Telephone scripts

From Hackerspace ACKspace
Revision as of 10:03, 14 November 2014 by Xopr (talk | contribs) (added autopatch and shoutcast stream links)
Jump to: navigation, search

These are scripts used by the Telephone system.

Number lookup

This script uses mod_cidlookup combined with a custom php page. It tests the number against a mySQL database, several reversed number lookup websites, the national telecommunications authority database and a coarse array of areas of the world, which are called in order of grannularity. The script is called both on outgoing and incoming calls

View the scripts at Telephone system:Number lookup active in ACKspace.

Simple intercom

This is a dialplan sample of using the secondary phone line put on auto answer, and bridge them in a conference call.

View the dialplan at Telephone system:Simple intercom

Space state

This script sets the spacestate variable according to what the webservice API returns. It then can be used in the dialplan to, for example, play a sound file or do call forwarding. Currently it is used to build a sound phrase for playing it as a greeting within our IVR

View the dialplan at Telephone system:Space state

Closing announcement

This collection of scripts was used to broadcast (closing) announcements using cron jobs

View the scripts at Telephone system:Closing announcement

Betamax credit notification

This collection of scripts is used to say the amount of credit left on one of the Betamax services (tested with VoipBuster and InterVoip). It is an altered version of a php script.

View the scripts at Telephone system:Betamax credit

Autopatch

To connect the PABX to an external communications channel, one needs a tiny bit of hardware and some scripting to get it working. A proof of concept is made that used the PMRs in the space and a couple of components on a perf-board.

View the schematic and script at Telephone_system:autopatch

Shoutcast streams

The most efficient way to have shoutcast streams as MOH is to set up a muted conference. This way, the fist connection will be made when the first listener joins, and no extra streams will be opened.

View the configuration files at Telephone_system:shoutcast_streams