@fastdrw wrote:

fail2ban will not start due to an error in Distro installation.
This is a repeatable error: (3 installs)

using the following command to debug fail2ban error:
fail2ban-client -vvvvvvvvvvvvvvvvvv start

ERROR Found no accessible config files for ‘filter.d/apache-api’ under /etc/fail2ban
ERROR Unable to read the filter
ERROR Errors in jail ‘apache-api’. Skipping…

If you go into the file jail.local and change the enabled = true to false

[apache-api]
enabled = true
filter = apache-api
action = iptables-multiport[name=api, protocol=tcp, port=“http,https”]
sendmail[name=api, dest=admin@totalconnect.ca, sender=uc252@tcci.cloud]
logpath = /var/log/httpd/*access_log

fail2ban will restart and run from the CLI.
BUT if you restart fail2ban in the GUI, it changes the false back to true and fail2ban STOPS RUNNING again.

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@sentinelace wrote:

We want to block IMCP for better security. We host PBX’s in the cloud and would like this for an extra layer of security. I read the link below, is that still the best method?

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@mitchmitchell wrote:

I’ve created pull requests for fixing several allowlist bugs and adding a few small features.

1. Allowlist grid is now sortable by both number and descrition. I have a request to include added date but that will require some reorganization of the database
2. Add action icon to transfer a number to the blacklist – this was requested to help manager auto added numbers
3. Fixed a backup and restore bug where none of the module settings were saved and only the last record of the allowlist was restored.
4. Added support for pausing the allowlist from the GUI or by dialing a feature code.
5. Removed old translation files leftover from blacklist
6. Changed the feature codes to eliminate collision with the PMS module

Still to be done are fixing up the audio files – @lgaetz any thoughts on getting a ‘proper’ set of messages for the allowlist?

Hopefully I created the pull requests correctly.

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@lilchub101 wrote:

Hello,

I’m having some issues with outgoing calls to a PJSIP trunk (flowroute).

Asterisk 16.17.0

Freepbx 15.0.17.55

When I make a call from a softphone (zoiper5) on the local network the call is immediately ended and I see a few interesting things in logs. Ultimately the call is rejected by flowroute with a 500 error. Looking at call logs on the flowroute side it looks like the invite is trying to make a call to my sip account number and not the dialed phone number. Looking at logs seems to confirm this <sip:42xxxx30@

Interestingly if I make a call to a custom dial plan and then have that dial plan connect me to the trunk directly PJSIP/<NUMBER>@fl avoiding the Freepbx outbound route, the call is works as expected.

My outbound route is very basic

It just allows all calls to go through the only trunk I have called fl
Dial patterns are
NXXXXXX
NXXNXXXXXX

Any help or suggestions is much appreciated.

I have had to remove all logs and screenshots because I keep getting an error telling me new users can’t post links. I’m assuming the flowroute url and IP address in the logs are causing this error.

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@spikegod wrote:

Hi mates!

I have just upgraded my FreePBX 15 to 16 using the GUI Tool.

Everything is working but in the System Update page is shown an error:

Undefined variable: pkglist
File:/var/www/html/admin/libraries/Builtin/SystemUpdates.php:435

Error.png1336×211 41.5 KB

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@sawgood1000 wrote:

We have a SangomaOS PBX (built from .ISO).
It sits on a LAN behind a NAT router (very common).
The SangomaOS firewall is disabled / stopped.

We have a few remote phones (home offices).
Endpoints can use either chan_SIP (5160) or PJSIP (5060)
No OpenVPN involved.

On the PBX (at the extension level) all remote phones use chan_SIP port 5160, and NAT is set to YES.

Here is the question part:
We have (2) phones at a remote site …
One Yealink and one Fanvil.

The Yealink phone registers remotely just fine (has 2 way audio).
The Fanvil phone will not register.

Watching (using) sngrep: shows both phone registration packets arriving to the PBX (also see this with tcpdump).

The Yealink registers fine (watching sngrep) (4 messages) (x201)
Register
401 Unauthorized:
2nd Registration (with digest):
200 OK

The Fanvil fails (watching sngrep) (2 messages) (x202)
Register
401 Unauthorized:
(this repeats with no 2nd registration (digest) or 200 OK)

I’ve tried using x202 on the Yealink, and it registers fine.
I’ve tried using x201 on the Fanvil, and it fails

I’ve not found anything in the UI of the Fanvil for for “NAT” which matters.

I’ve tried setting the x202 extension to be NAT: No (did not work)

I have not tried registering either phone as a PJSIP remote endpoint.

Side-Note (don’t want to start an argument):
I’ve never had any luck registering a PJSIP remote endpoint to a SangomaOS PBX when it is sitting behind a router (NAT), so I’ve stuck with chan_SIP.

Thanks for any tips.

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@jshrack wrote:

Hi All,

Encountered a strange outcome from a FreePBX 15->16 upgrade this morning on FPBX Distro. The upgrade reported successful in the upgrade log (using the 15-to-16 upgrade wizard), however, the outcome was fwconsole exists with this:

image.png1178×478 27.1 KB

And the UI shows only:

image.png3316×100 23.8 KB

freepbx.log outputs…

[2021-10-28 11:49:17] [freepbx.INFO]: Connection attmempt to AMI failed [] []
[2021-10-28 11:50:01] [freepbx.INFO]: Connection attmempt to AMI failed [] []
[2021-10-28 11:50:01] [freepbx.INFO]: Connection attmempt to AMI failed [] []
[2021-10-28 11:50:08] [freepbx.INFO]: Connection attmempt to AMI failed [] []
[2021-10-28 11:51:02] [freepbx.INFO]: Connection attmempt to AMI failed [] []
[2021-10-28 11:51:14] [freepbx.INFO]: Connection attmempt to AMI failed [] []

fwconsole is no help since it always returns ‘0’. Server rebooted, no other errors that I could identify in related logs.

Thoughts? Suggestions?

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@Powermatz wrote:

Hello,

I used the upgrade tool to get to 16.
After performing the update I cannot go to the GUI anymore. It shows only a page with this text:
_

0 High Availability Services 13.0.11 Copyright 2017 by Schmoozecom, Inc., All rights reserved By installing, copying, downloading, distributing, inspecting or using the materials provided herewith, you agree to all of the terms of use as outlined in our End User Agreement which can be found and reviewed at w w w. schmoozecom. com / cmeula

_

When I open a ssh session I can see that Asterisk is not started. Everytime I enter a fwconsole command I get the same text shown above

I can not get rid of this HA stuff. I never used it before.

I tried yum update and module updates already. No need for any updates.
Any clue on this?

Here is some output from the upgrade log file:

Bildschirmfoto 2021-10-28 um 21.15.43.jpg681×1359 348 KB

It seems that the upgrade was not completely performed.

Best regards
Matthias

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@inferno2 wrote:

I see something very strange. Using the UCP’s call history widget I’m trying to playback a recording of >1h long. The loader is loading, nothing is playing.
Further investigation:

Normally a recorded call is being converted to html5 formats, I believe to cover all browsers for the playback feature. In my test it’s being converted to wav, sln48 and ulaw. These are being saved to /var/spool/asterisk/tmp. When the convert ‘job’ is done, it’s streamed to the browser via javascript (an url is being called). After being streamed to the browser, the temp files are being removed.
To be honest, I really do not understand why this convert to 3 different formats is needed.

This works perfect for short calls, <1h. With long calls it’s broken. It looks like the convert failed (?) and nothing is being streamed to the browser (no url is being called from ajax), the temp files are generated but I think it stops at generating ulaw file… However I do not see any error messages in the browser console, not the asterisk logs or httpd logs…

Running the latest stable versions of UCP module, call recording module, cel module, cdr module, even the core and framework.
This all on the distro, FreePBX 15.0.17.24, Asterisk 18.6.0 (before it was asterisk 13, same issue).

Anyone else able to reproduce the same?

Thx

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@voiprocks1998 wrote:

I am trying to connecting my hosted FreePBX server to my Telnyx SIP Trunk. I’m using IP as a credential method, and no authentication on FreePBX for connecting the trunk (ip authentication). I have an Avaya 9621G phone logged into the FreePBX (even though when the extension is called it says it is “unavailable”), that I am trying to call the outside world (PSTN). When making a call after setting up my SIP trunk connection, and outbound rules, I get the following problem -> “Circuits are busy now, please try again later”. When checking the asterisk logs this is what I found.

[2021-10-28 14:12:24] VERBOSE[9314][C-00000027] pbx c: Executing [s@macro-dialout-trunk:34] Dial(“PJSIP/2000-00000044”, “PJSIP/PHONENUM@SIPPER,300,Tb(func-apply-sipheaders^s^1,(1))U(sub-send-obroute-email^PHONENUM^PHONENUM^1^ENUM^SIPPER^DID)”) in new stack
[2021-10-28 14:12:24] VERBOSE[9314][C-00000027] app_stack c: PJSIP/SIPPER-00000045 Internal Gosub(func-apply-sipheaders,s,1(1)) start
[2021-10-28 14:12:24] VERBOSE[9314][C-00000027] pbx c: Executing [s@func-apply-sipheaders:1] ExecIf(“PJSIP/SIPPER-00000045”, “0?Set(CHANNEL(hangup_handler_push)=crm-hangup,s,1)”) in new stack
[2021-10-28 14:12:24] VERBOSE[9314][C-00000027] pbx c: Executing [s@func-apply-sipheaders:2] NoOp(“PJSIP/SIPPER-00000045”, “Applying SIP Headers to channel PJSIP/SIPPER-00000045”) in new stack
[2021-10-28 14:12:24] VERBOSE[9314][C-00000027] pbx c: Executing [s@func-apply-sipheaders:3] Set(“PJSIP/SIPPER-00000045”, “TECH=PJSIP”) in new stack
[2021-10-28 14:12:24] VERBOSE[9314][C-00000027] pbx c: Executing [s@func-apply-sipheaders:4] Set(“PJSIP/SIPPER-00000045”, “SIPHEADERKEYS=Alert-Info”) in new stack
[2021-10-28 14:12:24] VERBOSE[9314][C-00000027] pbx c: Executing [s@func-apply-sipheaders:5] While(“PJSIP/SIPPER-00000045”, “1”) in new stack
[2021-10-28 14:12:24] VERBOSE[9314][C-00000027] pbx c: Executing [s@func-apply-sipheaders:6] Set(“PJSIP/SIPPER-00000045”, “sipheader=unset”) in new stack
[2021-10-28 14:12:24] VERBOSE[9314][C-00000027] pbx c: Executing [s@func-apply-sipheaders:7] ExecIf(“PJSIP/SIPPER-00000045”, “0?SIPRemoveHeader(Alert-Info:)”) in new stack
[2021-10-28 14:12:24] VERBOSE[9314][C-00000027] pbx c: Executing [s@func-apply-sipheaders:8] ExecIf(“PJSIP/SIPPER-00000045”, “1?Set(PJSIP_HEADER(remove,Alert-Info)=)”) in new stack
[2021-10-28 14:12:24] ERROR[10738] res_pjsip_header_funcs c: No headers had been previously added to this session.
[2021-10-28 14:12:24] VERBOSE[9314][C-00000027] pbx c: Executing [s@func-apply-sipheaders:9] ExecIf(“PJSIP/SIPPER-00000045”, “0?Set(sipheader=;info=unset)”) in new stack
[2021-10-28 14:12:24] VERBOSE[9314][C-00000027] pbx c: Executing [s@func-apply-sipheaders:10] ExecIf(“PJSIP/SIPPER-00000045”, “0?Set(sipheader=unset)”) in new stack
[2021-10-28 14:12:24] VERBOSE[9314][C-00000027] pbx c: Executing [s@func-apply-sipheaders:11] ExecIf(“PJSIP/SIPPER-00000045”, “0?SIPAddHeader(Alert-Info:unset)”) in new stack
[2021-10-28 14:12:24] VERBOSE[9314][C-00000027] pbx c: Executing [s@func-apply-sipheaders:12] ExecIf(“PJSIP/SIPPER-00000045”, “0?Set(PJSIP_HEADER(add,Alert-Info)=unset)”) in new stack
[2021-10-28 14:12:24] VERBOSE[9314][C-00000027] pbx c: Executing [s@func-apply-sipheaders:13] EndWhile(“PJSIP/SIPPER-00000045”, “”) in new stack
[2021-10-28 14:12:24] VERBOSE[9314][C-00000027] pbx c: Executing [s@func-apply-sipheaders:5] While(“PJSIP/SIPPER-00000045”, “0”) in new stack
[2021-10-28 14:12:24] VERBOSE[9314][C-00000027] pbx c: Executing [s@func-apply-sipheaders:14] Return(“PJSIP/SIPPER-00000045”, “”) in new stack
[2021-10-28 14:12:24] VERBOSE[9314][C-00000027] app_stack c: Spawn extension (from-pstn, PHONENUM, 1) exited non-zero on ‘PJSIP/SIPPER-00000045’
[2021-10-28 14:12:24] VERBOSE[9314][C-00000027] app_stack c: PJSIP/SIPPER-00000045 Internal Gosub(func-apply-sipheaders,s,1(1)) complete GOSUB_RETVAL=
[2021-10-28 14:12:24] VERBOSE[9314][C-00000027] app_dial c: Called PJSIP/PHONENUM@SIPPER
[2021-10-28 14:12:24] ERROR[18232] res_pjsip_outbound_authenticator_digest c: Endpoint: ‘SIPPER’: There were no auth ids available
[2021-10-28 14:12:24] VERBOSE[9314][C-00000027] app_dial c: Everyone is busy/congested at this time (1:0/0/1)
[2021-10-28 14:12:24] VERBOSE[9314][C-00000027] pbx c: Executing [s@macro-dialout-trunk:35] NoOp(“PJSIP/2000-00000044”, “Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 21”) in new stack
[2021-10-28 14:12:24] VERBOSE[9314][C-00000027] pbx c: Executing [s@macro-dialout-trunk:36] GotoIf(“PJSIP/2000-00000044”, “0?continue,1:s-CHANUNAVAIL,1”) in new stack
[2021-10-28 14:12:24] VERBOSE[9314][C-00000027] pbx_builtins c: Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
[2021-10-28 14:12:24] VERBOSE[9314][C-00000027] pbx c: Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] Set(“PJSIP/2000-00000044”, “RC=21”) in new stack
[2021-10-28 14:12:24] VERBOSE[9314][C-00000027] pbx c: Executing [s-CHANUNAVAIL@macro-dialout-trunk:2] Goto(“PJSIP/2000-00000044”, “21,1”) in new stack
[2021-10-28 14:12:24] VERBOSE[9314][C-00000027] pbx_builtins c: Goto (macro-dialout-trunk,21,1)
[2021-10-28 14:12:24] VERBOSE[9314][C-00000027] pbx c: Executing [21@macro-dialout-trunk:1] Goto(“PJSIP/2000-00000044”, “continue,1”) in new stack
[2021-10-28 14:12:24] VERBOSE[9314][C-00000027] pbx_builtins c: Goto (macro-dialout-trunk,continue,1)
[2021-10-28 14:12:24] VERBOSE[9314][C-00000027] pbx c: Executing [continue@macro-dialout-trunk:1] NoOp(“PJSIP/2000-00000044”, “TRUNK Dial failed due to CHANUNAVAIL HANGUPCAUSE: 21 - failing through to other trunks”) in new stack
[2021-10-28 14:12:24] VERBOSE[9314][C-00000027] pbx c: Executing [continue@macro-dialout-trunk:2] ExecIf(“PJSIP/2000-00000044”, “1?Set(CALLERID(number)=2000)”) in new stack
[2021-10-28 14:12:24] VERBOSE[9314][C-00000027] pbx c: Executing [1PHONENUM@from-internal:13] Macro(“PJSIP/2000-00000044”, “outisbusy,”) in new stack
[2021-10-28 14:12:24] VERBOSE[9314][C-00000027] pbx c: Executing [s@macro-outisbusy:1] Progress(“PJSIP/2000-00000044”, “”) in new stack
[2021-10-28 14:12:24] VERBOSE[9314][C-00000027] pbx c: Executing [s@macro-outisbusy:2] GotoIf(“PJSIP/2000-00000044”, “0?emergency,1”) in new stack
[2021-10-28 14:12:24] VERBOSE[9314][C-00000027] pbx c: Executing [s@macro-outisbusy:3] GotoIf(“PJSIP/2000-00000044”, “0?intracompany,1”) in new stack
[2021-10-28 14:12:24] VERBOSE[9314][C-00000027] pbx c: Executing [s@macro-outisbusy:4] Playback(“PJSIP/2000-00000044”, “all-circuits-busy-now&please-try-call-later, noanswer”) in new stack
[2021-10-28 14:12:24] VERBOSE[9314][C-00000027] file c: <PJSIP/2000-00000044> Playing ‘all-circuits-busy-now g722’ (language ‘en’)
[2021-10-28 14:12:26] VERBOSE[9314][C-00000027] file c: <PJSIP/2000-00000044> Playing ‘please-try-call-later ulaw’ (language ‘en’)

The line with
[2021-10-28 14:12:24] ERROR[18232] res_pjsip_outbound_authenticator_digest c: Endpoint: ‘SIPPER’: There were no auth ids available
is odd to me, because I’m not using the regular auth, it is based on both parties knowing each others IP address.

I’m not sure what other logs I could check to debug this problem. Any suggestions would be helpful.

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@lgaetz wrote:

This post was shared on reddit which I thought it was really clever. It’s the first time I can recall ever seeing an example of the pinsets module being used for anything other than it’s intent. I immediately wondered how I might incorporate this technique into dynroutes so that no custom dialplan was necessary.

The intent of the Pin Sets module is to restrict access to outbound routes and thus control which local extensions can dial which external numbers. But suppose you need to create a call flow where you want to challenge a caller for a PIN before they can continue (after hours support?). You could use an IVR, but since IVRs are static, that would probably mean sharing PINs across users and it won’t scale beyond a few PINs. Better is to create a Pin Set for this purpose with a complete list of acceptable codes and then refer to that list using a dynroute. Restricting or expanding access in the future is just a matter of editing the Pin Set.

Step 1 - Create a Pin Set

All that’s required is a name and a list of codes. It’s best if all codes are the same length, but not required. For our purposes here the ‘record in CDR’ option will be ignored. Once done, you must determine the index for the Pin Set, when editing note the trailing digit(s) in the URL that follow itemid=. We will use the index in the dynroute, and for this example we’re assuming the index=1.

Step 2 - Create the Dynamic Route

Create a Dynamic Route, enter a name and set “Enable DTMF Input” to yes. Fill out the fields for prompting and validating the input if desired. Set a name in “Saved input variable name”, in this example I’m using dtmf_in. For Source type select Asterisk Variable, Disable Substitutions and set Asterisk variable to:

$[${DB_EXISTS(PINSETS/1/${DYNROUTE_dtmf_in})}]

The above expression checks the AstDB for a PIN in index 1 that matches the caller DTMF input. If your PIn Set is another index, substitute that value. It’s required to disable substitutions for this config, because our Asterisk expression uses square brackets that are not intended to refer to a dynroute variable. Wrapping this in $[] makes it a condition which will return a 0 if false or 1 if true.

Step 3 - Create the Dynamic Route Entries

Create 2 entries, one with a match of 0 and one with a match of 1. Direct each case to the appropriate FreePBX destination, something like this:

image.png1093×348 14.4 KB

Limitations:

The method above as written will only allow the caller a signle attempt to enter a PIN. You might want to chain 2-3 dynroutes together to give them multiple attempts. One of the strenghths of using Pin Sets is to record the PIN in the CDR for record keeping purposes. If you need that feauture, it could be done with more Dynamic Routes, but prob easier just to follow the method linked at the top.

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@jimi61271 wrote:

I’m having an outbound issue that I’m unable to resolve. I’m using a Momentum Telecom trunk with outbound authentication and all calls placed from the test extension go to an All Circuits Busy message followed by a Busy signal.

The PBX server is remote. I’ve mirrored a server locally with the SIP trunk information and inbound / outbound works fine. I believe both servers have exactly the same settings but the remote server is behind a Netgate firewall. I did configure the firewall for the suggested settings from Momentum. Also, Inbound does work at the remote site.

I’ve posted a log of an example outbound attempt from the remote server:

https://pastebin.freepbx.org/view/aec88a04

Any help would be most appreciated.

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@Answerphone wrote:

Is there a way I can set up a trunk so that another PBX we have here can register to it? I want to use it kind of like a makeshift gateway device to the other PBX. Unfortunately, the other PBX will only take registrations and not passwordless peers.

I see when you define the trunk you can enter a registration string, but I believe that’s only for outbound registrations. What about inbound registrations like how I am trying to set up?

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@Afox wrote:

Hello,
is it possible to set the key size of a CSR or the self signed certificates/Let´s Encrypt?
If not, would it be possible to file a feature request for this?
Thanks & best regards,
Afox

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@dirk2358 wrote:

Hello!

Since some time I’m getting mails about a security warning regarding a not installed module. Well, security is important - but it’s really strange to be spammed about security problems which do have no relevance at all for the existing system.

not-installed.png1677×98 34.9 KB

Would be cool if you could fix this. I don’t want to get mails about problems regarding not installed modules!

Thanks

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@ptravel wrote:

Adding a remote extension to a FreePBX 15 installation using chan_sip. I’m getting this error:

[2021-10-30 07:45:52] NOTICE[2834] chan_sip.c: Registration from 'sip:1009@[outside IP of FreePBX]:5160 failed for ‘[remote extension IP]:5160’ - Wrong password

The passwords match. Any idea what’s causing this?

Thanks!

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@angelosn wrote:

Hi,
I am trying to setup my template for my users and I want the softphone on the side bar called Phone (Not zulu)
When I edit the template it is not there.
In order to edit a template I clicked on the button to add a user for editing the templates.
This user is under the group All Users group which it has enabled the phone
Group->UCP->Phone->Enable Phone->Yes
I also checked this user’s settings which wasn’t “Inherit” but “Yes”
Is this a bug or I miss something?

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@dicko wrote:

[split from here - mod]

What is the downside of not having the “digiumaddoninstaller”? I read something about missing G.729 for example.

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@bartwiggers wrote:

Hi,

I would like to change FOP2 port 4443 to something else.
Does anyone know where to do this ?

Gr. Bart.

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@Afox wrote:

Hello,
can I delete the files in /etc/asterisk/keys after import with the “Import locally” function?

Thanks and best regards,

Afox

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