Hello everybody, I am sorry to bring this up again if this kind of echo issue has ever discussed. Phone2 in below call path experiences quite annoying echo: Phone1 --> FXS (TDM400P) --> Asterisk --> SIP GW --> PSTN --> Phone2 It is annoying as on phone2, we can hear the whole words we say with the level of maybe 25% of the original sound. I can reduce the echo to maximum with the following settings for my FXS port on zapata.conf: rxgain=-8.0 txgain=2.0 echocancel=256 echotraining=500 But it is still not entirely eliminated as we still sometimes hear the last syllables, with the level of maybe 5% of the original sound. What I did was just playing around with the values of those parameters, use ztmonitor to have the FXS rx/tx signal visualised and use only my ears to check it. I think my ears are fine :), as I do this because my friends complain about the echo they hear. Does anybody know a better method to find the best value for those parameters? There is no echo on phone2 when I use softphone like this: PC(X-Lite) --> Asterisk --> SIP GW --> PSTN --> Phone2 The following is the version of asterisk I am using: CLI> show version Asterisk SVN-branch-1.2-r7999 built by root @ atvie-asterisk on a i686 running Linux on 2006-01-13 06:15:02 UTC And I set the echo canceller in zconfig.h to ECHO_CAN_MG2. Cheers, Anto -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060114/cfbeca69/attachment.htm
Try In chan_zap.c change the following line: #define READ_SIZE 160 to #define READ_SIZE 16 In zapata.conf jitterbuffers=40 This will also increase system load by a factor of 10. 2006/1/15, Aryanto Rachmad <aryanto.rachmad@chello.at>:> > Hello everybody, > > I am sorry to bring this up again if this kind of echo issue has ever > discussed. > > Phone2 in below call path experiences quite annoying echo: > > Phone1 --> FXS (TDM400P) --> Asterisk --> SIP GW --> PSTN --> Phone2 > > It is annoying as on phone2, we can hear the whole words we say with the > level of maybe 25% of the original sound. I can reduce the echo to maximum > with the following settings for my FXS port on zapata.conf: > > rxgain=-8.0 > txgain=2.0 > echocancel=256 > echotraining=500 > > But it is still not entirely eliminated as we still sometimes hear the > last syllables, with the level of maybe 5% of the original sound. > > What I did was just playing around with the values of those parameters, > use ztmonitor to have the FXS rx/tx signal visualised and use only my ears > to check it. I think my ears are fine :), as I do this because my friends > complain about the echo they hear. > > Does anybody know a better method to find the best value for those > parameters? > > There is no echo on phone2 when I use softphone like this: > > PC(X-Lite) --> Asterisk --> SIP GW --> PSTN --> Phone2 > > The following is the version of asterisk I am using: > > CLI> show version > Asterisk SVN-branch-1.2-r7999 built by root @ atvie-asterisk on a i686 > running Linux on 2006-01-13 06:15:02 UTC > > And I set the echo canceller in zconfig.h to ECHO_CAN_MG2. > > Cheers, > > Anto > > > _______________________________________________ > --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 > > >-- Giovanni Miano -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060116/2b7e0f96/attachment.htm
Aryanto Rachmad wrote:> Hello everybody, > > I am sorry to bring this up again if this kind of echo issue has ever discussed. > > Phone2 in below call path experiences quite annoying echo: > > Phone1 --> FXS (TDM400P) --> Asterisk --> SIP GW --> PSTN --> Phone2 > > It is annoying as on phone2, we can hear the whole words we say with the level of maybe 25% of the original sound. I can reduce the echo to maximum with the following settings for my FXS port on zapata.conf: > > rxgain=-8.0 > txgain=2.0 > echocancel=256 > echotraining=500 > > But it is still not entirely eliminated as we still sometimes hear the last syllables, with the level of maybe 5% of the original sound. > > What I did was just playing around with the values of those parameters, use ztmonitor to have the FXS rx/tx signal visualised and use only my ears to check it. I think my ears are fine :), as I do this because my friends complain about the echo they hear. > > Does anybody know a better method to find the best value for those parameters? > > There is no echo on phone2 when I use softphone like this: > > PC(X-Lite) --> Asterisk --> SIP GW --> PSTN --> Phone2Are you sure that X-Lite is not running an echo can? I'd say it's more likely that the SIP GW is causing the echo and that when you use X-Lite, it's echo cans are removing the echo. Try to make a call from Phone->FXS->Asterisk->X-Lite I suspect there will be no echo. BTW what is the SIP Gateway? -- Cheers, Matt Riddell _______________________________________________ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss)
I have been tasked with moving our office from our junky old Nortel Meridian system to Asterisk. I will be keeping the T-1 PRI Voice circuits for the immediate future. My current intent is to purchase a nice Dell server to run everything, with a Digium TE110P PRI card. I also intend to run some version of Centos as the operating system. I have a mix of Centos 3 and Centos 4 boxes here and would like to keep that consistent. There will be about 25 extensions to start and up to another 10 in the next year. More than that and we need more office space. I already have ethernet running to all of the locations where phones will be, but in most cases only 1 port, which is already being used by a PC, so I will likely need a phone with 2 ports so I can daisy-chain off of it. Customization will be likely as we are a technology-heavy company and would like to be able to link incoming phone numbers to orders and comments in the database for the sales and customer service reps eventually. We have a programming department (I am sysadmin) and will be able to write the code to do this wither on the phone or on the rep's screen (pushed based on static IPs). I would like to keep the phones under $300 apiece (well under if possible). Questions: (1) Any advantage of Centos 3 or 4? (2) What phones would be best to get? (3) Any recommendation on a Dell server? I was thinking a PE1850 because of the dual power supplies and hardware RAID in a 1U chassis. (4) If I get outside sales agents working from home, what would be a good phone for them to get to hook into our system as a local extension? Thanks a bunch! Warren