I’m new to asterisk and I’ve been reading the O’Reilly’s book for several days when I bump in to this issue.
Before everything else here is my configurations.
exten => 601,1,Noop() same => n,ConfBridge(room-one,default_bridge,default_user,sample_user_menu)
[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 now when I connect to the Asterisk and dial “601”
Asterisk CLI Error:
== 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[C-000000be]: app_confbridge.c:1187 join_conference_bridge: Conference 'room-one' [b]mixing bridge could not be created.[/b] == 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 also, it seems that google does not know much about this error.
If someone is more experienced in this matter, please try to help.
If anyother information is needed please let me know and I’ll provide it.
Thanks in advance