Forwarding of Call Request to external telephone
From Zenitel Wiki
This article describes how a Call Request can be forwarded to an external telephone via a SIP Gateway. Two types of call forwarding are described:
- Manual Call Forwarding: The Call Forwarding is switched on and off by the operator by dialling a code or pressing a DAK key.
- Automatic Call Forwarding: If the Call Request is not answered within a programmable time, the calling station is connected to the telephone
You can choose to use both types of forwarding, or only one of them.
The call to the telephone will be activated only if there is a free telephone line. If all available lines are busy, the system will retry every 5 seconds until a line is free.
There will be no redial if the telephone subscriber is busy. Also there is no option for dialling of alternative telephone numbers.
AlphaNet: The programming described in this document will also work in AlphaNet systems. The calling stations, the queing station and the SIP gateway can be located in the same node or in different nodes in the network.
Software requirement: AMC 11.2.3.3 or newer.
Manual Call Forwarding
The Call Forwarding is turned on and off from the Queuing Station by forwarding the Call Request to a dummy station. The dummy number can be any station directory number in the exchange that is not in use. In this article 5152 on physical 552 is used as an example.
- Call Forwarding On: Dial 7870 + Dummy Number (e.g. dial 7870 + 5152, or press a DAK key with the programming: I 7870 I 5152).
- Call Forwarding Off: Dial 70
In AlphaPro, Exchange & System -> Events, press Insert to add a new event.
Event 1: This event will start a 1 second timer when the dummy station receives a call request.
| Event Owner: | The Dummy Station |
| Event type: | 10 - Received Mail |
| Subevent: | 0 |
| When change to: | ON (When the call request is received) |
| When related to: | All |
| Action: | $ST L%1.dir W10 %2.ref |
Event 2: When the timer expires, and if there is any free phone line, the calling station will be connected to the SIP gateway (i.e. dial "0"), and the phone number 12345678 will be dialled.
If there are no free lines the timer is restarted, and a new attempt will be made 5 seconds later.
| Event Owner: | The Dummy Station |
| Event type: | 21 - Event Timeout |
| Subevent: | 0 |
| When change to: | ON |
| When related to: | All |
| Action: | IF %op(%tin(102),=,1) |
| $ST L%1.dir W50 %2.ref | |
| STOP | |
| ENDIF | |
| $DD %2.ref L0 | |
| pause | |
| pause | |
| $DD %2.ref G12345678 | |
| $CANM %2.ref L%1.dir |