@expresspotato wrote:
Hi,
So I decided to try out opus tonight, but for some reason I don’t seem to able to change the settings for it using codecs.conf.
I know asterisk / freepbx is at least able to read the codecs.conf file, as I named the codec to opustest and the call would not go through unless the extension allowed codecs included opuststest.
Regardless if calls are made to or from the phone seems like its always using opus @ 48Khz as shown below. On Zopier changing from opus wide to narrow etc doesn’t make a difference. Am I missing something?
Connected to Asterisk 13.13.1 currently running on localhost (pid = 1166) localhost*CLI> core show channel PJSIP/6000-0000000b -- General -- Name: PJSIP/6000-0000000b Type: PJSIP UniqueID: 1515721003.11 LinkedID: 1515721003.10 Caller ID: 6000 Caller ID Name: device Connected Line ID: 3000 Connected Line ID Name: Kevin Eff. Connected Line ID: 3000 Eff. Connected Line ID Name: Kevin DNID Digits: (N/A) Language: en State: Up (6) NativeFormats: (opus) WriteFormat: slin48 ReadFormat: slin48 WriteTranscode: Yes (slin@48000)->(opus@48000) ReadTranscode: Yes (opus@48000)->(slin@48000) Time to Hangup: 0 Elapsed Time: 0h0m25s Bridge ID: afb4e395-bcc1-467b-8e1d-42e794fb497b -- PBX -- Context: from-internal Extension: Priority: 1 Call Group: 0 Pickup Group: 0 Application: AppDial Data: (Outgoing Line) Call Identifer: [C-00000006] Variables:codecs.conf
[opus] type=opus fec=yes cbr=yes packet_loss=30 max_playback_rate=8000
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