Munir Hayat

I am currently working on different voip projects both at client end and sip provider end. I have more than 10 years of experience in VOIP solutions and more than 11 years of experience in Linux and Networking.
Asterisk, Freepbx, Vitalpbx, SwitchVox, Fusionpbx, Freeswitch,
--> SIP/IAX/PRI trunks, twilio elastic trunk, call termination and origination
--> dynamic ivr using text to speech engine and save DTMF feedback of users in Database..
Dialplans, Inbound, Outbound call Routing
Automatic Call distribution, Queues, dynamic logins, Call Recordings
Migration to different cloud technologies.
Customization of asterisk based Voicemail service
Realtime asterisk deployment
ASTPP/A2Billing installation and configuration
Astpp/A2billing origination and termination
Astpp/A2billing inbound and outbound routes tweaking
Astpp/A2billing rate groups and troubleshooting
Freeswitch Installation and Configuration
Freeswitch/Fusionpbx multitenant deployment for Telecom operators.