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 Vicidial Custom incoming Call Report

 Vicidial incoming Call Report create an all calls report in a nice table with the phone number, date and time of call, term reason, seconds per call, and agent on the call using this code Create call_report.php in /srv/www/htdocs <?php     $url1=$_SERVER['REQUEST_URI'];     header("Refresh: 40; URL=$url1");     echo "<h1>Calls Past 7 Days</h1>";     echo "<h5>*Refreshes every 40s</h5>";     $servername="localhost";     $username="cron";     $password="1234";     $dbname="asterisk";     $conn=new mysqli($servername, $username, $password, $dbname);     if ($conn->connect_error) {        die("Connection failed:  " . $conn->connect_error);     }     $query="select phone_number, call_date, status, term_reason, length_in_sec, user from vicidial_log where call_date > timestampadd(day, -7, now()) ;";   ...

Text to Speech Online

  Indian English https://ttsmp3.com/text-to- speech/Indian%20English/ http://www.fromtexttospeech. com http://ctrlq.org/listen/ Gujrati Language:- https://www.narakeet.com/app/text-to-audio/?projectId=d4bd805e-0cbf-4fb7-a7a4-006cebde8574 https://ttsfree.com/text-to-speech/gujarati-india Converter:-  https://g711.org/

Adding Listen, Whisper, and Barge to FreePBX or Asterisk

  If you are running a call center on FreePBX or Asterisk, most likely you will want the ability to listen in on agents calls, also known as joining multiple calls, or connected two calls to a manager, or other variations of barging in on a bridged channel. For the purpose of this article, we will define  manager as the callee who is the spying channel,  agent who is the spied-on channel, and  client who is the bridged channel / 3rd party. We define listen, whisper, and barge as follows: Listen: Monitor an agents call. The manager can hear both the agent and client channels, but no-one can hear the manager. Whisper:   Whisper to the agent. The manager can hear both the agent and client channels, and the agent can also hear the manager, but the client can only hear the agent, hence “whisper.” Barge: Barge in on both channels. The manager channel is joined onto the agent and client channels, and all parties can hear each other. Be warned, if the or...

Click to Call API for Asterisk/Freepbx

Click to Call API for Asterisk/Freepbx     Create call.php file in /var/www/html/ paste below mention code in call.php and replace context and manager username password <?php error_reporting(E_ALL); ini_set('display_errors', 1);   $strUser = "ccserver"; #specify the asterisk manager username you want to login with $strSecret = "ccserver"; #specify the password for the above user $strHost = "127.0.0.1"; //Clean up EXT $ext = $_GET['exten']; $ext = filter_var($ext, FILTER_SANITIZE_NUMBER_INT); $ext = preg_replace("/[^0-9,.]/", "", $ext); $strChannel = "SIP/".$ext; $strContext = $_GET['cnt']; $strWaitTime = "30"; #Wait Time before hangin up $strPriority = "1"; $strMaxRetry = "2"; #maximum amount of retries if(isset($_GET['number'])){ $number=strtolower($_GET[' number']);} elseif(isset($_GET['phone'])){ $number=strtolower($_GET[' phone']); } /...

TATA SIP WITH ISABEL PBX

  https://community.freepbx.org/ t/p-preferred-identity/14261 I added following to extensions_custom.conf but no lcuk! [from-internal-custom] exten => X,1,SipAddHeader(P-Preferred- Identity: sip:${CALLERIDNUM} ) I added followings to trunk settings. now working good!! sendrpid=pai trustrpid=no

Elastix/IssabelPBX/ASTERISK API

Elastix API Elastix and Asterisk API Provider to make FreePBX easy to manage, written in PHP4. Installation Open your Elastix (CentOS) terminal, and go to elastix directory, then download it: cd /var/www/html git clone https://github.com/101t/elastix-api.git Open api.php and generate token key then replace it in: $ this -> key = 'YOUR_SECRETKEY_50_RANDOM_CHARS' ; Now you have elastix api ready to use it. Useful API functions This api package can implement: Get Authentication Get SIP Peers Get SIP Extensions Check System Resources Get CDR Report Get *.wav files Get Hard Driver State Check IPTable Status Get Active Call (Live Calls) SIP Trunk / Extension Management (Create, Update, Delete, Read) Follow Me Extension Management (Create, Update, Delete, Read) Documentation Installation and developer guide here .

Allow recording link/filename on web with full name

  To enable web browsing of Recordings on Asterisk server, add the below line  at last line of httpd.conf   Alias /RECORDINGS/ “/var/spool/asterisk/ monitorDONE/” <Directory “/var/spool/asterisk/ monitorDONE”> Options Indexes MultiViews AllowOverride None Order allow,deny Allow from all <files *.mp3> Forcetype application/forcedownload </files> </Directory> Restart the Apache web server to apply the changes service httpd restart chkconfig httpd on

How to delete old call logs and other logs in vicidial or goautodial

  Description: How to delete old call logs and other logs in vicidial or goautodial. Step 1 : SSH to the server using the Putty Step 2:   login to mysql by typing   mysql -p                    (if you dont know password try below command )               mysql -ucron -p1234 Step 3 : select the asterisk database by typing               use asterisk step 4: Run the below command to check total disk occupied by asterisk database                SELECT table_schema AS "asterisk", ROUND(SUM(data_length + index_length) / 1024 / 1024, 2) AS "Size (MB)" FROM information_schema.TABLES GROUP BY table_schema; Step 5 : Run the below command to check disk space consumed by each table in asterisk database SELECT table_name AS "Table", ROUND(((data_length + index_length) / 1024 / 1024), 2) AS "Size (MB)" ...