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Apply Config - Registration Issues

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@MrXirtam wrote:

FreePBX 14.0.13.23
Current Asterisk Version: 13.29.2

I have noticed this issue for some time now and through at least a couple of FreePBX upgrades. Anytime I do a change, the Apply Config button shows. As soon as I press it to apply the changes, all phones drop registration and they will not register back again unless I either reboot the entire server, or I log into the CLI and issue the fwconsole restart command. Once I do that, phones register back up and everything is back to normal. I thought Apply Config was supposed to do the same thing, but doing it alone seems to break registrations. The trunks stay connected and registered, it’s just phones. I’ve noticed that this issue happens on multiple different FreePBX servers, both running different phones (Polycom and Yealink). Any ideas on this? Anything I can pay attention to while going through this process to help narrow down the cause?

Thanks!

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Noob questions about LDAP workflow

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@oliv2831 wrote:

Hello,

I’m completely new to LDAP.
I’ve just read [1] but I would appreciate to better understand how to use both FreePBX and LDAP.

General questions:
Do LDAP directories commonly implement helpers functions that keep people from attributing an already allocated or incorrect resource ?

FreePBX/OpenLDAP connector refers to “User Extension Link Attribute”. Does this attribute exists in every Directory based on OpenLDAP is it needed to create it ?

FreePBX/Active Directory connector refers to “ipphone” attribute. Does this attribute exists in every Active Directory ?

Is it possible to interface FreePBX with 389-ds ?

Scenario 1: Adding a new user and its telephony resources
Can you really set everything within LDAP Directory and let it create extension (I saw “Create Missing Extensions” in FreePBX/LDAP connector settings) ?
If positive, what is the event that triggers extension creation ?
Is there a Sync button within FreePBX GUI that create this extension or is the extension created whenever the LDAP Directory is searched by FreePBX and the need for extension creation is detected ?

Scenario 2: Removing user and its telephony resources
Is it possible from FreePBX GUI to reset some LDAP attributes for some LDAP entries ?

[1] https://wiki.freepbx.org/display/FPG/User+Management+with+OpenLDAP

Best regards

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PJSIP stops accepting calls and console flooded with WARNING messages

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@rymes wrote:

We have installed a new system that has been running reasonably well until recently. It is a virtual machine that is connected to a PRI via a Sangoma Vega device. All phones are Cisco 7960Gs running SIP firmware and the phones and the Vega are all connected using PJSIP. Three times in the last week we have had to restart the machine because all calls in and out fail.

When the system experiences the issue, the asterisk console is flooded with warning messages and the Asterisk full log (/var/log/asterisk/full) grows to multiple Gigabytes in a 24-hour period (one was 14 GB!). The console is completely useless when this is occurring, due to the amount of text scrolling by, so you have to run any commands via “asterisk -rx” to be able to read the output, even with verbosity set to 0. The Warnings look like this:

[2019-12-31 07:59:22] WARNING[13592][C-00000013]: taskprocessor.c:1110 taskprocessor_push: The 'stasis/m:cache_pattern:1/channel:all-000000ab' task processor queue reached 500 scheduled tasks again.
[2019-12-31 07:59:22] WARNING[13592][C-00000013]: taskprocessor.c:1110 taskprocessor_push: The 'stasis/m:cache_pattern:1/channel:all-000000ab' task processor queue reached 500 scheduled tasks again.
[2019-12-31 07:59:22] WARNING[13592]: logger: ***: Log queue threshold (1000) exceeded.  Discarding new messages.
[2019-12-31 07:59:22] WARNING[10968]: logger: ***: Logging resumed.  1055 messages discarded.
[2019-12-31 07:59:22] WARNING[13592]: logger: ***: Log queue threshold (1000) exceeded.  Discarding new messages.
[2019-12-31 07:59:22] WARNING[10968]: logger: ***: Logging resumed.  1399 messages discarded.
[2019-12-31 07:59:22] WARNING[13609][C-00000014]: taskprocessor.c:1110 taskprocessor_push: The 'stasis/m:cache_pattern:1/channel:all-000000ab' task processor queue reached 500 scheduled tasks again.
[2019-12-31 07:59:23] WARNING[13592][C-00000013]: taskprocessor.c:1110 taskprocessor_push: The 'stasis/m:cache_pattern:1/channel:all-000000ab' task processor queue reached 500 scheduled tasks again.
[2019-12-31 07:59:23] WARNING[13592]: logger: ***: Log queue threshold (1000) exceeded.  Discarding new messages.
[2019-12-31 07:59:23] WARNING[10968]: logger: ***: Logging resumed.  550 messages discarded.
[2019-12-31 07:59:23] WARNING[13592]: logger: ***: Log queue threshold (1000) exceeded.  Discarding new messages.
[2019-12-31 07:59:23] WARNING[10968]: logger: ***: Logging resumed.  1476 messages discarded.
[2019-12-31 07:59:23] WARNING[13592][C-00000013]: taskprocessor.c:1110 taskprocessor_push: The 'stasis/m:cache_pattern:1/channel:all-000000ab' task processor queue reached 500 scheduled tasks again.
[2019-12-31 07:59:23] WARNING[13592]: logger: ***: Log queue threshold (1000) exceeded.  Discarding new messages.
[2019-12-31 07:59:23] WARNING[10968]: logger: ***: Logging resumed.  519 messages discarded.
[2019-12-31 07:59:23] WARNING[13609][C-00000014]: taskprocessor.c:1110 taskprocessor_push: The 'stasis/m:cache_pattern:1/channel:all-000000ab' task processor queue reached 500 scheduled tasks again.
[2019-12-31 07:59:23] WARNING[13609][C-00000014]: taskprocessor.c:1110 taskprocessor_push: The 'stasis/m:cache_pattern:1/channel:all-000000ab' task processor queue reached 500 scheduled tasks again.
[2019-12-31 07:59:23] WARNING[13592]: logger: ***: Log queue threshold (1000) exceeded.  Discarding new messages.
[2019-12-31 07:59:23] WARNING[10968]: logger: ***: Logging resumed.  663 messages discarded.
[2019-12-31 07:59:23] WARNING[13592]: logger: ***: Log queue threshold (1000) exceeded.  Discarding new messages.
[2019-12-31 07:59:23] WARNING[10968]: logger: ***: Logging resumed.  952 messages discarded.
[2019-12-31 07:59:23] WARNING[13609][C-00000014]: taskprocessor.c:1110 taskprocessor_push: The 'stasis/m:channel:all-000000ac' task processor queue reached 500 scheduled tasks again.
[2019-12-31 07:59:23] WARNING[13592][C-00000013]: taskprocessor.c:1110 taskprocessor_push: The 'stasis/m:cache_pattern:1/channel:all-000000ab' task processor queue reached 500 scheduled tasks again.

I have found this blog post on the subject, and as a result I have disabled a number of modules we are not using, like Zulu, DUNDi Lookup Registry, iSymphony, Paging Pro, Sangoma Property Management, CRM, etc., but I don’t have any confidence that those changes will result in any progress, as the blog post and the log messages don’t make clear to me what is driving this. System load doesn’t seem to be an issue.

Can anyone point me to any details to what might be causing this? Which portions of FreePBX use Stasis?

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FreePBX Statistics shows exception error for undefined index in psi.memory.Swap

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@carriba wrote:

Have fired the usual command line update commands for FreePBX:

sudo fwconsole ma updateall
sudo fwconsole reload

Then when logging into the FreePBX web GUI, going to the “Dashboard” tab and incidentally selecting any time span under the “Memory” option of the “FreePBX Statistics” displet, I’m getting a red coloured pop-up window as depicted with:

Haven’t seen these red coloured pop-window in the past, thus not sure if it’s been introduced with the last FreePBX framework update.

With today’s update launched, the FreePBX framework version is now pumped up to 13.0.197.21.

Is this a bug or regression introduced with the latest update?

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Caller ID when Follow me to Mobile

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@inteleweb wrote:

HAPPY NEW YEAR TO ALL!!

I am using FreePBX 14

I have a provider that lets me use any CID that I want

I am forwarding incoming calls to a virtual extension that has follow me set up to forward to a list of mobile phones.

How can I get that the caller’s ID is used as the CID displayed on the mobile phones

Regards

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Music on Hold stream stopped playing suddenly

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@munozj wrote:

I use one of our DIDs to test line quality for remote employees and I use a stream from our broadcast encoder as the feed.

/usr/bin/mpg123 -q -r 8000 -f 8192 --mono -s http://10.255.96.50:8000

Today during some testing I was hearing the stream then after dialing back a few times the stream quit. I can listen to the stream directly from the url on my computer. Is there a service or something that may need to be restarted to get the stream flowing again?

I seam to remember this happening before and they only fix I could find to was to reboot the entire pbx.

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Is it possible to use your 'away' status to redirect calls?

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@dan_ce wrote:

Is it possible to somehow use your phone’s away/active status to redirect calls? Sort of like a day/night call flow toggle.

It might not be possible because somehow I guess you’d have to be able to insert the check on internal (extension to extension) calls

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Asterisk 13.23 PBX failed to create 2020 folder in /var/spool/asterisk/monitoring/ for Call Recordings

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@trevlyng wrote:

Hello All,

We use the Call Recordings and this morning Asterisk was failing to write the call recording files, checking Asterisk we got the message:

[2020-01-02 09:05:00] WARNING[29123][C-000266c8] file.c: Unable to open file /var/spool/asterisk/monitor/2020/01/02/q-500-6036777863-20200102-090459-1577973899.8005911.WAV: No such file or directory
[2020-01-02 09:05:00] ERROR[29123][C-000266c8] app_mixmonitor.c: Cannot open /var/spool/asterisk/monitor/2020/01/02/q-500-6036777863-20200102-090459-1577973899.8005911.WAV

Looking further into it we realized that there was no 2020 directory within /var/spool/asterisk/monitoring/ hence the title. After manually creating a 2020 directory and performing a “chown” to make sure it had the same permissions/owner/group Asterisk was able to auto-generate the month and day directories (e.g. /var/spool/asterisk/monitoring/2020/01/02) and everything is working as intended now. Looking in the logs I can’t seem to find anything specifically relating to attempting to create the directory or anything like that. Has anyone come across this before? Thanks!

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FreePBX 15 GUI does not configure Asterisk properly!

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@shetab wrote:

Hello All,
Happy New Year!

I’have recently deployed FreePBX 15 with Asterisk 16 and I have odd issues. When click on “Apply Config” sometimes it takes too long and then shows xhr empty respoonse error message which never seen in FreePBX 13. Also, it seems it does not configure asterisk properly because same config I used on FreePBX 13 does not work on 15. For example, I have deleted a trunk and when run asterisk -x “sip show peers”, it still shows deleted trunk.

Also I have some issues such sip accounts always raise forbidden 403 error even when username and secret is correct. When I set “Allow Annonymus” to enabled, incoming calls from trunk works but if I turn it off, it does work. I have used exactly same config for the trunk as I was using on 13. Bellow is my sip settings for the trunk:

Outgoing:

username=username
type=peer
transport=tcp
secret=password
qualify=yes
nat=yes
insecure=very
host=hostname
dtmfmode=rfc2833
context=from-internal
canreinvite=no
authenticate=username
allow=all

USER Context: username

Incomming:

username=username
secret=password
type=peer
qualify=yes
nat=yes
insecure=very
host=hostname
fromdomain=hostname
dtmfmode=rfc2833
context=from-internal
canreinvite=no
allow=all

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YUM Repo -- Not Using FreePBX Distro

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@hardocp wrote:

I am currently running a CentOS Linux release 7.7.1908 (Core) with both Asterisk v16 and Freepbx installed. When I originally setup my server I installed everything from scratch and source and its been working great for some time now. The issue I had / have is that in order to update asterisk – I need to rebuild from Source / scratch and honestly that is getting annoying. So I decided to try the following instructions in order to get away from building from source and move to a Repo update process with yum

After following the above – asterisk did appear to install / update to the latest version – however it was not running quite right –

One thing I noticed was that PJSIP did not get installed and so none of my phones or trunks would work – since that was not working – I did not really dig too deeply in what else might not be working

I then tried a yum install asterisk* (which appeared to state that PJSIP was installed and the latest) – followed by a yum install asterisk* --skip-broken (since a bunch of stuff was throwing off errors)

While this downloaded every sound file from every language – it still did not fix my asterisk

Ultimately – I downloaded and re-installed the latest version of asterisk 16 from source (16.7.0) which the yum update installed – had to run make install followed by make uninstall followed again by make install at the end of the process – then rebooted the server and it now looks like I got things back in order again (although I did loose my custom IVR recordings)

Anyway still after all that – since I do run quite a few asterisk instances – which were all built the same way – I would really like to get away from updates from source and onto a repo / yum update process

Can someone perhaps provide some guidance as to where I might have gone wrong or what I can do to use the repo you provide to keep my servers up to date

thanks

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EPM Licence on Virtual Machines

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@oliv2831 wrote:

Hello,

I’m about to install a new FreePBX VM that will use commercial EPM module.

Given license and other constraints, is it possible to use the following steps:
1- install and configure a new FreePBX setup as a KVM guest in a lab environment including its EPM module
2- export it as a VMDK image when configuration is over
3- maybe, converting it to VHDX image
4- import the VMDK or VHDX image into the target environment (with a different memory, CPU, …)
5- backup regularly target environment (snapshots, …)

Of course, at any time, only one VM either in lab or target environment is running or used.

Thoughts ?
Advices ?
Shall I impose myself to replicate some settings (MAC addresses, CPU, …) between both environments ?

Best regards

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I can’t remove the FreePBX modules that are unnecessary for me

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@plastilin wrote:

There are several modules that I do not plan to use in a small project. I delete them, and after a while they all recover again on their own. This has already been 3 times. How can you really remove unnecessary modules?

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New to VOIP

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@wryan wrote:

Hello. We upgraded our old phone system last year to FreePBX. The system runs good for a couple days and then incoming and outgoing calls start to get choppy. I’ve contacted xfinity, who has our internet service, and they see a lot of dropped packets from the modem. Our current service is 100mbps. When I restart the modem, the issue goes away for a couple days and then progressively gets worse. I’ve recently had the modem replaced and am still experiencing the same issues. The phones are on their own vlan. I’m thinking that there may be some network configuration issue. We are using Meraki devices on the network and have 20 phones at our main office. I’ve been in contact with the installer of the system and Xfinity. Xfinity has be restart the modem and the VOIP installer claims its an Xfinity issue. I’m thinking that it’s a network configuration issue. I’m new to this service. I would appreciate any help. Thank you.

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Is it realistic to target removing all ERRORS during startups and reloads?

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@oliv2831 wrote:

Hello,

On a newly installed FreePBX 15, I’m seeing a lot of various messages on console.
Among them many WARNING or ERROR as for instance:
WARNING[13421]: pbx.c:8732 ast_context_verify_includes: Context ‘macro-block-cf’ tries to include nonexistent context ‘macro-block-cf-custom’

Have you successfully removed all of them as this could help to quickly focus on real errors, if those ever happen ?

Best regards

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How to configure handset sidetone for Cisco 7945G on FreePBX 15.0.16.38

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@wgbecks wrote:

Hello.

I have a question regarding how I might configure my SEPconf.xml to enable handset sidetone for a Cisco 7945G working on FreePBX 15.0.16.38? I tried adding sideToneLevel to the vendorConfig section of my SEP.cnf.xm,l but no joy, it made no difference.

According to the Cisco documentation, handset/headset sidetone was support started at Firmware Version 9-3.x and my 7945G is at SIP45.9-4-2SR3-1S.

Any help would be appreciated!

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SSL Connect Error installing Sound Language

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@swalker2001 wrote:

I am getting an “SSL Connect Error” when attempting to install Allison’s English language pack (same error on any language pack). Sound Languages module is installed. I also removed it via cli and reinstalled it but no change.

This is FreePBX 13.0.197.14 and it is fully up to date. Full text of the error is:

Exception
SSL connect error
File:/var/www/html/admin/modules/soundlang/soundlang.class.php:1277

Not sure what to do. This is a fairly new install and I just need a single announcement set up but can’t do that until I have a language installed.

Any help would be appreciated.

Steve

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CID Superfecta for ConnectWise using ClientID REST API

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@ComputerBusines wrote:

I made a modification to the CID Superfecta for FreePBX Asterisk to incorporate the ConnectWise ClientID and REST API. Otherwise the SOAP API whichCID Superfecta uses is no longer working.

You will be required to request access to the ConnectWise Developer Network and create a private ClientID.

See developer https:// . connectwise . com/ClientID (Needed to break apart URL)

The modification is to the php script located on the FreePBX server at /var/www/html/admin/modules/superfecta/sources/source-ConnectWise.module

This will check the CW Companies first for the phone number match, returning the Company Name, and then, if not found, the CW contacts, returning the contact First & Last Name.

Cut & Paste the below into the source-ConnectWise.module. Watch out for line returns for lines that should be 1 line, pasting as 2.

Please freely share this outside of CW Forums as long as I am still provided credit. There is a way to have a developer ClientID for mutiple installs. Hopefully Superfecta Developers take this and use it with the Public Development ClientID so you do not have to create one for your own use.

‘’’

<?php

/***** ***** ***** ***** ***** ***** ***** ***** ***** ***** ***** *****
 * Module Dev notes:
 *
 *
 * Revision History:
 *    v0.1.0: Initial Release Version by myitguy
 *    v0.1.1: Minor bug fix by bushbomb
 *    v0.1.2: More bug fixes id'd by bushbomb
 *    v0.1.3: Initial migration to 2.11 by lgaetz
 *    2014-08-22  Added user param to set the API version in the URL
 *    2015-05-01  Add field for user supplied CNAM prefix
 *    2018-06-30  Fix for bug FREEPBX-17727
 *    2020-01-06  Modified by Clinton Pownall of Computer Business to
 *                accommodate the ConnectWise REST API and the ConnectWise
 *                Required ClientID.
***** ***** ***** ***** ***** ***** ***** ***** ***** ***** ***** *****/

class ConnectWise extends superfecta_base {

    public $description = "Look up data in local or remote ConnectWise CRM.";
    public $version_requirement = "2.11";
    public $source_param = array(
        'DB_Site' => array(
                'description' => 'ConnectWise Site URL not including the initial https://',
                'type' => 'text',
        ),
        'DB_Company' => array(
                'description' => 'ConnectWise Company ID',
                'type' => 'text',
        ),
        'DB_Public_Key' => array(
                'description' => 'Integration Public Key Created in ConnectWise: Members -> API Members -> API Keys',
                'type' => 'text',
        ),
        'DB_Private_Key' => array(
                'description' => 'Integration Private Key Created in ConnectWise: Members -> API Members -> API Keys',
                'type' => 'password',
        ),
        'DB_ClientID' => array(
                'description' => 'Client ID created using the ConnectWise Developer Site, See developer.connectwise.com/ClientID',
                'type' => 'password',
        ),
        'Search_Type' => array(
                'description' => 'The ConnectWise type of entries that should be used to match the number',
                'type' => 'select',
                'option' => array (
                    '1' => 'Companies Only',
                    '2' => 'Contacts Only',
                    '3' => 'Companies --> Contacts',
                ),
                'default' => '3',
        ),
        'Filter_Length' => array(
                'description' => 'The number of rightmost digits to check for a match. Enter zero to disable this setting',
                'type' => 'number',
                'default' => 10
        ),
        'CNAM_prefix' => array(
                'description' => 'This text will be prefixed to all CNAM returned by this module (optional)',
                'type' => 'text',
                'default' => null
        ),
        'API_Version' => array(
                'description' => 'The part of the URL excluding slashes following the host name that indicates the API version, e.g. "v2014_4" or "v4_6_release" (without quotes)',
                'type' => 'text',
                'default' => 'v2014_4',
        ),
    );


    function get_caller_id($thenumber, $run_param=array()) {
        $caller_id = null;
        
                $site = $run_param['DB_Site'];
                $companyid = $run_param['DB_Company'];
                $APIPublicKey = $run_param['DB_Public_Key'];
                $APIPrivateKey = $run_param['DB_Private_Key'];
                $varSearchType = $run_param['Search_Type'];
        $clientid = $run_param['DB_ClientID'];

        $varSearchType = $run_param['Search_Type'];
        if ($run_param['API_Version']) {
            $apiver = $run_param['API_Version'];
        } else {
            $apiver = 'v4_6_release';
        }

        $wquery_string = "";
        $wquery_result = "";
        $wresult_caller_name = "";

        $this->DebugPrint("Searching ConnectWise ... ");
        
        if ($run_param['Filter_Length'] != 0) {
            if (strlen($thenumber) > $run_param['Filter_Length']) $thenumber = substr($thenumber, -$run_param['Filter_Length']);  // keep only the filter_length rightmost digits
        }

        $APIEndpoint = 'https://' .$site. '/'.$apiver.'/apis/3.0';

        $APIkey = 'Basic ' . base64_encode($companyid . '+' . $APIPublicKey . ':' . $APIPrivateKey);


        // Search Companies
        if($run_param['Search_Type'] == 1 || $run_param['Search_Type'] == 3)  {
            $this->DebugPrint("Searching Companies ... ");
            /* COMPANY */
            $conditions='conditions=phoneNumber%20like%20%22'.$thenumber.'%22%20';

            $url = '/company/companies?'.$conditions.'&fields=name&pageSize=1000';

            $url = $APIEndpoint . $url;

            $curl = curl_init();
            curl_setopt($curl, CURLOPT_URL, $url);
            curl_setopt($curl, CURLOPT_HTTPHEADER, array('clientId:' . $clientid,'Authorization:'. $APIkey,    'Content-Type: application/json',));
               curl_setopt($curl, CURLOPT_RETURNTRANSFER, 1);
               curl_setopt($curl, CURLOPT_HTTPAUTH, CURLAUTH_BASIC);
               curl_setopt($curl, CURLINFO_HEADER_OUT, true);

            $result = curl_exec($curl);

            if(!$result){die("Connection Failure");}
            curl_close($curl);
        
            $name=preg_split('/"*"/i', $result, -1, PREG_SPLIT_NO_EMPTY | PREG_SPLIT_DELIM_CAPTURE);

            if (!empty($name[3])) {
                $wresult_caller_name = $name[3];
            }
            else
            {
                $this->DebugPrint("Not found in Companies... ");
            }
        }

        // Search Contacts
        if($run_param['Search_Type'] == 2 || $run_param['Search_Type'] == 3 && $wresult_caller_name =="")  {
            $this->DebugPrint("Searching Contacts ... ");
            /* CONTACT */
            $conditions='childconditions=communicationItems/value%20like%20%22'.$thenumber.'%22%20AND%20communicationItems/communicationType="Phone"';

            $url = '/company/contacts?'.$conditions.'&fields=&pageSize=1000';

            $url = $APIEndpoint . $url;

            $curl = curl_init();
            curl_setopt($curl, CURLOPT_URL, $url);
            curl_setopt($curl, CURLOPT_HTTPHEADER, array('clientId:' . $clientid,'Authorization:'. $APIkey,    'Content-Type: application/json',));
             curl_setopt($curl, CURLOPT_RETURNTRANSFER, 1);
            curl_setopt($curl, CURLOPT_HTTPAUTH, CURLAUTH_BASIC);
            curl_setopt($curl, CURLINFO_HEADER_OUT, true);
            $result = curl_exec($curl);

            if(!$result){die("Connection Failure");}
            curl_close($curl);

            $name=preg_split('/"*"/i', $result, -1, PREG_SPLIT_NO_EMPTY | PREG_SPLIT_DELIM_CAPTURE);;

            if (!empty($name[9])) {
                 $wresult_caller_name = $name[5]." ".$name[9]; }
            
            }
            if ($wresult_caller_name =="")  {
                $this->DebugPrint("Not found in Contacts... ");
            }

        
            if(strlen($wresult_caller_name) > 0) {
                $caller_id = $run_param['CNAM_prefix'].trim(strip_tags($wresult_caller_name));
                return $run_param['CNAM_prefix'].$caller_id;
            }
            else {
                $this->DebugPrint("Not found in ConneceWise");
            }
    }
}

?>

‘’’

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SSL Connect Error installing Sound Language

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@swalker2001 wrote:

I am getting an “SSL Connect Error” when attempting to install Allison’s English language pack (same error on any language pack). Sound Languages module is installed. I also removed it via cli and reinstalled it but no change.

This is FreePBX 13.0.197.14 and it is fully up to date. Full text of the error is:

Exception
SSL connect error
File:/var/www/html/admin/modules/soundlang/soundlang.class.php:1277

Not sure what to do. This is a fairly new install and I just need a single announcement set up but can’t do that until I have a language installed.

Any help would be appreciated.

Steve

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Call goes to voicemail but ask for a passcode

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@smwein wrote:

For some reason we get people calling into our system and when the call hits the voicemail it will ask them for a passcode. The system seems to be answer the call with a *98 code before the voicemail. I can’t figure out why this happens to only some calls. Here is snipit from the call cdr.

|Tue, 7 Jan 2020 13:47|CHAN_END|DEFAULT|h|ext-local||Local/301@from-queue-000034fa;2|

|—|---|—|---|—|---|—|
|Tue, 7 Jan 2020 13:47|BRIDGE_ENTER|DEFAULT|s|macro-dial-one|AppDial|Local/*988030@from-internal-00003500;1|
|Tue, 7 Jan 2020 13:47|BRIDGE_ENTER|DEFAULT|s|macro-dial-one|Dial|Local/324@from-queue-000034fe;2|
|Tue, 7 Jan 2020 13:47|BRIDGE_ENTER|DEFAULT|6507|from-queue|AppQueue|Local/324@from-queue-000034fe;1|
|Tue, 7 Jan 2020 13:47|BRIDGE_ENTER|DEFAULT|6507|ext-queues|Queue|SIP/sbc.questblue.com-00006b88|
|Tue, 7 Jan 2020 13:47|APP_START|DEFAULT|dvm8030|from-internal|VoiceMailMain|Local/*988030@from-internal-00003500;2|
|Tue, 7 Jan 2020 13:47|BRIDGE_EXIT|DEFAULT|6507|ext-queues|Queue|SIP/sbc.questblue.com-00006b88|
|Tue, 7 Jan 2020 13:47|BRIDGE_EXIT|DEFAULT|6507|from-queue|AppQueue|Local/324@from-queue-000034fe;1|
|Tue, 7 Jan 2020 13:47|HANGUP|DEFAULT|6507|from-queue|AppQueue|Local/324@from-queue-000034fe;1|
|Tue, 7 Jan 2020 13:47|CHAN_END|DEFAULT|6507|from-queue|AppQueue|Local/324@from-queue-000034fe;1|
|Tue, 7 Jan 2020 13:47|BRIDGE_EXIT|DEFAULT|s|macro-dial-one|Dial|Local/324@from-queue-000034fe;2|
|Tue, 7 Jan 2020 13:47|BRIDGE_EXIT|DEFAULT|s|macro-dial-one|AppDial|Local/*988030@from-internal-00003500;1|
|Tue, 7 Jan 2020 13:47|HANGUP|DEFAULT|s|macro-dial-one|AppDial|Local/*988030@from-internal-00003500;1|
|Tue, 7 Jan 2020 13:47|CHAN_END|DEFAULT|s|macro-dial-one|AppDial|Local/*988030@from-internal-00003500;1|
|Tue, 7 Jan 2020 13:47|APP_END|DEFAULT|dvm8030|from-internal|VoiceMailMain|Local/*988030@from-internal-00003500;2|
|Tue, 7 Jan 2020 13:47|HANGUP|DEFAULT|h|from-internal||Local/*988030@from-internal-00003500;2|
|Tue, 7 Jan 2020 13:47|CHAN_END|DEFAULT|h|from-internal||Local/*988030@from-internal-00003500;2|

Thanks

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Any security issues with modifying /var/www/html/.htaccess file?

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0
0

@gdesilva wrote:

Hi,

I had to modify the .htaccess file in order to get the directory service working on Cisco7970G phone. In particular, I just had to modify the mime type by adding one line, ‘AddType text/xml .xml’ as the phone does not like the default mime type of text/html.

Now FreePBX Framework Module warns that this file is tampered with and considers this as a critical security issue.

Is it safe to ignore this warning?

Thanks

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