Virtual Receptionist / Virtual Assistant

What is Virtual Receptionist ? Stay in touch with customers when the office is closed or staff members are out of the office by using Virtual Office. Incoming phone calls will be answered by one of our trained remote receptionists. The receptionist will then ask the caller who he or she is trying to reach … Read more

what you need to know, before you deploy the VoIP in your organization : QoS (Quality of Service) for VoIP

The organization(s) that migrate from legacy PABX system (traditional phone system) to IP based PBX (Private Branch eXchange), they require a very carefully planning before they deploy the VoIP. They need to ensure the QoS (Quality of Service) for VoIP in your organization. Step 1 Analyze your existing LAN (Local Area Network) and find if … Read more

Asterisk Server Consultant

Sample DAHDI installation After compiling and installing of dahdi and asterisk, you have to perform some further steps to use your hardware. This example will show you a few steps how to get asterisk and two Digium cards enabled: 1.) Detect your hardware   # (this will generate /etc/dahdi/system.conf and  /etc/asterisk/dahdi-channels.conf)  linux:~# dahdi_genconf 2.) Read … Read more

phpMyAdmin + CentOS 6.0 – Forbidden : phpmyadmin allow remote access centos

To remove and add access to phpmyadmin in asterisk : Add your IP as follow: vi /etc/httpd/conf.d/phpmyadmin.conf <Directory “/usr/share/phpmyadmin”> Order allow,deny Allow from all </Directory> Here in Allow from all and Allow from 192.168.0.10 Then issue the following command: # service httpd restart Now All done.      

SIP error code

SIP error code Code Meaning 1xx Informational — request received, continuing to process the request 2xx Success — the action was successfully received, understood, and accepted 3xx Redirection — further action needs to be taken in order to complete the request 4xx Client Error — the request contains bad syntax or cannot be fulfilled at … Read more

sip show channel command not found

my problem is that I cannot run ‘sip show peers’ or any other sip commands from the CLI.   Solution:- Check your /etc/asterisk/sip.conf    file carefully. Find the missing one and restart the asterisk server (service asterisk restart) and go to asterisk cli prompt (asterisk -r) and type the SIP command. 🙂        

call transfer or call forwarding in asterisk

call transfer or call forwarding in asterisk [test] exten => 1000,1,Dial(SIP/User1) exten => 1000,n,Hangup() exten => 2000,1,Dial(SIP/User2) exten => 2000,n,Hangup() exten => 3000,1,Dial(SIP/User3) exten => 3000,n,Hangup() ;call forwarding exten => _*21*X.,1,NoCDR exten => _*21*X.,2,Set(DB(CFIM/${CALLERID(NUM)}=${EXTEN:4}) exten => _*21*X.,3,Playback(vm-saved) exten => _*21*X.,4,Hangup()