Remote users report weak audio

Asterisk 1.8.15
SIP protocol

I’m getting a complaint from a remote user that the audio he receives is too faint. He is using X-Lite 4.5 on Windows 7.

He swears that he has all volume controls set to max, and that he runs into the same problem on two different computers. He also says that Skype works at a comfortable listening volume (not sure what relevance that has, but that’s the report I received).

I’m looking for a way to increase the audio level on this end to rule that out as a problem, or perhaps mitigate whatever problem he is having.

A few hours with google have yielded some sites like this:

…and this…

… which indicate that the fix involves editing /var/www/html/admin/modules/core/ and /etc/asterisk/extensions.conf. The latter is one of those files that we are warned to never edit by hand, so I assume there must be a way to edit extensions_custom.conf to do the same thing, and yet I can’t find a way to put that customization into a context that extensions.conf or extensions_additional.conf can see.

I saw that prior to Asterisk 1.6 there was no way to adjust SIP volume, but that was fixed in 1.6 and enhanced in 1.8. I just need to figure out the “right” way to do it within the FreePBX environment. Also… is editing /var/www/html/admin/modules/core/ really necessary? I don’t mind, if that’s what I have to do, but it seems like a huge disconnect with FreePBX’s objective to make it easy and idiot-proof for us dumbos, and I’m thinking there must be a more FreePBX-friendly way to do it.


Eric /