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Backup & Restore Module, ssh server, PHP Fatal error

June 21, 2018, 12:34 pm
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@cramermp wrote:

Entering up a FreePBX 14 Backup & Restore config. All modules are up to date. I created an SSH Server, generated the keys, tested scp connectivity and can transfer files using the key generated. So, I enter the settings into FreePBX, and when I run a backup I get:

Saving Backup 3...done!
Initializing Backup 3
Connecting to remote server...
bash: php: command not found
Something went wrong when connecting to remote server. Aborting!
PHP Fatal error: Function name must be a string in /var/www/html/admin/modules/backup/bin/backup.php on line 108
Whoops\Exception\ErrorException: Function name must be a string in file /var/www/html/admin/modules/backup/bin/backup.php on line 108
Stack trace:
1. Whoops\Exception\ErrorException->() /var/www/html/admin/modules/backup/bin/backup.php:108

Ran fwconsole chown, fwconsole ma downloadinstall backup, same problem. What gives?

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Problem whith SCCP Cisco CP6921

June 22, 2018, 3:03 am
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@felymania wrote:

Good morning,
i have some problem whith Cisco phome model CP6921, if i wont transfer call whith another device the button dos,t work.
See below my configuration :

;User XXX
[SEPXXXXXXXXXX]
button = line, XXX, default
button = line, XXX
description =USER XXX
devicetype = 6921
park = on
button = line, XXX, default
type = device
transfer = on
cfwdbusy = on
cfwdnoanswer = on
directed_pickup = on
directed_pickup_context = from-internal-xfer
directed_pickup_modeanswer = on
dnd = reject
directrtp = off
earlyrtp = progress
mwilamp = on
mwioncall = off
cfwdall = on
softkeyset = softkeydefaults
tzoffset = 2

[XXX]
id = XXX
type = line
pin = 1234
label = Name User
cid_name = Name User
cid_num = xxx
description = xxx Name User
incominglimit = 2
trnsfvm = 1000
callgroup = 2
pickupgroup = 2
secondary_dialtone_digits = 9
secondary_dialtone_tone = 0x22

Please send me feedback to solve this problem.
Best regards

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Please help a complete newb with SiPVicious

June 22, 2018, 12:48 pm
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@SiP1701 wrote:

I have fail2ban running, changed my SSH port number but I still have phantom calls from EXT 1000 non stop.
Do I need to create an IPtable like sonething below? If so, how would I do that?

-A INPUT -p udp -m udp --dport 5060 -m string --string “friendly-scanner” --algo bm --to 500 -j DROP

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Unable to get iSymphony V3.5.3 (Server) to install

June 22, 2018, 3:34 pm
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@jonathanrupp wrote:

I would believe this is a FreePBX Install topic because iSymphony is part of the package and the client is included. Now my issue is with the Server piece. I have tried downloading it from i9 without any problem. However, I can’t get it to install onto my Bistro version of FreePBX. A little background. I am running a Bistro version of the latest version of FreePBX. I have it licensed on several different levels for our needs. But I would like to see how iSymphony works. I’ve tried creating a stand alone VM client or something and even that’s hard to set up. But again it wont work since I don’t have the server up and running. I’m getting fed up with this as I’m not understanding the help wiki. Either its out dated or I am …

44%20PM
Screen Shot 2018-06-22 at 6.32.44 PM.png1235x784 113 KB

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Apply config going forever

June 22, 2018, 6:14 pm
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@carlyle705 wrote:

Platform: FreePBX 14 with asterisk 13/15 distro.
Hardware: Intel Dual-Core, 2T hard drive, 4GB RAM
Fresh Install, No extensions and trunks yet.

Symptoms: “apply config” hang after one or two successful one. I tried asterisk 15 and 13, both are same. Every time I have to “fwconsole restart” and it can’t kill asterisk. Then I kill the asterisk and retrieve_conf manually, then do “fwconsole restart” again, it will back to normal. After one or two “apply config”, same thing happens aggain. Sometimes retrieve_conf CPU usage 99%. I uninstalled all unused modules, such as digum module, and followed some instructions here, but all didn’t work, very frustrated. I have IncrediblePBX setup on my RaspberryPi3, it runs smoothly without any problem, I was just thinking that I’m using SD card instead of SSD/hard drive on the PI, so I would like to switch it to a desk-top. I also did the update. It’s been so many days. Any help would be appreciated!

Thanks!

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LetsEncrypt Error: Connection refused when requesting certificates

June 23, 2018, 1:09 pm
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@andy_woolford wrote:

There was an error updating the certificate: Error ‘Requested ‘http://xxxxxxx.co.uk//.freepbx-known/b7ee2c4268b04a1ab097d4055f193896’ - Failed connect to xxxxxxxx.co.uk:80; Connection refused’ when requesting http://xxxxxxxxx.co.uk//.freepbx-known/b7ee2c4268b04a1ab097d4055f193896

This is a new install of FreePBX14 and I am trying to obtain certificates for the first time.

There is no external firewall. I have a FQDN and a public fixed IP which resolves perfectly.
Port 80 is pointed to the LetsEncrypt /.freepbx-known directory using the System Admin > Ports.
Port 443 is set on the HTTPS admin interface
Firewall is configured correctly (shows green) for all the letsencrypt and freepbx mirrors. I have explicitly added these to the “Trusted” zone on the firewall.

I have tested that I can actually access the /.freepbx-known directory by manually placing a “hello world” htm file inside that directory and pointing an external browser to it.

I have also checked that mod_rewrite is enabled in the 00-base.conf file located in /etc/httpd/conf.modules.d/ directory, and that the following entry exists and is not commented.

LoadModule rewrite_module modules/mod_rewrite.so

I have also restarted httpd and rebooted the entire machine, however the error persists.

I would welcome any thoughts on this.

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Migrating paid licence for System Admin module

June 23, 2018, 1:15 pm
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@andy_woolford wrote:

I have recently created a new VM and migrated the old FreePBX to a new FreePBX using the convert tool here: http://convert.freepbx.org/

However, this has resulted in me creating a new deployment ID and my licences are still on the old FreePBX. Two of these licenses are free, (zulu 2 user and extension routes), and I have re-applied these from the purchase page without issue, however I have one paid licence for the System Admin Module, which I would like to transfer to the new deployment.

I would be grateful if someone could point me in the right direction.

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WAV File not uploading to MOW

June 23, 2018, 3:59 pm
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@chuckjuhl wrote:

I’m having a problem with an new install of FreePBX 14 uploading a .wav file to MOH. Initially I received the error message that the file size was too large (the file is a wav file, 3095kb in size). The wav file uploaded in an older version (11) of FreePBX MOH without any issues. After reading a previous thread I increased the limits in the php.ini file (to memory_limit=256m, upload_max_filesize=256m, post_max_size=256mb). However, that did not allow the file to be uploaded in MOH. Now there is no error message, but the file just does not upload either by browsing to the file or drag and drop.

What am i doing wrong?

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Mail queue (1 message is queued on this machine)

June 24, 2018, 9:50 am
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@angelino wrote:

Hi, my freepbx machine has a few days the following error: Mail queue 1 message is queued on this machine, and has not been deliverd.

How can I fix this error?

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Enabling the Jitter Buffer for all PJSIP extensions

June 20, 2018, 6:57 am
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@nabberuk wrote:

I have around 60 extensions (all extensions are and will be remote), i’ve been reading on how to enable the jitter buffer for PJSIP channels and that i have to add a context for each extension.

Is there now a way to add a single line to enable on all extensions?

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Move appliance Sangoma FreePBX to VM as a backup

June 25, 2018, 5:38 am
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@sjegambit wrote:

Good day,

We have two Sangoma 100 appliances (one was as a spare), one had issues, luckily we had the other as backup -all is working ok.
As further measures, I would like to find out if we can virutalize the current unit with all the setups and configs as is (purely for redundancy)

Furthermore - should this be possible, I would like more info on the following:

  1. Can we use the Sangoma virtual appliance image with our Sangoma hardware platform licences or what do we need to do to run it as a VM and how do I go about this?
  2. Can these (appliance and VM) run in primary/secondary mode - as failover should the appliance fail?

Any other suggestions on this?

Basically I want to virtualize my appliance, have it constantly updating with the VM, should the appliance fail, kick up the VM and have little to no interruptions on any calls.

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Unable to dial out

June 25, 2018, 5:56 am
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@ditkar wrote:

I am a newbie. Just installed FreePBX(see version details below) and created 2 sip extensions. Both the extensions(pj_sip) work well. I was able to dial and receive between the two extensions without any problem.

I have a sip provider who has given me only the ip address. I created a PJSIP trunk and added the information that I could(however, I am not 100% sure).

When I dial an outbound number, all I get is “could not complete as dialed”.
Executing [9xxxxxxxxxx@from-internal:5] Playback(“PJSIP/1002-0000000b”, “silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer”) in new stack
…(I had to replace the original number dialed with an “x”)

Any help is really appreciated.

My Configuration:
FreePBX 14.0.3.6
Asterisk 13.19.1

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External calls to extension forwarded. Internal calls to extension allowed

June 25, 2018, 9:44 am
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@Luke_johnson02 wrote:

So, I need a solution to use find me follow me for all external to be forwarded however, internal call should go to the extension dialed. This is a head scratcher for me. How can I make this work?

1 – If an outside call comes in and they dial the extension of Keith Howard (we’ll say he’s 201, I don’t remember without looking) then the call doesn’t go to 201, instead it goes to 220 which is his assistant Penny Polk.

2 – Internal users should be able to dial 201 without getting redirected to 220.

3 – Absolutely no external calls should get through to 201, whether its from the directory or directly dialing the extension, they all need to go to Penny Polk (220)

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Templates not available in Endpoint Manager

June 25, 2018, 10:59 am
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@rober925 wrote:

I am trying to get suggestions for configuring or using comparable templates for 2 SIP gateways that we want to register with our FreePBX.

The gateways don’t have available templates in Endpoint Manager, and we wouldn’t need to enable autoprovisioning.

The 2 models are – Patton SN4171 and Tenor AXG2400. There is already a trunk configured on our FreePBX that goes to our SBC.

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Help with WebRTC

June 25, 2018, 11:04 am
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@fearx wrote:

Hey so I’m using FreePBX 13 with Asteriks 11, I would like to know if I can use WebRTC on another softphone or just through the user panel (UCP).
Any help on setting up a webphone (it will use a WebSocket) will be very welcome.

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Cannot install DAHDI Config module

June 25, 2018, 12:11 pm
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@GeekBoy wrote:

On FreePBX 13.0.195.4, when attempting to install module version 13.0.33.13, I get this:

Checking tables…Done
Whoops\Exception\ErrorException: syntax error, unexpected $end in Unknown on line 18
in file /var/www/html/admin/modules/dahdiconfig/functions.inc.php on line 161
Stack trace:

  1. Whoops\Exception\ErrorException->() /var/www/html/admin/modules/dahdiconfig/functions.inc.php:161
  2. Whoops\Run->handleError() :0
  3. parse_ini_string() /var/www/html/admin/modules/dahdiconfig/functions.inc.php:161
  4. dahdi_config2array() /var/www/html/admin/modules/dahdiconfig/includes/dahdi_cards.class.php:1005
  5. dahdi_cards->read_dahdi_scan() /var/www/html/admin/modules/dahdiconfig/includes/dahdi_cards.class.php:589
  6. dahdi_cards->load() /var/www/html/admin/modules/dahdiconfig/includes/dahdi_cards.class.php:173
  7. dahdi_cards->__construct() /var/www/html/admin/modules/dahdiconfig/install.php:650
  8. include_once() /var/www/html/admin/libraries/modulefunctions.class.php:2516
  9. module_functions->_doinclude() /var/www/html/admin/libraries/modulefunctions.class.php:2468
  10. module_functions->_runscripts() /var/www/html/admin/libraries/modulefunctions.class.php:1999
  11. module_functions->install() /var/lib/asterisk/bin/module_admin:66
  12. doInstall() /var/lib/asterisk/bin/module_admin:845

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Transfer Management Feature Codes

June 25, 2018, 12:48 pm
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@jerryriggin wrote:

During an attended transfer there is no way to abort the transfer. Assume the target of the attended transfer just keeps ringing, there is no way (out of the box) that FreePBX will abort the transfer and reconnect the caller for further action. This seems like such a basic function I was amazed that I could google and experiment for a whole day and not be able to solve it.

Finally Sangoma support (lgaetz) solved it for me:

"Spent some time experimenting with attended transfers using the in-call feature code, and found the same thing you did. No matter what I tried, I was unable to trigger an abort using the *1 code. I did find a working solution though, and that is to populate the file

features_general_custom.conf

atxferabort = *5
atxfercomplete = *2
atxferthreeway = *3
atxferswap = *4

Once I changed the atxferabort from the default to *5, it worked just fine. I genuinely don’t see any reason why *1 doesn’t work, there were no conflicts that I can see. Also in case you are unaware, the old way of aborting an attended transfer was to use the ** in-call disconnect feature. That has been deprecated since Asterisk 11, starting in 12 the above 4 features are used to control attended transfers. I particularly like the *3 attended transfer three way feature and the *4 swap channel."

Thanks, Lorne.

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Update Error PHP

June 25, 2018, 12:50 pm
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@simpson wrote:

Hi, i’m using nethserver and installed voip PBX. It installed Freepbx and its working. Now i want to update and i get some error.
It starts with :
fwconsole
PHP Parse error: syntax error, unexpected ‘class’ (T_CLASS), expecting identifier (T_STRING) or variable (T_VARIABLE) or ‘{’ or ‘$’ in /var/www/html/freepbx/admin/libraries/Composer/vendor/symfony/translation/Translator.php on line 90
Whoops\Exception\ErrorException: syntax error, unexpected ‘class’ (T_CLASS), expecting identifier (T_STRING) or variable (T_VARIABLE) or ‘{’ or ‘$’ in file /var/www/html/freepbx/admin/libraries/Composer/vendor/symfony/translation/Translator.php on line 90
Stack trace:

  1. Whoops\Exception\ErrorException->() /var/www/html/freepbx/admin/libraries/Composer/vendor/symfony/translation/Translator.php:90
    [root@home translation]#

What to do? Someone has an hint?

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Trunk name appearing in INVITE to address

June 25, 2018, 7:44 pm
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@jcrawfordor wrote:

I’m working on setting up a SIP trunk to be used just for E911 termination. So, I have a chan_sip trunk called emergency_trunk set up with a very minimal configuration (below). A call dialed to ‘911’ should be sent to 911@sip.provider.com. However, when calls go out the INVITE line gives an address of 911%40emergency_trunk@sip.provider.com. That is, the phone number dialed is having @trunkname added onto it somewhere internally.

This seems like it could be related to SIP proxy support or similar, but I don’t have anything like that configured.

Here’s the PEER details for the SIP trunk called emergency_trunk:

host=sip.provider.com
type=peer
disallow=all
allow=ulaw,g729
qualify=yes

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Restore Freepbx 13 backup to Freepbx 14

June 25, 2018, 9:27 pm
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@cliffz wrote:

I restored the backup from 13 to 14. but found the error. any solution is welcome thanks.

Cliff

Error(s) have occured, the following is the retrieve_conf output:
exit: 1
Unable to continue. SQLSTATE[42S22]: Column not found: 1054 Unknown column ‘calendar_enable’ in ‘field list’ in /var/www/html/admin/modules/findmefollow/Findmefollow.class.php on line 756
#0 /var/www/html/admin/modules/findmefollow/Findmefollow.class.php(756): PDOStatement->execute(Array)
#1 /var/www/html/admin/modules/findmefollow/functions.inc.php(1144): FreePBX\modules\Findmefollow->get(‘111’)
#2 /var/www/html/admin/libraries/usage_registry.functions.php(300): findmefollow_getdestinfo(‘ext-local,111,d…’)
#3 /var/www/html/admin/libraries/usage_registry.functions.php(351): framework_identify_destinations(Array, Array)
#4 /var/lib/asterisk/bin/retrieve_conf(818): framework_list_problem_destinations(Array, false)
#5 {main}

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