It might be. I'm going to work with one of the remote users again tomorrow to see if we can get it working better. You're also right that the PSTN calls don't hear the echo, INSTEAD I hear a faint "static/waves on a beach" sound whenever I talk though a PSTN set through the system to this user. Pushing the packet size back to .03 makes direct calls better, but then MeetMe goes screwy again. ARG! :-)
Anyone have experience with the mentioned fix at: http://bugs.digium.com/view.php?id=5374 and Asterisk 1.2? Does it make call quality difference with SIP? I read the whole thing thinking it was going to end up saying this was a 1.2 feature, but looks like it got pushed to 1.3. Thoughts? Ryan Booz Director of IT Good Steward Software, LLC 111 Sowers Street, Suite 400 State College, PA 16801 Phone: 877-327-3702 x.26 (814-237-3744 x.26) Fax: 719-623-0577 Visit us at www.energycap.com -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wolfgang S. Rupprecht Sent: Thursday, December 08, 2005 4:27 PM To: [email protected] Subject: [Asterisk-Users] Re: Meetme and Sipura SPA-941 - badjitter/distortion "Ryan Booz" <[EMAIL PROTECTED]> writes: > Now, however, there is a (very) slight echo introduced into any calls made > to this extension. So obviously the way that the phone sends packets is > causing some issues. Anyone have a resource or guide to point me to on best > way to debug packet transmission for good calls? Are you sure the echo isn't acoustic echo from the handset itself? Its older sibling, the SPA-841 was really bad in this regard. On a purely sip call between two SPA-841's, if you bumped the earphone gain past halfway on the display the other side would invariably complain about the echo. I always wanted to fill the Sipura handset with modeling clay and see if that helped things any. (The echo was only a problem on direct sip-to-sip calls. Any calls going into the PSTN seemed to always be processed by an echo-can, so it wasn't noticed there.) -wolfgang -- Wolfgang S. Rupprecht http://www.wsrcc.com/wolfgang/ Direct SIP URL Dialing: http://www.wsrcc.com/wolfgang/phonedirectory.html _______________________________________________ --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 _______________________________________________ --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
