We moved our TDM800P card to a newer machine. We have several SIP users and all are now experiencing intermittent echo and sometimes it’s very annoying. The local sip user hears echo but no echo heard by user on other end (pstn). The odd thing is that it’s not always there, even in the same phone call sometimes. I’ve looked online and googled trying to find a fix. I’m hoping someone will have some ideas here.
Details:
[ul]
[li]TDM800P analog card[/li]
[li]SLES 11 SP1 linux[/li]
[li]2.6.32.43-0.4-default x86_64 GNU/Linux kernel[/li]
[li]Asterisk/1.6.2.20[/li]
[li]Asterisk GUI-version : SVN–rexported[/li]
[li]SIP & IAX softphone clients on local side[/li]
[li] * connected to PSTN trunk[/li]
[li]started with mg2 EC and then moved to oslec EC; [/li]
[li]tried different values but currently: echocancel=1024, echocancelwhenbridged=no, echotraining not set[/li]
[li]ran fxotune[/li]
[li]followed online instructions for adjusting rx/tx gains and tried different values; currently rx set to 3 and tx set to -10[/li]
[li]local sip/iax users are using various softphone clients (voix, voiper, blink) on various OSs (win, mac, linux)[/li]
[li]local sip/iax users are using various model headsets[/li]
[li]irq looks good[/li]
[li]result with dahdi_test > 99.99%[/li][/ul]
I’d appreciate any suggestions and help.