You need to be using firmware 1.0.1.12 on the GXP2000 There is a known issue with feedback/echo on the GXP2000 with earlier versions. It was fixed with .12 firmware and works fine.
Well mine does anyway... Cheers, Mark -----Original Message----- From: Shawn Iverson [mailto:[EMAIL PROTECTED] Sent: Saturday, 12 November 2005 9:32 AM To: [EMAIL PROTECTED] Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Wits end with echo On Wednesday, November 09, 2005 5:57 PM, Jon Reynolds wrote >Hello, > >I have an AAH-1.5 with a TMD400P with four lines, 8 >Grandstream GXP-2000 >phones, I am having echo issues on the GXP-2000 side. I have evaluated a similar setup as yours involving the Granstream 2000. I was able to isolate two sources of echo. 1. The Grandstream 2000 when the volume is up will cause echo because the microphone picks up the speaker on the handset. Don't even attempt to use speakerphone as you will cause full echo that will drive the remote party nuts. This problem is specific to the phone and doesn't relate to Asterisk. (Perhaps a newer firmware will resolve this?) > >Here is what I have tried so far: > >The server has everything in the bios turned off except what >is needed, >USB, LPT, Serial etc,etc. > >I have uncommented Echo Suppresion in zconfig.h and shutdown >and turned >back on the asterisk box. > >I have updated the phones to 1.0.12 firmware, I have echotraining=800, >echocancel=yes, echowhenbridged=yes, in my sip.conf file. I am using >Mark2 as the echo suppresion and still I have echo. 2. Try the following settings in your zapata.conf. These seem to work well for me. echocancel=yes echocancelwhenbridged=yes echotraining=yes ; Use ztmonitor to adjust your gain to levels that work for you. rxgain=-4.0 txgain=-4.0 > >All the phones have been wired straight to the cisco 2950 >switch and all >cables have been tested and found to be good. > >I am completely at a loss at this point as to where to start >looking and >working to fix the problem. I would like to switch from Mark2 >to MG1 but >I don't know how I would acomplish that with AAH. I have >played with the >rx and tx gain but after reading multiple docs on it am still >unsure how >this would help and how to adjust it using /usr/bin/ztmonitor 1 -v. When you place a call outbound, launch it and watch your gain as you speak. If you can humm a tone at around normal speaking voice to the far side, you can adjust the tx gain up or down to get it about halfway. Have the far end party do the same for the rx gain. It is trial and error. I was surprised to find that my setup worked best by turning the gain down. Check out this link for more info: http://www.voip-info.org/wiki/view/Asterisk+x100p+echotraining > >If anybody could point me in a new direction or something else to look >at or something more to read that I may have missed I would be very >appreciative. > >Thanks for any help, > >Jon BTW, Digium recently released a new card with hardware-based echo cancellation. It may be worth a try. http://www.digium.com/index.php?menu=product_detail&category=hardware&pr oduct=TE411P&tab=details You may still hear echo at the first moment a call is placed, but it should completely disappear in a few seconds. -- Shawn _______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
