Hello everyone, I get reports that all I/C & O/G calls are low in aduio. This is reported by the Called party or the calling party. Audio levels at the FreePbx extn are OK. Is there a setting(s) that I can adjust to increase the audio level. Many thanks
What channel technology types are you using. DAHDI/analogue is the only one where audio levels can possibly be blamed on Asterisk. In particular, for pure SIP, it is the phones that are responsible for achieving the correct level.
Oh yes should have explained, the I/f to the PSTN (BT - OPENREACH) is a Sangoma Vega 60G (FX) PORT) which is colocated witht tht eFreePbx server and the tech used between the two boxes is SIP.
This is the SIP setting for Outgoing;
There are no settings for the Incoming side.
The correct place to fix this is on the Vega.
Ok thanks, any idea as to where?
The same settings will be used, In fact, “insecure” is an incoming only setting, although insecure=invite will be ignored here as you have no secret. You may well need insecure=port.
Here are the Vaga GW settings, any guidance as what seetings to change and what to set them to would be appreciated, thanks Peter
Try increasing both gains by 3 (dB) or maybe even 6. If you set them too high you may have echo problems.
I’m not familiar with Vega, but if their terminology is consistent with other vendors, Tx affects what the remote party hears and Rx what the extension user hears. You can confirm this by making a big change in only one gain.
If the remote party has plenty of volume, increase only Rx, as this is less likely to cause echo problems. Approximately speaking, the propensity for echo is the sum of the two gains.
covered here if using the Vega management module in freepbx
generically from the gateway ui
- FXO/FXS profile under POTS
- TDM profile for T1/E1
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