That has always been a problem with analog lines without hardware assisted echo-detection/suppression, at least make sure the line impedance is matched to your providers ‘copper pair’ and your RX/TX levels are the same as the providers.
Possibly, you don’t have an echo problem at all. If you were testing to your own mobile (or some other phone within earshot), some sound from one receiver (earpiece) may have acoustically coupled into the other transmitter (microphone), resulting in an echo effect.
Next, the echo may be related to the extension device. Make/model or app name/version? If a softphone on Windows, Mac or Linux, the may be sound leakage from speaker to mic. Confirm that calls between extensions do not have echo.
Is Asterisk transcoding? Try enabling only ulaw (a.k.a. G.711u or PCMU) and alaw for the device and extension.
Possibly, an impedance mismatch on the FXO. What country are you in? I couldn’t guess, because your server timezone seems to be UTC, and many countries have mobile numbers in the form 07XXXXXXXX. Try setting AC Termination Model to Auto-Detected, or a setting that most closely matches what’s used in your country.
You may be able to reduce or elminate the echo by reducing TX and/or RX gain. Of course, that will also reduce the volume heard by the remote party and/or you, so some experimentation may be needed.
If you still have trouble, please post: Echo heard by PBX user, remote party or both? Device used as extension? Have you tried any others? What is FXO port connected to (copper pair from CO, cable MTA, fiber ONT, etc.)? If your system also has a SIP trunk, is that free from echo?
Analog phone lines will be better understood after reading
and you will understand why intrinsically two channel RX/TX communications, mixed into a single current loop (pstn line) without perfect subtraction of the TX signal on the (non instantaneous) RX signal and the corollary subtraction of RX from TX will be the cause of echo, clever FFT filters can ‘tail out’ echo, cheap ATA’s aren’t generally so endowed. So your best effort is to match the transmission lines’ impedance and balance your TX/RX levels as best as you can.
appreciate your help. yup tried another application as windows softphone is working now.
so i had microsip , and now using zoiper , the voice is good after adjusting gain as you proposes.
the issue with zoiper that free version can allow only one sip account while i have 4 accounts , do you know free windows softphone can work great with no issue
I have generally had better results with grandstream devices, but no longer have any copper pair analog services. Best results have used Digium hardware with added HWEC, but that was a long time ago
