I have the following situation: A softphone on the Asterisk server is making calls to local sip phones. These phones are on Linksys SPA2102 interfaces. The phones are old analogue phones and these have low microphone gain. I tried increasing the TX levels in Asterisk to compensate however this cause a great deal of echo. Instead I adjusted the “FXS Port Input Gain” to 16 in the SPA20102 config. This solved the problem. Calls from the Asterisk-based softphone to these analogue phones gives no noticeable echo. When, however, the operator on the Asterisk server-based softphone makes a call to one of the analogue sips and then makes a 3-way attended transfer (atxferthreeway) to an outside line (via VoIP service) or to another local analogue phone, there is a great deal of echo on the transferee’s phone. If the operator makes a atxfercomplete transfer the echo is largely taken care of, it seems, by the SPA2102’s echo cancellations
I’ve tested with zero gain on the SPA and the echo remains however is reduced and better handled by the SPA’s echo cancellation.
I need to use these old phones and the question of the gain on the voice is important. Can anyone please suggest a solution?
As I understand it, Asterisk echo cancellation only applies to DAHDI and Zaptel cards. I’m open to try another interface, however it would need to be external like the SPA2102.