Hello,
My system:
PBX Firmware:
12.7.5-1807-1.sng7
FreePBX 14.0.3.13
Current Asterisk Version: 13.22.0
Please help more experienced users how to set codecs.
Incoming calls to the FreePBX are G.711a
Telephones use G.722
The “core show channel …” command shows something like this
Incoming call:
– General –
Name: PJSIP/incoming-gw-00000111
Type: PJSIP
UniqueID: 1537443591.557
LinkedID: 1537443591.557
Caller ID: 222222222
Caller ID Name: 222222222
Connected Line ID: 555555555
Connected Line ID Name: 555555555
Eff. Connected Line ID: 555555555
Eff. Connected Line ID Name: 555555555
DNID Digits: 555555555
Language: en
State: Up (6)
NativeFormats: (alaw)
WriteFormat: slin16
ReadFormat: slin16
WriteTranscode: Yes (slin@16000)->(slin@8000)->(alaw@8000)
ReadTranscode: Yes (alaw@8000)->(slin@8000)->(slin@16000)
Time to Hangup: 0
Elapsed Time: 0h0m38s
Connection to the phone:
– General –
Name: PJSIP/555555555-00000112
Type: PJSIP
UniqueID: 1537443591.558
LinkedID: 1537443591.557
Caller ID: 555555555
Caller ID Name: 555555555
Connected Line ID: 222222222
Connected Line ID Name: 222222222
Eff. Connected Line ID: 222222222
Eff. Connected Line ID Name: 222222222
DNID Digits: (N/A)
Language: en
State: Up (6)
NativeFormats: (g722)
WriteFormat: slin16
ReadFormat: slin16
WriteTranscode: Yes (slin@16000)->(g722@16000)
ReadTranscode: Yes (g722@16000)->(slin@16000)
Is not too much transcoding?
Maybe phones should also use only G.711a