Asterisk 1.4, dahdi 2.2, Centos 2.6, Processor is Xeon 3200 with AEX2400 and TE220 installed (although the TE220 is not in use).
I’m getting echo on SIP-to-SIP calls (SIP phone to Asterisk to SIP trunk to the PSTN). The echo must be bouncing back from the far end – when I use one of the SIP phones to call my cell phone and then mute the cell phone, the echo stops. My first impulse is to reduce the rxgain parameter in sip.conf so that the echo will be less noticeable, but before I reinvent this wheel, does anyone know if this sounds like a good strategy, and if so, what’s a good starting value to try for rxgain?
Jim Shilliday
That is a problem with whoever provides the SIP to PSTN interface, if the echo is coming form the PSTN side, and with the phone if it is coming from the phone side. These are the places where echo cancellation should be applied. (Note that the use of speaker phones on the PSTN side may make echo cancellation very difficult.)
Hi
Firstly what sip phones are you using ?
Secondly make a call to a person on a mobile the ISNT yours and is in a different room to you so that you cannot hear them.
Then let us know how it was
Ian
Hello and thanks for the replies – the SIP phones are Cisco 7940’s, and I should have been clearer – someone else answered my cell phone in a different room, and I could hear echo unless they muted the phone. No speaker phones were involved.
Jim