@tonyg wrote:

hi, I just looked up the difference between these two commands and I understand that update updates ONLY while upgrade updates and removes unused packages.
In the forums, I see people talking about both, If I am on FPBX v14, does it matter which one I use? is one preferable to the other?

thanks

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@johnjces wrote:

I have been using FreePBX Distro14.0.4.5, (12.7.5-1807-1.sng7), using Asterisk 13.22.0 without any issues and it has always just worked as long as it is set up correctly. In this case, I have to be missing something simple.

When I installed this distro I immediately disabled PJSIP, set Chan_SIP (Sip) back to the old port we all know and love, 5060, have a couple old GrandStream phones and an HT704 for a FAX and two analog wireless phones. All works GREAT on good old SIP on port 5060. I have yet to re-add PJSIP (Both) to the mix.

(I am trying to make this short)… In preparing to move over to the new FlowRoute POPS, which use PJSIP, I figured I would move my Sip to 5160, the new default. I have made the necessary port changes to my external firewall and had no troubles registering to FlowRoute’s legacy POPS on port 5160. Set the BIND port under Chan SIP Settings to 5160. Saved and applied. Set the HT704 to use 5160 and rebooted the HT704 AND did an fwconsole restart to ensure all has been applied.

No registration to FreePBX by the HT704. I can dial in and ring any of the phones hooked to the HT704, (not registered?) and talk. Any dial out is just “dead air”… no announcement about the call, i.e., can’t be completed as dialed or… anything.

Forgot to add that I then set the extensions to the HT704 to 5160. Same thing. No registration.

I am on a static Internet IP. All devices on the internal network all also static.

Setting everything back to 5060 and we are back to working like a charm.

What am I missing?

John

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@Mercury1 wrote:

When doing an fwconsole restart I get a segmentation fault error 139. Everything still seems to work but annoying.

Error message below

==========

In Process.php line 239:

The command “runuser ‘asterisk’ -s ‘/bin/bash’ -c ‘cd /var/www/html/admin/modules/pm2/nod
e && mkdir -p /home/asterisk/.pm2 && mkdir -p /var/www/html/admin/modules/pm2/node/logs &
& export HOME=/home/asterisk && export PM2_HOME=/home/asterisk/.pm2 && export ASTLOGDIR=/
var/log/asterisk && export ASTVARLIBDIR=/var/lib/asterisk && export PATH=$HOME/.node/bin:
$PATH && export NODE_PATH=$HOME/.node/lib/node_modules:$NODE_PATH && export MANPATH=$HOME
/.node/share/man:$MANPATH && /var/www/html/admin/modules/pm2/node/node_modules/pm2/bin/pm
2 jlist’” failed.

Exit Code: 139(Segmentation violation)

Working directory: /root

Output:

Error Output:

Segmentation fault

===============

Is there a simple fix for this problem? It does not seem uncommon, but I am unsure how to proceed without breaking something.

Thanks.

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@artello73 wrote:

Hello,

I’ve a FreePBX v13 running in VM and I’m trying to get it upgraded to v14 for a quite some time. So I cloned running system into new VM and after few attempts successfully completed upgrade. However, once I tried to put traffic in into the new instance I noticed RTP issues (no audio or one way audio).

Does anyone faced such issues right after upgrade? The configuration of both v13 and v14 VM’s is fully identical, so looks like the problem is in the newer asterisk version.

Worth to say that i have running quite specific setup (like SBC) - I have 3 voice trunks and each of them using own interface and the packets should be originated from the interfaces (there is no single external IP address which sources all SIP peer connections). So I’m using PJSIP and per-interface transports configured. This works with FreePBX13 (with asterisk 13.18.3) and doesn’t work in up-to-date FreePBX14.

Any clues that could be the reason? How to debug this?

Thanks in advance!

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@kendalldever15 wrote:

I have an immediate need for Ad-Hoc call recording. Currently all you get a is a beep when you start recording but NO visual light on the key pressed. Whats more is it possible to send to a mailbox vs having to check in UCP. They want one mailbox for all Ad-hoc call recordings.

They do not want continuous call recording.

The system we replaced gave them a fluttering Red key when pressed and two soft keys appeared. Pause and End. If Pause was pressed it changed the soft key to Resume and the fluttering red key when solid red. End if pressed ended the recording.

Can anything close to this be done?

Any help would be greatly appreciated.

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@mcomtec wrote:

I have one question,

I installed the distro "STABLE
SNG7-PBX-64bit-1805-1 "with Asterisk 15, but I can not receive faxes 99% of the time the reception fails and incoming faxes do it by G.711 without using Spandsp T.38.

I have tried to change from 9600 bps to 14400 bps, but I can not get it to work.

I have tried many things, and I have been trying and testing for more than a month without any results. Can you help me?

Thank you!

fax show stats

FAX Statistics:

Current Sessions : 0
Reserved Sessions : 0
Transmit Attempts : 0
Receive Attempts : 18
Completed FAXes : 18
Failed FAXes : 17

Spandsp G.711
Success : 1
Switched to T.38 : 0
Call Dropped : 6
No FAX : 0
Negotiation Failed : 0
Train Failure : 0
Retries Exceeded : 7
Protocol Error : 0
TX Protocol Error : 0
RX Protocol Error : 4
File Error : 0
Memory Error : 0
Unknown Error : 0

Spandsp T.38
Success : 0
Call Dropped : 0
No FAX : 0
Negotiation Failed : 0
Train Failure : 0
Retries Exceeded : 0
Protocol Error : 0
TX Protocol Error : 0
RX Protocol Error : 0
File Error : 0
Memory Error : 0
Unknown Error : 0

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@johnjces wrote:

I am using FreePBX Distro 14.0.5.2 and I received emails about the vulnerable modules and when I logged in the above showed up as well… which is really cool. The Apply config button was visable and I hit Apply.

In my Dashboard, it continues to advise me that the vulnerable module was auto upgraded and I cannot get rid of. I hit the - (minus) sign and a refresh or going back from another page and it still shows up.

I went to 'resolve" which takes me Module Admin and nothing is new to update since it apparently auto updated. I also tried a fwconsole ma upgrade framework since that was the vulnerable module and nothing was available since it was already updated.

How do i get rid of this allegedly fixed vulnerability warning from my Dashboard?

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@jerryriggin wrote:

I have FreePBX 10.13.66-22 and System Firewall 13.0.57.1 with Responsive Firewall enabled hosted at FreePBXHosting.com with one NIC assigned as Internet, with whitelisted networks for admin access. All services are local or local and other (default), including SIP. I am not sure this is correct configuration.

I have blacklisted IPs that are not being blocked. Here is an example:

image.png1010x448 50.4 KB

BTW: the traffic IS blocked outbound apparently. There is only the INVITE received and apparently no reply sent.

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@Stuart wrote:

This is a new install of FreePBX V.14.0.5.2 (around 2 months old) and is installed on a Citrix 7.5 virtual machine, FreePBX has been running well. I am able to update modules without issues but when I ran ‘yum update’ to update the system files I get the following. I need some advice on how to proceed, has anyone has experienced this. I think that it’s saying that the MySQL version is obsolete, should I upgrade MySQL or would that cause more issues?

______                   ______ ______ __   __
|  ___|                  | ___ \| ___ \\ \ / /
| |_    _ __   ___   ___ | |_/ /| |_/ / \ V /
|  _|  | '__| / _ \ / _ \|  __/ | ___ \ /   \
| |    | |   |  __/|  __/| |    | |_/ // /^\ \
\_|    |_|    \___| \___|\_|    \____/ \/   \/

Current Network Configuration
+-----------+-------------------+--------------------------+
| Interface | MAC Address       | IP Addresses             |
+-----------+-------------------+--------------------------+
| eth0      | 8A:1A:B2:F0:08:85 | 192.168.0.11             |
|           |                   | fe80::881a:b2ff:fef0:885 |
| eth1      | 96:D2:27:14:88:0A |                          |
| eth2      | 8A:AB:ED:25:36:76 |                          |
| eth3      | FE:81:9D:3F:45:C2 |                          |
+-----------+-------------------+--------------------------+

Please note most tasks should be handled through the GUI.
You can access the GUI by typing one of the above IPs in to your web browser.
For support please visit:
    http://www.freepbx.org/support-and-professional-services

+------------------------------------------------------------+
| There are 14 System updates available.                     |
|   Run yum update to update them.                           |
| Your PBX is up to date.                                    |
|   Also 3 Uninstalled modules.                              |
+------------------------------------------------------------+
[root@freepbx ~]# yum update
Loaded plugins: fastestmirror, versionlock
Loading mirror speeds from cached hostfile
 * sng-base: package1.sangoma.net
Resolving Dependencies
--> Running transaction check
---> Package asterisk13.x86_64 0:13.22.0-1.sng7 will be updated
---> Package asterisk13.x86_64 0:13.23.1-1.shmz65.1.204 will be an update
--> Processing Dependency: asterisk-sounds-core-en-gsm for package: asterisk13-13.23.1-1.shmz65.1.204.x86_64
---> Package asterisk13-addons.x86_64 0:13.22.0-1.sng7 will be updated
---> Package asterisk13-addons.x86_64 0:13.23.1-1.shmz65.1.204 will be an update
---> Package asterisk13-addons-bluetooth.x86_64 0:13.22.0-1.sng7 will be updated
---> Package asterisk13-addons-bluetooth.x86_64 0:13.23.1-1.shmz65.1.204 will be an update
---> Package asterisk13-addons-core.x86_64 0:13.22.0-1.sng7 will be updated
---> Package asterisk13-addons-core.x86_64 0:13.23.1-1.shmz65.1.204 will be an update
---> Package asterisk13-addons-mysql.x86_64 0:13.22.0-1.sng7 will be updated
---> Package asterisk13-addons-mysql.x86_64 0:13.23.1-1.shmz65.1.204 will be an update
--> Processing Dependency: libmysqlclient.so.16(libmysqlclient_16)(64bit) for package: asterisk13-addons-mysql-13.23.1-1.shmz65.1.204.x86_64
Package mysql-libs-5.1.73-5.el6_6.x86_64 is obsoleted by 1:mariadb-libs-5.5.56-2.el7.x86_64 which is already installed
--> Processing Dependency: libmysqlclient.so.16()(64bit) for package: asterisk13-addons-mysql-13.23.1-1.shmz65.1.204.x86_64
Package mysql-libs-5.1.73-5.el6_6.x86_64 is obsoleted by 1:mariadb-libs-5.5.56-2.el7.x86_64 which is already installed
---> Package asterisk13-addons-ooh323.x86_64 0:13.22.0-1.sng7 will be updated
---> Package asterisk13-addons-ooh323.x86_64 0:13.23.1-1.shmz65.1.204 will be an update
---> Package asterisk13-core.x86_64 0:13.22.0-1.sng7 will be updated
---> Package asterisk13-core.x86_64 0:13.23.1-1.shmz65.1.204 will be an update
--> Processing Dependency: libtiff.so.3()(64bit) for package: asterisk13-core-13.23.1-1.shmz65.1.204.x86_64
--> Processing Dependency: libical.so.0()(64bit) for package: asterisk13-core-13.23.1-1.shmz65.1.204.x86_64
---> Package asterisk13-curl.x86_64 0:13.22.0-1.sng7 will be updated
---> Package asterisk13-curl.x86_64 0:13.23.1-1.shmz65.1.204 will be an update
---> Package asterisk13-dahdi.x86_64 0:13.22.0-1.sng7 will be updated
---> Package asterisk13-dahdi.x86_64 0:13.23.1-1.shmz65.1.204 will be an update
---> Package asterisk13-doc.x86_64 0:13.22.0-1.sng7 will be updated
---> Package asterisk13-doc.x86_64 0:13.23.1-1.shmz65.1.204 will be an update
---> Package asterisk13-odbc.x86_64 0:13.22.0-1.sng7 will be updated
---> Package asterisk13-odbc.x86_64 0:13.23.1-1.shmz65.1.204 will be an update
---> Package asterisk13-ogg.x86_64 0:13.22.0-1.sng7 will be updated
---> Package asterisk13-ogg.x86_64 0:13.23.1-1.shmz65.1.204 will be an update
---> Package asterisk13-resample.x86_64 0:13.22.0-1.sng7 will be updated
---> Package asterisk13-resample.x86_64 0:13.23.1-1.shmz65.1.204 will be an update
---> Package asterisk13-voicemail.x86_64 0:13.22.0-1.sng7 will be updated
---> Package asterisk13-voicemail.x86_64 0:13.23.1-1.shmz65.1.204 will be an update
--> Running transaction check
---> Package asterisk-sounds-core-en-gsm.noarch 0:1.4.25-94_centos5 will be installed
---> Package asterisk13-addons-mysql.x86_64 0:13.23.1-1.shmz65.1.204 will be an update
--> Processing Dependency: libmysqlclient.so.16(libmysqlclient_16)(64bit) for package: asterisk13-addons-mysql-13.23.1-1.shmz65.1.204.x86_64
Package mysql-libs-5.1.73-5.el6_6.x86_64 is obsoleted by 1:mariadb-libs-5.5.56-2.el7.x86_64 which is already installed
--> Processing Dependency: libmysqlclient.so.16()(64bit) for package: asterisk13-addons-mysql-13.23.1-1.shmz65.1.204.x86_64
Package mysql-libs-5.1.73-5.el6_6.x86_64 is obsoleted by 1:mariadb-libs-5.5.56-2.el7.x86_64 which is already installed
---> Package asterisk13-core.x86_64 0:13.23.1-1.shmz65.1.204 will be an update
--> Processing Dependency: libical.so.0()(64bit) for package: asterisk13-core-13.23.1-1.shmz65.1.204.x86_64
---> Package compat-libtiff3.x86_64 0:3.9.4-11.el7 will be installed
--> Finished Dependency Resolution
Error: Package: asterisk13-addons-mysql-13.23.1-1.shmz65.1.204.x86_64 (pbx)
           Requires: libmysqlclient.so.16(libmysqlclient_16)(64bit)
           Available: mysql-libs-5.1.73-3.el6_5.x86_64 (base)
               libmysqlclient.so.16(libmysqlclient_16)(64bit)
           Available: mysql-libs-5.1.73-5.el6_6.x86_64 (updates)
               libmysqlclient.so.16(libmysqlclient_16)(64bit)
Error: Package: asterisk13-addons-mysql-13.23.1-1.shmz65.1.204.x86_64 (pbx)
           Requires: libmysqlclient.so.16()(64bit)
           Available: mysql-libs-5.1.73-3.el6_5.x86_64 (base)
               libmysqlclient.so.16()(64bit)
           Available: mysql-libs-5.1.73-5.el6_6.x86_64 (updates)
               libmysqlclient.so.16()(64bit)
Error: Package: asterisk13-core-13.23.1-1.shmz65.1.204.x86_64 (pbx)
           Requires: libical.so.0()(64bit)
           Available: libical-0.43-6.el6.x86_64 (base)
               libical.so.0()(64bit)
           Installed: libical-1.0.1-1.el7.x86_64 (@anaconda/1805)
              ~libical.so.1()(64bit)
 You could try using --skip-broken to work around the problem
 You could try running: rpm -Va --nofiles --nodigest
[root@freepbx ~]#

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@Chimeara wrote:

I have installed FreePBX 14 and followed all instructions and Youtube videos regarding configuration. Have installed Zulu server and 2 Zulu UC soft phones as 2 extensions. When I call from one Zulu I get no ringing on the one making the call and a disconnect while immediately get a missed call on the other Zulu.
What do i need to configure these to work correctly?

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@tazugre wrote:

Hello!

I’m currently testing a freePBX update from version 13.0.188.8 to SNG7.

Following the upgrade guide i get this info:
Guide: https://wiki.freepbx.org/display/PPS/Upgrading+from+FreePBX+10.13.66+to+SNG7

Using distro-upgrade:
Checking prerequsites…
Checking FreePBX Version [ ✘ ]
Current FreePBX version is 13.0.188.8 - Must be 13.0.191 or higher

Yesterday i tried updating all plugins/modules and it was able to get past the version compatibility check. All checks were green and normally a selection with y/n to approve should come up. But nothing happened. I tried to reboot the system (normally it should start the upgrade automatically, but the upgrade source didnt show up.

Does anyone have an idea?

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@dengel wrote:

I have an A200 with 2-FXO, 2-FXS installed. The second FXO port is connected to POTS. The card itself is set up as a PCI passthrough device to a VM running under ESXI 6.7.

Everything worked great with FreePBX 13 (AsteriskNow Distro 10.13). I had no issue making or receiving calls. However, I broke that installation trying to upgrade to FreePBX 14 using the built-in upgrader.

Now, I have a clean install of FreePBX 14 (FreePBX Distro SNG7-FPBX-64bit-1805-1), but the card doesn’t work anymore. It’s detected, and I can even see the ring voltage on incoming calls, but asterisk never picks up. Outgoing calls just hold forever in silence.

When everything works under FreePBX 13, an outgoing call gives me (excerpt):

-- Executing [s@macro-dialout-trunk:24] Dial("PJSIP/225-00000003", "DAHDI/2/wXXXXXX9971,300,T") in new stack
-- Called DAHDI/2/wXXXXXX9971
-- DAHDI/2-1 answered PJSIP/225-00000003
-- Channel DAHDI/2-1 joined 'simple_bridge' basic-bridge <88a799b0-1058-4fb6-be4e-2c6081b49184>
-- Channel PJSIP/225-00000003 joined 'simple_bridge' basic-bridge <88a799b0-1058-4fb6-be4e-2c6081b49184>
-- Channel PJSIP/225-00000003 left 'simple_bridge' basic-bridge <88a799b0-1058-4fb6-be4e-2c6081b49184>
  == Spawn extension (macro-dialout-trunk, s, 24) exited non-zero on 'PJSIP/225-00000003' in macro 'dialout-trunk'
  == Spawn extension (from-internal, 5035629971, 6) exited non-zero on 'PJSIP/225-00000003'
-- Executing [h@from-internal:1] Macro("PJSIP/225-00000003", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("PJSIP/225-00000003", "1?theend") in new stack
-- Goto (macro-hangupcall,s,3)
-- Channel DAHDI/2-1 left 'simple_bridge' basic-bridge <88a799b0-1058-4fb6-be4e-2c6081b49184>
-- Hanging up on 'DAHDI/2-1'
-- Hungup 'DAHDI/2-1'

With 14, the comparable excerpt is:

-- Executing [s@macro-dialout-trunk:25] Dial("PJSIP/123-00000001", "DAHDI/2/wXXXXXX9971,300,Tb(func-apply-sipheaders^s^1)") in new stack
-- DAHDI/2-1 Internal Gosub(func-apply-sipheaders,s,1) start
-- Executing [s@func-apply-sipheaders:1] NoOp("DAHDI/2-1", "Applying SIP Headers to channel") in new stack
-- Executing [s@func-apply-sipheaders:2] Set("DAHDI/2-1", "SIPHEADERKEYS=") in new stack
-- Executing [s@func-apply-sipheaders:3] ExecIf("DAHDI/2-1", "0?Set(Rheader=1)") in new stack
-- Executing [s@func-apply-sipheaders:4] While("DAHDI/2-1", "0") in new stack
-- Jumping to priority 8
-- Executing [s@func-apply-sipheaders:9] ExecIf("DAHDI/2-1", "0?SIPRemoveHeader(Alert-Info:)") in new stack
-- Executing [s@func-apply-sipheaders:10] ExecIf("DAHDI/2-1", "0?Set(PJSIP_HEADER(remove,Alert-Info)=)") in new stack
-- Executing [s@func-apply-sipheaders:11] Return("DAHDI/2-1", "") in new stack
  == Spawn extension (from-analog, 5035629971, 1) exited non-zero on 'DAHDI/2-1'
-- DAHDI/2-1 Internal Gosub(func-apply-sipheaders,s,1) complete GOSUB_RETVAL=
-- Called DAHDI/2/wXXXXXX9971
-- Hanging up on 'DAHDI/2-1'
-- Hungup 'DAHDI/2-1'

The biggest difference seems to be the lines relating to ‘simple_bridge’, but I don’t know what this is. Adding “w” in the trunk dialing prefix doesn’t change the result.

Of course, since both of these are in VMs, it’s trivial to go back and forth. If I have to I can just keep using 13, but I’d like to make 14 work if I can. I’ve compared configs: the DAHDI and wanpipe configs generated by sangoma-setup are identical except for the PCIBUS IDs that ESXI assigns to the different VMs. The asterisk configs are markedly different, especially ‘extensions_additional.conf’, but I think this would be expected. Besides extensions, I haven’t done much to modify the configs on either system.

Has anybody else had this issue, or gotten an A200 to work with FreePBX 14?

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@freak wrote:

I’m trying to run distro-upgrade on 10.13.66-22 and it fails. I get a bunch of these errors in the log file. I can download the files just fine with curl at the command line. The machine in question does not use a proxy. Partial error log below. Any tips?

Downloading failed: Errors were encountered while downloading packages.
avahi-libs-0.6.31-17.el7.x86_64: failure: Packages/avahi-libs-0.6.31-17.el7.x86_64.rpm from cmdline- instrepo: [Errno 12] Timeout on http://sng7.com/distro-upgrade/Packages/avahi-libs-0.6.31-17.el7.x86_64.rpm: (28, 'Operation too slow. Less than 1 bytes/sec transferred the last 30 seconds')

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@esarant wrote:

Hello,

I switched to asterisk 16 with (asterisk-version-switch) and during the installation I got the following from yum:
No package asterisk16-g729 available.

Running core show translation I get the following:

           ulaw  alaw   gsm  g726 g726aal2 adpcm slin8 slin12 slin16 slin24 slin32 slin44 slin48 slin96 slin192 lpc10  ilbc  g722 siren7 siren14 testlaw  opus silk8 silk12 silk16
     ulaw     -  9150 15000 15000    15000 15000  9000  17000  17000  17000  17000  17000  17000  17000   17000 15000 15000 17250  23000   23000   15000 23000 15000  23000  23000
     alaw  9150     - 15000 15000    15000 15000  9000  17000  17000  17000  17000  17000  17000  17000   17000 15000 15000 17250  23000   23000   15000 23000 15000  23000  23000
      gsm 15000 15000     - 15000    15000 15000  9000  17000  17000  17000  17000  17000  17000  17000   17000 15000 15000 17250  23000   23000   15000 23000 15000  23000  23000
     g726 15000 15000 15000     -    15000 15000  9000  17000  17000  17000  17000  17000  17000  17000   17000 15000 15000 17250  23000   23000   15000 23000 15000  23000  23000
 g726aal2 15000 15000 15000 15000        - 15000  9000  17000  17000  17000  17000  17000  17000  17000   17000 15000 15000 17250  23000   23000   15000 23000 15000  23000  23000
    adpcm 15000 15000 15000 15000    15000     -  9000  17000  17000  17000  17000  17000  17000  17000   17000 15000 15000 17250  23000   23000   15000 23000 15000  23000  23000
    slin8  6000  6000  6000  6000     6000  6000     -   8000   8000   8000   8000   8000   8000   8000    8000  6000  6000  8250  14000   14000    6000 14000  6000  14000  14000
   slin12 14500 14500 14500 14500    14500 14500  8500      -   8000   8000   8000   8000   8000   8000    8000 14500 14500 14000  14000   14000   14500 14000 14500   6000  14000
   slin16 14500 14500 14500 14500    14500 14500  8500   8500      -   8000   8000   8000   8000   8000    8000 14500 14500  6000   6000   14000   14500 14000 14500  14500   6000
   slin24 14500 14500 14500 14500    14500 14500  8500   8500   8500      -   8000   8000   8000   8000    8000 14500 14500 14500  14500   14000   14500 14000 14500  14500  14500
   slin32 14500 14500 14500 14500    14500 14500  8500   8500   8500   8500      -   8000   8000   8000    8000 14500 14500 14500  14500    6000   14500 14000 14500  14500  14500
   slin44 14500 14500 14500 14500    14500 14500  8500   8500   8500   8500   8500      -   8000   8000    8000 14500 14500 14500  14500   14500   14500 14000 14500  14500  14500
   slin48 14500 14500 14500 14500    14500 14500  8500   8500   8500   8500   8500   8500      -   8000    8000 14500 14500 14500  14500   14500   14500  6000 14500  14500  14500
   slin96 14500 14500 14500 14500    14500 14500  8500   8500   8500   8500   8500   8500   8500      -    8000 14500 14500 14500  14500   14500   14500 14500 14500  14500  14500
  slin192 14500 14500 14500 14500    14500 14500  8500   8500   8500   8500   8500   8500   8500   8500       - 14500 14500 14500  14500   14500   14500 14500 14500  14500  14500
    lpc10 15000 15000 15000 15000    15000 15000  9000  17000  17000  17000  17000  17000  17000  17000   17000     - 15000 17250  23000   23000   15000 23000 15000  23000  23000
     ilbc 15000 15000 15000 15000    15000 15000  9000  17000  17000  17000  17000  17000  17000  17000   17000 15000     - 17250  23000   23000   15000 23000 15000  23000  23000
     g722 15600 15600 15600 15600    15600 15600  9600  17500   9000  17000  17000  17000  17000  17000   17000 15600 15600     -  15000   23000   15600 23000 15600  23500  15000
   siren7 23500 23500 23500 23500    23500 23500 17500  17500   9000  17000  17000  17000  17000  17000   17000 23500 23500 15000      -   23000   23500 23000 23500  23500  15000
  siren14 23500 23500 23500 23500    23500 23500 17500  17500  17500  17500   9000  17000  17000  17000   17000 23500 23500 23500  23500       -   23500 23000 23500  23500  23500
  testlaw 15000 15000 15000 15000    15000 15000  9000  17000  17000  17000  17000  17000  17000  17000   17000 15000 15000 17250  23000   23000       - 23000 15000  23000  23000
     opus 23500 23500 23500 23500    23500 23500 17500  17500  17500  17500  17500  17500   9000  17000   17000 23500 23500 23500  23500   23500   23500     - 23500  23500  23500
    silk8 15000 15000 15000 15000    15000 15000  9000  17000  17000  17000  17000  17000  17000  17000   17000 15000 15000 17250  23000   23000   15000 23000     -  23000  23000
   silk12 23500 23500 23500 23500    23500 23500 17500   9000  17000  17000  17000  17000  17000  17000   17000 23500 23500 23000  23000   23000   23500 23000 23500      -  23000
   silk16 23500 23500 23500 23500    23500 23500 17500  17500   9000  17000  17000  17000  17000  17000   17000 23500 23500 15000  15000   23000   23500 23000 23500  23500      -

Also running g729 I get the following:

[root@pbx-octo ~]# g729
The Open Source G729 code is not installed.
You can install it with the following command:
         yum -y install asterisk16-g729
[root@pbx-octo ~]# yum -y install asterisk16-g729
Loaded plugins: fastestmirror, versionlock
Loading mirror speeds from cached hostfile
No package asterisk16-g729 available.
Error: Nothing to do

So, g729 is not installed. Is this intended or I need to make a ticket to freepbx issue tracker?
PBX Version is: 12.7.5-1807-1.sng7

Thanks,
esarant

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@Cricchetto wrote:

Hi, I need information on XactView.

• Can it help to see and manage microphones for conference participants?
• Is it a commercial form connected to the purchase of iSymphony or can it be used freely on its own?
• I tried to see how to activate it, but maybe it’s not in my pbx version. Can you add?
  My purpose is only to manage the on-off of the conference microphones, the switchboard does not have to do anything else, it’s very trivial. I think it is useless to use the fop2 which has too many useless things in this case.

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@MAA wrote:

Hello,

For example i have:
Trunks: trunk1, trunk2, trunk3
Inbound routes: route1, route2, route3

now I need to send all calls from trunk2 through route2.
how to do it?

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@supertin wrote:

I have an old 32 bit FreePBX install on a little rackmount server that I need to upgrade to SNG7. I don’t mind a couple of hours downtime while upgrading, but I need to continue using the same hardware.

Is my best option to run a backup from within FreePBX web GUI, then reinstall and restore the backup? Will SNG7 restore a 10.x backup, or will I need to make sure I’m running a matching version of the 64, and restore to that before upgrading?

I gather the migration script is the preferred method, but I also gather this needs 2 machines. I can arrange this, but would need to run the migration twice… And I also want to keep the call history, which apparently needs to be done manually anyway.

Thoughts?

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@GeekBoy wrote:

In advanced settings, the Create Voicemail Hints is already set to yes.

Is res_mwi_blf.so still being used for this?

freepbx*CLI> module show like res_mwi_blf.so
Module Description Use Count Status Support Level
0 modules loaded

freepbx*CLI> module load res_mwi_blf.so
Unable to load module res_mwi_blf.so
Command ‘module load res_mwi_blf.so’ failed.

This is all the hints I see

> freepbx*CLI> core show hints
> 
>     -= Registered Asterisk Dial Plan Hints =-
> 107@ext-local       : PJSIP/107&Custom:DND  State:Idle            Presence:not_set         Watchers  4
> 103@ext-local       : PJSIP/103&Custom:DND  State:Idle            Presence:not_set         Watchers  0
> 108@ext-local       : PJSIP/108&Custom:DND  State:Idle            Presence:not_set         Watchers  4
> 109@ext-local       : PJSIP/109&Custom:DND  State:Idle            Presence:not_set         Watchers  1
> *8575@park-hints    : park:75@parkedcalls   State:Idle            Presence:not_set         Watchers  0
> *8574@park-hints    : park:74@parkedcalls   State:Idle            Presence:not_set         Watchers  0
> *8577@park-hints    : park:77@parkedcalls   State:Idle            Presence:not_set         Watchers  0
> *8576@park-hints    : park:76@parkedcalls   State:Idle            Presence:not_set         Watchers  0
> *8571@park-hints    : park:71@parkedcalls   State:Idle            Presence:not_set         Watchers  0
> *8573@park-hints    : park:73@parkedcalls   State:Idle            Presence:not_set         Watchers  0
> *8572@park-hints    : park:72@parkedcalls   State:Idle            Presence:not_set         Watchers  0
> *8578@park-hints    : park:78@parkedcalls   State:Idle            Presence:not_set         Watchers  0
> 111@ext-local       : PJSIP/111&Custom:DND  State:Unavailable     Presence:not_set         Watchers  4
> 110@ext-local       : PJSIP/110&Custom:DND  State:Unavailable     Presence:not_set         Watchers  4
> 72@park-hints       : park:72@parkedcalls   State:Idle            Presence:not_set         Watchers  0
> 73@park-hints       : park:73@parkedcalls   State:Idle            Presence:not_set         Watchers  0
> 70@park-hints       : park:71@parkedcalls&  State:Idle            Presence:not_set         Watchers  0
> 71@park-hints       : park:71@parkedcalls   State:Idle            Presence:not_set         Watchers  0
> 76@park-hints       : park:76@parkedcalls   State:Idle            Presence:not_set         Watchers  0
> 77@park-hints       : park:77@parkedcalls   State:Idle            Presence:not_set         Watchers  0
> 74@park-hints       : park:74@parkedcalls   State:Idle            Presence:not_set         Watchers  0
> 75@park-hints       : park:75@parkedcalls   State:Idle            Presence:not_set         Watchers  0
> 78@park-hints       : park:78@parkedcalls   State:Idle            Presence:not_set         Watchers  0
> _*96X.@ext-cf-hints : Custom:DEVCF${EXTEN:  State:Unavailable     Presence:                Watchers  0
> 299@ext-local       : PJSIP/299&Custom:DND  State:Unavailable     Presence:not_set         Watchers  0
> _*76X.@ext-dnd-hints: Custom:DEVDND${EXTEN  State:Unavailable     Presence:                Watchers  0
> _*80X.@ext-local    : ${DB(AMPUSER/${EXTEN  State:Unavailable     Presence:                Watchers  0
> _*21X.@ext-findmefol: Custom:FOLLOWME${EXT  State:Unavailable     Presence:                Watchers  0
> _*98X.@app-dialvm   : MWI:${EXTEN:3}@${DB(  State:Unavailable     Presence:                Watchers  0
> ----------------
> - 29 hints registered

Ideas on this one?

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@srcfreepbx wrote:

Hi,
I’m getting flags from my nessus vulnerability scanner for my FreePBX 14 about php 5.6.36, regarding CVE-2018-19935 and CVE-2018-19158. Recommended action is to upgrade to php >= 5.6.39, I’m wondering if this is something that should be done manually, or will the upgrade come through official channels.
Anyone else having the same issue?

Thanks

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@pelikana wrote:

Hello everyone, this is my first post. Currently using FreePBX 14.0.5.25 in our organisation. I am new to FreePBX systems at all. I want to achieve and use FreePBX to call and speak (not robot massege or etc) with my clients. The idea is this. Configurate PBX to auto take numbers from (csv, datebase) and dial them for me, after the call end to dial next number in this list lets say after 5m for example. What i need to make and use as modules, and is this possible to make? Ty in advance.

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