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Type:
Bug
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Status: Closed
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Priority:
Minor
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Resolution: Third Party Issue
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Affects Version/s: 15
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Fix Version/s: None
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Component/s: Conferences
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Labels:None
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ToDo:
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Asterisk Version:13 and 16
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Distro Version:15
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Distro:FreePBX Distro
When I am calling into a conference bridge and my channel is using Opus full band (48kHz), Asterisk will transcode to slin@16000.
– General –
Name: PJSIP/1002-00000000
Type: PJSIP
UniqueID: 1557576564.0
LinkedID: 1557576564.0
Caller ID: 1002
Caller ID Name: Zoiper
Connected Line ID: (N/A)
Connected Line ID Name: (N/A)
Eff. Connected Line ID: (N/A)
Eff. Connected Line ID Name: (N/A)
DNID Digits: 1100
Language: en
State: Up (6)
NativeFormats: (opus)
WriteFormat: slin16
ReadFormat: opus
WriteTranscode: Yes (slin@16000)>(slin@48000)>(opus@48000)
ReadTranscode: No
Time to Hangup: 0
Elapsed Time: 0h0m3s
Bridge ID: (Not bridged)
– PBX –
Context: from-internal
Extension: STARTMEETME
Priority: 5
Call Group: 0
Pickup Group: 0
Application: ConfBridge
Data: 1100,,,
Call Identifer: [C-00000000]
Variables:
DB_RESULT=
GOSUB_RETVAL=
CALLFILENAME=1100-1100-never-20190511-140926-1557576564.0
REC_POLICY_MODE_SAVE=
MON_FMT=wav
FROMEXTEN=1002
TIMESTR=20190511-140926
YEAR=2019
MONTH=05
DAY=11
NOW=1557576566
REC_STATUS=INITIALIZED
MAX_PARTICIPANTS=0
MEETME_MUSIC=
MEETME_ROOMNUM=1100
MACRO_DEPTH=0
TTL=64
CALLEE_ACCOUNCODE=
DIAL_OPTIONS=HhTtr
AMPUSERCID=1002
AMPUSERCIDNAME=Zoiper
AMPUSER=1002
REALCALLERIDNUM=1002
TOUCH_MONITOR=1557576564.0
CDR Variables:
level 1: cnum=1002
level 1: cnam=Zoiper
level 1: dnid=1100
level 1: clid="Zoiper" <1002>
level 1: src=1002
level 1: dst=STARTMEETME
level 1: dcontext=from-internal
level 1: channel=PJSIP/1002-00000000
level 1: lastapp=ConfBridge
level 1: lastdata=1100,,,
level 1: start=1557576564.937299
level 1: answer=1557576564.993498
level 1: end=0.000000
level 1: duration=2
level 1: billsec=2
level 1: disposition=1
level 1: amaflags=3
level 1: uniqueid=1557576564.0
level 1: linkedid=1557576564.0
level 1: sequence=0