GSM gateway (Mobile VoIP)

GSM Gateway MV-370

This article describes the setup of the GSM gateway MV-370 (Mobile VoIP) from PORTech Communications Inc The GSM Gateway must be equipped with a SIM card and registered to the GSM network as a regular mobile phone subscriber. Via the gateway calls can be made from the AlphaCom E to the GSM network, as well as from the GSM network in to the AlphaCom E. The AlphaCom must be equipped with a license for SIP Trunk.

It is also possible to send SMS messages from the AlphaCom via the GSM gateway.

Configuration example

Configuration of the MV-370

The MV-370 unit is configured via a web interface. The default IP address is 192.168.0.100, mask 255.255.255.0. The default IP address can be set by pressing the reset button (located next to the SIM card) for 3 seconds. Other settings will be kept. Before accessing the web page, confirm that the IP address of the configuration PC is on the same subnet, e.g. 192.168.0.x.

• Username: voip
• Password: 1234

Network settings

Network Settings

Change the network settings according to the network environment. Select Network > WAN Settings:

• IP Type = Enable Fixed IP
• IP = IP address of the Mobile VoIP unit
• Mask = Network mask
• Gateway = IP address of the network gateway
  

SIP settings

SIP Settings

In the menu SIP Settings > Service Domain, enter information for "Realm 1":

• Active = ON
• User Name = Any text, used for Caller ID. This text will be shown in the display on incoming calls from the GSM network, together with the telephone number
• Proxy Server = IP address of the AlphaCom

Status will show Not Registered.

Enable DTMF signalling by SIP INFO method:

• SIP Settings > DTMF Setting: Enable Send DTMF SIP Info

Route

Routing Mobile to LAN

Select Route > Mobile to LAN Settings
Alternative 1 (default):

• CID = *
• URL = *

With this setting there will be a second Dial-Tone presented on incoming calls from the GSM network. The user must dial the intercom number. The dialling can be terminated by '#', or alternatively one can wait for 5 seconds and the call will be established.

Alternative 2:

• CID = *
• URL = 101

With this setting incoming calls will automatically be connected to station 101 (or any other number you enter) in the AlphaCom.

Mobile settings

Mobile settings

• Mobile > Settings > SIP from: Select User/Tel (Not Reg)
• Mobile > Settings > Mobile PIN Code: If the mobile needs to be unlocked by a pin code you must enable ON, and enter the pin code, and confirm the pin code

• Mobile > Status: Shows that the SIM card in in place, and that the Mobil VoIP unit is registered on the GSM network.

DTMF Input from outside line

During a conversation between a Pulse station and a telephone, the telephone operator can activate an output (e.g. trigger a relay for door opening) in the Pulse System by pressing a digit on the phone.

To enable digit actions from the telephone line during conversation, go to set SIP Settings > DTMF Setting, and enable "SIP INFO":

Set DTMF by 'SIP INFO'

Configuration of AlphaCom XE

Define SIP Trunk

The AlphaCom XE needs to be configured with a SIP trunk node.

To access the GSM Gateway from the AlphaCom, a directory number with feature 83/<SIP node> must be created in the Directory & Features window of AlphaPro.

Example: Directory number "0" used for accessing the GSM Gateway

Set SIP Dial Delay parameter

In order to call to preprogrammed phone numbers from a station DAK key, substation Call Button, or via Call Forwarding, set "SIP dial delay" = 20 (2 sec.) in Exchange & System -> System -> VoIP

Features

Outgoing calls from AlphaCom:

• Selective: Prefix + phone number: Yes
• Dialled number shows in display: Yes
• Put call on hold and transfer: Yes
• Call to predefined phone number from DAK: Yes (Set "SIP dial delay" = 20)
• Call to predefined phone number from Substation: Yes (Set "SIP dial delay" = 20)
• Door Opening (6) from line: Yes (from firmware 669f in gateway)
• Cancel when remote phone hangs up: Yes
• Automatic cancel when remote phone is busy: Yes
• On-hook when intercom cancel the call: Yes
• Signal when dialing prefix and no lines are available : Busy tone (Camp on Busy)
• Call Forward (71) from intercom to external phone: Yes
• Forward Call Requests to phone: Yes
• Forward if phone do not answer: No (The gateway responds with "200 OK" immediately)
• Call to remote service requiring DTMF signalling (e.g. Call Center): Yes
• Call to remote service: DAK 0 transmit DTMF "*", DAK 1 = DTMF "#": Yes (from firmware 669f in gateway)
• Call from analogue phone (ATLB) to external phone: Yes
• Call from subscriber in remote AlphaNet node: Yes
• Call from SIP extension to external phone: Yes (X-Lite)
• Call from IP Substation: Yes
• Call from IP Master: Yes
• Access restriction for SIP Phones: Yes (OK with firmware 6.693t, was not OK with 6.690f)

Incoming calls from GSM network:

• Two-step (selective) inward dialling - second dialtone: Yes
• Two-step (selective) inward dialling - voice prompt: No
• Automatic call to predefined station: Yes
• Delayed automatic call to predefined station: No
• Caller ID: Yes. The display will show <Phone number> + <User Name>, "User Name" is the text entered in SIP Settings. E.g. "73905391 (GSM)"
• Call to a remote node (AlphaNet) using Area Code or Global number: Yes
• Call to a SIP extension: Yes (X-Lite)
• Make group call, with answer Meet Me (99): Yes
• M-key control from line: Yes (from firmware 669f in gateway)
• Door Opening (6) from line: Yes (from firmware 669f in gateway)
• Cancel call when on-hook from line: Yes
• Cancel call when cancel at intercom station: Yes
• Call to intercom station which is transfered (71) to an other station: Yes
• Put calls on hold and transfer (from intercom station): Yes
• Force calls from GSM network to be in Private: Yes (Set AlphaPro flag "Incoming calls from SIP in private ringing mode")
• Call to IP substation: Yes
• Call to IP Masterstation: Yes

Related articles

• Send SMS messages from AlphaCom