@jtharvey wrote:

Though I know every installation is unique, does anyone have – or know where I can get my hands on a VoIP onboarding/provisioning checklist - at least a starter?

Thanks in advance,

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@TecScott wrote:

Upgrading from 13 to 14 and running into a lot of issues.

1. Initially asterisk did not automatically start following the server starting, so I had to manually start the service
2. The ‘manager’ password mismatched which resulted in FreePBX being unable to communicate with Asterisk (as a result ‘Can not connect to Asterisk’ was shown on PBX)
3. Details for all PJSIP extensions have disappeared (ext + name is still recorded, but Voicemail settings, Call Waiting, DND, etc details have all disappeared)
4. My log is plagued with errors such as;

[2019-05-01 09:03:18] ERROR[9295] config_options.c: Unable to load config file ‘acl.conf’
[2019-05-01 09:03:18] WARNING[9295] named_acl.c: Could not reload ACL config
[2019-05-01 09:03:18] WARNING[9295] pbx.c: Extension ‘s’ priority 13 in ‘macro-user-logon’, label ‘gotpass’ already in use at priority 9
[2019-05-01 09:03:18] ERROR[9295] pbx.c: You have to be kidding-- add exten ‘’ to context clean? Figure out a name and call me back. Action ignored.
[2019-05-01 09:03:18] WARNING[9295] pbx_config.c: Unable to register extension at line 7976 of /etc/asterisk/extensions_additional.conf
[2019-05-01 09:03:18] ERROR[9295] pbx.c: You have to be kidding-- add exten ‘’ to context clean? Figure out a name and call me back. Action ignored.
[2019-05-01 09:03:18] WARNING[9295] pbx_config.c: Unable to register extension at line 7978 of /etc/asterisk/extensions_additional.conf
[2019-05-01 09:03:18] ERROR[9295] pbx.c: You have to be kidding-- add exten ‘’ to context mini-bar? Figure out a name and call me back. Action ignored.
[2019-05-01 09:03:18] WARNING[9295] pbx_config.c: Unable to register extension at line 8004 of /etc/asterisk/extensions_additional.conf
[2019-05-01 09:03:18] ERROR[9295] pbx.c: You have to be kidding-- add exten ‘’ to context housekeeping-service? Figure out a name and call me back. Action ignored.
[2019-05-01 09:03:18] WARNING[9295] pbx.c: Context ‘from-did-direct’ tries to include nonexistent context ‘ext-findmefollow’
[2019-05-01 09:03:18] WARNING[9295] pbx.c: Context ‘from-internal-xfer’ tries to include nonexistent context ‘from-internal-custom’
[2019-05-01 09:03:18] WARNING[9295] pbx.c: Context ‘from-internal-noxfer’ tries to include nonexistent context ‘from-internal-noxfer-custom’
[2019-05-01 09:03:18] ERROR[9295] res_sorcery_config.c: Unable to load config file ‘pjproject.conf’
[2019-05-01 09:03:18] VERBOSE[9295] loader.c: Reloading module ‘res_pjsip.so’ (Basic SIP resource)
[2019-05-01 09:03:18] ERROR[8783] res_pjsip_config_wizard.c: Unable to load config file ‘pjsip_wizard.conf’
[2019-05-01 09:03:18] NOTICE[8783] sorcery.c: Type ‘system’ is not reloadable, maintaining previous values
[2019-05-01 09:03:18] WARNING[9295] app_voicemail.c: maxsilence should be less than minsecs or you may get empty messages
[2019-05-01 09:03:18] NOTICE[9295] app_queue.c: No queuerules.conf file found, queues will not follow penalty rules

Prior to the upgrade I see no such messages.

I’ve also apparently lost ‘Asterisk Info’…

Any ideas what’s gone so terribly wrong? Extensions are obviously not registering likely due to the errors above and all the info missing for each extension.

Would a fresh install and FreePBX migration be a tidier way of migrating?

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@slip_cougan wrote:

Greetings Community,
I’m new to FreePBX/Asterisk and will have a few slightly more technical challenges to query later.

I’ve set up FreePBX on a Raspberry Pi for my home system. This is more of an exercise for me to become familiar with this technology as well as ongoing technical hobbies.

My system currently consists of:
4 x Snom 821
Various mobile devices running Sipnetic softphone app (which I really like)
X-Lite softphone running on MacBook Pro
3 x Crestron TSW-752 touchpanels with built-in SIP/Rava (integrated into my Crestron automation system)
1 x Linksys SPA3102 trunk (not currently working - I think I have a duff unit but will cover this in a separate post)

I have set this all up and I’ve found it pretty straightforward. All phones work and the system performs really well with one exception - I cannot get the Snom’s to dial a page group reliably. All other devices on the system work flawlessly.

The symptom:

• Dial page group (3012)
• Snom shows ‘Calling 3012’ but no ring tone
• After about a minute the Snom indicates ‘Network Failure 3012’
• FreePBX logs show no activity at all, like the request never even gets to FreePBX
• Dialling all other extensions work as expected.

If I reboot the Snom’s one of them will dial the ring group once and fail on all subsequent attempts.
The other 3 Snom’s will never dial.

This is looking like an issue with the Snom’s to me, does anyone have any suggestions how I may rectify this?

Many thanks

-slip

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@slip_cougan wrote:

Hi Guys
So until my local UK telco can provide VoIP trunks I’d like to try and get a Linksys SPA3102 working.
I’ve followed this excellent guide:
http://www.aoakley.com/articles/2008-01-08.php

Connection to the PSTN is using a 4-wire straight through BT to RJ-11 cable salvaged from a working BT phone. Cable is fully tested and there are no issues there.
Line voltage measures 49.6V on-hook.
However I cannot get the SPA to work (even in bypass mode with power off)

Logging into the /admin/voice/advanced info page I see the following:
Line Voltage: 4 (V)
Registration State: Not Registered

I would expect to see a line voltage ~50V

Line impedance is set:
FXS & FXO Port Impedance: 270+750 150n

Do I have a duff SPA?

Thanks
-slip

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@fleex2017 wrote:

we need any solution missed calls or abandoned class

send email or callback automatic

???

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@tfcarlin wrote:

Is there a way to have a call stop ringing and immediately transfer to VM when any extension in a Ring Group presses a soft button defined as Decline/603? Is there a different strategy to implement (e.g. blind transfer?).

Thanks!

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@GSnover wrote:

I don’t want to put the ID here for security reasons, but here is what activation on the box shows:

image.png1058×207 22.2 KB

And here is what the portal shows for the same Deployment ID:

image.png1803×253 23.1 KB

Same thing on the other box:

image.png1061×182 16.6 KB

Sangoma Portal:

image.png1838×248 22.2 KB

Why all the sudden did this go away? How do I get it back?

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@venizia03 wrote:

Hello!

I have freepbx 14.0.11. I needed this morning to add a new user. I so used User management to do so. When I clicked on password field or if I try to submit (when login name field is populated), the page crashed. I can update existing user. I tried to find an error in error log but without success. Any idea on what could be the reason of this crash? is there a way to create user with command line?

Thx in advance!

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@rodriguesbeast wrote:

Anyone experienced this issue already?

Webpage error:

Fatal error : Allowed memory size of 134217728 bytes exhausted (tried to allocate 71 bytes) in /var/www/html/recordings/modules/VmX.module on line 0

More details;
FreePBX 2.11.0.43
3 SIP Channels
2 Trunks
100 SIP Peers
VM Server

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@Basildane wrote:

I know this is an Asterisk issue, but I hope someone here has some tips for me to check.
I have a system with excellent audio on calls, however recordings and paging are choppy - consistently.

I do have a digium card installed for timing.
System load is nearly zero. The choppyness occurs even with just one call.

It’s Asterisk 16 and FPBX 14.0.12.2.

What should I check?

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@Smitty6504 wrote:

We are using Freepbx, Avaya, and Skype for business. They all tie into a Sonus GW. We have four digit dialing between all systems working great. I have created an IVR on Freepbx and i would like to be able to dial by extension for local and remote extensions. Local extension (6xxx) dialing is working fine, but having a hard time figuring out how to get the system to dial my 4xxx extentions that live on my SFB and 8xxx on my Avaya system. Is this possible to do and if so can you point me in the direction to get this working?

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@shivansps wrote:

The problem is simple, when i call from a external extension to any of my internal extensions, the call will always end at 32 seconds, BUT if i call from my internal extension to my external extension everything is fine.
On top of this i also have a sip trunk that works whiout issues (on my internal extensions).

This is a call started from my external extension (801)to my internal extension 300

ptqFfXu.png1360×856

As you can see it saying “invite from sip 801@my external pbx dns and port” this is definately wrong. If i call from internal to external extension the correct sip 801@external extension ip" is used and call is not cut.

And im not sure what im doing wrong, on my sip settings i have my dns on the external adress, and my local networks setup correctly.
On my router i have both the non-standart sip port and the UDP port range forwarded to my pbx.

I need help because i run out of ideas. Not sure what to do.

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@fer_18 wrote:

Hello guys.
I’m new here, I researched before but found nothing that could help me.
I’m trying to set up Jolt Select Dial PBX. I edited the call.php file, I can make calls between internal extensions, but when trying to make calls to a stern number I get the message that the number is incorrect.
The report follows:

[2019-05-03 11:29:51] VERBOSE[24681] dial.c: Called 210
[2019-05-03 11:29:51] VERBOSE[46821] netsock2.c: Using SIP RTP Audio TOS bits 184
[2019-05-03 11:29:51] VERBOSE[46821] netsock2.c: Using SIP RTP Audio TOS bits 184 in TCLASS field.
[2019-05-03 11:29:51] VERBOSE[46821] netsock2.c: Using SIP RTP Audio CoS mark 5
[2019-05-03 11:29:51] VERBOSE[24681] dial.c: PJSIP/210-00000368 is ringing
[2019-05-03 11:29:51] VERBOSE[24681] dial.c: PJSIP/210-00000368 is ringing
[2019-05-03 11:29:54] VERBOSE[24681] dial.c: PJSIP/210-00000368 answered
[2019-05-03 11:29:54] VERBOSE[24681][C-00000162] pbx.c: Executing [0971442523@from-internal:1] Macro("PJSIP/210-00000368", "user-callerid,LIMIT,EXTERNAL,") in new stack
[2019-05-03 11:29:54] VERBOSE[24681][C-00000162] pbx.c: Executing [s@macro-user-callerid:1] Set("PJSIP/210-00000368", "TOUCH_MONITOR=1556893791.1003") in new stack
[2019-05-03 11:29:54] VERBOSE[24681][C-00000162] pbx.c: Executing [s@macro-user-callerid:2] Set("PJSIP/210-00000368", "AMPUSER=0971442523") in new stack
[2019-05-03 11:29:54] VERBOSE[24681][C-00000162] pbx.c: Executing [s@macro-user-callerid:3] GotoIf("PJSIP/210-00000368", "0?report") in new stack
[2019-05-03 11:29:54] VERBOSE[24681][C-00000162] pbx.c: Executing [s@macro-user-callerid:4] ExecIf("PJSIP/210-00000368", "1?Set(REALCALLERIDNUM=0971442523)") in new stack
[2019-05-03 11:29:54] VERBOSE[24681][C-00000162] pbx.c: Executing [s@macro-user-callerid:5] Set("PJSIP/210-00000368", "AMPUSER=") in new stack
[2019-05-03 11:29:54] VERBOSE[24681][C-00000162] pbx.c: Executing [s@macro-user-callerid:6] GotoIf("PJSIP/210-00000368", "0?limit") in new stack
[2019-05-03 11:29:54] VERBOSE[24681][C-00000162] pbx.c: Executing [s@macro-user-callerid:7] Set("PJSIP/210-00000368", "AMPUSERCIDNAME=") in new stack
[2019-05-03 11:29:54] VERBOSE[24681][C-00000162] pbx.c: Executing [s@macro-user-callerid:8] ExecIf("PJSIP/210-00000368", "0?Set(__CIDMASQUERADING=TRUE)") in new stack
[2019-05-03 11:29:54] VERBOSE[24681][C-00000162] pbx.c: Executing [s@macro-user-callerid:9] GotoIf("PJSIP/210-00000368", "1?report") in new stack
[2019-05-03 11:29:54] VERBOSE[24681][C-00000162] pbx_builtins.c: Goto (macro-user-callerid,s,16)
[2019-05-03 11:29:54] VERBOSE[24681][C-00000162] pbx.c: Executing [s@macro-user-callerid:16] NoOp("PJSIP/210-00000368", "Macro Depth is 1") in new stack
[2019-05-03 11:29:54] VERBOSE[24681][C-00000162] pbx.c: Executing [s@macro-user-callerid:17] GotoIf("PJSIP/210-00000368", "1?report2:macroerror") in new stack
[2019-05-03 11:29:54] VERBOSE[24681][C-00000162] pbx_builtins.c: Goto (macro-user-callerid,s,18)
[2019-05-03 11:29:54] VERBOSE[24681][C-00000162] pbx.c: Executing [s@macro-user-callerid:18] GotoIf("PJSIP/210-00000368", "1?continue") in new stack
[2019-05-03 11:29:54] VERBOSE[24681][C-00000162] pbx_builtins.c: Goto (macro-user-callerid,s,37)
[2019-05-03 11:29:54] VERBOSE[24681][C-00000162] pbx.c: Executing [s@macro-user-callerid:37] Set("PJSIP/210-00000368", "CALLERID(number)=0971442523") in new stack
[2019-05-03 11:29:54] VERBOSE[24681][C-00000162] pbx.c: Executing [s@macro-user-callerid:38] Set("PJSIP/210-00000368", "CALLERID(name)=Web Call") in new stack
[2019-05-03 11:29:54] VERBOSE[24681][C-00000162] pbx.c: Executing [s@macro-user-callerid:39] GotoIf("PJSIP/210-00000368", "0?cnum") in new stack
[2019-05-03 11:29:54] VERBOSE[24681][C-00000162] pbx.c: Executing [s@macro-user-callerid:40] Set("PJSIP/210-00000368", "CDR(cnam)=Web Call") in new stack
[2019-05-03 11:29:54] VERBOSE[24681][C-00000162] pbx.c: Executing [s@macro-user-callerid:41] Set("PJSIP/210-00000368", "CDR(cnum)=0971442523") in new stack
[2019-05-03 11:29:54] VERBOSE[24681][C-00000162] pbx.c: Executing [s@macro-user-callerid:42] Set("PJSIP/210-00000368", "CHANNEL(language)=pt-br") in new stack
[2019-05-03 11:29:54] VERBOSE[24681][C-00000162] pbx.c: Executing [0971442523@from-internal:2] NoCDR("PJSIP/210-00000368", "") in new stack
[2019-05-03 11:29:54] VERBOSE[24681][C-00000162] pbx.c: Executing [0971442523@from-internal:3] Progress("PJSIP/210-00000368", "") in new stack
[2019-05-03 11:29:54] VERBOSE[24681][C-00000162] pbx.c: Executing [0971442523@from-internal:4] Wait("PJSIP/210-00000368", "1") in new stack
[2019-05-03 11:29:55] VERBOSE[24681][C-00000162] pbx.c: Executing [0971442523@from-internal:5] Playback("PJSIP/210-00000368", "silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer") in new stack
[2019-05-03 11:29:55] VERBOSE[24681][C-00000162] file.c: <PJSIP/210-00000368> Playing 'silence/1.slin16' (language 'pt-br')
[2019-05-03 11:29:56] VERBOSE[24681][C-00000162] file.c: <PJSIP/210-00000368> Playing 'cannot-complete-as-dialed.slin16' (language 'pt-br')

If anyone can help me, I’ll be very grateful.
Sorry for my english, I’m using google translator, lol.

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@ozarktech wrote:

Have a pbxact system on 14.0.11 Firmware 12.7.6-1904-1.sng7

image.png1872×938 76.1 KB

Everytime I click on UCP, I get the Whoops \ Exception \ ErrorException Maximum execution time of 30 seconds exceeded on “/var/ww/html/admin/modules/ucp/htdocs/vendor/tedivm/jshrink/src/JShrink/Minifier.php The spinning gear is in the middle of the page. I’ve switched versions of asterisk and it makes no difference. System is up to date. What is making it timeout?

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@markprotec wrote:

I am utilizing an OpenVPN server and client configuration to connect a remote computer to my network. The OpenVPN server is not the integrated version on the PBXact appliance. The server is an additional appliance. I have a PBXact 25 appliance. The local network at the office is 192.168.1.x, the VPN network is 128.10.0.x. I am able to connect to all other resources at the office from the remote machine through the VPN. However, I am not able to access any resources on the PBXact appliance from the remote network. I have put the VPN subnet into the firewall, responsive firewall, intrusion detection and asterisk SIP network. All without any desired change. Is there any where else that I can look to figure out why traffic is blocked coming from the additional local subnet?

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@Smurfturf wrote:

I’m currently running Freepbx distro (14) and I can’t seem to get past the following error when I run yum update:

Transaction check error:
file /boot/efi/EFI/centos from install of grub2-efi-x64-1:2.02-0.76.el7.centos.1.x86_64 conflicts with file from package grub2-common-1:2.02-0.65.el7.centos.2.noarch

I’ve run yum clean several times and I can’t seem to get past it. Any advice would be appreciated.

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@rohitg76 wrote:

This is on Asterisk 13 and FreePBX 13.

There are two errors I have started facing, likely after upgrading to Voicemail 13.0.59.3 (I am not sure and it may just be coincidence that there is an aliases related fix in 13.0.59.2. Not sure if it broke some compatibility)

The behaviour gets exposed when after upgrade to Voicemail 13.0.59.3, I try to do one of the following:

1) After importing extensions, and then try to export the extensions csv file via Bulk Handler. I get a Whoops\Exception “Cannot unset string offsets” (picture enclosed) in voicemail.class.php. On closer look I find that in voicemail.conf, under general settings, this new line is getting added post the upgrade “aliasescontext=pbxaliases”, and all extensions being added as “299@device=299@default” under a new context [pbxaliases]. If I delete the context and the newly added line “aliasescontext=pbxaliases”, then I am able to successfully export without exception error.

FreePBX-voicemail-exception-1.jpg1280×643 133 KB

2) Remove one or more extensions from Extensions page, I get Whoops\Exception undefined variable var (picture enclosed). Unable to work around this one - when I am required to bulk remove say 100 extensions. However, if I edit the extension and then click Delete from the edit page, then it removes the extension without throwing any exception.

FreePBX-voicemail-exception-2.jpg1280×277 112 KB

After two days of debugging, I am yet to find the root cause though it does seem to point towards the voicemail module and the handling of aliases. Looked at app_voicemail.c but not very sure how the aliasescontext is being handled and if anything has changed recently.

Here’s what I am trying to do - please review and point if I am doing or expecting something incorrect.

We ship our box with predefined extensions and users 200-299. Some clients want a different numbering scheme, in which case, we export out the extensions csv, then we delete 2XX series, then import back with new numbering scheme. This was working as expected until we recently upgraded about 39 packages, including the Voicemail 13.0.59.3.

Appreciate any help please. Thanks.

Regards,
Rohit

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@andy_woolford wrote:

For the past couple of days my access and error logs are full of hacking attempts. I use a couple of non-standard ports for http and https interface, but it looks like the hacker has found these and is blasting away at them.

Having had a previous PBX 14 broken into, I am anxious to stop these. Obviously blocking the IP will have little effect, but I had thought the “intrusion detection” (Fail2Ban) would have identified and blocked these. However, the Fail2Ban blocked list is empty.

Any suggestions?

Many thanks

Andy Woolford

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@mvogel4949 wrote:

I feel like at one point there was a zap barge feature code 888. Is that now deprecated?

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@ptsung wrote:

Hi,

I am having an issue using Attended Transfer. I am running PBX Firmware 10.13.66-22. The endpoints are Yealink T23G and T27G. When I try to use the TRAN button on the phones, it parks the active call, then I can dial the new number and connect with the third party. Then I would hit the TRAN button again, but it will just drop both calls.

Thanks in advance for any help.

Paul

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