hi Lars, does el-cheapo BRI mean chan_modem_i4l? the echo you hear is caused by todays very sensitive pstn phones. the mic picks up the sound from the speaker and sends it back to you. try to reduce the rx and tx volumes of your snoms (if that is possible), or add rx/tx gain support to chan_modem_i4l like i have for chan_capi. regards kapejod -- Klaus-Peter Junghanns CEO,CTO Junghanns.NET GmbH Breite Strasse 13 - 12167 Berlin - Germany fon: +49 30 79705392 fax: +49 30 79705391 iaxtel: 1-700-157-8753 email: kpj@junghanns.net http://www.junghanns.net/asterisk Am Die, 2003-05-27 um 14.49 schrieb Lars Boegild Thomsen:> Hi Everybody, > > Got a weird problem here I think. Got a setup with an asterisk (current > from cvs as of a few hours ago) in a box with an el-cheapo ISDN BRI card > connected to the PSTN network and two Snom phones internally (one Snom-100 > and one Snom-200). Dialing between the snom phones or dialing out to PSTN > from any of the snom phones works perfectly. > > But when I receive a call FROM the PSTN network to any of the Snom phones, > the user on the Snom phone is hearing a bad echo of himself. This echo is > however NOT heard on the PSTN side. Any hints? > > Regards, > > Lars... > > -- > Lars Boegild Thomsen > Technical Director > JustIT Sdn. Bhd. > Cell Phone (MY): +60 (16) 323 1999 > ICQ: 6478559 > Yahoo Chat: lars_boegild_thomsen@yahoo.com > MSN Chat: lars_boegild_thomsen@hotmail.com > http://www.justit.ws > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users
Hi Everybody,
Got a weird problem here I think. Got a setup with an asterisk (current
from cvs as of a few hours ago) in a box with an el-cheapo ISDN BRI card
connected to the PSTN network and two Snom phones internally (one Snom-100
and one Snom-200). Dialing between the snom phones or dialing out to PSTN
from any of the snom phones works perfectly.
But when I receive a call FROM the PSTN network to any of the Snom phones,
the user on the Snom phone is hearing a bad echo of himself. This echo is
however NOT heard on the PSTN side. Any hints?
Regards,
Lars...
--
Lars Boegild Thomsen
Technical Director
JustIT Sdn. Bhd.
Cell Phone (MY): +60 (16) 323 1999
ICQ: 6478559
Yahoo Chat: lars_boegild_thomsen@yahoo.com
MSN Chat: lars_boegild_thomsen@hotmail.com
http://www.justit.ws
I have been struggling with echo cancellation for the last few days. It seems to me that it would be useful to start up a technical discussion of the issue so that we don't have to solve the problem empirically. My system is SIP (Grandstream) <=> Asterisk <=> Adtran TSU600 <=>FXO <=>POTS. From what I can tell from testing on my system the echo has the following characteristics: 1) It varies over time. It is worst at the start of a call and after periods of silence. From this I conclude that the echo canceller is adaptive, adjusting delay and amplitude based on real time signal analysis. 2) The echo is only heard on the SIP phones, not on the far end of the POTS connection. In spite of seeming somewhat mysterious at first, I concluded that this is actually the expected behavior. Note: Voice is digitized at SIP phone, transmitted (with delay) to the FXO card and converted to analog. The analog signal is placed on the 2-wire POTS line, where it loops back to the analog receiver in the FXO card and is digitized and sent back to the SIP phone as an echo. The POTS end hears no echo because there is no pathway for the echoed packets to get back to the POTS. From this, it seems to me that the ideal place to deal with the echo would be at the point of conversion (from A to D and back), since there would be virtually no delay at this point and it would be simpler to determine the correct amplitude correction. If this is the case, then it explains why adjusting receiver and transmitter gains would aid in cancelling echo, with the caveat that it should only work in those cases where the Rx ant Tx gains that are being adjusted are those which are right at the A/D and D/A converters. In other words, adjustment of these values in Zaptel.conf should have no effect if I am using an outboard channel bank. What I would like is for the real experts to jump into this discussion, lay out the real theory and technical details of the echo cancellation used in asterisk so that we can all make more intelligent attacks on the echo problem. How about it? Stephen R. Besch
I've got an X100P & a cisco 7960. if i call from an analog line via the x100p to the cisco, there is an overly audible echo on the cisco. If i make a call from a cisco to cisco, there is no echo. zapata.conf has echocancel=yes & echocancelwhenbridged=yes set. Any ideas? I'm currently using the default implementation of echo cancellation...which one should I try next? elijah chancey
I am also facing this issue between softphone <-> softphone and cisco <-> softphone. I don't have zaptel, so is there any other configuration to apply echo cancellation. Regards, Arslan. -----Original Message----- From: Martin Pycko [mailto:martinp@digium.com] Sent: Thursday, November 20, 2003 4:01 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] echo cancellation Did you place echocancel=yes before the definition of the channel with channel keyword in zapata.conf ? regards Martin On Wed, 19 Nov 2003, Elijah Chancey wrote:> I've got an X100P & a cisco 7960. if i call from an analog line viathe> x100p to the cisco, there is an overly audible echo on the cisco. If i > make a call from a cisco to cisco, there is no echo. zapata.conf has > echocancel=yes & echocancelwhenbridged=yes set. Any ideas? > > I'm currently using the default implementation of echo > cancellation...which one should I try next? > > elijah chancey > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users >_______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users
What is the status on echo cancellation in Asterisk/CAPI? I know Zaptel drivers will do echocancel, and chan_capi does have support for Eicon's echo cancellation, but what about the rest? I found in mailing list archives a patch description that will mute RX channel whenever signal level is 4 times lower than TX channel - has this patch made into chan_capi or Asterisk CVS yet? Peter -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031124/a756d988/attachment.htm
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Nicolas Gudino wrote: | If the terminating tail circuit has cancelable echos and if the echo | canceler is enabled, you will hear echo for the first few utterances and | then it will die away. After a few seconds of speech, the echo should be | gone or at least very quiet compared to the echo level at the beginning of | the call. This is the signature of a working echo canceler. " In my situation where we use VoIP softphones connecting to an X101P card to the PSTN, the other end hears us just fine and there is no echo. However, there is substantial echo on the VoIP clients. I did one extended test with X-Lite (or maybe it was DIAX, but I have the feeling the performance would have been the same) where I was in an extended conversation. Echo was quite bad for the first portion of the call, but after approximately 1 - 2 minutes, it progressively got better until after that time, it was still there, but at a very controlled, curtailed level. The beginning and end of the echoed portions were chopped off and the volume level of the echo was quite a bit lower than the audio level of the two parties, to the point that it was no longer distracting for the VoIP client to talk. This was using the MARK2 with AGGRESSIVE enabled suppressor. |>"Headsets are particularly notorious for poor echo performance.". This |>is due to lack of acoustic isolation. Perhaps you could test using |>headphones and a mic. What does this mean? That the earphones are feeding back into the wrap-around mic? I have a USB Plantronics DSP 400 headset running under ALSA sound system. Is it feeding through the plastic parts and entering the microphone sitting out near my mouth? - -- Jason A. Pattie pattieja@xperienceinc.com Xperience, Inc. (http://www.xperienceinc.com) -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.2.3 (GNU/Linux) Comment: Using GnuPG with Debian - http://enigmail.mozdev.org iD8DBQE/y7ULuYsUrHkpYtARAkxjAJ4jPSeS6vbcnLpEkUrYS/VrVs0klACfVuF7 npoIHQr3q6n/5H4zyrGoaWw=XjPE -----END PGP SIGNATURE----- -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. MailScanner thanks transtec Computers for their support.
echo cancellation is activated in /etc/asterisk/zapata.conf However, how to confirm it? Does "zap show channel 1" confirm the existence of echo cancellation? -- David Kwok Iaxtel/FWD # 17001813482 -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/x-pkcs7-signature Size: 1878 bytes Desc: S/MIME Cryptographic Signature Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20040120/c9c3a049/smime.bin
I could kiss you, this resolved my entire echo problem with the X100P. I have a bottle of scotch with your name on it. Jeff>From: dkwok <dkwok@iware.com.au> >Reply-To: asterisk-users@lists.digium.com >To: asterisk-users@lists.digium.com >Subject: [Asterisk-Users] Echo cancellation >Date: Fri, 20 Feb 2004 12:43:59 +0100 > >I had problem with echo on my GS phone and it is now resolved. What happens >was that the order of settings were incorrect in the zapata.conf. > >All the echo cancellation settings and other settings to effect on a >particular channel have to be set out before the occurence of the channel >no. > >see zapata.conf below: >; Zapata telephony interface >; >; Configuration file > >[channels] >; >language=en >context=incoming >amaflags=default >cancallforward=yes >callwaiting=no >busydetect=no >callprogress=no >usecallingpres=yes >musiconhold=classic >rxgain=0.0 >txgain=0.0 >immediate=no >echocancel=yes >echocancelwhenbridged=yes >echotraining=yes >usecallerid=yes >callerid=asreceived >callwaitingcallerid=yes >threewaycalling=yes >signalling=fxs_ks >callreturn=yes >channel=1 > >After the change the voice quality of GS is superior I think it is value >for money. I would not spend my money on snom or other phones. I would >consider cisco 7960 is money is not an object though. > >-- >David Kwok > >Iaxtel/FWD # 17001813482 ext 1002 ><< smime.p7s >>_________________________________________________________________ Stay informed on Election 2004 and the race to Super Tuesday. http://special.msn.com/msn/election2004.armx
I had problem with echo on my GS phone and it is now resolved. What happens was that the order of settings were incorrect in the zapata.conf. All the echo cancellation settings and other settings to effect on a particular channel have to be set out before the occurence of the channel no. see zapata.conf below: ; Zapata telephony interface ; ; Configuration file [channels] ; language=en context=incoming amaflags=default cancallforward=yes callwaiting=no busydetect=no callprogress=no usecallingpres=yes musiconhold=classic rxgain=0.0 txgain=0.0 immediate=no echocancel=yes echocancelwhenbridged=yes echotraining=yes usecallerid=yes callerid=asreceived callwaitingcallerid=yes threewaycalling=yes signalling=fxs_ks callreturn=yes channel=1 After the change the voice quality of GS is superior I think it is value for money. I would not spend my money on snom or other phones. I would consider cisco 7960 is money is not an object though. -- David Kwok Iaxtel/FWD # 17001813482 ext 1002 -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/x-pkcs7-signature Size: 1878 bytes Desc: S/MIME Cryptographic Signature Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20040219/417f477c/smime.bin
hi... in my zapata conf i have echocancel=yes but when i do zap show channel 2 i have echocancel=128 taps, currently off I am have big echo problems from the remote user....the remote user can hear himself talk. Thanks, Peter -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040825/5167ecd6/attachment.htm
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I am having problems with echo, first let me explain my setup: I have a Gateway box, which basically is an Asterisk with a PRI card. It's only job is to interface with 2 incoming ISDN PRI connections. Then there are two other asterisk boxes to which my users are registered. Dialing from a phone it hits the first asterisk which forwards it to the gateway box and then on to the PSTN. What are the general causes of echo? When calling from my SIP phone I hear no echo but the other end, the PSTN end, hears a lot of echo. What could cause this? Kristian.
> I am having problems with echo, first let me > explain my setup: > > I have a Gateway box, which basically is an > Asterisk with a PRI card. It's only job is to > interface with 2 incoming ISDN PRI connections. > Then there are two other asterisk boxes to which > my users are registered. > Dialing from a phone it hits the first asterisk > which forwards it to the gateway box and then on > to the PSTN. > > What are the general causes of echo? > When calling from my SIP phone I hear no echo but > the other end, the PSTN end, hears a lot of echo. > What could cause this? > > Kristian.You need to google echo on the wiki. There are so many causes for echo and possible fixes that work on some installations but not others. Some keywords to search for are "echo pri" and "echo avoidance">From reading the list, there is no echo introduced by a PRI and the echois created by the far side. Nonetheless, it is still your problem. I would first try have a phone register directly to the box with the PRI card and see if there is any difference. I would then check your Zapata.conf settings and adjust gains and also try different echo can settings. Make sure to change one thing at a time and restart asterisk and test. Write down your results so you can get an idea for what is working. Finally, if that is still not helping, you can change the echo can method. Trial and error. Digium sells a hardware echo can upgrade for their cards as well but I think it may only be an option for the quad port cards. Finally, if that still is not working, then you may want to see what 3rd party devices others have used. I have seen success stories posted but am not sure what was used. I think the reason for the 3rd party devices working when * software echo can cannot is the size of the tail. Thanks, Steve
I've got a slight problem with echo. Basically, most of the outgoing phone calls on our system echo, but as far as I can tell, the incoming echo has been relatively fixed, with just a bit of work left to do on it. I read somewhere that asterisk doesn't echo cancel on outgoing calls, am I wrong in that assumption, and if I am, what else can be done when the echo training and echo cancel tapping isn't working? Aaron
Hi there I am using asterisk version 1.2.4. I have clients based on the iax client library dialling into meetme sessions. I am experiencing echo in the case where one or more users has speakers instead of headphones. So the audio from me is fed from the other participant's speakers into their mic and back to me. What is the best way to fix this? Is there an echo cancel facility in asterisk which will sort this out? Many thanks Steven
I've got exactly the same problem with echo, where the mic feeds into the speakers. I'm looking at purchasing the Tellabs 2572 64ms T1 echo cancellation card to see if it will help. Anyone have any experience with this hardware and how it deals with echo? I've read on the wiki that it's supposed to work pretty well.
On Wednesday, March 15, 2006 9:49 PM Hagen Rode wrote:> I've got exactly the same problem with echo, where the mic feeds into > the speakers. I'm looking at purchasing the Tellabs 2572 64ms T1 echo > cancellation card to see if it will help.Are we still talking about people attending MeetMe conferences with laptops but without headsets? If so: Personally I doubt that a 64ms echo can will help you with this issue. I have not yet measured this but am willing to bet that the echo produced by such a speaker/mic setup will exceed 64 and even 128ms. Perhaps you might want to record a test and have a look at the WAV file to at least get an idea of how long the echo really is before you invest in any sort of echo can hardware. You might very well find out that your only option are headets. Kind regards, JP -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/x-pkcs7-signature Size: 3104 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20060316/76e7b21a/smime.bin
Hi all, I'm using bristuff 0.2.0 RC8o with a HFC pci card and on several calls I saw that the echo cancellation is on OFF Echo Cancellation: 0 taps, currently OFF (the result of "zap show channel 1-1" for example) I cannot control it and on my zapata i configured it switchtype = euroisdn signalling = bri_cpe_ptmp pridialplan = unknown prilocaldialplan = unknown echocanel = yes echocancelwhenbridged = yes echotraining = yes musiconhold = default immediate = yes group = 1 context=incoming channel => 1-2 Anyone can explain me what happen ? Thanks Giordano Grandis Le informazioni contenute nella presente e-mail e nei documenti eventualmente allegati possono essere confidenziali e sono comunque riservate al destinatario della stessa. La loro diffusione, distribuzione e/o copiatura da parte di terzi ? proibita. Se avete ricevuto questa comunicazione per errore, Vi preghiamo di informare immediatamente il mittente del messaggio e di distruggere questa e-mail. This e-mail may contain confidential and/or privileged information. If you are not the intended recipient (or have received this e-mail in error) please notify the sender immediately and destroy this e-mail. Any unauthorised copying, disclosure or distribution of the material in this e-mail is strictly forbidden. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060328/25b4b8c7/attachment.htm
General question. If you install a Digium card in an Asterisk system, and install zaptel drivers, do this give any benefit of echo cancellation? Our PSTN gateway is a separate Audiocodes box, so the zaptel card wouldn't actually be connected to anything. I'm wondering though doing this would help, in general, with echo cancellation. Doug.
> -----Original Message----- > From: Jean-Michel Hiver [mailto:jhiver@ykoz.net] > Sent: Wednesday, June 28, 2006 4:27 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Echo Cancellation > > > Douglas Garstang a ?crit : > > >General question. > > > >If you install a Digium card in an Asterisk system, and > install zaptel drivers, do this give any benefit of echo > cancellation? Our PSTN gateway is a separate Audiocodes box, > so the zaptel card wouldn't actually be connected to > anything. I'm wondering though doing this would help, in > general, with echo cancellation. > > > > > a) No it won't unless you connect it to a TDM circuit > > b) I have an audiocodes too (mediant 2000 4E1), and I've > found the echo > cancellation to be superb. I'm surprised to see you're having issues!Me too, given we're on fiber gigabit ethernet, with only a few test calls in progress!