Hello,
When creating a Bridge using ARI, it looks like the bridge sampling frequency is locked to 192 kHz. There’s no argument on the ARI REST API call that I can see to specify it, or the context for the bridge.
Asterisk is being used on FreePBX.
ANCK-PC-MELE-01*CLI> core show channel PJSIP/1000-00000000
– General –
Name: PJSIP/1000-00000000
Type: PJSIP
UniqueID: 1719262327.4
LinkedID: 1719262326.0
Caller ID: 1000
Caller ID Name: test
Connected Line ID: 1234lemon
Connected Line ID Name: ABC
Eff. Connected Line ID: 123lemon
Eff. Connected Line ID Name: ABC
DNID Digits: (N/A)
Language: en_GB
State: Up (6)
NativeFormats: (opus)
WriteFormat: slin192
ReadFormat: slin192
WriteTranscode: Yes (slin@192000)->(slin@48000)->(opus@48000)
ReadTranscode: Yes (opus@48000)->(slin@48000)->(slin@192000)
Time to Hangup: 0
Elapsed Time: 0h0m35s
Bridge ID: c08799bb-bfdd-44ff-9c15-c0bdcd588891
– PBX –
Context: from-internal
Extension:
Priority: 1
Call Group: 0
Pickup Group: 0
Application: AppDial
Data: (Outgoing Line)
Call Identifer: [C-00000001]
Variables:
BRIDGEPEER=Local/1000@from-internal-00000000;2
GOSUB_RETVAL=
sipkey=
SIPHEADERKEYS=
TECH=PJSIP
DIALEDPEERNUMBER=1000/sip:1000@192.168.8.153:57016;transport=TCP;ob
KEEPCID=TRUE
CWIGNORE=
MON_FMT=wav
FROMEXTEN=unknown
TIMESTR=20240624-205207
YEAR=2024
MONTH=06
DAY=24
REC_STATUS=INITIALIZED
PICKUPMARK=1000
EXTTOCALL=1000
TTL=64
RINGTIMER=15
– Streams –
Name: audio-0
Type: audio
State: sendrecv
Group: -1
Formats: (opus)
Metadata:
Manually editing (just for testing) confbridge_additional.conf and restarting asterisk yielded no change in the bridge sampling frequency.
[general]
;This section reserved for future use
internal_sample_rate=48000
[default_user]
type = user
internal_sample_rate=48000
[default_bridge]
type = bridge
internal_sample_rate=48000