellation=yes (128), and echowhenbridged in zapata.conf, to no avail.

I'm looking for a good, reliable, and cheap 4-port FXS ATA. Does anyone know of one that works with Asterisk? Thanks, Waldo

Hello, I have read the information on echo and SIP in the FAQ and I have scoured the mailing list for possible solutions, but as yet I have not been able to get rid of this echo. I have a Sip phone, Digium 4 Port FXS, and 3 Digium FXO cards installed into an asterisk server. If I call between the Sip Phone (Budgettone-100) and the 4 FXS ports everything sounds great. If I call out to the PSTN

Hi all, I have a problem with echo and silence in the middle of calls. the echo problem is that in the first 5 to 10 seconds of a call there is echo on the sip side but not on the PSTN side, also the echo will randomly come back in the call sometimes, I'd say 3 out of 10 calls. the other problem I have is that sometimes ( like maybe 4 times a day ) we will be talking to PSTN calls and one

Hi Everybody, Has anybody tried the above (or indeed any other 4XBRI cards) successfully with Asterisk. As far as I can see the above mentioned card is an active ISDN card but supported by it's own I4L driver. This leads to interesting questions particularly regarding echo cancellations (which usually doesn't work on the cheap passive cards with one exception as far as I can see).

Hi, I have a box with Debian Sarge, Asterisk 1.2.1 (and zaptel 1.2.1) and a TDM2400P with echo canceller. I installed the card but no echo cancellation is being made...seems like the echo canceller module does not work, infact the software cancellation is working. My zapata.conf has echocancel = 128 and echocancelwhenbridged = yes but no echotraining parameter which gives a warning. I found

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

I am using a SIP softphone (X-lite, SIPPS or Firefly) connected to an Asterisk v 1.0.3 PBX. The PBX is also connected via a ISDN-PRI crossover cable to a Avaya Definity Generic 3 PBX via a TE405P card. All outside of the office calls go through the Definity. Here's the issue: Calls to internal SIP extensions, Definity extensions, other offices within our private network (through the

I just implemented a system using a TE411P hardware echo cancellation card. Per Digium, I setup zaptel.conf, and zapata.conf the same way as I always have. To my surprise calls out to the PSTN had a terrible echo. 1 - 2 second delay, and quite clear. The echo was so bad that I had to remove the hardware echo cancellation module from the card. We are only using the 1st span of this card right

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

We're deploying an asterisk-based phone system at all of our branch offices in an effort to eliminate long-distance costs incurred from the constant branch to branch calls. We're using the Snom 300's at all offices for the desk phones and X100P cards to interface to 2 analog lines. I'm having a problem tuning all the echo out of the system. So far two branches are using the

Hi, all. I'm a programmer lived in Korea. This time, I was investingating echo cancellation and found speex echo cancellation related API. But echo stilll remains. 1. echo cancellation using only files works propertly. 2. echo cancellation using sound device works not propertly. You can download test files as followed. http://www.vdream.co.kr/temp/ec/echocancel.wmv ( testing screen capture