Issues missing PJSIP
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Ring multiple extensions at once
I use more Queues (set to ringall) than Ring Groups. Queues have a lot more options as far as caller and agent interaction. The feature I like mostly is the agent extensions can have the option to sign in and out. So for your example, if someone calls Department A, and none of the extensions are signed in, you can have it do something specific, like go directly to voicemail or return to an IVR after an announcement.
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Error when I Apply Config
I had this years back. I’m thinking I had to run two commands from CLI.
fwconsole chown
fwconsole reload
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Error when I Apply Config
Hi Kevin,
Thank you for your reply. I tried that now and I got a bunch of errors. Here is the output of both commands:
[root@ip-172-31-32-120 ~]# fwconsole chown
Taking too long? Customize the chown command, See http://wiki.freepbx.org/display/FOP/FreePBX+Chown+Conf
Setting Permissions...
In Chown.class.php line 167:
Undefined index: byconfig
chown [-f|--file FILE] [--] [<args>]...
[root@ip-172-31-32-120 ~]# fwconsole reload
Reloading FreePBX
Error(s) have occured, the following is the retrieve_conf output:
exit: 1
Unable to continue. Only variables should be assigned by reference in /var/www/html/admin/modules/core/functions.inc.php on line 5454
#0 /var/www/html/admin/modules/core/functions.inc.php(5454): Whoops\Run->handleError(2048, 'Only variables ...', '/var/www/html/a...', 5454, Array)
#1 /var/www/html/admin/modules/core/functions.inc.php(2239): general_generate_indications()
#2 /var/www/html/admin/libraries/BMO/DialplanHooks.class.php(95): core_do_get_config('asterisk')
#3 /var/lib/asterisk/bin/retrieve_conf(860): FreePBX\DialplanHooks->processHooks('asterisk', Array)
#4 {main}
Any idea what is going on?
Best regards,
Aécio
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Issues missing PJSIP
Yep found that. So stopped looking that direction. Still can’t figure out why it’s not loading
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PJSIP Error with NO PJSIP configured
I had seen that after the PJSIP errors and maybe did a bad thing and assume that was the result of the PJSIP issue. I tried to remove all forwards, DND’s etc, but still happens.
I may simply delete and recreate the extension to see if I can force a change.
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Issues missing PJSIP
try an iota of pjsip maybe
module load res_pjsip
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External Voicemail Server
Alright, so it seems i’ve got it resolved. I was able to track down the change to the voicemail app that invalidated the old config from that original post. Here’s the working config I put together just in case anyone is interested:
On the main PBX I added the following to my extensions_custom.conf file:
[macro-vm]
exten => s-NOANSWER,1,Dial(IAX2/Mailbox/${ARG1}u)
exten => s-NOANSWER,2,Goto(default,s,1)
exten => s-BUSY,1,Dial(IAX2/Mailbox/${ARG1}b)
exten => s-BUSY,2,Goto(default,s,1)
and I added the following to my extensions_custom.conf file:
[from-internal]
exten => _XXXXb,1,Voicemail(${EXTEN:0:-1}@default,b)
exten => _XXXXu,1,Voicemail(${EXTEN:0:-1}@default,u)
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One way audio
I think it was unhappy with two localnet settings so I used one CIDR notation that covered both subnets.
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Undo activation to create master image
I want to completely remove the deployment ID so I can re-activate the system as a new deployment and not interfere with the previous deployment. If not, I can’t purchase any commercial modules for the new system.
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One way audio
If you do that then the subnet’s broadcasts won’t work 
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Undo activation to create master image
After you restore your image on a different machine you get a new deployment id and freepbx is not activated anymore.
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Implementing and Testing Naf's GVsip Google Voice Sip Ubuntu 18.04 Freepbx
Is there a ticket on the issue tracker we can follow?
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FreePBX 14 - Cannot update Let's Encrypt certificate!
I don’t recall seeing two // how come it adds it?
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Bulk/Global DPMA Configuration
Never touched these phones, are they not configurable via EPM?
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Asterisk and/or FreePBX will crash several times a week
There’s newer Asterisk versions, try upgrading.
Also, restart asterisk by running
fwconsole stop
fwconsole start↧
One way audio
localnet= impacts network interface configurations? Routing must be working properly otherwise I wouldn’t have had two way audio on my last test call, which needs to cross a router boundary.
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FreePBX 14 - Cannot update Let's Encrypt certificate!
Two // doesn’t matter. It will resolve to one.
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Implementing and Testing Naf's GVsip Google Voice Sip Ubuntu 18.04 Freepbx
No there is not.
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One way audio
It does not, only pjsip and chan_sip are affected and these stacks do not use broadcasts.
I don’t understand. It is working properly now (both incoming and outgoing), or do only some calls have two way audio?
If the problem is solved, please post enough detail to help other readers of this thread.
If you are still having trouble, answer the questions asked earlier.
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