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How to convert all wav files into mp3 extensions in centos/linux at one go? is there any command?

shell script #!/bin/bash SAVEIF=$IFS IFS=$(echo -en "\n\b") for file in $(/usr/bin/find /usr/src/wav-folder/ -type f -name "*.wav") do   name=${file%%.wav}   lame -V0 -h -b 160 --vbr-new $name.wav $name.mp3 done IFS=$SAVEIFS

copy directory on a remote server via SCP

scp -rp source-directory root@IP:/path-of-directory -r means recursive -p preserves modification times, access times, and modes from the original file.

Run MySQL query from Asterisk Dialplan and get values

[MySQL] exten => 1234,1,Answer() exten => 1234,n,NoOp(${CALLERID(num)}) exten => 1234,n,Read(caller,,10) exten => 1234,n,Set(CID=${CALLERID(num)}) exten => 1234,n,NoOp(${caller}) exten => 1234,n,Set(userCID=${caller}) exten => 1234,n,MYSQL(Connect connid YOUR_IP username pwd Database_name) exten => 1234,n,MYSQL(Query resultid ${connid} select * from customer where caller='${userCID}') exten => 1234,n,MYSQL(Fetch fetchid ${resultid} id name) exten => 1234,n,SayAlpha(${name}) exten => 1234,n,Saydigits(${id}) exten => 1234,n,MYSQL(Clear ${resultid}) exten => 1234,n,MYSQL(Disconnect ${connid}) exten => 1234,n,Playback(vm-goodbye) Tested on Asterisk 11.22

Memory leaks in chan_sip.c with realtime peers

  Just comment the below line as I did     31402                 /* Startup regular pokes */   31403                 if (!devstate_only && enablepoke) {   31404                         /*sip_ref_peer(peer, "schedule qualify");*/   31405                         sip_poke_peer(peer, 0);   31406                 }

Increase import limit in vtiger

Under /modules/Import $ImportConfig = array( 'importTypes' => array( 'csv' => array('reader' => 'Import_CSV_Reader', 'classpath' => 'modules/Import/readers/CSVReader.php'), 'vcf' => array('reader' => 'Import_VCard_Reader', 'classpath' => 'modules/Import/readers/VCardReader.php'), 'default' => array('reader' => 'Import_File_Reader', 'classpath' => 'modules/Import/readers/FileReader.php') ), 'userImportTablePrefix' => 'vtiger_import_', // Individual batch limit - Specified number of records will be imported at one shot and the cycle will repeat till all records are imported 'importBatchLimit' => '2500', // Threshold record limit for immediate import. If record count is more than this, then the import is scheduled through cron job 'immediateImportLimi...

pbx.c:4926 pbx_extension_helper: No application 'MeetMe' for extension

Solution: pbx*CLI> module load app_meetme.so

res_config_odbc.c:1250 require_odbc: Realtime table booking@meetme requires column 'members', but that column does not exist!

Solution: The members column is not used by WMM, and you can disable it with a logmembercount=no in meetme.conf

Dial an exten 157, record the message, hang up, then create call files to dial out to overhead page unit and announce message.

exten => 157,1,Goto(overhead-pager,s,1) [overhead-pager] ;------record the over head page file----------------------------------------- exten => s,1,Answer() exten => s,n,Wait(2) exten => s,n,Record(pager/currentpage.ulaw,5,60,)    ;mkdir /var/lib/asterisk/sounds/pager exten => s,n,Wait(2) exten => s,n,Hangup exten => t,1,Hangup() exten => h,1,Goto(page-app,ohp,1) [page-app] ;------over head page (ohp) call file setup------------------------------------ exten => ohp,1,Set(spooldir=/var/spool/asterisk) exten => ohp,n,System(echo "Channel: SIP/102" > ${spooldir}/page.call)    ;set the proper sip device to call exten => ohp,n,System(echo "MaxRetries: 0" >> ${spooldir}/page.call) exten => ohp,n,System(echo "RetryTime: 60" >> ${spooldir}/page.call) exten => ohp,n,System(echo "WaitTime: 45" >> ${spooldir}/page.call) exten => ohp,n,System(echo "Context: page-app...

How to convert all wav files into mp3 in one go on centos?

cat abc.sh #!/bin/bash SAVEIF=$IFS IFS=$(echo -en "\n\b") for file in $(ls *wav) do name=${file%%.wav} lame -V0 -h -b 160 --vbr-new $name.wav $name.mp3 done IFS=$SAVEIFS sh abc.sh run in that folder where wav file exists and want to convert it into mp3 OUTPUT LAME 3.99.5 64bits (http://lame.sf.net) polyphase lowpass filter disabled Encoding 1515151515.226426-030327-123456789.wav       to 1515151515.226426-030327-123456789.mp3 Encoding as 8 kHz single-ch MPEG-2.5 Layer III VBR(q=0)     Frame          |  CPU time/estim | REAL time/estim | play/CPU |    ETA   5871/5871  (100%)|    0:01/    0:01|    0:01/    0:01|   361.29x|    0:00  64 [5871] ******************************************************************************************************************************************************************************************...

Error in SugarCRM

Currently we are not able to locate the GD library, as a result you will not be able to use the CSS Sprite functionality Solution: yum install gd gd-devel php-gd service httpd restart

Suite CRM permission problem

chmod -R 777 cache custom modules themes data upload config_override.php

How to install lame in centos 6,7 with Asterisk 11,13

cd /usr/local/src wget https://sourceforge.net/projects/lame/files/lame/3.99/lame-3.99.5.tar.gz/download tar -xvf lame-3.99.5.tar.gz rm lame-3.99.5.tar.gz cd lame-3.99.5 ./configure make make install

Click to Call - Jquery with PHP (GUI)

PHP part <span class="contact_number"><?=$row['contact_no'];?></span> PHP Jquery part <script src="js/alertify.js"></script> <script type="text/javascript"> function reset () { $("#toggleCSS").attr("href", "css/alertify.default.css"); alertify.set({ labels : { ok     : "OK", cancel : "Cancel" }, delay : 5000, buttonReverse : false, buttonFocus   : "ok" }); } </script> <script type="text/javascript"> $(document).ready(function(){ $('.contact_number').click(function(){ var number = $(this).text(); var res = number.substr(-10); var option = $('#ext').val(); $.get('http://IP/click2call/cc.php',{num: 0+res, user: option}, function(data){ if(data){ reset(); alertify.success("Dialing 0" +res); return false; ...

Click to Call from PHP via Asterisk Manager

<?php $strHost = "127.0.0.1"; $strUser = "username";#specify the asterisk manager username you want to login with $strSecret = "password";#specify the password for the above user $strChannel = "SIP/".$_GET['user']; $strContext = "call-file"; $strWaitTime = "30"; $strPriority = "1"; $number=$_GET['num']; $oSocket = fsockopen ($strHost, 5038, &$errno, &$errstr, 20); fputs($oSocket, "Action: login\r\n"); fputs($oSocket, "Events: off\r\n"); fputs($oSocket, "Username: $strUser\r\n"); fputs($oSocket, "Secret: $strSecret\r\n\r\n"); fputs($oSocket, "Action: originate\r\n"); fputs($oSocket, "Channel: $strChannel\r\n"); fputs($oSocket, "WaitTime: $strWaitTime\r\n"); fputs($oSocket, "Exten: $number\r\n"); fputs($oSocket, "Context: $strContext\r\n"); fputs($oSocket, "Priority: $strPriorit...