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 7-8 seconds, until the "Mobile" and "LAN" leds start to flash. 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

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
  

Network 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

SIP Settings

Status will show Not Registered.

Configure incoming calls from the GSM network

There are two options:

• Two-stage dialing
• Automatic dialing

Two-stage dialing

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.

Select Route > Mobile to LAN Settings, and set:

• CID = *
• URL = *

Routing Mobile to LAN

Automatic dialing

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

• CID = *
• URL = 101

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. The signal should be in the range of 10 - 31.

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

1. Create a SIP trunk node in the AlphaCom. Follow the instructions in this article: SIP trunk node
2. 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

3. 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