@eknudhol wrote:

This Freepbx system has been been running well for about 6 months. I have 3 local extensions working fine and three remote phones s500, s700 and new s300. The 500 & 700 have been connected via OpenVPN for about 6-9 months and I just added the s300 last week. Freepbx is running on a VM so I took a snapshot and performed the 14 to 15 Upgrade, which went well.

Now that the upgrade has completed, s500 would no longer register (via VPN) but the s700 and s300 would. The only only difference is that I did a Factory Reset on the s500, so assuming there is something blocking the tunnel from being built and spent most of the last 2 days looking at the firewall.

So I tested the the theory that the phone would join as long as it had an existing config and did the factory reset to the s300. It will no longer register. The phones also get the base config and never load the extension config, the phones stop and never reboot.

Not a real fan of OpenVPN so looking for options, or suggestions on what may have changed going to v15.

Quick Update: After spending many hours looking at this, it appears the firewall settings to allow provisioning via Internet had somehow gotten changed, not sure how I missed it. Although for some reason the s500 is still not getting a VPN IP - I am sure that is unrelated.

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@JeCtech wrote:

I have been reading up on different ways to connect analog lines to FreePBX and found lots of devices on the net, mostly ones that say they do this but when looking at documentation I find it is an FXS device.
I found some that have 1-FXS and 1-FXO port but in my case I have two analog lines I want to connect to my FreePBX setup.
I have seen all the recommended devices and majority have only 1-FXO only, My setup is for personal use and only for purpose of testing, not for commercial use so I am trying to keep my costs down.

I have found this device for a reasonable price and was wondering if anyone had any advise as to it’s ability to easily configure with FreePBX, I really don’t plan on using the FXS ports at this time.

Model: Netgen Smart ATA with 2 FXS Ports 2FXS/2FXO

Thank You

James

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@schuyler wrote:

In FreePBX 15 it looks like there is no way to select an S3 AWS Region in Filestore besides those provided in the dropdown box. I use a third party S3 compatible solution (DigitalOcean Spaces) and would like to backup to that location instead of S3 on AWS.

Is there any way to add a custom S3 endpoint url instead of picking one of the provided S3 regions?

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@douggoens wrote:

Because of the pandemic, I have taken on a new customer that has an Avaya phone system with standard rj-11 type connections. I’m using a SPA8000 device to convert sip to analog. Now that I’ve moved them from copper, is there a simple way I can cause the first number dialed in to go to the first extension, and if that line is in use, it will roll to the next line/extension and so on? I could possibly be doing this wrong, but it was sort of an emergency transfer as the prior provider went out of business.
Any help or advice is appreciated.

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@mjrudol wrote:

I have a conference on which I want to simplify the menu. I want users to just be able to use the toggle mute feature without enabling the whole default menu. Since I can’t edit the confbridge_additional.conf file and change the default, I edited the confbridge_custom.conf file like this:
[user_menu]
type = menu
147 = toggle_mute

This works. I can now use the code 147 to mute/unmute.
But… the whole default menu is also available.
I tried putting this same three lines in a confbridge_override_freepbx.conf file, but that seemed to do nothing.
What should I try next?

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@darvinm wrote:

I am experiencing an issue with PJSIP extensions and the FOP2 flash operator panel and was wondering if anyone can shed some light on the situation. The problem is that the PJSIP extensions do not show CID information of the call an extension is currently on, or of any calls that are ringing any of those extensions. When the call is answered, the extension on the panel does turn orange indicating they are on a call (see screenshot below) but with no CID info.

If I perform a “service fop2 restart” in SSH, it will start working but only for a few hours before the PJSIP extension will again show no CID info. I do not have this issue with CHAN_SIP extensions. All of these PJSIP extensions were setup originally as CHAN_SIP but then converted to PJSIP.

Has anyone else had this issue? If so, have you found a fix?

PS: I’m a newbie, so pardon any ignorance on my part!

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@ilouiemiami wrote:

Hi, can someone please advice what happens after an installation in a vps. Can I still install other programs on it?. Websites?..ETC… TY

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@wesleys wrote:

We have a self-hosted FreePBX server (Currently on version 14.0.13.28)

Today our FreePBX system apparently made a few hundred outgoing calls to the Dominican Republic, there are no logs of anyone SSHing into our server, and our web portal is behind a firewall only allowing specific traffic to access it.
We utilize SIP trunks, hooked up with Twilio for our inbound and outbound calls.

Immediately after this happened, the server crashed, and upon reboot I ran the fpbx updates. This appears to have stopped for the time being, but I’m worried that the issues came from an exploit or hack that has permanently opened our FreePBX server in a way I’m not familiar with.

Are there areas I should check in the FreePBX server for why this would have happened? Is there anything other than updating the OS and modules that I should be doing past this point? How likely was this just an exploit on a vulnerability that was patched with my updates? (I believe the last time I updated the modules on the server was 3-4 months prior).

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@rponkrashov wrote:

Hi community.

I will say in advance that a great many such topics were created on the forum, but nevertheless, I decided to try my luck, because I did not find a discussion of a similar problem that I had. I also tried various methods for resolving problem situations, which I will report below in a post.

So, to the problem:

I am trying to create a FreePBX distribution, in which I have included my scripts, the software I need. Asterisk 16.9.0 is installed from source, FreePBX 15 too, but I ran into a serious problem when I tried to install modules that I had already downloaded in advance.

When installing FreePBX, I use the following command:

./install -n --dbuser = user --dbpass = pass

Installation is successful, then I try to install modules for FreePBX, which were already downloaded earlier and included in the distribution.

I move the necessary modules to /var/www/html/admin/modules/ and then install using the fwconsole ma installlocal command.

Installation is successful, after which I do:

fwconsole chown
fwconsole reload

and get:

[root @ headstack ~] # fwconsole reload
Reload Started

In php-asmanager.php line 495:

fclose () expects parameter 1 to be resource, boolean given

reload [–json] [–dry-run] [–skip-registry-checks] [–dont-reload-asterisk]

Moreover, I noticed a similar behavior during tests, when I completely used fwconsole to install the necessary modules with the command:

fwconsole ma downloadinstall

The result is the same anyway.

When I get to FreePBX on the Web-interface, I see that an exclamation mark appears next to Asterisk, which says in the text that Asterisk was launched as early as 10 minutes, supposedly not using FreePBX startup scripts.

I ask for any help in resolving this problem, or your thoughts on how to resolve this incident. Ready to provide any information for debug.

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@dencoby wrote:

Hi All,
Ive been having some strange behavior since upgrading to 15
Everything works perfectly fine after setup until a reboot has occurred then I get this error in the logs.
chan_sip.c: Call from ‘09******’ (SIP Provider IP:5060) to extension ‘09******’ rejected because extension not found in context ‘default’
Both the 09 numbers are the same which is my voip username with my provider.

I can get it working again by deleting and recreating the incoming call route, but it breaks again if I need to do a reboot.

Any Suggestions? Am I missing something?

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@invictamix wrote:

Hi everybody,

i hope you can help me.
I see this notice in asterisk

[2020-04-15 10:39:40] NOTICE[2745][C-000000b1]: chan_sip.c:25904 handle_request_invite: Failed to authenticate device sip:1001@127.0.0.0:5060;tag=d11f24229f52d2472c23756e2e0effbc

i don’t understant why because right now i don’t have any sip or device with number 1001.
I restarted all the system but this notice persistes.

Can you help me please what should i do?

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@mayro wrote:

Hello!

Please help solve the problem. Thanks.

Logs:

[2020-04-15 12:26:34] WARNING[14666]: file.c:663 ast_openstream_full: File cannot-complete-as-dialed does not exist in any format
[2020-04-15 12:26:34] WARNING[14666]: file.c:958 ast_streamfile: Unable to open cannot-complete-as-dialed (format 0x4 (ulaw)): No such file or directory
[2020-04-15 12:26:34] WARNING[14666]: app_playback.c:475 playback_exec: ast_streamfile failed on Local/89312303162@from-internal-670f;2 for silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer
[2020-04-15 12:26:34] WARNING[14666]: file.c:663 ast_openstream_full: File check-number-dial-again does not exist in any format
[2020-04-15 12:26:34] WARNING[14666]: file.c:958 ast_streamfile: Unable to open check-number-dial-again (format 0x4 (ulaw)): No such file or directory
[2020-04-15 12:26:34] WARNING[14666]: app_playback.c:475 playback_exec: ast_streamfile failed on Local/89312303162@from-internal-670f;2 for silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer

ls -l /usr/share/asterisk/sounds/en/silence/1.gsm
-rw-r–r-- 1 root root 1650 окт. 5 2011 /usr/share/asterisk/sounds/en/silence/1.gsm

PBX Core settings

Version: 1.8.13.1~dfsg-3ubuntu3
Build Options: LOADABLE_MODULES
Maximum calls: Not set
Maximum open file handles: Not set
Verbosity: 0
Debug level: 0
Maximum load average: 0.000000
Minimum free memory: 0 MB
Startup time: 10:02:55
Last reload time: 12:05:57
System: Linux/3.2.0-37-generic built by buildd on x86_64 2013-10-16 23:58:01 UTC
System name:
Entity ID: 0c:c4:7a:01:b4:91
Default language: en
Language prefix: Enabled
User name and group: /
Executable includes: Disabled
Transcode via SLIN: Enabled
Internal timing: Enabled
Transmit silence during rec: Disabled
Generic PLC: Enabled

• Subsystems

  Manager (AMI): Enabled
  Web Manager (AMI/HTTP): Disabled
  Call data records: Enabled
  Realtime Architecture (ARA): Disabled

• Directories

  Configuration file:
  Configuration directory: /etc/asterisk
  Module directory: /usr/lib/asterisk/modules
  Spool directory: /var/spool/asterisk
  Log directory: /var/log/asterisk
  Run/Sockets directory: /var/run/asterisk
  PID file: /var/run/asterisk/asterisk.pid
  VarLib directory: /var/lib/asterisk
  Data directory: /usr/share/asterisk
  ASTDB: /var/lib/asterisk/astdb
  IAX2 Keys directory: /usr/share/asterisk/keys
  AGI Scripts directory: /usr/share/asterisk/agi-bin

sudo -u asterisk cat /usr/share/asterisk/sounds/en/silence/1.gsm
▒ ▒▒ZPI$▒I$PI$▒I$PI$▒I$PI$▒I$▒ ▒▒ZPI$▒I$PI$▒I$PI$▒I$PI$▒I$▒ ▒▒ZPI$▒I$PI$▒I$PI$▒I$

ls /usr/lib/asterisk/modules | grep codec

codec_adpcm.so
codec_alaw.so
codec_a_mu.so
codec_dahdi.so
codec_g722.so
codec_g726.so
codec_gsm.so
codec_lpc10.so
codec_speex.so
codec_ulaw.so

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@AndreL wrote:

Hi,

When the public ip changes, a script is used to update the “externerip” value in the kvstore database.
This update, correctly refected in the GUI sip settings, is followed by a “fwconsole reload”.

But the trunk using the PJSIP driver is not updated : SIP/SDP is still using previous IP.

My current workaround is a full “fwconsole restart”.

What’s the best way to manage public ip change with PJSIP trunk ?

That’s for your help,
André.

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@jbaron wrote:

Hello,

I have tried to sign my own module with my own key, following instructions on the wiki (Requesting a Key to be Signed and Signing your own modules)

# ./sign.php /opt/freepbx/www/admin/modules/droitappels/ 
Signing with D7669362454060A6
Generating file list...
Signing /opt/freepbx/www/admin/modules/droitappels/module.sig..gpg: using "D7669362454060A6" as default secret key for signing

Done
#

But I still get “Module has been signed with an invalid key” in the “Module Admin” page and dashboard.

I tried to

#fwconsole ma refreshsignatures

I tried packaging the module and reinstalling it,

I tried updating “FreePBX Framework” to version 15.0.16.49,

Also tried to change (temporarily) the keyservers hardcoded in “FreePBX Framework” to one where my key is published (hkp://keyserver.ubuntu.com:80) but it doesn’t change the status of my own modules.

Is there something I am missing?

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@jamesg224 wrote:

Hi everyone,

I have a working HT813 with ChanSIP trunk to FreePBX. The other day we had an internet outage which made the pstn line stop working as chansip stops when there is no internet. Does anyone have a working PJSIP setup for the FXO line? Or any instructions/ideas of how to set this up? I’m unfamilair with PJSIP trunking.

Thanks in advance

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@joelones wrote:

Was just wondering if it’s my setup (RPi3) but I notice when I hit the recording preview icon to quick listen to a recording it tends to just spin there if the file is of rather long length >10mins and I then get a undefined message. Shorter messages tend to play fine and the download link just works fine.
Any thoughts here? Coming from an older version (IncrediblePBX) and this would not happen.
Thanks

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@HypnotiXDMP wrote:

So, from what I can tell, the PBX is not even checking the blacklist (this would be the default one), and just lets the call go right through. Any advice?

FreePBX Version: 15.0.16.44
Asterisk Version: 16.6.2
Blacklist Version: 15.0.2.10

Here are the logs for when the call came in (I have put ########## where my number is, but it does show mu number properly).

[2020-04-15 11:42:20] VERBOSE[24623][C-0000080c] pbx.c: Executing [in@sub-record-check:3] ExecIf(“SIP/Vitel-Default-In-00002594”, “10?Set(FROMEXTEN=##########)”) in new stack
[2020-04-15 11:42:20] VERBOSE[24623][C-0000080c] pbx.c: Executing [8506089068@from-trunk:9] ExecIf(“SIP/Vitel-Default-In-00002594”, “0 ?Set(CALLERID(name)=##########)”) in new stack
[2020-04-15 11:42:20] VERBOSE[24623][C-0000080c] pbx.c: Executing [8506089068@from-trunk:21] Set(“SIP/Vitel-Default-In-00002594”, “__CRM_SOURCE=##########”) in new stack
[2020-04-15 11:42:21] VERBOSE[24623][C-0000080c] pbx.c: Executing [s@macro-user-callerid:2] Set(“SIP/Vitel-Default-In-00002594”, “AMPUSER=##########”) in new stack
[2020-04-15 11:42:21] VERBOSE[24623][C-0000080c] pbx.c: Executing [s@macro-user-callerid:9] ExecIf(“SIP/Vitel-Default-In-00002594”, “1?Set(REALCALLERIDNUM=##########)”) in new stack
[2020-04-15 11:42:21] VERBOSE[24623][C-0000080c] pbx.c: Executing [s@macro-user-callerid:43] Set(“SIP/Vitel-Default-In-00002594”, “CALLERID(number)=##########”) in new stack
[2020-04-15 11:42:21] VERBOSE[24623][C-0000080c] pbx.c: Executing [s@macro-user-callerid:47] Set(“SIP/Vitel-Default-In-00002594”, “CDR(cnum)=##########”) in new stack
[2020-04-15 11:42:21] VERBOSE[24623][C-0000080c] pbx.c: Executing [exten@sub-record-check:1] NoOp(“SIP/Vitel-Default-In-00002594”, “Exten Recording Check between ########## and 401”) in new stack
[2020-04-15 11:42:21] VERBOSE[24623][C-0000080c] pbx.c: Executing [s@macro-dial-one:2] Set(“SIP/Vitel-Default-In-00002594”, “__CRM_SOURCE=##########”) in new stack
[2020-04-15 11:42:21] VERBOSE[24623][C-0000080c] pbx.c: Executing [ctset@macro-dial-one:1] Set(“SIP/Vitel-Default-In-00002594”, “DB(CALLTRACE/401)=##########”) in new stack

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@ccts wrote:

Ok, setup a new fpbx 15 for a client, using bandwidth as the upstream carrier and they send calls in as e164 format, so the phones (yealink sip-t29g) get the call with the +1 on it, so if after a call is hung up, i go to history and try to redial a number from the history, the call fails cuz it has the full +1xxxxxxxxxx in there. If I hand dial just the number, it works fine, but if i select it from the history, with the +1, it doesn’t work. How to fix this?

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@abenezer wrote:

[2012-04-02 05:46:57] WARNING[3429][C-00000036] channel.c: Prodding channel ‘SIP/cisco2800-00000025’ failed
[2012-04-02 05:46:57] VERBOSE[3429][C-00000036] pbx.c: Spawn extension (from-internal, 0115180676, 7) exited non-zero on ‘SIP/cisco2800-00000025’
[2012-04-02 05:46:57] VERBOSE[3429][C-00000036] pbx.c: Executing [h@from-internal:1] Macro(“SIP/cisco2800-00000025”, “hangupcall”) in new stack
[2012-04-02 05:46:57] VERBOSE[3429][C-00000036] pbx.c: Executing [s@macro-hangupcall:1] GotoIf(“SIP/cisco2800-00000025”, “1?theend”) in new stack
[2012-04-02 05:46:57] VERBOSE[3429][C-00000036] pbx_builtins.c: Goto (macro-hangupcall,s,3)
[2012-04-02 05:46:57] VERBOSE[3429][C-00000036] pbx.c: Executing [s@macro-hangupcall:3] ExecIf(“SIP/cisco2800-00000025”, “0?Set(CDR(recordingfile)=)”) in new stack
[2012-04-02 05:46:57] VERBOSE[3429][C-00000036] pbx.c: Executing [s@macro-hangupcall:4] NoOp(“SIP/cisco2800-00000025”, " montior file= ") in new stack
[2012-04-02 05:46:57] VERBOSE[3429][C-00000036] pbx.c: Executing [s@macro-hangupcall:5] GotoIf(“SIP/cisco2800-00000025”, “1?skipagi”) in new stack
[2012-04-02 05:46:57] VERBOSE[3429][C-00000036] pbx_builtins.c: Goto (macro-hangupcall,s,7)
[2012-04-02 05:46:57] VERBOSE[3429][C-00000036] pbx.c: Executing [s@macro-hangupcall:7] Hangup(“SIP/cisco2800-00000025”, “”) in new stack
[2012-04-02 05:46:57] VERBOSE[3429][C-00000036] app_macro.c: Spawn extension (macro-hangupcall, s, 7) exited non-zero on ‘SIP/cisco2800-00000025’ in macro ‘hangupcall’
[2012-04-02 05:46:57] VERBOSE[3429][C-00000036] pbx.c: Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/cisco2800-00000025’

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@jcadman wrote:

I have started to use freepbx with Cisco 8841 IP Phones and would like to make sure that the phones are secure. At the moment, all extensions are on the local network to where the PBX is located. In the future I would like to have remote extensions that connect to the PBX however I need to make sure that I have the correct settings to secure the connection without need of a VPN. What are the best steps to secure the freepbx as well how how to setup a secure port for the port?

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