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Would it be possible to change PABX settings via CLI or changing files manually?

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Personally I don’t think docker is a good container for pbxes, For multiple small clients on one machine, look to FusionPBX which is a gui front end to FreeSwitch, most simple components of a PBX are included , there is also a t38 fax server.

Perhaps best in your case, you can have multiple URLs all pointing to its IP and each client is totally separate, so reusing end point numbers is not a problem.

All of the provisioning and is stored in SQL ( postgres by default) so easily scriptable.

If you need more than that then add a cluster of dsiprouters (a simple gui in front of kamailio, a SIP proxy) before them.


Zulu plugin dial prefix

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Try to place the 0 in the prefix section, that way you’ll send to your provider the correct number.

Transferring calls with the BLF keys proper method?

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Depend on the phone manufacture. They all do it differently.

Please help - updated modules and now unable to access freepbx GUI

Please help - updated modules and now unable to access freepbx GUI

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I don’t. I was in the process of updating quite a few modules via the GUI when the system timed out and I was unable to access the GUI. I was told the better way is via the CLI.

13 to 14 upgrade

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Trying other mirror.
rpmforge | 1.9 kB 00:00
schmooze-commercial | 2.9 kB 00:00
updates | 3.4 kB 00:00
Setting up Install Process
No package schmooze-release available.
Error: Nothing to do

V14 Automatic Updates Not Occurring - Modules or System

V14 Automatic Updates Not Occurring - Modules or System

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We have looked at it. At the current moment it’s not a priority. - this makes me laugh and sad at the same time


V14 Automatic Updates Not Occurring - Modules or System

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Not to start a debate here, but I’m not a fan at all of automatic updates. (besides security updates)

I don’t think having automatic updates on any server in general is a good idea…

V14 Automatic Updates Not Occurring - Modules or System

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I can understand that but if you have hundreds of systems deployed the module updates can become impossible if they are done only manually.

I have an issue with freePBX dashboard

13 to 14 upgrade

V14 Automatic Updates Not Occurring - Modules or System

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You can vote for this issue, maybe it’ll become a priority with more votes. :slight_smile:

V14 Automatic Updates Not Occurring - Modules or System

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This has never worked. Broken module or not.

Looping a prerecorded page

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I recognize that it is in bad taste to reply to an old abandoned post, but this thread fits what I was looking for exactly.

I was not able to find adequate explanation in the manuals to help me figure out how to accomplish this using @cynjut’s recommendation (Custom Context + Miscellaneous Destination + Multicast Paging device extension). If someone wants to provide additional pointers to documentation for looping a prerecorded audio file in that manner, I would be very appreciative.

So instead this turned into my first FreePBX programming project. I’m posting the result here 1) in case it can help someone else 2) so that someone can tell me that there is a better way.

I set up a custom extension that:

  1. originates a call between a multicast device and an extension that quietly joins a conference bridge
  2. originates a call between an extension that loops the desired audio file and an extension that joins the same conference bridge
  3. connects the caller to the conference bridge muted to prevent feedback and as admin so that the paging stops when the caller disconnects

And it worked.

;;;
;;; Contents of extensions_custom.conf (abbreviated)
;;;

[from-internal-custom]
...
; include the context containing the emergency
; paging extensions
include => custom-emergency

[custom-emergency]
exten => *9001,1,Wait(1)
    same => n,Set(PAGE_LOOP_FILE_ID="all_clear")
    same => n,Goto(custom-page-loop,page_loop,1)

exten => **9002,1,Wait(1)
    same => n,Set(PAGE_LOOP_FILE_ID="severe_weather_drill")
    same => n,Goto(custom-page-loop,page_loop,1)

exten => *9002,1,Wait(1)
    same => n,Set(PAGE_LOOP_FILE_ID="severe_weather")
    same => n,Goto(custom-page-loop,page_loop,1)

[custom-page-loop]
exten => page_loop,1,NoOp()
    same => n,Answer()
    ; connect multicast devices to conference bridge
    same => n,Originate(MulticastRTP/basic/X.X.X.X:XXXXX//c(ulaw),exten,custom-page-loop,page_loop_enter,1)
    ; connect looping audio to conference bridge
    same => n,Originate(Local/page_loop_play_${PAGE_LOOP_FILE_ID}@custom-page-loop,exten,custom-page-loop,page_loop_enter_play,1)
    ; connect caller to conference bridge
    same => n,Set(CONFBRIDGE(user,template)=page_loop_admin)
    same => n,Set(CONFBRIDGE(bridge,template)=page_loop_bridge)
    same => n,Set(CONFBRIDGE(menu,template)=page_loop_admin_menu)
    same => n,ConfBridge(page_loop_1)
    same => n,Hangup()

; output to multicast devices
exten => page_loop_enter,1,NoOp()
    same => n,Answer()
    same => n,Set(CONFBRIDGE(user,template)=page_loop_user)
    same => n,Set(CONFBRIDGE(bridge,template)=page_loop_bridge)
    same => n,Set(CONFBRIDGE(menu,template)=page_loop_user_menu)
    same => n,ConfBridge(page_loop_1)
    same => n,Hangup()

; input from looping audio
exten => page_loop_enter_play,1,NoOp()
    same => n,Answer()
    same => n,Set(CONFBRIDGE(user,template)=page_loop_play)
    same => n,Set(CONFBRIDGE(bridge,template)=page_loop_bridge)
    same => n,Set(CONFBRIDGE(menu,template)=page_loop_user_menu)
    same => n,ConfBridge(page_loop_1)
    same => n,Hangup()

; looping audio with alert tones
exten => _page_loop_play_.,1,NoOp()
    same => n,Answer()
    same => n,Set(PAGE_LOOP_FILE_ID=${EXTEN:15})
    same => n(loop),Wait(1)
    same => n,Playback(/var/lib/asterisk/sounds/en/custom/3tones)
    same => n,Wait(1)
    same => n,Playback(/var/lib/asterisk/sounds/en/custom/${PAGE_LOOP_FILE_ID})
    same => n,Goto(loop)
    same => n,Hangup()


;;;
;;; Contents of confbridge_custom.conf
;;;

[page_loop_user]
type = user
quiet = yes
announce_user_count = no
wait_marked = no
end_marked = yes
dsp_drop_silence = yes
announce_join_leave = no
admin = no
marked = no
startmuted = yes

[page_loop_play]
type = user
quiet = yes
announce_user_count = no
wait_marked = no
end_marked = yes
dsp_drop_silence = yes
announce_join_leave = no
admin = no
marked = no
startmuted = no

[page_loop_admin]
type = user
quiet = yes
announce_user_count = no
wait_marked = no
end_marked = yes
dsp_drop_silence = yes
announce_join_leave = no
admin = yes
marked = yes
startmuted = yes

[page_loop_bridge]
type = bridge

[page_loop_user_menu]
type = menu
* = no_op

[page_loop_admin_menu]
type = menu
* = no_op

PJSIP and call files

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This topic was automatically closed 7 days after the last reply. New replies are no longer allowed.

Would it be possible to change PABX settings via CLI or changing files manually?

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Hello @dicko Many thanks for your valuable tips :smiley:
I’m sure they’ll be useful.

Sangoma FXO / FXS Card Won't Start + WebSocket Error

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Hello
Is there any other steps I can take before logging a ticket? Whom should I go to; Sangoma or FreePBX or are they both working off the same ticketing system?

Thanks in advance for any help that can be provided.

Michael

13 to 14 upgrade

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It was originally installed with the FreePbx iso… However, it seems like I’m running into a lot of issues like this.

Ring group and CDR reports

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Thanks for your reply. I can not write perl or php. I am not a programmer. It would be easier to have something ready, even paid program, to do so.
I don’t know how queues reporting works and if it does what I want. It would be nice to have a trial version for testing purposes for a few days.
Thank you all for your replies.

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