I have recently moved an asterisk system to a new location. This location is experiencing terrible echo. I installed the HPEC from Digium but that has caused a new problem. When HPEC is enabled and more that 16 taps are used, the audio from the outside caller gets clipped. Instead of hearing: Hello, my name is Mike one hears He o, m ame ike If the taps are set to less than 16, there is no clipping, but there is significant echo. I don't know if this is relevant, but asterisk does report "No Zaptel transcoder support" on startup. I am at my wits end; any advice would be greatly appreciated. My setup follows: Hardware, AMD Athlon(tm) 64 Processor 3000+, 512MB ram, 2 TDM400p with a total of 5 fxo channels, snom 320 phones OS: gentoo 2006.1 amd64 Software: Asterisk 1.2.17 and Zaptel 1.2.16 (fxotune does not seem to work in this configuration) Also tried Asterisk 1.4.2 and Zaptel 1.4.1 (fxotune does work, but doesn't seem to help in this configuration) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070405/1c8958be/attachment.htm
Michael Boers wrote:> I have recently moved an asterisk system to a new location. This location > is experiencing terrible echo. I installed the HPEC from Digium but that > has caused a new problem. > > When HPEC is enabled and more that 16 taps are used, the audio from the > outside caller gets clipped. Instead of hearing: > > Hello, my name is Mike > > one hears > > He o, m ame ikeI am experiencing the same thing. I assumed that I just didn't have a fast enough CPU (2.4 Ghz Celeron Ghz, also tried on a 1.8 Ghz Pentium 4). I am using a T400P with an Adtran TA750 Channel Bank rather than the Digium analog cards. I'm not doing any VoIP on this system, strictly analog and I get echo on calls.
Same issue here as well. Running Centos 4.4 (x86_64) with all updates installed on a Pentium D 2.8 GHz, 1.5GB RAM and TDM800P with 1 x Quad FXS and 2 x FXO ports. I installed the new 64 bit HPEC module earlier in the week when it was released and have been unable to get it to work. I have the same symptoms as the OP and also noticed that there is much more line noise (soft static) than when HPEC is disabled. I sent a support request in several days ago and I have not yet seen a reply. Kind Regards Greg -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070406/170ba3f7/attachment.htm
Hi: Greg Siemon wrote:> Same issue here as well. > > Running Centos 4.4 (x86_64) with all updates installed on a Pentium D > 2.8 GHz, 1.5GB RAM and TDM800P with 1 x Quad FXS and 2 x FXO ports. > > I installed the new 64 bit HPEC module earlier in the week when it was > released and have been unable to get it to work. I have the same > symptoms as the OP and also noticed that there is much more line noise > (soft static) than when HPEC is disabled. > > I sent a support request in several days ago and I have not yet seen a > reply.I have observed very similar behaviour in an HPEC installation. Some observations: the clipping happens if receive gain is cranked up. If I set the gain to defaults, the clipping goes away, but then I have the problem of users complaining that the remote caller is too quiet. Another recommendation -- whether or not you are using Zaptel 1.2.x or 1.4.x, you should always the fxotune from 1.4.x. A question - do the people who are experiencing this problem have non-default gain settings, either in zapata.conf or in the telephone sets? -Stephen-
In my case, the rx and tx gains are 0. I will try the 1.4 version for fxotune to see if that helps. Thanks for the suggestions! -- Michael Boers On 4/6/07, Stephen Bosch <posting@vodacomm.ca> wrote:> > Hi: > > Greg Siemon wrote: > > Same issue here as well. > > > > Running Centos 4.4 (x86_64) with all updates installed on a Pentium D > > 2.8 GHz, 1.5GB RAM and TDM800P with 1 x Quad FXS and 2 x FXO ports. > > > > I installed the new 64 bit HPEC module earlier in the week when it was > > released and have been unable to get it to work. I have the same > > symptoms as the OP and also noticed that there is much more line noise > > (soft static) than when HPEC is disabled. > > > > I sent a support request in several days ago and I have not yet seen a > > reply. > > I have observed very similar behaviour in an HPEC installation. > > Some observations: the clipping happens if receive gain is cranked up. > If I set the gain to defaults, the clipping goes away, but then I have > the problem of users complaining that the remote caller is too quiet. > > Another recommendation -- whether or not you are using Zaptel 1.2.x or > 1.4.x, you should always the fxotune from 1.4.x. > > A question - do the people who are experiencing this problem have > non-default gain settings, either in zapata.conf or in the telephone sets? > > -Stephen- > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070406/e0f73042/attachment.htm
>From: Bob Smither <smither@c-c-i.com> >Date: Fri, 06 Apr 2007 20:22:34 -0500 > >Warning - novice question ahead! > >Dear List, > >I have installed Asterisk 1.4.2 on an AMD dual core x86-64 box running >CentOS 4.4. Compilation and installation were straightforward. > >The box only supports IAX connections so I have no zap hardware. > >My question is this - where do I set the txgain and rxgain parameters >for the IAX channels? With a previous setup I used settings in >zapata.conf, but I believe these are not used with the IAX connections >(?).You are right. zapata.conf is not used in IAX connections. My reading has led me to believe that manipulating gain on an IP PBX is neither necessary nor practical in VoIP channels, so Asterisk does not devise such settings. Yuan Liu>Thanks for any insight. > >-- >Bob Smither <smither@c-c-i.com>
On Fri, 2007-04-06 at 21:42 -0700, Yuan LIU wrote: <snip>> You are right. zapata.conf is not used in IAX connections. My reading has > led me to believe that manipulating gain on an IP PBX is neither necessary > nor practical in VoIP channels, so Asterisk does not devise such settings.Thanks Yuan. I beg to differ with the developers if there really is no amplitude control on IP channels. I have an application where I am studying the spectrum of recorded voice. When I call into my Asterisk box I have to hold the phone away and speak softly to avoid clipping the recorded waveform - clipped waveforms play havoc with the spectrum. I guess it is time to study the source code (ugh!). Best regards, -- Bob Smither <smither@c-c-i.com>
Bob Smither wrote:> On Fri, 2007-04-06 at 21:42 -0700, Yuan LIU wrote: > > <snip> > >> You are right. zapata.conf is not used in IAX connections. My reading has >> led me to believe that manipulating gain on an IP PBX is neither necessary >> nor practical in VoIP channels, so Asterisk does not devise such settings. > > Thanks Yuan. I beg to differ with the developers if there really is no > amplitude control on IP channels. I have an application where I am > studying the spectrum of recorded voice. When I call into my Asterisk > box I have to hold the phone away and speak softly to avoid clipping the > recorded waveform - clipped waveforms play havoc with the spectrum. > > I guess it is time to study the source code (ugh!).The device doing the IP/TDM conversion should be the device that sets the gains correctly. The same applies to echo canceling.
On Sat, 2007-04-07 at 23:52 -0500, Eric "ManxPower" Wieling wrote: <snip>> The device doing the IP/TDM conversion should be the device that sets > the gains correctly. The same applies to echo canceling.As I stated, this started with the warning of Novice Question :-). Eric, can you elaborate on the above? Is the device you are referring to within Asterisk or somewhere else in VOIP land? I am not sure what to do with this information. If it matters - the clipping behavior I see is in voices recorded on Asterisk 1.4.2 from a call placed over Packet8 and routed back to my Asterisk box through NuFone.net. Same happens from a POTS call routed back to my Asterisk box through NuFone.net. Thanks, -- Bob Smither <smither@c-c-i.com>
Kevin P. Fleming wrote:> Eric "ManxPower" Wieling wrote: > >> Any updates on this? > > The code is done and initially tested; it is being reviewed internally > and should be available on Friday or Monday.Under what circumstances would this clipping be present? Is this patch going to be recommended for anybody using HPEC? -Stephen-