Stuck in ConfBridge()

Asterisk Asterisk Support
1

Hi all

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.

extensions.conf

exten => 601,1,Noop()
    same => n,ConfBridge(room-one,default_bridge,default_user,sample_user_menu)

confbridge.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 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[16686][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

2

This configuration should work, if not it looks like a problem with non-loaded modules …

Please check the output of the Asterisk-CLI-command

(It should look like

Module Description Use Count app_confbridge.so Conference Bridge Application 1 1 modules loaded

If You can’t see a similar output You shoud issue in the CLI

and post the output here.

3

Thanks for the fast response.

I have already loaded that module:

Asterisk*CLI> module show like app_confbridge.so 
Module                         Description                              Use Count 
app_confbridge.so              Conference Bridge Application            0         
1 modules loaded

Also here is my modules.conf:

[modules]
autoload=no
load=pbx_config.so

;Channels
load=chan_sip.so
load=chan_iax2.so
load=chan_bridge.so

;Resources
load=res_rtp_asterisk.so
load=res_musiconhold.so

;Applications
load=app_hangup.so
load=app_playback.so
load=app_stack.so
load=app_dial.so
load=app_voicemail.so
load=app_confbridge.so

;Codecs      
load=codec_ulaw.so
load=codec_gsm.so
load=codec_alaw.so
load=codec_g722.so 
    
;FileFormats
load=format_gsm.so 
load=format_h264.so
load=format_wav.so
load=format_mp3.so

I don’t know if the order at witch the modules are loaded can play a role in this?
The only difference that I can see is that in your output the number for Use Count is 1 and in mine it is 0.

I also ran the CLI in debug 9 and captured this output, I hope it can be helpful:

    -- Executing [601@internal:3] ConfBridge("SIP/nasko-0000002a", "room-one,default_bridge,default_user,sample_admin_menu") in new stack
[Sep 27 13:09:11] DEBUG[10782][C-0000003f]: app_confbridge.c:1141 join_conference_bridge: Trying to find conference bridge 'room-one'
[Sep 27 13:09:11] DEBUG[10782][C-0000003f]: app_confbridge.c:879 destroy_conference_bridge: Destroying conference bridge 'room-one'
[Sep 27 13:09:11] ERROR[10782][C-0000003f]: app_confbridge.c:1187 join_conference_bridge: Conference 'room-one' mixing bridge could not be created.
[Sep 27 13:09:11] DEBUG[10782][C-0000003f]: pbx.c:6316 __ast_pbx_run: Spawn extension (internal,601,3) exited non-zero on 'SIP/nasko-0000002a'
  == Spawn extension (internal, 601, 3) exited non-zero on 'SIP/nasko-0000002a'
[Sep 27 13:09:11] DEBUG[10782][C-0000003f]: channel.c:2661 ast_softhangup_nolock: Soft-Hanging up channel 'SIP/nasko-0000002a'
[Sep 27 13:09:11] DEBUG[10782][C-0000003f]: channel.c:2661 ast_softhangup_nolock: Soft-Hanging up channel 'SIP/nasko-0000002a'
[Sep 27 13:09:11] DEBUG[10782][C-0000003f]: pbx.c:4633 pbx_extension_helper: Launching 'Hangup'
    -- Executing [h@internal:1] Hangup("SIP/nasko-0000002a", "") in new stack

For some reason it first tries to find this bridge and after that issues a delete command ?!?
I’m totally messing something, why it is trying to do that?

4

Ok, as You load only selected modules (autoload = no) You’re missing an auxiliary module.
Just include

in Your modules.conf and the error should go away …

5

I loaded that one as well but the error persisted.
A little breakthrough I went to the modules.conf and set the autoload to yes and now it is working.
Do you know what else I might need to load, because as far as I heard the best practice is to leave the autoload to no?

I’ve also tried to load the rest bridge modules but the error came again:

load=bridge_builtin_features.so load=bridge_simple.so load=bridge_multiplexed.so load=bridge_softmix.so

6

If You prefer autoload=no You can debug into missing things by

a) looking in the CLI-Output for a failed application or function (Like ConfBridge)
b) step-by-step loading/unloading non-loaded modules from /usr/lib/asterisk/modules (or /usr/lib64/asterisk/modules depending on Your distro/architecture)

To be honest, I don’t know which module(s) are needed for app_confbrifge.so to work properly. I always go the other way: autoload = yes plus noload-directives for all modules I don’t want to have loaded (it’s easier :unamused: )

7

Hi, I’m struggling with pretty much the same problem here. The only difference is that I’m running Asterisk on OpenWRT based router. Asterisk is ported in a very light manner and only a number of modules are available here.
Did you manage to figure out what module is required to make confbridge() to work?

Thanks, Andrzej