Hello, I am a longtime FreePBX non distro user.
Recently I built a CentOS 7 (64) server with FreePBX 14 & Asterisk 13.24.0. Using PJSIP*
Everything seems to work fine with the following exception: After setting extension and trunk and sip / PJSIP settings as well as endpoint phone to use g722 it still seems to transcode it.
Using core show channel it shows native is g722 but read and write format is slin16. When I change them back to g711 then native and read and write are all g711 like expected.
I am curious if anyone has thoughts on what could be causing this.
As I stated in my OP:
Everything is set to g722 and yet read & write formats are slin16. I included a picture to help illustrate.
The picture is an output of an active channel using core show channel ‘channel-id’
This is how it has been for a while now. You are looking at how Asterisk will Read and/or Write to/from the disk but the actual audio is still in its native format. The reason that you are not seeing this with ulaw (g711) is because it is an 8kHz/16kHz sampled coded which is the native sampling for Asterisk. When codecs, like g722, have differently sample rate (like 7kHz) you’ll see that.
Does your trunking provider meaningfully support g722 (for example when interoperating with a mobile carrier on a VoLTE call)? The vast majority either don’t support it at all, or support is limited to calls to/from other customers of the provider. It’s likely that they will fall back to ulaw or alaw on most calls to/from the PSTN.
My provider does support G722. I realize the vast majority of legacy telco gear is only ulaw at best. I have spoke with my provider about interop with VoLTE providers which I believe could happen.
