[Asterisk-Users] Channelized T1, SIP phones, HW Echo Canceller

[Steve Creel]

I have a channelized T1 coming in from our telco, terminated onto a TE405.
There are three channelbanks serving internal analog extensions, and about
10 Cisco 7960s.

I have no reports of echo on the analog extensions (as expected).  The
7960 users complain of occasional echo (seems like 1 in 5 calls).  Only
the SIP user hears the echo, not the caller.

I have echocancel=yes, echotraining=yes, echocancelwhenbridged=yes.
Changes in the taps of echotraining have made things worse, so I have left
it alone.

I have backed the txgain down, as audio going out on the telco T1 is
really hot.  Even at -6dB gain, it is still notably louder from outside
than other audio (comparing the ring generated by the telco when calling
into asterisk with the ring generated by asterisk calling a station from
the auto-attendant).  If I drop gain to anything less that -6, I lose all
audio.

Would a hardware echo canceller deal with this type of echo?  My
understanding is that it is a result of sip being non-realtime and
introducing latency (the latency being half the difference from the
original utterance and the echo).  Is this correct, or do I have it all
wrong?