I’ve been experiencing this issue for over a year across dozens of servers with FPBX 15-16.

FPBX iCal calendar syncing is a mess when repeating events are set up in google calendar.

My biggest issue is that if an event repeats in any way, it is completely unreliable on if freepbx syncs following events. Some are missing, some are repeated extra days, some don’t actually trigger the time condition.

Has anyone experienced similar issues and if so, would you be willing to go and upvote my untouched yearlong bug report in github?

Here is a more detailed description of all the issues;

Multiple calendar-related bugs occur when syncing a Google Calendar with FreePBX (versions 15 and 16) via Remote iCal. The issues include:

1. Repetition Interval Errors
Events set to repeat more than every 1 or 2 weeks (e.g., every 3 weeks, every 4 weeks) incorrectly show up and behave as weekly events in FreePBX.

2. Multi-Day Event Time Handling
- Multi-day events that begin and end at the same time on different days default to starting at midnight in FreePBX, rather than displaying or respecting the actual set times.
- Changing start/end times by at least one minute apart fixes the display, indicating a parsing or validation issue when the times are identical.

3. BLF Hint “Stuck” Status
- Events repeating more than every 30 days display correctly and route calls as expected, but the BLF keys never revert to “unmatched” after the event ends. Toggling the feature code manually is required to reset the BLF state.

4. Missing Events with End Dates
- When an event repeats until a specific end date, some occurrences between the start and end may be missing or not displayed in FreePBX at all.

Additionally, we ran into a strange issue that manually running fwconsole job --run can cause calendars to refresh every second instead of following the configured interval. These issues appear consistently across multiple FreePBX instances, Asterisk versions, and browser environments.

Instant Sync:
https://github-production-user-asset-6210df.s3.amazonaws.com/119361408/403166283-c2b9683c-106f-477f-b1d8-f05792b68a44.mp4?X-Amz-Algorithm=AWS4-HMAC-SHA256&X-Amz-Credential=AKIAVCODYLSA53PQK4ZA%2F20250123%2Fus-east-1%2Fs3%2Faws4_request&X-Amz-Date=20250123T201945Z&X-Amz-Expires=300&X-Amz-Signature=ef0c9c5f0af09bdb392da1b0b817724ebe198496cc02c73912fe1d3c7824ec9e&X-Amz-SignedHeaders=host

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I need architecture & configuration advice. I’m currently supporting an environment that has a high inbound call volume on a FreePBX instance hosted in the United States. Latency, RTT, and others are becoming an issue with our users in the APAC region.

This post ended up being too long, so I’ve broken up my thoughts into a bullet list.

• SIPStation with ULAW Codecs
• APAC Users MOS of 1-3, high RTT
• AMI integrations on existing FreePBX (I have concerns regarding integration across multiple FreePBX instances - specifically, we make recordings public via randomized GUIDs for CRM linking)
• My goal: have a primary server handling configuration with it propagated to other instances (versus having to manager X configurations) - hub and spoke? (maybe? probably wrong term.)
• Heavy inbound call flow to Call Queues (with custom IVRs through AGI)
• Very frequent transfers for calls (all remote users), I don’t want to make users dial an extension with a prefix (the solution I’ve seen for multi-office setups with IAX trunks)
• Heavy reliance on call recordings
• Currently working on deploying dSIPRouter in APAC region (not sure if proxying the traffic would provide any tangible value)

Maybe IAX trunks are the solution here - I think I just need to talk it through with some people smarter then me and once I have an understanding of best practices in this scenario I can execute.

The core of the problem is call quality, happy to be told I’m over complicating things and need to start somewhere more tame & simple.

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How to display which extension no. Calling I have panasonic ns300 epbax analog betel m59 landline phone

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I have two systems: my FreePBX server (hosted in the cloud) and Bitrix24.
I created a SIP number in Bitrix24 so that calls are routed to FreePBX.
Then I created an outbound trunk in FreePBX.
I created a second inbound trunk so that Bitrix24 connects to FreePBX. When I redirect to this trunk, it should ring in Bitrix24.

The first trunk (outbound) works, but the second trunk (inbound) does not. The following error appears in the terminal:

[2025-01-24 16:39:47] WARNING[21288]: res_pjsip_registrar.c:1166 find_registrar_aor: AOR '' not found for endpoint '<ENDPOINT>' (<IP>)

When I remove the first trunk (outbound), the second trunk (inbound) works fine.

Note: For some reason, when I try to connect Bitrix24 to the second trunk (inbound), Asterisk shows it as if it were the first trunk (outbound). But since the first trunk is outbound, FreePBX did not create an endpoint in the pjsip.endpoint.conf file.

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Not sure if this is the correct place to post this. I am getting this message appear in the fail2ban logs while looking in the Asterisk Log Files. It appears every 2 seconds or so.

11063 [2025-01-24 15:37:45] SECURITY[3756] res_security_log.c: SecurityEvent=“SuccessfulAuth”,EventTV=“2025-01-24T15:37:45.395-0500”,Severity=“Informational”,Service=“AMI”,EventVersion=“1”,AccountID=“admin”,SessionID=“0x7f42e80022c8”,LocalAddress=“IPV4/TCP/0.0.0.0/5038”,RemoteAddress=“IPV4/TCP/127.0.0.1/43090”,UsingPassword=“0”,SessionTV=“2025-01-24T15:37:45.395-0500”
11064 [2025-01-24 15:37:48] SECURITY[3756] res_security_log.c: SecurityEvent=“SuccessfulAuth”,EventTV=“2025-01-24T15:37:48.469-0500”,Severity=“Informational”,Service=“AMI”,EventVersion=“1”,AccountID=“admin”,SessionID=“0x7f42f400d438”,LocalAddress=“IPV4/TCP/0.0.0.0/5038”,RemoteAddress=“IPV4/TCP/127.0.0.1/43096”,UsingPassword=“0”,SessionTV=“2025-01-24T15:37:48.469-0500”
11065 [2025-01-24 15:37:51] SECURITY[3756] res_security_log.c: SecurityEvent=“SuccessfulAuth”,EventTV=“2025-01-24T15:37:51.553-0500”,Severity=“Informational”,Service=“AMI”,EventVersion=“1”,AccountID=“admin”,SessionID=“0x7f42fc002068”,LocalAddress=“IPV4/TCP/0.0.0.0/5038”,RemoteAddress=“IPV4/TCP/127.0.0.1/43102”,UsingPassword=“0”,SessionTV=“2025-01-24T15:37:51.553-0500”
11066 [2025-01-24 15:37:54] SECURITY[3756] res_security_log.c: SecurityEvent=“SuccessfulAuth”,EventTV=“2025-01-24T15:37:54.619-0500”,Severity=“Informational”,Service=“AMI”,EventVersion=“1”,AccountID=“admin”,SessionID=“0x7f42fc002068”,LocalAddress=“IPV4/TCP/0.0.0.0/5038”,RemoteAddress=“IPV4/TCP/127.0.0.1/43108”,UsingPassword=“0”,SessionTV=“2025-01-24T15:37:54.619-0500”
11067 [2025-01-24 15:37:57] SECURITY[3756] res_security_log.c: SecurityEvent=“SuccessfulAuth”,EventTV=“2025-01-24T15:37:57.686-0500”,Severity=“Informational”,Service=“AMI”,EventVersion=“1”,AccountID=“admin”,SessionID=“0x7f42fc002068”,LocalAddress=“IPV4/TCP/0.0.0.0/5038”,RemoteAddress=“IPV4/TCP/127.0.0.1/43114”,UsingPassword=“0”,SessionTV=“2025-01-24T15:37:57.686-0500”
11068 [2025-01-24 15:38:00] SECURITY[3756] res_security_log.c: SecurityEvent=“SuccessfulAuth”,EventTV=“2025-01-24T15:38:00.752-0500”,Severity=“Informational”,Service=“AMI”,EventVersion=“1”,AccountID=“admin”,SessionID=“0x7f42fc002068”,LocalAddress=“IPV4/TCP/0.0.0.0/5038”,RemoteAddress=“IPV4/TCP/127.0.0.1/43120”,UsingPassword=“0”,SessionTV=“2025-01-24T15:38:00.752-0500”
11069 [2025-01-24 15:38:03] SECURITY[3756] res_security_log.c: SecurityEvent=“SuccessfulAuth”,EventTV=“2025-01-24T15:38:03.678-0500”,Severity=“Informational”,Service=“AMI”,EventVersion=“1”,AccountID=“admin”,SessionID=“0x7f42fc002068”,LocalAddress=“IPV4/TCP/0.0.0.0/5038”,RemoteAddress=“IPV4/TCP/127.0.0.1/43126”,UsingPassword=“0”,SessionTV=“2025-01-24T15:38:03.678-0500”
11070 [2025-01-24 15:38:03] SECURITY[3756] res_security_log.c: SecurityEvent=“SuccessfulAuth”,EventTV=“2025-01-24T15:38:03.822-0500”,Severity=“Informational”,Service=“AMI”,EventVersion=“1”,AccountID=“admin”,SessionID=“0x7f42fc002068”,LocalAddress=“IPV4/TCP/0.0.0.0/5038”,RemoteAddress=“IPV4/TCP/127.0.0.1/43132”,UsingPassword=“0”,SessionTV=“2025-01-24T15:38:03.822-0500”
11071 [2025-01-24 15:38:06] SECURITY[3756] res_security_log.c: SecurityEvent=“SuccessfulAuth”,EventTV=“2025-01-24T15:38:06.897-0500”,Severity=“Informational”,Service=“AMI”,EventVersion=“1”,AccountID=“admin”,SessionID=“0x7f42fc002068”,LocalAddress=“IPV4/TCP/0.0.0.0/5038”,RemoteAddress=“IPV4/TCP/127.0.0.1/43138”,UsingPassword=“0”,SessionTV=“2025-01-24T15:38:06.897-0500”

I am not sure where to start to look as the IP address referenced is the loopback IP and there is no user called admin on the PBX. This has been occurring for some time now.

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Hey guys, having an issue and not be able to login to Sangoma Chat on my mobile. With the same credentials I can login fine to UCP, SangomaConnect and etc. What should I look for?

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Welp, i hosed an upgrade from 15 to 16. I had to wipe it and start with 16 and now i will be doing a restore. This isnt my issue as to why i am writing. I will be doing this for 8 more PBXes. but before I do this, i assume it has to do with an out of date versionupgrade. What can i check to get this thing to update?

versionupgrade | 15.0.32 | Enabled; Not available online | Commercial | Sangoma

Thanks

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Hi,

My name is Malik Rihan, and I recently upgraded my FreePBX from version 11.13.0 to 18.16.0. On the old version, I had a Dinstar PRI line configured, which worked smoothly without any issues. However, after the upgrade, the SIP trunk configuration is not getting established, and the status shows as fault.

Here’s what I have already tried:

1. Checked and updated the Trunk Settings to simple, default settings for easy compatibility.
2. Verified the network and connectivity between the FreePBX server and the Dinstar PRI gateway. (Connectivity is happening)
3. Tested with basic configurations, but the SIP registration fails, and no calls are going through.
4. Firewall and everything disabled for test purpose still same issue
   I would appreciate any insights or suggestions to resolve this issue. Please let me know if additional logs or configurations are needed.
   Thank you for helping me in this in advance!

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So I have a AWS FreePBX v16 that’s been in production for a few years now. Running the AMI on an EC2 instance. With AWS FreePBX v17 available, the upgrade workflow requires launching the new v17 AMI and restoring the v16 backup over there. Since v17 is Debian-based and v16 is CentOS-based, a new AMI was required. I have all of those steps down-pat in terms of base FreePBX.

The one curveball I have is the v16 instance has Flash Operator Panel 2 installed and configured on it. A very helpful tool for receptionists at a couple of our sites. I haven’t really touched FOP2 in years. Just updating the group buttons to display/hide extensions and that’s about it.

Any suggestions on how to move FOP2 over to the v17 instance? Last time I ran into a FOP2 issue was a few years back. And vendor support really didn’t respond IIRC.

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Hello,

I am using FreePBX with Asterisk, and I am trying to disable the default prompts played before the beep in the voicemail application. Specifically, I want to bypass the following messages:

• if-correct-press
• digits/1

I only want to play my custom voicemail greeting (e.g., custom/segreteria-interno) followed directly by the beep (beep.alaw), without any intermediate prompts asking the caller to press a key.

What I’ve Tried:

1. Customized Dialplan in extensions_override_freepbx.conf:
   I tried overriding the dialplan for app-vmblast using the following code:

[app-vmblast]
exten => vmblast,1,NoOp(Skip confirmation messages)
exten => vmblast,n,Playback(custom/segreteria-interno)
exten => vmblast,n,Playback(beep)
exten => vmblast,n,Return()

However, this caused the call to hang up immediately, and no audio was played.
2. Modified Voicemail Settings:
I disabled review=yes, tempgreetwarn=yes, and other related settings in voicemail configuration files (e.g., voicemail.conf), but the default prompts still play.
3. File Overrides:
I considered replacing the audio files for the prompts (if-correct-press, digits/1) with empty files, but I would prefer a proper dialplan solution to keep these prompts intact for other use cases.

Logs:

Here’s an example log showing the prompts being played:

[2025-01-26 02:49:33] VERBOSE[154850][C-00000014] file.c: <PJSIP/TIM-00000013> Playing 'if-correct-press.alaw' (language 'it')
[2025-01-26 02:49:35] VERBOSE[154850][C-00000014] file.c: <PJSIP/TIM-00000013> Playing 'digits/1.alaw' (language 'it')
[2025-01-26 02:49:39] VERBOSE[154850][C-00000014] file.c: <PJSIP/TIM-00000013> Playing 'beep.alaw' (l

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Hello,

After a look at the documentation I can’t understand how permissions assigned to groups are merged to make the permission of users belonging to several groups.

For example, I can assign the user U to groups A (priority 0), B (priority 1), C (priority 2) and so on.

But it seems that my user U doesn’t get the permissions I assigned to group C.

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Using the Sipnetic (https://www.sipnetic.com) Android app, registers fine and can make and receive calls when connected to my local WIFI network, but from outside it won’t register.
Asterisk 22.1.0
FreePBX 17.0.19.23
RaspberryPi 5 8gb

No firewalls enabled just yet as this is preliminary testing. I have not include any logging (including “pjsip show registrations”) because that extension 1121 doesn’t show up anywhere.

Anyone have any suggestions? I’m going to guess ahead-of-time that it’s something really stupid that is staring me in the face, but I’m mentally exhausted and would really appreciate another perspective. Thank you.

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Hi everyone,

I’m setting up a FreePBX system integrated with a FRITZ!Box to handle incoming and outgoing calls. My goal is to play a welcome message and then route incoming calls to a single extension or all extensions connected to the FRITZ!Box (both IP and analog devices).

Current Setup:

1. I’ve configured a PJSIP trunk between FreePBX and the FRITZ!Box, and it successfully handles incoming calls.
2. I’ve created a welcome message using Announcements in FreePBX, which plays correctly when a call is received.
3. I can successfully route calls to an extension configured on FreePBX, but I cannot route calls to a single extension (e.g., **610) or to all extensions (e.g., **9) on the FRITZ!Box.

Question:

How can I configure FreePBX to route calls, after the welcome message, to a single extension or all extensions on the FRITZ!Box (e.g., by using the code **610 for a specific extension or **9 for all extensions)?

Current Configuration:

• FreePBX Version: 17
• Asterisk Version: 21.6.0
• SIP Trunk: Configured with the FRITZ!Box extension credentials.
• Welcome Announcement: Created via Announcements and linked to an Inbound Route.

What I’ve Tried So Far:

1. I configured a Misc Destination with the value **610 and another one with **9, but neither seems to work.
2. I verified that the FRITZ!Box accepts calls routed to individual extensions (**610, **620) and to the group (**9), but FreePBX doesn’t seem to route the call properly through the trunk.

Any advice or guidance on how to properly configure FreePBX to route calls to a single extension or all FRITZ!Box extensions would be greatly appreciated.

Thank you in advance for your help!

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This is the second time that the Asterisk crontab gets emptied somehow, so anything that is dependant on Cron jobs doesn’t work, like backups, calendar resync, jobs, etc.
I fixed it with redownloading and reinstalling all modules but this is getting tedious.

I’m running FreePBX Distro version 15, all modules updated.

Just in case, none of the server’s interfaces are exposed to the internet, apart from SIP and RTP ports forwarded to the FreePBX server.

Has anyone experienced this behaviour before?

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Hello All ,
I am about to install 2 new server with debian with FreePBX 17.
My question is ,
Does FreePBX 17 is fully supported on Debian ARM64.
I must say that I am likely to have all my linux server to migrate to ARM because of price & my green computing effort.

Thank you all
Koby Peleg Hen

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Hi everyone,

Iam having an issue with my FreePBX system where the GUI shows a completely white screen after I updated one of the modules. The dashboard worked perfectly before the update, but now the interface won’t load, and I cannot access any settings.

I have tried clearing my browser cache, using different browsers, rebooting the server, and running basic FreePBX commands like fwconsole reload, but nothing seems to help.

Has anyone else experienced a similar issue after an update, or does anyone know how to fix this? Any advice would be greatly appreciated!

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I also got an email saying that Microsoft Permanently Disabling SMTP AUTH
I am not sure but I suspect this might be the issue.
Any idea what the settings need to be to get this working again?

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I have an odd UCP issue. My end user has been using UCP for Call History monitoring of 5 extensions for months with no issues. Recently the call history widgets stopped showing data and I get a “There was an error.See console log for more details”. The Call history widgets say “Loading, please wait…” except for one that works. The one that works is the newest extension added to the system. I have created another test user and given them access to the call history of the same extensions and get the same error. Here is the JS error that Google Chrome is spitting out. I am on FreePbx 16.0.40.11 and am updated to the newest updates. Server is cloud hosted as well. Please let me know if more information is needed. Any help would be greatly appreciated.

Uncaught TypeError: Cannot read properties of undefined (reading ‘transcriptionURL’)
at Object.formatActions (jsphp_7da9f1371eacc6a4f2d4bb483e6377c7.js?load_version=v16.0.38.2:392:24)
at h (jsphpg_b76ec9325a1a5caca04f28721d9cc414.js?load_version=v16.0.38.2:548:1213)
at String. (jsphpg_b76ec9325a1a5caca04f28721d9cc414.js?load_version=v16.0.38.2:548:30129)
at Function.each (jquery-3.1.1.min.js?load_version=v16.0.38.2:2:2815)
at p.initBody (jsphpg_b76ec9325a1a5caca04f28721d9cc414.js?load_version=v16.0.38.2:548:29331)
at p.load (jsphpg_b76ec9325a1a5caca04f28721d9cc414.js?load_version=v16.0.38.2:549:10368)
at f.load (jsphpg_b76ec9325a1a5caca04f28721d9cc414.js?load_version=v16.0.38.2:567:3630)
at Object.success (jsphpg_b76ec9325a1a5caca04f28721d9cc414.js?load_version=v16.0.38.2:549:3241)
at i (jquery-3.1.1.min.js?load_version=v16.0.38.2:2:27983)
at Object.fireWith [as resolveWith] (jquery-3.1.1.min.js?load_version=v16.0.38.2:2:28749)
formatActions @ jsphp_7da9f1371eacc6a4f2d4bb483e6377c7.js?load_version=v16.0.38.2:392
h @ jsphpg_b76ec9325a1a5caca04f28721d9cc414.js?load_version=v16.0.38.2:548
(anonymous) @ jsphpg_b76ec9325a1a5caca04f28721d9cc414.js?load_version=v16.0.38.2:548
each @ jquery-3.1.1.min.js?load_version=v16.0.38.2:2
p.initBody @ jsphpg_b76ec9325a1a5caca04f28721d9cc414.js?load_version=v16.0.38.2:548
p.load @ jsphpg_b76ec9325a1a5caca04f28721d9cc414.js?load_version=v16.0.38.2:549
f.load @ jsphpg_b76ec9325a1a5caca04f28721d9cc414.js?load_version=v16.0.38.2:567
success @ jsphpg_b76ec9325a1a5caca04f28721d9cc414.js?load_version=v16.0.38.2:549
i @ jquery-3.1.1.min.js?load_version=v16.0.38.2:2
fireWith @ jquery-3.1.1.min.js?load_version=v16.0.38.2:2
A @ jquery-3.1.1.min.js?load_version=v16.0.38.2:4
(anonymous) @ jquery-3.1.1.min.js?load_version=v16.0.38.2:4
load
send @ jquery-3.1.1.min.js?load_version=v16.0.38.2:4
ajax @ jquery-3.1.1.min.js?load_version=v16.0.38.2:4
jQuery.ajax @ jsphpg_b76ec9325a1a5caca04f28721d9cc414.js?load_version=v16.0.38.2:21
p.initServer @ jsphpg_b76ec9325a1a5caca04f28721d9cc414.js?load_version=v16.0.38.2:549
p.refresh @ jsphpg_b76ec9325a1a5caca04f28721d9cc414.js?load_version=v16.0.38.2:549
e @ jquery-3.1.1.min.js?load_version=v16.0.38.2:2
dispatch @ jquery-3.1.1.min.js?load_version=v16.0.38.2:3
q.handle @ jquery-3.1.1.min.js?load_version=v16.0.38.2:3Understand this errorAI

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Thank you to @btbutts and @danb35 for keeping this going. I’m here to update this as per Brian’s question about scripting the sysadmin module to update the cert along with certman. I can confirm that it IS possible. the command is as follows:

fwconsole sa ihc default

I had a slightly different use-case as I have a different server creating wildcard certs and I share those among my several different servers using NFS. My script renews the cert by comparing the cert (hourly through cron.hourly folder) and then copying the new cert and updating the various modules. The script is as follows:

#!/bin/bash

LOGFILE="/var/log/asterisk/letsencrypt_update.log"
CERT_DIR="/path/to/nfs/mounted/cert"
ASTERISK_KEY_DIR="/etc/asterisk/keys"
SSL_PRIVATE_DIR="/etc/ssl/private"
DOMAIN="domain.local"

# Log function
log_message() {
  local MESSAGE="$1"
  echo "$(date '+%Y-%m-%d %H:%M:%S') - $MESSAGE" >> "$LOGFILE"
  logger -t "certificate_update" "$MESSAGE"
}

log_message "Starting certificate update process."

# Check if mounted certificates are newer
if [[ "$CERT_DIR/fullchain.pem" -nt "$ASTERISK_KEY_DIR/$DOMAIN.crt" ]]; then
  log_message "New certificates found. Updating..."

  # Backup existing certificates
  cd "$SSL_PRIVATE_DIR" || exit 1
  tar -cvf cert-backup_$(date +%Y-%m-%d_%H.%M.%S).tar asterisk* 2>/dev/null || true

  # Convert and prepare certificates
  log_message "Converting certificates to required formats..."

  # Copy new certs to SSL private directory
  cp "$CERT_DIR/fullchain.pem" "$SSL_PRIVATE_DIR/asterisk19-pub.crt"
  cp "$CERT_DIR/privkey.pem" "$SSL_PRIVATE_DIR/asterisk19-priv.key"

  # Convert to PKCS12 and PEM formats
  openssl pkcs12 -export -in "$SSL_PRIVATE_DIR/asterisk19-pub.crt" \
    -inkey "$SSL_PRIVATE_DIR/asterisk19-priv.key" \
    -out "$SSL_PRIVATE_DIR/asterisk19.p12" \
    -name freepbx -password pass:freepbx-lets-encrypt

  # Convert private key to RSA format
  openssl pkcs8 -topk8 -nocrypt \
    -in "$SSL_PRIVATE_DIR/asterisk19-priv.key" \
    -out "$SSL_PRIVATE_DIR/asterisk19-priv_rsa.key"

  # Convert PKCS12 to PEM
  openssl pkcs12 -in "$SSL_PRIVATE_DIR/asterisk19.p12" \
    -out "$SSL_PRIVATE_DIR/asterisk19.pem" \
    -nodes -password pass:freepbx-lets-encrypt

  # Set permissions on SSL private directory
  chown root:root "$SSL_PRIVATE_DIR"/*
  chmod 664 "$SSL_PRIVATE_DIR"/*

  # Copy to FreePBX directory
  cd "$ASTERISK_KEY_DIR" || exit 1
  tar -cvf cert-backup_$(date +%Y-%m-%d_%H.%M.%S).tar * 2>/dev/null || true

  # Copy converted certificates
  cp "$SSL_PRIVATE_DIR/asterisk19-priv_rsa.key" "$ASTERISK_KEY_DIR/$DOMAIN.key"
  cp "$SSL_PRIVATE_DIR/asterisk19-pub.crt" "$ASTERISK_KEY_DIR/$DOMAIN.crt"
  cp "$SSL_PRIVATE_DIR/asterisk19.pem" "$ASTERISK_KEY_DIR/$DOMAIN.pem"

  # Set permissions
  chown asterisk:asterisk "$ASTERISK_KEY_DIR/$DOMAIN".*
  chmod 640 "$ASTERISK_KEY_DIR/$DOMAIN".*

  # Import to FreePBX Certman Module
  log_message "Importing certificates to FreePBX..."
  fwconsole certificates --import
  fwconsole certificates --default=0
  # Update SysAdmin Module
  fwconsole sa ihc default

  # Reload FreePBX
  log_message "Reloading FreePBX configuration..."
  fwconsole reload

  log_message "New certificates imported, set as default, and deployed successfully."
else
  log_message "No new certificates to import."
fi

log_message "Certificate update process completed."

Feel free to update this for your own use-case or just let me know if you think I’m doing something wrong here. It sure would be nice if Sangoma actually updated Certman to do this natively.

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Hello everybody

I have freepbx server and i need to enable sms chatting in my server what’s the steps that i need to enable that?

Thank you.

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