I'm receiving an error when I try to call an extension that leads to ConfBridge() app in my dialplan
Here is the extensions.conf
exten => 601,1,Noop()
same => n,ConfBridge(room-one,default_bridge,default_user,sample_user_menu)
here are the definitions in conferance.conf
[default_user]
type=user
[default_bridge]
type=bridge
[sample_user_menu]
type=menu
*=playback_and_continue(conf-usermenu)
*1=toggle_mute
1=toggle_mute
*4=decrease_listening_volume
4=decrease_listening_volume
*6=increase_listening_volume
6=increase_listening_volume
*7=decrease_talking_volume
7=decrease_talking_volume
*8=leave_conference
8=leave_conference
*9=increase_talking_volume
9=increase_talking_volume
And here is the error that I receive in the asterisk console when I dial 601:
== Using SIP RTP CoS mark 5
-- Executing [601@internal:1] NoOp("SIP/nasko-00000047", "") in new stack
-- Executing [601@internal:2] ConfBridge("SIP/nasko-00000047", "room-one,default_bridge,default_user,sample_user_menu") in new stack
[Sep 26 00:43:06] ERROR[16686][C-000000be]: app_confbridge.c:1187 join_conference_bridge: Conference 'room-one' mixing bridge could not be created.
== Spawn extension (internal, 601, 2) exited non-zero on 'SIP/nasko-00000047'
-- Executing [h@internal:1] Hangup("SIP/nasko-00000047", "") in new stack
== Spawn extension (internal, h, 1) exited non-zero on 'SIP/nasko-00000047'
The version of the asterisk is 11.5. I've looked at a several tutorials on the internet and this was the basic configuration for all of them, but I can't get it to work and it seems that google does not know much about this error.
If someone is more experianced in this matter, please try to help.
Thanks in advance
Best Answer
This works for me:
or a simpler config with less possibility of failing: