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 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 > 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.

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 or substation call button, 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

GSM gateway (Mobile VoIP)

From Zenitel Wiki

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