How to prevent a pjsip endpoint to be able to send audio

Hi, I am trying to setup a conference, and I want a specific pjsip endpoint to not be able to speak (= send audio to the conference).

Is it something to do on the endpoint configuration or on the dialplan ? or in the conf bridge file ?

Thanks.

The MUTEAUDIO dialplan function[1] can be used to mute incoming audio from a channel. ConfBridge also allows you to start a user as muted[2].

[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Function_MUTEAUDIO
[2] https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Configuration_app_confbridge

Thanks for your help.

My confbridge file is as follow :

[general]

[default_user]
type=user
admin=no
send_events=no
echo_events=no
startmuted=no
music_on_hold_when_empty=no
quiet=yes
announce_user_count=no
announce_user_count_all=no
announce_only_user=no
wait_marked=no
dsp_drop_silence=yes
announce_join_leave=no

[default_bridge]
type=bridge
max_members=50
record_conference=no
internal_sample_rate=8000
mixing_interval=20
video_mode=none
regcontext=rtmonitor
enable_events=no

and my dialplan is like that :

[general]
static=yes
writeprotect=yes
autofallthrough=no

[rtmonitor]
exten => _rt-.,1,NoOp(conf)
same => n,Answer()
same => n,ConfBridge(${EXTEN})
same => n,Hangup()

I only want to mute on specific endpoint. How can I modify my confbridge for that ?

You can also use AMI https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+ManagerAction_MuteAudio

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