Actions

GSM gateway (Mobile VoIP)

From Zenitel Wiki

Revision as of 16:41, 6 March 2008 by Asle (talk) (→‎Features)

This article describes the setup of the GSM gateway MV-370 (Mobile VoIP) from PORTech Communications Inc

Configuration example

The GSM Gateway must be registered to the GSM network as a regular mobile phone subscriber, and needs to be equipped with a SIM card. 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.

Because the MV-370 needs to registrer to a SIP server, the AlphaCom must be equipped with license for SIP phone (not SIP Trunk license!).

Configuration of the MV-370

The MV-370 unit is configured via a web interface. The default IP address is 192.168.0.100. Before accessing the web page, confirm that the IP address of the configuration PC is on the same subnet, e.g. 192.168.x.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
  • Display Name = Any text
  • User Name = Access number to the gateway. This is the directory number you must dial on an intercom station to access the GSM gateway. The directory number must be programmed in the AlphaCom directory table with feature 83/<SIP node>.
  • Register Name = Access number to the gateway
  • Domain Server = IP address of the AlphaCom
  • Proxy Server = IP address of the AlphaCom

Status will show Registered when the gateway is registered at the SIP Registrar server in the AlphaCom.

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 = *

This setting will give a second Dial-Tone on incoming calls to AlphaCom. The user must then dial the intercom number. The Mobile VoIP unit will connect the call 5 seconds after having received the last digit in the intercom number.

Alternative 2:

  • CID = *
  • URL = 101

With this setting incoming calls will automatically be connected to station 101 in the AlphaCom.

Mobile settings

Mobile settings
  • Mobile > Settings > Mobile PIN Code: If the mobile needs to be unlocked by a pin code you must choose Enable, 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.


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" = 10)
  • Call to predefined phone number from Substation: Yes (Set "SIP dial delay" = 10)
  • M-key control from line: No
  • Door Opening (6) from line: No
  • 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)
  • 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): No (DTMF not sent)
  • Call to remote service: DAK 0 transmit DTMF "*", DAK 1 = DTMF "#": No (DTMF not sent)
  • 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


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
  • 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: No (No SIP INFO sent)
  • Door Opening (6) from line: No (No SIP INFO sent)
  • Cancel call when on-hook from line: Yes
  • On-hook 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


Configuration of AlphaCom E

The AlphaCom E needs to be configured with a SIP registrar node.

Retrieved from "https://wiki.zenitel.com/w/index.php?title=GSM_gateway_(Mobile_VoIP)&oldid=64099"