Substations calling external telephone(s)

This article describes how substations in the AlphaCom can be configured to call external telephone number(s).

• If all lines in the SIP gateway are busy, the call will be queued. When the line becomes free, the call will automatically progress.
• Multiple phone numbers can be defined. If the first phone doesn't answer within a preset time, the call can be routed to a second (and a third...) phone number.

The AlphaCom and SIP Gateway must be configured to use One Stage (Enbloc) dialing to achive the described functionality

Prerequisites

The AlphaCom and SIP Gateway must be configured as described in relevant articles.

• AudioCodes MP114/118
• GSM gateway (MV370)

Configure the Call Button of the substation

The call button can be configured in two different ways to dial a preconfigured phone number.

Let's assume that the prefix to the external phone line is "0", and the phone number to call is 987654321.

Alternative 1:

Call Button configuration

Start the string with a "I", followed by the prefix number "0". Then "P", followed by the phone number.

Alternative 2:

Call Button configuration

Start the string with a "T", followed by the prefix number "0". Then "w" or "W", followed by the phone number. "w" introduces a 0.5 second pause, while "W" introduces a 1.0 second pause. See Play DAK Feature for further details.

Configure the Prefix Number

One Stage dialing ("enbloc dialing") must be used. The digit are collected in the AlphaCom and sent to the SIP gateway in the INVITE message. The prefix number must be programmed in the AlphaCom directory table with Feature 81 and Node = SIP Trunk node number. In the "Parameter 2" field you must enter the maximum number of digits in a phone number.

Prefix number is 0, SIP trunk node is 100, collect up to 16 digits

When the prefix is dialed, the AlphaCom will collect the number of digits specified in the “Parameter 2" before sending the INVITE. If fewer digits are dialed, the AlphaCom will send the INVITE after a predefined timeout. The timeout is by default 3 seconds, and can be configured in Exchange & System > System VoIP: SIP digit collection timeout. The call setup time will be faster if this timer is set to a low value.

Digit collection timeout set to 1.0 seconds

Set SIP Gateway in One Stage dialing mode

GSM Gateway MV370

Set the GSM Gateway to One Stage dialing mode by selecting Route > LAN to Mobile Stettings

GSM Gateway set to One Stage dialing