@Pietropf wrote:

Hello,
I'm currently having a problem when I try to install the module XMPP. I've tried using the console and the interface, but none of them worked.
This message appears on the console:

    [npm-cache] [INFO] [npm] running [npm install]...
      /tmp/shelljs_c84f103a11c07b6890e2:4
        fs.writeFileSync('/tmp/shelljs_cd560880cf12f1f05bb7', err ? err.code.toString() : '0');
                                                                            ^

      TypeError: Cannot read property 'toString' of null
          at /tmp/shelljs_c84f103a11c07b6890e2:4:71
          at ChildProcess.exithandler (child_process.js:205:5)
          at emitTwo (events.js:106:13)
          at ChildProcess.emit (events.js:191:7)
          at maybeClose (internal/child_process.js:891:16)
          at Process.ChildProcess._handle.onexit (internal/child_process.js:226:5)
      shell.js: internal error
      Error: Command failed: "/usr/bin/node" /tmp/shelljs_c84f103a11c07b6890e2
          at checkExecSyncError (child_process.js:472:13)
          at Object.execSync (child_process.js:512:13)
          at execSync (/home/asterisk/.node/lib/node_modules/npm-cache/node_modules/shelljs/src/exec.js:81:11)
          at Object._exec (/home/asterisk/.node/lib/node_modules/npm-cache/node_modules/shelljs/src/exec.js:214:12)
          at Object.exec (/home/asterisk/.node/lib/node_modules/npm-cache/node_modules/shelljs/src/common.js:182:23)
          at CacheDependencyManager.installDependencies (/home/asterisk/.node/lib/node_modules/npm-cache/cacheDependencyManagers/cacheDependencyManager.js:49:13)
          at CacheDependencyManager.loadDependencies (/home/asterisk/.node/lib/node_modules/npm-cache/cacheDependencyManagers/cacheDependencyManager.js:237:18)
          at startManager (/home/asterisk/.node/lib/node_modules/npm-cache/index.js:117:15)
          at /home/asterisk/.node/lib/node_modules/npm-cache/node_modules/async/lib/async.js:181:20
          at Object.async.forEachOf.async.eachOf (/home/asterisk/.node/lib/node_modules/npm-cache/node_modules/async/lib/async.js:233:13)

      Finished updating libraries!
      Starting new Xmpp Process...


        [Symfony\Component\Process\Exception\ProcessFailedException]
        The command "runuser 'asterisk' -c 'cd /var/www/html/admin/modules/xmpp/node/node_modules/lets-chat && mkdir -p /home/asterisk/.pm2
        && mkdir -p /var/www/html/admin/modules/xmpp/node/node_modules/lets-chat/logs && export HOME="/home/asterisk" && export PM2_HOME="/h
        ome/asterisk/.pm2" && export ASTLOGDIR="/var/log/asterisk" && export ASTVARLIBDIR="/var/lib/asterisk" && export PATH="$HOME/.node/bi
        n:$PATH" && export NODE_PATH="$HOME/.node/lib/node_modules:$NODE_PATH" && export MANPATH="$HOME/.node/share/man:$MANPATH" && /var/ww
        w/html/admin/modules/pm2/node/node_modules/pm2/bin/pm2 start /var/www/html/admin/modules/xmpp/node/node_modules/lets-chat/app.js  --
        name '\''xmpp'\'' -e '\''/var/log/asterisk/xmpp_err.log'\'' -o '\''/var/log/asterisk/xmpp_out.log'\'' --merge-logs --log-date-format
         '\''YYYY-MM-DD HH:mm Z'\'''" failed.
        Exit Code: 1(General error)
        Output:
        ================
        Error Output:
        ================
        bash: line 0: cd: /var/www/html/admin/modules/xmpp/node/node_modules/lets-chat: No such file or directory

Can you guys help me?

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@alenguav wrote:

Hello,
I was wondering if there is any tutorial for installing FreePBX on Alibaba cloud.
Time ago I installed it on a VULTR VPS using these feature https://www.vultr.com/features/uploadiso/
but it seems that I can't do the same on Alibaba cloud

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@KingBeetle wrote:

How I implemented Flowroute's SMS services into my X-Lite softphone's IM feature.

Introduction and background

As many of you are no doubt aware, SMS messaging is not really an integrated part of FreePBX/Asterisk. In fact as far as I know, SIPStation is the only SIP Trunk provider that offers integrated SMS capabilities. Other providers offer SMS, but they don't work out of the box like SIPStation. However for my basic home VoIP needs, their offerings aren't a lot cheaper than the services I already had through my provider.

I did some shopping around and decided on Flowroute for my SIP trunk. Their basic services were within my budget, and they offer SMS. Now came the fun part; making it work with the X-Lite instant messenger. From what I understand, the SIP SIMPLE protocol used by the X-Lite softphone also works with a variety of SIP phones as well, but I don't own any SIP phones and don't really have a way to verify that my solution works on those phones as well.

How Flowroute's SMS services work

Flowroute's SMS service works via a basic RESTful HTTP interface, using simple POST and GET HTTP requests.

Through Flowroute's account manager, their SMS API can be "pointed" to an SMS Callback URL, a specific address on your network. Any incoming SMS messages they receive for one of your DIDs is sent to this callback URL in the form of an HTTP request. Software on your end receives the message, and it's up to you to take it from there.

They also provide a URL on their network to receive outgoing SMS messages, utilizing an HTTP request as well. Again, it's up to software on your end to format, prepare, and send the message.

How I handle incoming SMS messages

Using Apache and a perl script, I receive an incoming SMS message and use the information to create a .call file that I drop into Asterisk's outgoing call directory. Asterisk takes it from there and sends the message to a specific softphone extension as an IM.

How I handle outbound SMS messages

Using a custom context stored in Asterisk's custom extension file, instant messages are examined to determine if the destination is a local extension or a 10 digit phone number. If it's a 10 digit phone number, the message data is passed on to a bash script where it's formatted into a valid HTTP request and sent to Flowroute through cURL.

Overall, it's a working solution. It's a bit of a kludge, but it does work. I have been considering writing up a more detailed explanation with an explanation of the underlying code, if anyone is interested.

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@zara wrote:

Dear all
I have installed asterisk 13.17 and freepbx 13 according to
FOP-InstallingFreePBX13onUbuntuServer14.04.2LTS-120817-1342-4146.pdf on my zynq SoC.
after installation, connecting from browser to freepbx is right but freepbx can not connect to asterisk.
the error is FreePBX cannot connect to Asterisk via AMI.
could you help me please.
best regard.

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@digitalb wrote:

We use the update procedures located in the wiki (https://wiki.freepbx.org/display/PPS/Upgrading+from+Distro+6) and on phase two the system gets stuck in a loop when trying to install/upgrade through yum.

I see several errors simular to this one:

Error: Package: sangoma-pbx-1707-3.sng7.noarch (sng-pkgs)
Requires: asterisk-sounds-extra-en-ulaw
Removing: asterisk-sounds-extra-en-ulaw-1.5-2.sng7.noarch (installed)
asterisk-sounds-extra-en-ulaw = 1.5-2.sng7
Available: asterisk-sounds-extra-en-ulaw-1.5-1.2.noarch (sng-pkgs)
asterisk-sounds-extra-en-ulaw = 1.5-1.2
Available: asterisk-sounds-extra-en-ulaw-1.5-1.3.noarch (sng-pkgs)
asterisk-sounds-extra-en-ulaw = 1.5-1.3

As it continues the proccess it gets to "Running transaction test" then provides the error:

Transaction check error:
file /var/lib/asterisk/sounds/en/core-sounds-en.txt from install of asterisk-sounds-core-en-wideband-1.5-2.sng7.noarch conflicts with file from package asterisk-sounds-core-en-gsm-1.4.25-94_centos5.noarch

Any suggestions on where to start to trouble shoot this issue?

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@alenguav wrote:

Hello,
I was wondering if anybody has managed to install FreePBX on Alibabacloud
I tried an ISO mounting method in VULTR
but it seems that this facility is not available in Alibaba

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@kyiu wrote:

I've just installed FreePBX-32bit-10.13.66 on my system. The system is stuck on "upgrade your machine to version 10.13.66-21" days. I tried to run the update on the commend-line "sysadmin_update_system"...it come back with "Update already in progress!". How can fix this issue?

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@teleute wrote:

We had a user discover that her calls were not coming through. We looked, and in the FreePBX GUI, it showed her extension as being set for DND. We tried toggling DND on and off on her phone, with no change. Tried resetting the phone to factory, and then a new phone - still no change. Clearly it wasn't set on the phone, but I couldn't find any place on the server to toggle this for the extension. Finally I tried logging into the UCP as the user (which we do not use, and the passwords were just auto-set from the initial setup - I can 100% guarantee this user did not go into this page at any point), and I could see that DND was showing as on in here. I set it to Off, and now everything is fine. This is bringing up a number of questions:

1. This can't be the only place that this is controlled, right? Surely there's some central/admin access to this?
2. Any other ways (key strokes, specific error conditions, etc...) this could get toggled if we did not do it?
3. How is this expected to interact with the DND codes from the phone? Having it on here appears to override whatever the phone is saying, rather than them working together...is this expected behaviour?
4. The extension list in FreePBX only seems to show the extension as DND if it's turned on in this fashion, rather than on the phone directly - is this also expected?

I'm not sure I'll be able to figure out what happened in this one odd possible glitch, but I'd like at a minimum to understand how these two different DND implementations are expected to work together.

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@KingBeetle wrote:

At the thrift store today I scored an 8x8 DTA310 ATA brand new in the box for $7. It came with proprietary firmware for use with an VoIP provider that doesn't even exist anymore, with no way to even configure the SIP settings. Strike one.
Thanks to the trusty internet, I was able to determine that the actual manufacturer of the hardware was a company called Leadtek, they called it a BVA8051, and their firmware was unlocked. Unfortunately, they haven't manufactured it for about 10 years, and no longer offered the firmware on their website. Strike two!
Back to the trusty internet I went, scouring Google for answers. Lo and behold, I found the firmware online!
I had to actually downgrade the firmware to an earlier version of 8x8's firmware before I could then upgrade it to the Leadtek firmware, but I managed to install the unlocked Leadtek firmware.
I set up an extension, entered the appropriate SIP connection information into the ATA and....... it wouldn't register despite my best efforts. Strike three! (or so I thought)
Turns out that the SIP password is limited to 20 characters. I couldn't tell that when I was cutting and pasting the secret from the FreePBX GUI, since it was replacing the entered characters with dots. I had spent HOURS looking for answers online, changing untold configuration settings both in the ATA and in the extension setup in FreePBX, and it turned out to be a d%^n 20 character password limit in the device.
I deleted the extension and created a new one with default settings and chopped the newly created secret back to the first 20 characters, and hardware reset the ATA (I had made so many changes trying to make it work, I lost track of what the defaults were!).
Logged into the ATA, entered my SIP information including the chopped off 20 character secret, and BOOM! Registered and ready to go.
If anyone has one of these devices or finds one cheap and wants to use it with Asterisk, that's the answer. I have both the 8x8 firmware needed to install the Leadtek firmware, and the latest Leadtek firmware if anyone finds them useful. I know this is an old piece of hardware and not something that would even see use in a professional environment, but perhaps someone like me looking for a cheap ATA to add another analog phone to their home network might find them of value.
Note that if you are dealing with the 8x8 DTA 310, be sure and install the 8x8 firmware BEFORE trying to install the Leadtek firmware, or you stand a pretty good chance of BRICKING the device forever.

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@btomjack wrote:

Hello,

I'm running FreePBX 14.0.1.1 and Asterisk 13.17.0 and seeing some odd behavior with a ring group. The group has just two members one is an internal extension (1010) and the other is an outside phone number. The confirm calls option is set to yes.

When a call comes into the ring group the phones will ring as expected. If i answer before the 15 second ring timeout I get the call confirm and its acts normally. If i do not answer the call rings and the stops for a split second (or if i ignore the call on both phones, after the 15 second ring timeout) both phones start ringing again. This time if i answer it's a direct connection no confirmation needed and if i don't answer the external cell phone's voicemail grabs the call.

Anyone else seen this kind of behavior?

What logs might I have that would be helpful to dig deeper?

thanks,
Brian

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@kristiandg wrote:

Good evening, all (well, at least for some it might be)...

Ran an all module update this evening, and apparently in the middle of it, the HD became 100% full, causing the update to fail. Once I figured out what was causing it, I was able to remove the excess logs and clear space, but by then the update had already been stopped and the system is now essentially down.

Running the fwconsole command merely responds with:
SELECT value FROM admin WHERE variable = 'version'

SQLSTATE[HY000]: General error: 1017 Can't find file: 'admin' (errno: 2)::

My hope was to try installing all modules again (with a FORCE switch), but that's all contingent upon "fwconsole", which fails. I then thought, well maybe I can run the upgrade script, but that also failed because it essentially updates the modules via "fwconsole".

Does anyone have any thoughts on this one?

Thanks.

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@mitterhuemer wrote:

Hello,

A new Hotel with 45 Rooms wants a PBX Solution. I dont know if FreePBX is the right for this project.

I know there is a Hotel Wakeup module, but how can they bill the customer for external calls they made?

I dont want to build a own Sandbox 3rd Party Solution for the customer.

Is there an supported way from Sangoma howto build a Hotel PBX Solution with PBXAct?

Thank you for all answers.

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@cinhau wrote:

Guys, I've got problem when do fresh install FreePBX 13 (10.13.66-64bit) with Asterisk 13 (latest download from website).. Got error "driver not found" message with option select driver, select driver from disk and cancel if I'm not wrong.. possibly the problem with ethernet driver..

I do installation on HP Prodesk 600 G2 MD and Intel NUC6i3, both using USB Thumb Drive and DVD, neither is working..

Is there any solution available for this error? Have tried SNG7 and got error fwconsole not found..

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@datatechnv wrote:

I have 3 PBX's connected via IAX(trough VPN).
Site-1 extensions 11xx. site-2 extensions 12xx. site-3 extensions 13xx
extension-to-extension dialing between the 3 pbx's works fine.

All have separate incoming PSTN lines and have IVR on these incoming lines.
IVR setup has "enable direct dial" ENABLED.

External callers can dial anyone's entension in the PBX they called into. But they cannot dial an extension on another PBX.(the IVR responds: We have not received a valid response)

How do I get this to work?

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@ryanhs wrote:

I just installed from wiki.freepbx.org FreePBX 13 CentOS 7 following the direction exactly and it i was able to access the web interface at first however and restarting services I am getting:

On the web page
Whoops \ Exception \ ErrorException (E_WARNING)
file(/etc/asterisk/asterisk.conf): failed to open stream: Permission denied

In freepbx log:
CRITICAL admin/bootstrap.php:270 - Connection attmempt to AMI failed

This is my second install getting the exact same result. I assumed i did something wrong during the first install so I installed centos7 fresh and performed the install again.

I have run fwconsole chown
I have performed every step as root user
fwconsole reload / restart are successful
/etc/amportal.conf, /etc/freepbx.conf /etc/asterisk/*.conf are owned by asterisk:asterisk and are 664 -rw-rw-r--

I am not sure where to go from here and am not having success finding a resolution for this problem. I would appreciate any help you could throw my way. Thank you in advance.

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@sorvani wrote:

I upgraded my FreePBX 13 instance to FreePBX 14 on Friday night. I have a weird issue with TLS registration not working well, but otherwise everything seemed to be working.

Today I finally setup my UCP because I will be working from the back porch for a while and I use the UCP to start calls (I wear a DECT Headset).

Well the UCP works and I setup the widgets I want, but the data is all from prior to the upgrade.

This is a small system and I can easily blow it up and install clean, but I wanted to test the upgrade process.

image.png1101x774 45.6 KB

I also no longer see a way to dial from the UCP. So I will have to use FOP2 for that.

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@travisryan123 wrote:

Running FreePBX 10.13.66-21

I am trying to add an external number to be part of a ring group. I did test adding it directly to the ring group extension list (ex. 5555555555#) and it worked correctly, ringing the incoming call to the external number (my cell phone).

However I cannot add it to the ring group this way because I want to be able to prepend a number to the CallerID number (which I can do via Set CallerID). So I tried adding it to the ring group another way after reading the forums.

I created a Misc Application- Feature code: 701 Destination: Misc Destination. The Misc Destination dial is set to my cell phone (ex. 15555555555).

When I dial 701 from a phone in the office it calls my cell phone, however when I add 701 to the ring group my cell phone doesn't ring.

I also tried adding a custom extension with FindMe/FollowMe enabled with initial ring time of 0 with my cell phone in the follow-me list (ex. 5555555555#). Again when I dial that extension from an office phone it calls my cell phone. But when I add it to the ring group, the incoming call does not ring to my cell phone.

I've tested just about every setting and can't seem to get it to work.

Please help.

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@parny wrote:

I recently did a test run of the FreePBX migration tool from the FreePBX.org website.

My old system is running Elastix 2.5, which is running on top of FreePBX 2.11.10 r17.

I installed the new system using FreePBX-SNG7-FPBX-64bit-1707-1 on a bootable USB key. I had purchased the System Builder Starter Bundle and the Endpoint manager before installing and activating the new system.

The migration went fine. I had to use the -insecure option on the old system (curl --insecure...).

After the install I noticed a few issues that I think I've resolved.

The dashboard said I had a number of invalid destinations in my IVR. Turns out voice mail is deactivated for all extensions on the new FreePBX system.

Anyone know how to enable voicemail for multiple extensions at a time?

When I tried to enable voicemail for an extension, I got an error message from the parkpro module -- something about parkpro.class.php (sorry, didn't take a screenshot).

Turns out my parking lot didn't copy over correctly. I created a new parking lot in the parking application in FreePBX with the same settings as on our old Elastix boxe. After that I was able to turn on voice mail and set a password for an extension without getting the error message.

I then checked my ring groups and my IVR, both of which were using voicemails as destinations, and made sure they were pointed to the correct voicemails. This cleared the bad destinations error in the FreePBX dashboard.

Got a few more things to check and then I hope to do a live migration next week after doing a fresh install of FreePBX onto the new system.

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@bkost wrote:

I realize this may be an asterisk installation issue. I have already posted this question on the asterisk forums. But I wish to cover my bets and ensure that it is not a freepbx issue.

Fresh install of asterisk 14 and Freepbx 14 succeeds without errors but it does not install binaries.

Actually the directory '/var/lib/asterisk/bin' is not installed at all. After installing and reinstalling many times from source and wasting a week it remains an issue. I've read some place that there may be a bug that ignores or silently hides an error during installation.

Asterisk does start and run. I can also access it this way 'asterisk -rvvv'.

Could this be a asterisk issue? probably because I'm under the impression that fwconsole and other binaries are installed by the asterisk installation. If nothing else, confirm for me that this is an asterisk issue.

System Info -
Debian: 9.1 Stretch (x86_64)
php: 7.1.8
mysql: 14.14 dist 5.7.19
apache2: Server version: Apache/2.4.25 (Debian), Server built: 2017-07-18T18:37:33

directory structure after successful install: /var/lib/asterisk# l
total 48
drwxr-xr-x 2 root root 4096 Aug 17 16:58 agi-bin
drwxr-xr-x 3 root root 4096 Aug 17 16:58 documentation
drwxr-xr-x 3 root root 4096 Aug 17 16:58 firmware
drwxr-xr-x 2 root root 4096 Aug 17 16:58 images
drwxr-xr-x 2 root root 4096 Aug 17 16:58 keys
drwxr-xr-x 2 root root 4096 Aug 17 16:58 moh
drwxr-xr-x 2 root root 4096 Aug 17 17:01 phoneprov
drwxr-xr-x 2 root root 4096 Aug 17 16:58 rest-api
drwxr-xr-x 2 root root 4096 Aug 17 16:58 scripts
drwxr-xr-x 3 root root 4096 Aug 17 16:58 sounds
drwxr-xr-x 2 root root 4096 Aug 17 16:58 static-http
drwxr-xr-x 3 root root 4096 Aug 17 16:58 third-party
mybox:/var/lib/asterisk#

Followed installation instructions:
here: https://wiki.asterisk.org/wiki/display/AST/Installing+Asterisk+From+Source (success no bin directory)
and here: http://www.francoconidi.it/asterisk-14-freepbx-14-su-debian-stretch-9-1/ (success no bin directory)

compiled with the --with-pjproject-bundled

All comments and help are welcomed. Perhaps this is something very simple that I'm overlooking after staring at this for a week.

Sincerely,
Ben

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@parny wrote:

I'm testing the migration script from FreePBX.org.

My old system is running Elastix 2.5, which is running on top of FreePBX 2.11.10 r17.

I installed the new system using FreePBX-SNG7-FPBX-64bit-1707-1 on a bootable USB key. I had purchased the System Builder Starter Bundle and the Endpoint manager before installing and activating the new system.

I'm working on figuring out how to copy the CDR (call records) from the Elastix system to the FreePBX system. I don't know mysql very well, so I'm learning as I go.

I think I've figured it out. Can anyone see any problems with the below approach?

On Elastix system at Linux prompt:

mysqldump -p asteriskcdrdb cdr > elastix.sql (I needed to use a password here, hence the -p option)

Copy elastix.sql to another PC with mysql installed.
Create a new database, and run/import elastix.sql into that database. This recreates the cdr table from Elastix. I used the HeidiSQL graphical interface on a windows PC.

On FreePBX system at the linux prompt:

mysqldump asteriskcdrdb cdr > freepbx.sql (I didn't need to use a password here)

Compare the table definitions from elastix.sql and freepbx.sql; the FreePBX table had a few new columns.

In HeidiSQL on the Windows PC run the following SQL commands to add the extra columns to the table:

alter table cdr
add did varchar(50) COLLATE utf8mb4_unicode_ci NOT NULL DEFAULT ''
after userfield;
alter table cdr
add linkedid varchar(32) COLLATE utf8mb4_unicode_ci NOT NULL DEFAULT '';
alter table cdr
add peeraccount varchar(80) COLLATE utf8mb4_unicode_ci NOT NULL DEFAULT '';
alter table cdr
add sequence int(11) NOT NULL DEFAULT '0';

Sill in HeidiSQL on the windows PC, right click on the cdr table and choose "export as sql" to newfreepbx.sql.

Edit newfreepbx.sql to comment out any "drop table" "drop database" "create database" and "create table" SQL commands at the beginning of the file. I did this because I wanted to keep using the same database and table that FreePBX originally created.

copy newfreepbx.sql to the FreePBX system. At the linux prompt, in the same directory you copied newfreepbx.sql to:

mysql
use asteriskcdrdb
source newfreepbx.sql

And this imports the cdr records! I tried a couple of reports from the FreePBX reports/cdr reports and they look ok. Note that reports are limited to 100 results by default -- I kept wondering why I was only getting 100 call records until I noticed this setting on the right of the report query screen.

If you can see anything in the above that will cause me problems when I do a live migration, please let me know. Thanks!

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