BT Call Server (powered by Asterisk PBX) provides basic call routing capabilities for TPO devices.
It is designed for small, non-complex deployments where cost is a key consideration.
Usage is limited to configuring TPO lines and trunks, allowing the TPO to place and receive calls with the external world.
Desk phones are not supported on this package.
Deployments are limited to a maximum of 200 TPO registered SIP extensions.
There is no predefined limit on the number of supported SIP trunks.
However, where complex call flows are required across multiple SIP trunks, the design, testing, and implementation of the associated routing rules is the customer's responsibility.
BT has provided enough examples to satisfy a basic system setup
See: Single Node Trunk, Dual Node Trunk and Dual Trunk Routing
Please also see Limitations for further information.
The setup is made in 3 easy steps:
1.Installation 2.Extensions Configuration 3.Trunks Configuration 4.Route (Dial Plan) Configuration
Example configurations are provided to illustrate what needs to be defined on the BT Call Server side.
Configuration of the remote end (the other PBX or trunk provider) is the reader's responsibility, as this varies too significantly between systems to be covered with universal examples.
Notes for BT Call Server v3.x and greater
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Due to a major change in the SIP stack module (see below), which requires a new configuration syntax, and which is NOT directly compatible with the previous syntax.
Care should be taken when upgrading from versions <= v2.0 to versions >= v3.x
It is therefore highly recommended that lab testing is done to ensure you have the correct configurations prior to any upgrade.
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•Major change on Asterisk SIP stack, Asterisk now uses the newer chan_pjsip SIP stack (See: PJSIP-pjproject) “Asterisk currently contains two SIP stacks: the original chan_sip SIP channel driver which is a complete standalone implementation,
has been present in all previous releases of Asterisk and no longer receives core support, and the newer chan_pjsip SIP stack”
“The chan_sip module is now deprecated, users should migrate to the replacement module chan_pjsip.”
•Due to the SIP stack change, the back end configuration files now need to use the required syntax of the pjsip stack and its configuration files.
Please consult the official Asterisk Documentation for any additional information. Notes for BT Call Server v4.x and greater
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From V4.x and greater, Extensions and Trunks configuration are separated into their own configuration pages.
If you are upgrading from v3.x and restoring configuration files, make sure to correctly separate the extension and trunk configurations into their respective configuration pages.
NOTE: Configuration syntax is the same as v3.0 and greater
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