I received this error after running the installer script for FreePBX 17. I have another install on another server running and I am waiting to see if it fails. Is anyone else having this issue? We have done about six installs successfully so far, so this is new to us.

SangomaConnect Server is not running
SangomaConnect module installed but not activated. Please buy a license and activate the module
Generating CSS…Done
Module sangomaconnect version 17.0.1.35 successfully installed
Updating Hooks…Done
Chowning directories…Done
Verifying GPG…
Done
Checking sangomacrm…
Signature Invalid
Refreshing sangomacrm
Starting module download from https://mirror.freepbx.org/modules/packages/sangomacrm/8.2/sangomacrm-17.0.1.16.tgz.gpg …
Processing
Downloading…
0/820398 [>---------------------------] 0%
820398/820398 [============================] 100%
Finished downloading
Extracting…The following error(s) occured:

• File Integrity failed for /var/www/html/admin/modules/_cache/sangomacrm-17.0.1.16.tgz.gpg - aborting (GPG Verify File check failed)
  2024-08-29 10:44:20 - ****** INSTALLATION FAILED *****
  2024-08-29 10:44:20 - Installation failed at step Wrapping up the installation process. Please check log /var/log/pbx/freepbx17-install-2024.08.29-10.28.54.log for details.
  2024-08-29 10:44:20 - Error at line: 1197 exiting with code 2 (last command was: fwconsole ma refreshsignatures >> “$log” 2>&1)
  2024-08-29 10:44:20 - Exiting script

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We’re almost done moving our incredibly old and decrepit Comdial phone system over to a hosted FreePBX instance, but for the life of me, I can’t get the the Valcom Universal Paging interface V-9972 to register as a SIP extension. I’ve called support, been given a fairly old configuration doc and entered the fairly simple values but the Paging Adapter never tries to connect - the reports show the extension as “unavailable” - all of the things I’ve read say to set it up as a generic SIP device but I’m using Endpoint Manager which had no option for generic.

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I would like to begin migrating our FreePBX 60 appliance to Debain/FreePBX 17. The FreePBX 60 appliance spec sheet states that an Intel Celeron processor is used which has an x86 architecture. Which Debian download do I use for this…

• Installation Guide for 64-bit PC (amd64)
• Installation Guide for 64-bit ARM (AArch64)
• Installation Guide for EABI ARM (armel)
• Installation Guide for Hard Float ABI ARM (armhf)
• Installation Guide for 32-bit PC (¡386)
• Installation Guide for MIPS (little endian)
• Installation Guide for 64-bit MIPS (little endian)
• Installation Guide for POWER Processors
• Installation Guide for IBM System z

I suspect the first item in the list, but I would like for someone to confirm this for me.

Thanks!

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Hi,

I’m expierincing some troubles with outgoing calls, I’m runnning FreePBX 17 on the newest Asterisk (21.4.2) on an Raspberry Pi 5.
When registrating my pjsip Trunk with A1 Telekom I can only receive incoming calls or make internal calls in between extensions. Outbound calls don’t work and are terminated with the message “All circuits are busy now, please try again later”.

Tbh I’m not that familiar with VoIP or PBX’s generally speaking, when putting the same user credentials directly into a SNOM D715 both In- and Outgoing calls worked perfectly, but due to some changes in the Fire Department we want to switch to a small scale pbx.

Thanks in advance for any tips / help

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Hey folks,
I’ve tried setting up an S3 bucket in the filestore module. However when clicking the “Add S3 bucket” button my frontend crashes as seen in this screenshot:

I’m using FreePBX 17.0.19.7 on Debian Bookworm.

Is there anything I’m missing or I can try to rectify this situation?

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Hello colleagues, I wanted to mask the director’s line (FreePBX 13) 277278305 as the assistant’s line 277278300, but it does not work. I have a SIP trunk 2772783xx, I edited the CID, deleted the CalerID in out. route…
For calls inside FreePBX ok, for calls outside FreePBX no.
Does anyone have an idea? Thank you Peter

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Hi Everyone.

I updated core module and added a field media_address in the PJSIP extensions.
This would help to fix some NAT issues with PJSIP extensions.

Available on:
core version 16.0.68.30
core version 17.0.16

Have a nice day guys.

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Hello!

I’m using (or at least I try to) Freepbx 17, and want to give a Windows sip client access to asterisks call log. So he can see his calls while the PC was off.

I used the quite short doc of the sip client (in german: Asterisk CDR auslesen ) but with not much luck so far.

I grant permissions to the db, and installed the mariadb odbc connector. Windows didn’t let me use it as file-DSN like the phonesuite doc wanted. But I could get it to connect with System-DSN. Don’t know if that makes a difference.

The ODBC connector can connect to mariadb, and also shows the correct db (asteriskcdrdb).
But if I try to load the data in the phone client, it gets only three older entries from one single extension. But none of the calls that I wanted…

At this point I’m confused and don’t know whether the problem is on the client or server side. Or in front of the keyboard… (SQL is not really my friend)

In the call log of the FreePBX gui I can see all calls (a quite long list), so it seems to be recorded.

“cdr show status” gives:

Call Detail Record (CDR) settings
----------------------------------
Logging:                    Enabled
Mode:                       Simple
Log calls by default:       Yes
Log unanswered calls:       Yes
Log congestion:             Yes

Ignore bridging changes:    No

Ignore dial state changes:  No

* Registered Backends
-------------------
cdr_manager
Adaptive ODBC

Edit:
“odbc show” says “Logging: Disabled”. Is that my problem?

ODBC DSN Settings
-----------------

Name:   asteriskcdrdb
DSN:    MySQL-asteriskcdrdb
Number of active connections: 2 (out of 5)
Logging: Disabled

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Dear Community,

after a new installation with version 17 i want to give it another try with the webphone on the UCP.
I already enabled the plugin for a user and it showes up. As i know that a https connection is nessesary i’m stucked to get it working.

I only want to use it in our internal network, and i’m not sure if i need a “let’s encrypt” or a self signed" certificate for it and what i do after it.

I tried it in the past but after enableing the certificate i could not reach the GUI anymore i want to avoid this situation again.

I only found this instructions which refer to a older FreePBX i think?
https://sangomakb.atlassian.net/wiki/spaces/PG/pages/38634428/WebRTC+Phone-UCP

Did exist a newer one i can use?

Thank You
Stefan

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Dear you,
Why i don’t see menu Add sip(chain_sip) trunk
How can enabled it?
Thanks

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I operate a FreePBX with version 16 in the cloud directly connected to the public IP. The phones are connected via the Internet and are located behind a firewall. They are all registered with the customer’s public IP address, each with the same IP address but with a different port number.

Now the SDP sometimes contains the internal IP address instead of the public IP address when the phone says 200 OK. As a result, the audio stream does not arrive.

The extension is configured with RTP Symmetric yes, Rewrite Contact yes and Force rport yes. Transport ist UDP.

Does anyone know the problem? How can this be prevented? As mentioned, it is not always, only sometimes.

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Description: During the installation of FreePBX 17, an error occurs after entering the database credentials. The error message indicates that the class Ramsey\Uuid\Exception\UnsatisfiedDependencyException cannot be found. This issue arises consistently when attempting to install FreePBX on an ARM64 architecture system using Asterisk 20.

Steps to Reproduce:

1. Clone the FreePBX 17 repository:

git clone https://github.com/FreePBX/framework.git -b release/17.0

2. • Compile and install Asterisk 20 from source.
3. • Begin the FreePBX installation:

./install

2. Follow the installation prompts until asked to enter database details.
3. After submitting the database information, the installation fails with the mentioned error.

Expected Behavior: The installation should proceed without errors, properly initializing the database and continuing with the FreePBX setup.

Actual Behavior: The installation fails immediately after entering the database details, with the following error:

In FreePBX.class.php line 19:
Class "Ramsey\Uuid\Exception\UnsatisfiedDependencyException" not found

System Information:

• Operating System: [e.g., Debian 12 ARM64]
• Asterisk Version: 20.9.2
• FreePBX Version: 17.0 (release/17.0 branch)
• PHP Version: 8.2.22
• Composer Version: 2.5.5

Greetings
Niclas

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i have a freepbx and i have dahdi trunk
on i call external internatunal i got error busy
this is astrisik debug

[2024-08-31 11:09:01] VERBOSE[13517][C-00000113] netsock2.c: Using SIP RTP CoS mark 5
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [00xxxxxxxxxx@from-internal:1] Macro("SIP/7001-0000012d", "user-callerid,LIMIT,EXTERNAL,") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [s@macro-user-callerid:1] Set("SIP/7001-0000012d", "TOUCH_MONITOR=1725088141.542") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [s@macro-user-callerid:2] Set("SIP/7001-0000012d", "AMPUSER=7001") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [s@macro-user-callerid:3] GotoIf("SIP/7001-0000012d", "0?report") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [s@macro-user-callerid:4] ExecIf("SIP/7001-0000012d", "1?Set(REALCALLERIDNUM=7001)") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [s@macro-user-callerid:5] Set("SIP/7001-0000012d", "AMPUSER=7001") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [s@macro-user-callerid:6] GotoIf("SIP/7001-0000012d", "0?limit") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [s@macro-user-callerid:7] Set("SIP/7001-0000012d", "AMPUSERCIDNAME=7001") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [s@macro-user-callerid:8] GotoIf("SIP/7001-0000012d", "0?report") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [s@macro-user-callerid:9] Set("SIP/7001-0000012d", "AMPUSERCID=7001") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [s@macro-user-callerid:10] Set("SIP/7001-0000012d", "__DIAL_OPTIONS=Ttr") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [s@macro-user-callerid:11] Set("SIP/7001-0000012d", "CALLERID(all)="7001" <7001>") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [s@macro-user-callerid:12] GotoIf("SIP/7001-0000012d", "0?limit") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [s@macro-user-callerid:13] ExecIf("SIP/7001-0000012d", "1?Set(GROUP(concurrency_limit)=7001)") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [s@macro-user-callerid:14] ExecIf("SIP/7001-0000012d", "0?Set(CHANNEL(language)=)") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [s@macro-user-callerid:15] GotoIf("SIP/7001-0000012d", "1?continue") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx_builtins.c: Goto (macro-user-callerid,s,29)
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [s@macro-user-callerid:29] Set("SIP/7001-0000012d", "CALLERID(number)=7001") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [s@macro-user-callerid:30] Set("SIP/7001-0000012d", "CALLERID(name)=7001") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [s@macro-user-callerid:31] GotoIf("SIP/7001-0000012d", "0?cnum") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [s@macro-user-callerid:32] Set("SIP/7001-0000012d", "CDR(cnam)=7001") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [s@macro-user-callerid:33] Set("SIP/7001-0000012d", "CDR(cnum)=7001") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [s@macro-user-callerid:34] Set("SIP/7001-0000012d", "CHANNEL(language)=en") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [00xxxxxxxxxx@from-internal:2] Gosub("SIP/7001-0000012d", "sub-record-check,s,1(out,00xxxxxxxxxx,dontcare)") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [s@sub-record-check:1] GotoIf("SIP/7001-0000012d", "0?initialized") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [s@sub-record-check:2] Set("SIP/7001-0000012d", "__REC_STATUS=INITIALIZED") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [s@sub-record-check:3] Set("SIP/7001-0000012d", "NOW=1725088141") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [s@sub-record-check:4] Set("SIP/7001-0000012d", "__DAY=31") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [s@sub-record-check:5] Set("SIP/7001-0000012d", "__MONTH=08") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [s@sub-record-check:6] Set("SIP/7001-0000012d", "__YEAR=2024") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [s@sub-record-check:7] Set("SIP/7001-0000012d", "__TIMESTR=20240831-110901") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [s@sub-record-check:8] Set("SIP/7001-0000012d", "__FROMEXTEN=7001") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [s@sub-record-check:9] Set("SIP/7001-0000012d", "__MON_FMT=wav") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [s@sub-record-check:10] NoOp("SIP/7001-0000012d", "Recordings initialized") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [s@sub-record-check:11] ExecIf("SIP/7001-0000012d", "0?Set(ARG3=dontcare)") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [s@sub-record-check:12] Set("SIP/7001-0000012d", "REC_POLICY_MODE_SAVE=") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [s@sub-record-check:13] ExecIf("SIP/7001-0000012d", "0?Set(REC_STATUS=NO)") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [s@sub-record-check:14] GotoIf("SIP/7001-0000012d", "3?checkaction") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx_builtins.c: Goto (sub-record-check,s,17)
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [s@sub-record-check:17] GotoIf("SIP/7001-0000012d", "1?sub-record-check,out,1") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx_builtins.c: Goto (sub-record-check,out,1)
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [out@sub-record-check:1] NoOp("SIP/7001-0000012d", "Outbound Recording Check from 7001 to 00xxxxxxxxxx") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [out@sub-record-check:2] Set("SIP/7001-0000012d", "RECMODE=force") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [out@sub-record-check:3] ExecIf("SIP/7001-0000012d", "0?Goto(routewins)") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [out@sub-record-check:4] ExecIf("SIP/7001-0000012d", "0?Goto(routewins)") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [out@sub-record-check:5] Gosub("SIP/7001-0000012d", "recordcheck,1(force,out,00xxxxxxxxxx)") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [recordcheck@sub-record-check:1] NoOp("SIP/7001-0000012d", "Starting recording check against force") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [recordcheck@sub-record-check:2] Goto("SIP/7001-0000012d", "force") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx_builtins.c: Goto (sub-record-check,recordcheck,5)
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [recordcheck@sub-record-check:5] Set("SIP/7001-0000012d", "__REC_POLICY_MODE=FORCE") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [recordcheck@sub-record-check:6] GotoIf("SIP/7001-0000012d", "1?startrec") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx_builtins.c: Goto (sub-record-check,recordcheck,16)
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [recordcheck@sub-record-check:16] NoOp("SIP/7001-0000012d", "Starting recording: out, 00xxxxxxxxxx") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [recordcheck@sub-record-check:17] Set("SIP/7001-0000012d", "AUDIOHOOK_INHERIT(MixMonitor)=yes") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [recordcheck@sub-record-check:18] Set("SIP/7001-0000012d", "__CALLFILENAME=out-00xxxxxxxxxx-7001-20240831-110901-1725088141.542") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [recordcheck@sub-record-check:19] MixMonitor("SIP/7001-0000012d", "2024/08/31/out-00xxxxxxxxxx-7001-20240831-110901-1725088141.542.wav,abi(LOCAL_MIXMON_ID),") in new stack
[2024-08-31 11:09:01] VERBOSE[24584][C-00000113] app_mixmonitor.c: Begin MixMonitor Recording SIP/7001-0000012d
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [recordcheck@sub-record-check:20] Set("SIP/7001-0000012d", "__MIXMON_ID=0x7f84bc7b8ae0") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [recordcheck@sub-record-check:21] Set("SIP/7001-0000012d", "__RECORD_ID=SIP/7001-0000012d") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [recordcheck@sub-record-check:22] Set("SIP/7001-0000012d", "__REC_STATUS=RECORDING") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [recordcheck@sub-record-check:23] Set("SIP/7001-0000012d", "CDR(recordingfile)=out-00xxxxxxxxxx-7001-20240831-110901-1725088141.542.wav") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [recordcheck@sub-record-check:24] Return("SIP/7001-0000012d", "") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [out@sub-record-check:6] Return("SIP/7001-0000012d", "") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [00xxxxxxxxxx@from-internal:3] ExecIf("SIP/7001-0000012d", "0 ?Set(CDR(accountcode)=)") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [00xxxxxxxxxx@from-internal:4] Set("SIP/7001-0000012d", "MOHCLASS=default") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [00xxxxxxxxxx@from-internal:5] ExecIf("SIP/7001-0000012d", "1?Set(TRUNKCIDOVERRIDE=023067000)") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [00xxxxxxxxxx@from-internal:6] Set("SIP/7001-0000012d", "_NODEST=") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [00xxxxxxxxxx@from-internal:7] Macro("SIP/7001-0000012d", "dialout-trunk,2,00xxxxxxxxxx,,off") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [s@macro-dialout-trunk:1] Set("SIP/7001-0000012d", "DIAL_TRUNK=2") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [s@macro-dialout-trunk:2] GosubIf("SIP/7001-0000012d", "0?sub-pincheck,s,1()") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [s@macro-dialout-trunk:3] GotoIf("SIP/7001-0000012d", "0?disabletrunk,1") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [s@macro-dialout-trunk:4] Set("SIP/7001-0000012d", "DIAL_NUMBER=00xxxxxxxxxx") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [s@macro-dialout-trunk:5] Set("SIP/7001-0000012d", "DIAL_TRUNK_OPTIONS=Ttr") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [s@macro-dialout-trunk:6] Set("SIP/7001-0000012d", "OUTBOUND_GROUP=OUT_2") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [s@macro-dialout-trunk:7] GotoIf("SIP/7001-0000012d", "0?nomax") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [s@macro-dialout-trunk:8] GotoIf("SIP/7001-0000012d", "0?chanfull") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [s@macro-dialout-trunk:9] GotoIf("SIP/7001-0000012d", "0?skipoutcid") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [s@macro-dialout-trunk:10] Set("SIP/7001-0000012d", "DIAL_TRUNK_OPTIONS=T") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [s@macro-dialout-trunk:11] Macro("SIP/7001-0000012d", "outbound-callerid,2") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [s@macro-outbound-callerid:1] ExecIf("SIP/7001-0000012d", "0?Set(CALLERPRES(name-pres)=)") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [s@macro-outbound-callerid:2] ExecIf("SIP/7001-0000012d", "0?Set(CALLERPRES(num-pres)=)") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [s@macro-outbound-callerid:3] ExecIf("SIP/7001-0000012d", "0?Set(REALCALLERIDNUM=7001)") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [s@macro-outbound-callerid:4] GotoIf("SIP/7001-0000012d", "1?normcid") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx_builtins.c: Goto (macro-outbound-callerid,s,7)
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [s@macro-outbound-callerid:7] Set("SIP/7001-0000012d", "USEROUTCID=") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [s@macro-outbound-callerid:8] Set("SIP/7001-0000012d", "EMERGENCYCID=") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [s@macro-outbound-callerid:9] Set("SIP/7001-0000012d", "TRUNKOUTCID=023067000") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [s@macro-outbound-callerid:10] GotoIf("SIP/7001-0000012d", "1?trunkcid") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx_builtins.c: Goto (macro-outbound-callerid,s,15)
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [s@macro-outbound-callerid:15] ExecIf("SIP/7001-0000012d", "1?Set(CALLERID(all)=023067000)") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [s@macro-outbound-callerid:16] ExecIf("SIP/7001-0000012d", "0?Set(CALLERID(all)=)") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [s@macro-outbound-callerid:17] ExecIf("SIP/7001-0000012d", "1?Set(CALLERID(all)=023067000)") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [s@macro-outbound-callerid:18] ExecIf("SIP/7001-0000012d", "0?Set(CALLERPRES(name-pres)=prohib_passed_screen)") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [s@macro-outbound-callerid:19] ExecIf("SIP/7001-0000012d", "0?Set(CALLERPRES(num-pres)=prohib_passed_screen)") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [s@macro-outbound-callerid:20] Set("SIP/7001-0000012d", "CDR(outbound_cnum)=023067000") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [s@macro-outbound-callerid:21] Set("SIP/7001-0000012d", "CDR(outbound_cnam)=") in new stack
[2024-08-31 11:09:01] WARNING[13454] func_cdr.c: CDR requires a value (CDR(variable)=value)
)[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [s@macro-dialout-trunk:12] GosubIf("SIP/7001-0000012d", "1?sub-flp-2,s,1()") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [s@sub-flp-2:1] ExecIf("SIP/7001-0000012d", "1?Return()") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [s@macro-dialout-trunk:13] Set("SIP/7001-0000012d", "OUTNUM=00xxxxxxxxxx") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [s@macro-dialout-trunk:14] Set("SIP/7001-0000012d", "custom=DAHDI/r1") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [s@macro-dialout-trunk:15] ExecIf("SIP/7001-0000012d", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default)T)") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [s@macro-dialout-trunk:16] ExecIf("SIP/7001-0000012d", "0?Set(DIAL_TRUNK_OPTIONS=TM(confirm))") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [s@macro-dialout-trunk:17] Macro("SIP/7001-0000012d", "dialout-trunk-predial-hook,") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/7001-0000012d", "") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [s@macro-dialout-trunk:18] GotoIf("SIP/7001-0000012d", "0?bypass,1") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [s@macro-dialout-trunk:19] ExecIf("SIP/7001-0000012d", "1?Set(CONNECTEDLINE(num,i)=00xxxxxxxxxx)") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [s@macro-dialout-trunk:20] ExecIf("SIP/7001-0000012d", "1?Set(CONNECTEDLINE(name,i)=CID:023067000)") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [s@macro-dialout-trunk:21] ExecIf("SIP/7001-0000012d", "0?Set(CONNECTEDLINE(name,i)=CID:(Hidden)023067000)") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [s@macro-dialout-trunk:22] GotoIf("SIP/7001-0000012d", "0?customtrunk") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] pbx.c: Executing [s@macro-dialout-trunk:23] Dial("SIP/7001-0000012d", "DAHDI/r1/00xxxxxxxxxx,300,T") in new stack
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] sig_pri.c: Requested transfer capability: 0x00 - SPEECH
[2024-08-31 11:09:01] VERBOSE[24583][C-00000113] app_dial.c: Called DAHDI/r1/00xxxxxxxxxx
[2024-08-31 11:09:04] VERBOSE[24583][C-00000113] app_dial.c: DAHDI/i1/00xxxxxxxxxx-ee is ringing
[2024-08-31 11:09:04] VERBOSE[24583][C-00000113] app_dial.c: DAHDI/i1/00xxxxxxxxxx-ee is making progress passing it to SIP/7001-0000012d
[2024-08-31 11:09:09] VERBOSE[24583][C-00000113] app_dial.c: DAHDI/i1/00xxxxxxxxxx-ee is busy
[2024-08-31 11:09:09] VERBOSE[24583][C-00000113] chan_dahdi.c: Hungup 'DAHDI/i1/00xxxxxxxxxx-ee'
[2024-08-31 11:09:09] VERBOSE[24583][C-00000113] app_dial.c: Everyone is busy/congested at this time (1:1/0/0)
[2024-08-31 11:09:09] VERBOSE[24583][C-00000113] pbx.c: Executing [s@macro-dialout-trunk:24] NoOp("SIP/7001-0000012d", "Dial failed for some reason with DIALSTATUS = BUSY and HANGUPCAUSE = 0") in new stack
[2024-08-31 11:09:09] VERBOSE[24583][C-00000113] pbx.c: Executing [s@macro-dialout-trunk:25] GotoIf("SIP/7001-0000012d", "0?continue,1:s-BUSY,1") in new stack
[2024-08-31 11:09:09] VERBOSE[24583][C-00000113] pbx_builtins.c: Goto (macro-dialout-trunk,s-BUSY,1)
[2024-08-31 11:09:09] VERBOSE[24583][C-00000113] pbx.c: Executing [s-BUSY@macro-dialout-trunk:1] NoOp("SIP/7001-0000012d", "Dial failed due to trunk reporting BUSY - giving up") in new stack
[2024-08-31 11:09:09] VERBOSE[24583][C-00000113] pbx.c: Executing [s-BUSY@macro-dialout-trunk:2] PlayTones("SIP/7001-0000012d", "busy") in new stack
[2024-08-31 11:09:09] WARNING[24583][C-00000113] translate.c: no samples for alawtolin
[2024-08-31 11:09:09] VERBOSE[24583][C-00000113] pbx.c: Executing [s-BUSY@macro-dialout-trunk:3] Busy("SIP/7001-0000012d", "20") in new stack
[2024-08-31 11:09:09] VERBOSE[24583][C-00000113] app_macro.c: Spawn extension (macro-dialout-trunk, s-BUSY, 3) exited non-zero on 'SIP/7001-0000012d' in macro 'dialout-trunk'
[2024-08-31 11:09:09] VERBOSE[24583][C-00000113] pbx.c: Spawn extension (from-internal, 00xxxxxxxxxx, 7) exited non-zero on 'SIP/7001-0000012d'
[2024-08-31 11:09:09] VERBOSE[24583][C-00000113] pbx.c: Executing [h@from-internal:1] Macro("SIP/7001-0000012d", "hangupcall") in new stack
[2024-08-31 11:09:09] VERBOSE[24583][C-00000113] pbx.c: Executing [s@macro-hangupcall:1] GotoIf("SIP/7001-0000012d", "1?theend") in new stack
[2024-08-31 11:09:09] VERBOSE[24583][C-00000113] pbx_builtins.c: Goto (macro-hangupcall,s,3)
[2024-08-31 11:09:09] VERBOSE[24583][C-00000113] pbx.c: Executing [s@macro-hangupcall:3] ExecIf("SIP/7001-0000012d", "0?Set(CDR(recordingfile)=)") in new stack
[2024-08-31 11:09:09] VERBOSE[24583][C-00000113] pbx.c: Executing [s@macro-hangupcall:4] Hangup("SIP/7001-0000012d", "") in new stack
[2024-08-31 11:09:09] VERBOSE[24583][C-00000113] app_macro.c: Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/7001-0000012d' in macro 'hangupcall'
[2024-08-31 11:09:09] VERBOSE[24583][C-00000113] pbx.c: Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/7001-0000012d'
[2024-08-31 11:09:09] VERBOSE[24584][C-00000113] app_mixmonitor.c: MixMonitor close filestream (mixed)
[2024-08-31 11:09:09] VERBOSE[24584][C-00000113] app_mixmonitor.c: End MixMonitor Recording SIP/7001-0000012d

so can any one help me to solve it by astrisik command line

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Hey everyone - anyone who can help?

So, I have this page where it outputs some data we want to have playback to the caller (to update them on their order status)

1. https://friendlystranger.gr/check_order.php?order_number=1184

2. We then setup a Dynamic route in FreePBX:

3. Added a Custom Destination:

4. And finally added this in the “extensions_custom.conf” file:

[read-order-status]
exten => s,1,NoOp(Processing Order Status)
exten => s,n,Set(order_details=${dynroute})

; Extract the information
exten => s,n,Set(status=${CUT(CUT(order_details,|,1),:,2)})
exten => s,n,Set(courier=${CUT(CUT(order_details,|,2),:,2)})
exten => s,n,Set(tracking=${CUT(CUT(order_details,|,3),:,2)})

; Log the extracted details for debugging
exten => s,n,NoOp(Status: ${status}, Courier: ${courier}, Tracking: ${tracking})

; Playback the details
exten => s,n,Playback(your-order-status-is) 
exten => s,n,SayAlpha(${status})

exten => s,n,Playback(your-courier-is) 
exten => s,n,SayAlpha(${courier})

exten => s,n,Playback(your-tracking-number-is) 
exten => s,n,SayDigits(${tracking})

exten => s,n,Hangup()

However, although it’s pulling the data, its just going into a blackhole:

107097	[2024-09-01 00:58:17] VERBOSE[21117][C-00000137] app_read.c: Accepting a maximum of 4 digits.	
107098	[2024-09-01 00:58:17] VERBOSE[21117][C-00000137] file.c: <PJSIP/anonymous-0000027e> Playing 'custom/Call_originating_from_other_store.slin' (language 'gr')	
107099	[2024-09-01 00:58:24] VERBOSE[21117][C-00000137] app_read.c: User entered '1184'	
107100	[2024-09-01 00:58:24] VERBOSE[21117][C-00000137] pbx.c: Executing [s@dynroute-1:3] Set("PJSIP/anonymous-0000027e", "dynroute=") in new stack	
107101	[2024-09-01 00:58:24] VERBOSE[21117][C-00000137] pbx.c: Executing [s@dynroute-1:4] Set("PJSIP/anonymous-0000027e", "CURLOPT(dnstimeout)=5") in new stack	
107102	[2024-09-01 00:58:24] VERBOSE[21117][C-00000137] pbx.c: Executing [s@dynroute-1:5] Set("PJSIP/anonymous-0000027e", "CURLOPT(conntimeout)=5") in new stack	
107103	[2024-09-01 00:58:24] VERBOSE[21117][C-00000137] pbx.c: Executing [s@dynroute-1:6] Set("PJSIP/anonymous-0000027e", "CURLOPT(ftptimeout)=5") in new stack	
107104	[2024-09-01 00:58:24] VERBOSE[21117][C-00000137] pbx.c: Executing [s@dynroute-1:7] Set("PJSIP/anonymous-0000027e", "CURLOPT(httptimeout)=5") in new stack	
107105	[2024-09-01 00:58:24] VERBOSE[21117][C-00000137] pbx.c: Executing [s@dynroute-1:8] Set("PJSIP/anonymous-0000027e", "dynroute=status:wc-failed|courier:ACS|tracking:2548778541") in new stack	
107106	[2024-09-01 00:58:24] VERBOSE[21117][C-00000137] pbx.c: Executing [s@dynroute-1:9] Set("PJSIP/anonymous-0000027e", "dynroute=status:wc-failed|courier:ACS|tracking:2548778541") in new stack	
107107	[2024-09-01 00:58:24] VERBOSE[21117][C-00000137] pbx.c: Executing [s@dynroute-1:10] GotoIf("PJSIP/anonymous-0000027e", "0?dynroute-1,1,1") in new stack	
107108	[2024-09-01 00:58:24] VERBOSE[21117][C-00000137] pbx.c: Executing [s@dynroute-1:11] GotoIf("PJSIP/anonymous-0000027e", "0?read-order-status,s,1") in new stack	
107109	[2024-09-01 00:58:24] VERBOSE[21117][C-00000137] pbx.c: Executing [s@dynroute-1:12] Goto("PJSIP/anonymous-0000027e", "dynroute-1,1,1") in new stack	
107110	[2024-09-01 00:58:24] VERBOSE[21117][C-00000137] pbx_builtins.c: Goto (dynroute-1,1,1)	
107111	[2024-09-01 00:58:24] VERBOSE[21117][C-00000137] pbx.c: Executing [1@dynroute-1:1] Goto("PJSIP/anonymous-0000027e", "app-blackhole,busy,1") in new stack	
107112	[2024-09-01 00:58:24] VERBOSE[21117][C-00000137] pbx_builtins.c: Goto (app-blackhole,busy,1)	
107113	[2024-09-01 00:58:24] VERBOSE[21117][C-00000137] pbx.c: Executing [busy@app-blackhole:1] NoOp("PJSIP/anonymous-0000027e", "Blackhole Dest: Busy") in new stack	
107114	[2024-09-01 00:58:24] VERBOSE[21117][C-00000137] pbx.c: Executing [busy@app-blackhole:2] Progress("PJSIP/anonymous-0000027e", "") in new stack	
107115	[2024-09-01 00:58:24] VERBOSE[21117][C-00000137] pbx.c: Executing [busy@app-blackhole:3] Busy("PJSIP/anonymous-0000027e", "20") in new stack	
107116	[2024-09-01 00:58:27] VERBOSE[21117][C-00000137] pbx.c: Spawn extension (app-blackhole, busy, 3) exited non-zero on 'PJSIP/anonymous-0000027e'	
107117	[2024-09-01 00:58:27] VERBOSE[21118][C-00000137] app_mixmonitor.c: MixMonitor close filestream (mixed)	
107118	[2024-09-01 00:58:27] VERBOSE[21118][C-00000137] app_mixmonitor.c: End MixMonitor Recording PJSIP/anonymous-0000027e

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At the moment, on FreePBX16, I’m rotating only asterisk logs.

/var/log/asterisk/debug
/var/log/asterisk/messages
/var/log/asterisk/full
/var/log/asterisk/*_log {
        weekly
        missingok
        rotate 6
        sharedscripts
        create 0640 asterisk asterisk
        su asterisk asterisk
        postrotate
                /usr/sbin/invoke-rc.d asterisk logger-reload > /dev/null 2> /dev/null
        endscript
}

Does the same postrotate commands cover rotating these files:

/var/log/asterisk/freepbx.log
/var/log/asterisk/freepbx_dbug
/var/log/asterisk/freepbx_security.log
/var/log/asterisk/backup.log

and what about node processes’ logs

/var/log/core-*.log

Many thanks

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I have a Cisco VG224 and a server running FreePBX. I would like to use the VG224 to register each analog port to an extension in FreePBX. I guess I effectively want a large ATA?

I’ve seen a few post alluding to it. They talk about doing so in this post: Issue registering old cisco device and a user gives another some instructions in this post: Use a Cisco router FXS as PBX subscriber - #7 by michelpy

I guess I just need a little more to get me over the finish line. Any ideas?

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I have FreePBX 17 . Does anybody might know why I’m having two IP address. Here are the pictures.

I tried to run cd /etc/sysconfig/network-scripts/ but no such file or directory.

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I recently built a new FreePBX 17 system to replace an older FreePBX 16 system that was built using the custom .iso image. The installation of Debian 12, updates to the OS and then the Github script to download and install FreePBX went really smoothly.
Now that OS updates are handled from Debain, I have patched it from the CLI (sudo apt-get update, sudo apt-get upgrade etc). Yesterday I noticed a couple of updates were held back, one of them being ‘freepbx17’. I took a snapshot of the VM and then proceeded to update the held back packages. This worked, or at least it downloaded and installed the updates, however FreePBX was broken. I tried a couple of things, but didn’t get anywhere, so restored the snapshot.
I’m guessing if I did a clean install, the latest update to freepbx17 would be installed by the script.
Any ideas how we get this update?

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We’ve upgraded to FreePBX 17 and our configuration has worked except where we have two endpoints registered on an extension, the busy lamps don’t light. (We were using chan_sip before so this is new configuration.) Our Snom D715 deskphones subscribe and show BLF hints, but not for those of us with DECT handsets as well. The deskphone subscribes, but it doesn’t get hints.

Has anyone found a solution to this, or have I missed some configuration setting in PJSIP? I’ve not found any posts relating to this even in the wider asterisk community.

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I just installed the Tailscale Addon for Home Assistant… Everything is running fine. I enable SUBNET ROUTES on the server so i have remote access to devices to my local network including Home Assistant server.

I Also have a Freepbx server running on the same local network for my home voip phone… everything on my PBX system is working fine aslong that its on local… the problem is when i try to make a call using a softphone app “linphone” outside my network, my local voip phone rings and can answer the call and also hear the caller from the softphone… but when i speak thru the voip phone the other end cannot hear me…

Troubleshooting i tried to connect my softphone to local wifi… then make a call… only then 2 way audio without issue… i dont know where could the problem be… i dont know if its on tailscale side or maybe the freepbx side… maybe someone here came across the same issue?

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