Hello I meet very strange behaviour when there are just 2 participants in conference.
If one of the members leave the conference , there are no announce for the leaving user just music on hold start immediately after the user quit.
This is my configuration
extensions.conf
exten => 990,1,Ringing
exten => 990,2,Wait(3)
exten => 990,3,Answer()
exten => 990,4,ConfBridge(1234,default_bridge,default_user,sample_user_menu)
exten => 990,5,Ringing
exten => 990,6,Wait(5)
exten => 990,n,Hangup()
the steps after 4 was for some testing…
confbridge.conf
[general]
[default_user]
type=user
;pin=1010
marked=yes
music_on_hold_when_empty=yes
music_on_hold_class=default
announce_user_count=yes
announce_user_count_all=yes
announce_join_leave=yes
dsp_drop_silence=yes
denoise=yes
[102030]
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
[sample_admin_menu]
type=menu
*=playback_and_continue(conf-adminmenu)
*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=no_op
8=no_op
*9=increase_talking_volume
9=increase_talking_volume
Output from the console
== Using SIP RTP CoS mark 5
– Executing [990@GoldTelecom-new-idea:1] Ringing(“SIP/100-00000021”, “”) in new stack
– Executing [990@GoldTelecom-new-idea:2] Wait(“SIP/100-00000021”, “3”) in new stack
– Executing [990@GoldTelecom-new-idea:3] Answer(“SIP/100-00000021”, “”) in new stack
> 0x7fafd4002a60 – Strict RTP qualifying stream type: audio
> 0x7fafd4002a60 – Strict RTP switching source address to 192.168.0.222:8000
– Executing [990@GoldTelecom-new-idea:4] ConfBridge(“SIP/100-00000021”, “1234,default_bridge,default_user,sample_user_menu”) in new stack
– <SIP/100-00000021> Playing ‘vm-rec-name.alaw’ (language ‘en’)
> 0x7fafd4002a60 – Strict RTP learning complete - Locking on source address 192.168.0.222:8000
– <SIP/100-00000021> Playing ‘beep.alaw’ (language ‘en’)
– x=0, open writing: /var/spool/asterisk/confbridge/confbridge-name-1234-1538057391.117 format: sln, 0x7fb05c005bc8
– User ended message by pressing #
– <SIP/100-00000021> Playing ‘auth-thankyou.alaw’ (language ‘en’)
– Channel CBAnn/1234-00000017;2 joined ‘softmix’ base-bridge <98e6fae9-3b6c-4f0d-99ee-3fcebc806fdb>
– <CBAnn/1234-00000017;1> Playing ‘/var/spool/asterisk/confbridge/confbridge-name-1234-1538057391.117.slin’ (language ‘en’)
– <CBAnn/1234-00000017;1> Playing ‘conf-hasjoin.slin’ (language ‘en’)
– Started music on hold, class ‘default’, on channel ‘SIP/100-00000021’
– Channel SIP/100-00000021 joined ‘softmix’ base-bridge <98e6fae9-3b6c-4f0d-99ee-3fcebc806fdb>
– <CBAnn/1234-00000017;1> Playing ‘confbridge-join.slin’ (language ‘en’)
== Using SIP RTP CoS mark 5
> 0x7fafd401a500 – Strict RTP learning after remote address set to: 192.168.168.50:63434
– Executing [990@GoldTelecom-new-idea:1] Ringing(“SIP/222-00000022”, “”) in new stack
– Executing [990@GoldTelecom-new-idea:2] Wait(“SIP/222-00000022”, “3”) in new stack
– Executing [990@GoldTelecom-new-idea:3] Answer(“SIP/222-00000022”, “”) in new stack
> 0x7fafd401a500 – Strict RTP switching to RTP target address 192.168.168.50:63434 as source
– Executing [990@GoldTelecom-new-idea:4] ConfBridge(“SIP/222-00000022”, “1234,default_bridge,default_user,sample_user_menu”) in new stack
– <SIP/222-00000022> Playing ‘vm-rec-name.alaw’ (language ‘en’)
> 0x7fafd401a500 – Strict RTP learning complete - Locking on source address 192.168.168.50:63434
– <SIP/222-00000022> Playing ‘beep.alaw’ (language ‘en’)
– x=0, open writing: /var/spool/asterisk/confbridge/confbridge-name-1234-1538057409.120 format: sln, 0x7fafac01fad8
– User ended message by pressing #
– <SIP/222-00000022> Playing ‘auth-thankyou.alaw’ (language ‘en’)
– Stopped music on hold on SIP/100-00000021
– <SIP/222-00000022> Playing ‘conf-onlyone.alaw’ (language ‘en’)
– <CBAnn/1234-00000017;1> Playing ‘conf-onlyone.slin’ (language ‘en’)
– <CBAnn/1234-00000017;1> Playing ‘/var/spool/asterisk/confbridge/confbridge-name-1234-1538057409.120.slin’ (language ‘en’)
– <CBAnn/1234-00000017;1> Playing ‘conf-hasjoin.slin’ (language ‘en’)
– Channel SIP/222-00000022 joined ‘softmix’ base-bridge <98e6fae9-3b6c-4f0d-99ee-3fcebc806fdb>
– <CBAnn/1234-00000017;1> Playing ‘confbridge-join.slin’ (language ‘en’)
– Channel SIP/222-00000022 left ‘softmix’ base-bridge <98e6fae9-3b6c-4f0d-99ee-3fcebc806fdb>
– Started music on hold, class ‘default’, on channel ‘SIP/100-00000021’
– Executing [990@GoldTelecom-new-idea:5] Ringing(“SIP/222-00000022”, “”) in new stack
– Executing [990@GoldTelecom-new-idea:6] Wait(“SIP/222-00000022”, “5”) in new stack
– <CBAnn/1234-00000017;1> Playing ‘/var/spool/asterisk/confbridge/confbridge-name-1234-1538057409.120.slin’ (language ‘en’)
– <CBAnn/1234-00000017;1> Playing ‘conf-hasleft.slin’ (language ‘en’)
– <CBAnn/1234-00000017;1> Playing ‘confbridge-leave.slin’ (language ‘en’)
– Executing [990@GoldTelecom-new-idea:7] Hangup(“SIP/222-00000022”, “”) in new stack
== Spawn extension (GoldTelecom-new-idea, 990, 7) exited non-zero on ‘SIP/222-00000022’
– Stopped music on hold on SIP/100-00000021
– Channel SIP/100-00000021 left ‘softmix’ base-bridge <98e6fae9-3b6c-4f0d-99ee-3fcebc806fdb>
– <CBAnn/1234-00000017;1> Playing ‘/var/spool/asterisk/confbridge/confbridge-name-1234-1538057391.117.slin’ (language ‘en’)
– <CBAnn/1234-00000017;1> Playing ‘conf-hasleft.slin’ (language ‘en’)
– <CBAnn/1234-00000017;1> Playing ‘confbridge-leave.slin’ (language ‘en’)
– Channel CBAnn/1234-00000017;2 left ‘softmix’ base-bridge <98e6fae9-3b6c-4f0d-99ee-3fcebc806fdb>