I am looking for an experienced Asterisk 1.4 PBX developer/ installer/ configuration person to produce a fairly complex Asterisk operational configuration. Work only in either AEL or {Asterisk Configuration File language} required. I don’t believe any custom C/C++ or other compiled programs are required. Completion will consist of supplying me acceptable .conf and .ael files which will implement required functionality for my existing Asterisk installation on a Ubuntu Linux server which has 1 x 4-line Voicetronix fxo card installed plus an internet connection for voip installed and tested. Preferrably the main configuration should be implemented in AEL language, unless a good reason exists for not using that.
Operations to be implemented are to be those of a standard landline-to-voip service, connecting my two private Asterisk PBX servers using sip protocol (or recommend alternative), and providing local call connections via landline to my employees or customers holding PIN identity numbers provided to them by me. The Asterisk system shall first collect their PIN number, then validate that PIN from a shared database (developers choice) operating on one of the Ubuntu servers. It shall then determine the amount of time allowed / remaining on the PIN and inform the caller of that time limit verbally. It shall then prompt for and acquire the desired telephone number, and pass the call and the desired number off to the remote server which shall then dial that number in it’s local calling area using it’s landlines, and on answering, connect the two parties and begin recording time. When only one minute of available time remains on the provided PIN, the PBX should warn the caller with a beep sequence or voice prompt. When no time remains, the PBX should disconnect the call.
Developer is free to define whatever sound files I need to provide, their names, locations and wording.
Questions:
(1) Years experience with Asterisk
(2) Years experience with Linux, Ubuntu Server
(3) Any relational database experience