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Difference between revisions of "GSM gateway (Mobile VoIP)"

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[[Image:2200003000 RS.jpg|thumb|250px|GSM Gateway MV-370]]
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{{AI}}
 +
[[Image:2200003000 RS.jpg|thumb|600px|GSM Gateway MV-370]]
  
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This article describes the setup of the GSM gateway MV-370 (Mobile VoIP) from [http://www.portech.com.tw PORTech Communications Inc]
+
Alternative solution: [[IP Telephony Service (ICX-AlphaCom)]]
−
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. The AlphaCom must be equipped with a [[Licenses#SIP_trunk_license|license for SIP Trunk]].
 
  
−
It is also possible to [[Send SMS messages from AlphaCom|send SMS messages from the AlphaCom]] via the GSM gateway.
+
__TOC__
  
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__TOC__
+
This article describes the setup of the GSM gateway MV-370 (Mobile VoIP) for communication with the ICX system or the AlphaCom.
−
[[Image:SIP GSM Gateway.jpg|thumb|left|500px|Configuration example]]
+
 
 +
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 ICX or AlphaCom to the GSM network, as well as from the GSM network in to the ICX/AlphaCom.
 +
 
 +
[[Image:MV370 ICX.png|thumb|left|500px|GSM Gateway]]
 
<br style="clear:both;" />
 
<br style="clear:both;" />
  
−
==Additional Documentation==
+
It is also possible to [[Send SMS messages from ICX-AlphaCom|send SMS messages from the ICX/AlphaCom]] via the GSM gateway.
−
For more documentation please see http://www.zenitel.com/product/gsm-media-gateway
 
  
−
== GSM Gateway Configuration ==
+
==Licenses==
 +
 
 +
*'''AlphaCom''': [[Licenses#SIP_trunk_license|License for SIP Trunk, 100 9642 001]]. One license for each connected gateway.
 +
*'''ICX''': [[Licenses for ICX-500 and ICX-AlphaCom Core|ILI-SIP2 SIP Trunking, 2 lines, 1002602101]]. One license can serve two gateways.
 +
 
 +
==GSM Gateway Configuration==
 
'''Important:''' If you have a SIM card with PIN code activated, '''DO&nbsp;NOT'''&nbsp;insert the SIM card yet.  
 
'''Important:''' If you have a SIM card with PIN code activated, '''DO&nbsp;NOT'''&nbsp;insert the SIM card yet.  
  
  
−
=== Load Factory Default values ===
+
===Load Factory Default values===
−
The GSM Gateway comes with default network parameters (factory default parameters). The default IP address is '''191.168.0.100'''.
+
The GSM Gateway comes with default network parameters (factory default parameters). The default IP address is '''192.168.0.100'''.
  
 
You can load factory network parameters and reset the username and password to its default settings (username: '''voip''', password: '''1234''') by following these steps:
 
You can load factory network parameters and reset the username and password to its default settings (username: '''voip''', password: '''1234''') by following these steps:
  
−
# Press and hold the button SW1 located at the bottom of the unit (you have to remove the lid) for about 7-8 seconds
+
#Press and hold the button SW1 located at the bottom of the unit (you have to remove the lid) for about 7-8 seconds
−
# Wait for the "Mobile" and "LAN" leds to start blinking
+
#Wait for the "Mobile" and "LAN" leds to start blinking
  
 
[[File:GSM FactoryDef.PNG|left|500px|thumb|Set default IP Address: Press SW1 for 7-8 seconds]]
 
[[File:GSM FactoryDef.PNG|left|500px|thumb|Set default IP Address: Press SW1 for 7-8 seconds]]
Line 29: Line 36:
  
  
−
* The GSM Gateway will now get the IP address '''191.168.0.100''', subnet mask 255.255.255.0.
+
*The GSM Gateway will now get the IP address '''192.168.0.100''', subnet mask 255.255.255.0.
 +
 
 +
===Accessing the gateway===
  
−
===Accessing the gateway ===
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*Change the IP address of your '''PC to 192.168.0.99''', subnet mask 255.255.255.0.
−
* Change the IP address of your '''PC to 191.168.0.99''', subnet mask 255.255.255.0.
+
*Connect the LAN port of the PC to the Ethernet port of the GSM Gateway.
−
* Connect the LAN port of the PC to the Ethernet port of the GSM Gateway.
+
*Start your Web Browser and type 192.168.0.100 in the URL field.
−
* Start your Web Browser and type 191.168.0.100 in the URL field.
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*Type in username '''voip''' and password '''1234'''. (Case-sensitive!)
−
* Type in username '''voip''' and password '''1234'''. (Case-sensitive!)
 
  
 
[[File:GSM accessing.PNG|left|500px|thumb|Accessing the gateway]]
 
[[File:GSM accessing.PNG|left|500px|thumb|Accessing the gateway]]
Line 41: Line 49:
  
  
−
*The Home page of the Web Interface will appear:  
+
*The Home page of the Web Interface will appear:
  
 
[[File:GSM Home.PNG|thumb|left|500px|Homepage of the GSM Gateway]]
 
[[File:GSM Home.PNG|thumb|left|500px|Homepage of the GSM Gateway]]
 
<br style="clear:both;" />
 
<br style="clear:both;" />
  
−
=== Network settings ===
+
===Network settings===
 
Change the network settings according to the network environment. Select '''Network > WAN Settings''':
 
Change the network settings according to the network environment. Select '''Network > WAN Settings''':
 +
 
:*'''IP Type''' = Enable ''Fixed IP''
 
:*'''IP Type''' = Enable ''Fixed IP''
 
:*'''IP''' = IP address of the Mobile VoIP unit
 
:*'''IP''' = IP address of the Mobile VoIP unit
 
:*'''Mask''' = Network mask
 
:*'''Mask''' = Network mask
 
:*'''Gateway''' = IP address of the network gateway <br style="clear:both;" />
 
:*'''Gateway''' = IP address of the network gateway <br style="clear:both;" />
 +
 
[[Image:MV-370-WAN.PNG|left|thumb|500px|Network Settings]]
 
[[Image:MV-370-WAN.PNG|left|thumb|500px|Network Settings]]
 
<br style="clear:both;" />
 
<br style="clear:both;" />
  
  
−
=== Mobile settings ===
+
===Mobile settings===
 +
Go to '''Mobile''' > '''Settings''' and define the following parameters:
 +
 
 +
*'''SIP From''' = User/Tel (Not Reg)
 +
*'''Mobile PIN Code''' = ON. Code: Enter the PIN code for the SIM card. Confirmed: Enter the PIN code again.
 +
 
 
[[Image:MV-370-MobileSettings.PNG|left|thumb|500px|Mobile settings]]
 
[[Image:MV-370-MobileSettings.PNG|left|thumb|500px|Mobile settings]]
 
<br style="clear:both;" />
 
<br style="clear:both;" />
−
:*'''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.
+
 
 +
 
 +
The SIM card can now be inserted. The gateway uses Standard (Mini) SIM card (15x25mm).
 +
[[File:GSM SIM Placement.PNG|thumb|left|300px|SIM card placement]]
 
<br style="clear:both;" />
 
<br style="clear:both;" />
  
−
=== SIP settings ===
+
After the SIM card is inserted, you can check that the gateway is successfully registered to the GSM network in '''Mobile''' > '''Status'''.
 +
 
 +
Some parameters to look for are:
 +
 
 +
*'''Operator''' = The telecom provider
 +
*'''Signal Quality''' = Shows the signal strength. The signal should be in the range of 10 - 31.
 +
 
 +
[[File:GSM Registered.PNG|thumb|left|500px|Gateway successfully registered to the GSM network]]
 +
<br style="clear:both;" />
 +
 
 +
If these parameters does not show, your gateway is not registered to the GSM network. Check that SIM card is properly inserted, and that the PIN code is correctly entered.
 +
 
 +
'''TIP''': Try the SIM card in a regular mobile phone.
 +
 
 +
===SIP settings===
 
In the menu '''SIP Settings > Service Domain''', enter information for "Realm 1":
 
In the menu '''SIP Settings > Service Domain''', enter information for "Realm 1":
 +
 
:*'''Active''' = ON
 
:*'''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
 
:*'''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
+
:*'''Proxy Server''' = IP address of the ICX or AlphaCom
  
 
[[Image:MV-370-SIP.PNG|left|thumb|500px|SIP Settings]]
 
[[Image:MV-370-SIP.PNG|left|thumb|500px|SIP Settings]]
Line 76: Line 107:
 
'''Status''' will show ''Not Registered''.
 
'''Status''' will show ''Not Registered''.
  
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=== Configure outgoing calls to the GSM network ===
+
===Configure outgoing calls to the GSM network===
  
−
==== Two-stage dialing ====
+
The configuration is done in '''Route''' > '''Lan To Mobile Settings'''.
 +
 
 +
====Two-stage dialing====
 
This is the default-setting after a factory-reset.  
 
This is the default-setting after a factory-reset.  
  
−
In this mode the caller is presented with a second dial-tone generated in the GSM-gateway and the user must dial the destination number wich is then sent as DTMF to the gateway.
+
The call progress is like this:
  
−
[[File:GSM mv370-two-stage-lantomobile.PNG|left|thumb|400px|Two-stage lan to Mobile]]
+
#The caller dials a prefix
 +
#A connection will be established to the GSM Gateway, and a second dial-tone is generated from the GSM-gateway
 +
#The caller dials the destination phone number
 +
 
 +
[[File:GSM mv370-two-stage-lantomobile.PNG|left|thumb|400px|Two-stage Lan to Mobile]]
 
<br style="clear:both;" />
 
<br style="clear:both;" />
  
−
==== One-stage dialing ====
+
''Additional information:'' <br>
−
In this mode the outgoing number can either be configured directly in the GSM gateway or be sent to the GSM-gateway in the SIP invite from the Alphacom.  
+
As soon as the prefix is dialed, a voice connection is established between the ICX/AlphaCom and the GSM gateway. The GSM gateway will respond with the SIP message OK. Further dialing is signaled as DTMF signals to the GSM gateway.
  
−
'''Defined in the GSM gateway'''<br>
+
====One-stage dialing====
 +
In this mode the outgoing number can either be configured directly in the GSM gateway or be sent to the GSM-gateway in the SIP invite from the ICX/AlphaCom.
 +
 
 +
''Phone number defined in the GSM gateway:''<br>
 
In this mode the call is set up to the GSM gateway and the defined number is dialed automatically.  
 
In this mode the call is set up to the GSM gateway and the defined number is dialed automatically.  
−
[[File:GSM mv370-one-stage-in-mv370.PNG|left|thumb|400px|Two-stage lan to Mobile]]
+
[[File:GSM mv370-one-stage-in-mv370.PNG|left|thumb|400px|One-stage Lan to Mobile]]
 
<br style="clear:both;" />
 
<br style="clear:both;" />
  
−
'''Recieved from SIP-invite'''<br>
+
''Phone number received in SIP-INVITE from ICX/AlphaCom:''<br>
−
In this mode the entire number is sent from the Alphacom.   
+
In this mode the entire number is sent from the ICX/AlphaCom.   
  
 
'''N.B. If using MV-370 together with an ATLB card and analogue stations this is the recommended method.'''
 
'''N.B. If using MV-370 together with an ATLB card and analogue stations this is the recommended method.'''
−
[[File:GSM mv370-one-stage-sipinvite.PNG|left|thumb|400px|Two-stage lan to Mobile]]
+
[[File:GSM mv370-one-stage-sipinvite.PNG|left|thumb|400px|One-stage lan to Mobile]]
 
<br style="clear:both;" />
 
<br style="clear:both;" />
  
−
=== Configure incoming calls from the GSM network ===
+
''Additional information:'' <br>
 +
When One-Stage Dialing is used, the GSM gateway will respond with the SIP message RINGING while the call is being established. The caller will hear the ringing tone generated by the ICX/AlphaCom. When the remote party answers the call, the GSM Gateway will send SIP message OK, and the voice communication will be established between the ICX/AlphaCom and the GSM gateway.
 +
 
 +
===Configure incoming calls from the GSM network===
  
 
There are two options:
 
There are two options:
−
* Two-stage dialing
 
−
* Automatic dialing
 
  
−
==== Two-stage 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.  
+
*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 dialing 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:<br>  
 
Select '''Route > Mobile to LAN Settings''', and set:<br>  
 +
 
:*'''CID''' = *
 
:*'''CID''' = *
 
:*'''URL''' = *
 
:*'''URL''' = *
  
−
==== Automatic dialing ====  
+
====Automatic dialing====  
−
With this setting incoming calls will automatically be connected to station 101 (or any other number you enter) in the AlphaCom.
+
With this setting incoming calls will automatically be connected to station 101 (or any other number you enter) in the ICX/AlphaCom.
  
  
 
Select '''Route > Mobile to LAN Settings''', and set:<br>  
 
Select '''Route > Mobile to LAN Settings''', and set:<br>  
 +
 
:*'''CID''' = *
 
:*'''CID''' = *
 
:*'''URL''' = 101
 
:*'''URL''' = 101
Line 127: Line 173:
  
  
−
=== DTMF Input from outside line ===
+
===DTMF Input from outside line===
−
During a conversation between a intercom station and a telephone, the telephone operator can activate an output (e.g. trigger a relay for door opening) in the AlphaCom by pressing a digit on the phone.  
+
During a conversation between a intercom station and a telephone, the telephone operator can activate an output (e.g. trigger a relay for door opening) in the ICX/AlphaCom 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":  
 
To enable digit actions from the telephone line during conversation, go to set '''SIP Settings''' > '''DTMF Setting''', and enable "SIP INFO":  
Line 135: Line 181:
 
<br style="clear:both;" />
 
<br style="clear:both;" />
  
−
== Configuration of AlphaCom XE ==
+
==Configuration of ICX/AlphaCom==
  
−
Create a [[SIP trunk node - configuration|SIP trunk node]] in the AlphaCom. Follow the instructions in this article: [[SIP trunk node - configuration|SIP trunk node]]
+
Create a SIP trunk node in the ICX/AlphaCom. Follow the instructions in this article: [[SIP trunk node - configuration|SIP trunk node]]
  
−
=== Two-stage dialing===
+
===Two-stage dialing===
 
This section is valid when using two-stage dialing for outgoing calls.
 
This section is valid when using two-stage dialing for outgoing calls.
  
−
# 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.
+
#To access the GSM Gateway from the ICX/AlphaCom, a directory number with feature 83/<SIP node> must be created in the Directory & Features window of AlphaPro.
 +
 
 
[[Image:AlphaPro Create SIP AccessCode.jpg|left|thumb|500px|Example: Directory number "0" used for accessing the GSM Gateway]]
 
[[Image:AlphaPro Create SIP AccessCode.jpg|left|thumb|500px|Example: Directory number "0" used for accessing the GSM Gateway]]
 
<br style="clear:both;" />
 
<br style="clear:both;" />
  
 
:2. In order to automatically call a fixed telephone number from a DAK key, from a Call Button, or via Call Forwarding or AutoSearch, set "SIP dial delay" = 20 (2 sec.) in '''Exchange & System -> System -> VoIP''':
 
:2. In order to automatically call a fixed telephone number from a DAK key, from a Call Button, or via Call Forwarding or AutoSearch, set "SIP dial delay" = 20 (2 sec.) in '''Exchange & System -> System -> VoIP''':
 +
 
[[File:SIP Delay.PNG|left|thumb|500px|Set "SIP Dial Delay" when using automatic call to fixed phone number]]
 
[[File:SIP Delay.PNG|left|thumb|500px|Set "SIP Dial Delay" when using automatic call to fixed phone number]]
 
<br style="clear:both;" />
 
<br style="clear:both;" />
Line 154: Line 202:
 
<br style="clear:both;" />
 
<br style="clear:both;" />
  
−
=== One-stage dialing===
+
===One-stage dialing===
 
This section is valid when using One-stage dialing.
 
This section is valid when using One-stage dialing.
  
−
# To send an invite with the dialed number from the Alphacom, a directory number with feature 81/<SIP node> must be created in the Directory & Features window of AlphaPro. Parameter 2 defines the max amount of numbers.  
+
*To collect the phone number in ICX/AlphaCom, a directory number with feature 81/<SIP node> must be created in the Directory & Features window of AlphaPro. Parameter 2 defines the max amount of digits in the phone number.
−
[[Image:Alphapro Feature 81 one stage.jpg|left|thumb|600px|Example: Directory number "0" is used to start collecting numbers in Alphacom. Parameter 2 defines the maximum amount of numbers]]
+
*If less digits than "max" is dialed, you can press M-key to send the number, or wait for timeout (3 sec by default).
 +
 
 +
[[Image:Alphapro Feature 81 one stage.jpg|left|thumb|600px|Example: Directory number "0" is used to start collecting numbers in ICX/AlphaCom. Parameter 2 defines the maximum amount of numbers]]
 
<br style="clear:both;" />
 
<br style="clear:both;" />
  
−
== Features ==
+
==Features==
−
Outgoing calls from AlphaCom:
+
Outgoing calls from ICX/AlphaCom:
 +
 
 
:*Selective: Prefix + phone number: Yes
 
:*Selective: Prefix + phone number: Yes
−
:*Dialled number shows in display: Yes
+
:*Dialed number shows in display: Yes
 
:*Put call on hold and transfer: Yes
 
:*Put call on hold and transfer: Yes
 
:*Call to predefined phone number from DAK: Yes (Set "[[New fields (AlphaPro 10.27)|SIP dial delay]]" = 20)
 
:*Call to predefined phone number from DAK: Yes (Set "[[New fields (AlphaPro 10.27)|SIP dial delay]]" = 20)
Line 176: Line 227:
 
:*Forward Call Requests to phone: Yes
 
:*Forward Call Requests to phone: Yes
 
:*Forward if phone do not answer: '''''No''''' (The gateway responds with "200 OK" immediately)
 
:*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 requiring DTMF signaling (e.g. Call Center): Yes
 
:*Call to remote service: DAK 0 transmit DTMF "*", DAK 1 = DTMF "#": Yes (from firmware 669f in gateway)
 
:*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 analogue phone (ATLB) to external phone: Yes
Line 187: Line 238:
 
 
 
Incoming calls from GSM network:
 
Incoming calls from GSM network:
−
:*Two-step (selective) inward dialling - second dialtone: Yes
+
 
−
:*Two-step (selective) inward dialling - voice prompt: No
+
:*Two-step (selective) inward dialing - second dial tone: Yes
 +
:*Two-step (selective) inward dialing - voice prompt: No
 
:*Automatic call to predefined station: Yes
 
:*Automatic call to predefined station: Yes
 
:*Delayed automatic call to predefined station: No
 
:*Delayed automatic call to predefined station: No
Line 199: Line 251:
 
:*Cancel call when on-hook from line: Yes
 
:*Cancel call when on-hook from line: Yes
 
:*Cancel call when cancel at intercom station: Yes
 
:*Cancel call when cancel at intercom station: Yes
−
:*Call to intercom station which is transfered (71) to an other station: Yes
+
:*Call to intercom station which is transferred (71) to an other station: Yes
 
:*Put calls on hold and transfer (from intercom 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 "[[New fields (AlphaPro 10.27)|Incoming calls from SIP in private ringing mode]]")
 
:*Force calls from GSM network to be in Private: Yes (Set AlphaPro flag "[[New fields (AlphaPro 10.27)|Incoming calls from SIP in private ringing mode]]")
 
:*Call to  IP substation: Yes
 
:*Call to  IP substation: Yes
 
:*Call to  IP Masterstation: Yes
 
:*Call to  IP Masterstation: Yes
 +
 
<br>
 
<br>
  
−
== Related articles ==
+
==Related articles==
−
* [[Send SMS messages from AlphaCom]]
+
 
 +
*[[Send SMS messages from ICX-AlphaCom]]
 +
*More documentation: https://www.zenitel.com/product/mv-370l-4g
  
−
[[Category:SIP]]
+
[[Category: ICX-AlphaCom - SIP Integration]]
−
[[Category:3rd party integration]]
+
[[Category: AlphaCom - SIP Integration]]
 +
[[Category: 3rd party integration]]

Latest revision as of 14:14, 16 April 2024

AI.png
GSM Gateway MV-370

Alternative solution: IP Telephony Service (ICX-AlphaCom)

This article describes the setup of the GSM gateway MV-370 (Mobile VoIP) for communication with the ICX system or the AlphaCom.

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 ICX or AlphaCom to the GSM network, as well as from the GSM network in to the ICX/AlphaCom.

GSM Gateway


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

Licenses

GSM Gateway Configuration

Important: If you have a SIM card with PIN code activated, DO NOT insert the SIM card yet.


Load Factory Default values

The GSM Gateway comes with default network parameters (factory default parameters). The default IP address is 192.168.0.100.

You can load factory network parameters and reset the username and password to its default settings (username: voip, password: 1234) by following these steps:

  1. Press and hold the button SW1 located at the bottom of the unit (you have to remove the lid) for about 7-8 seconds
  2. Wait for the "Mobile" and "LAN" leds to start blinking
Set default IP Address: Press SW1 for 7-8 seconds



  • The GSM Gateway will now get the IP address 192.168.0.100, subnet mask 255.255.255.0.

Accessing the gateway

  • Change the IP address of your PC to 192.168.0.99, subnet mask 255.255.255.0.
  • Connect the LAN port of the PC to the Ethernet port of the GSM Gateway.
  • Start your Web Browser and type 192.168.0.100 in the URL field.
  • Type in username voip and password 1234. (Case-sensitive!)
Accessing the gateway



  • The Home page of the Web Interface will appear:
Homepage of the GSM Gateway


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



Mobile settings

Go to Mobile > Settings and define the following parameters:

  • SIP From = User/Tel (Not Reg)
  • Mobile PIN Code = ON. Code: Enter the PIN code for the SIM card. Confirmed: Enter the PIN code again.
Mobile settings



The SIM card can now be inserted. The gateway uses Standard (Mini) SIM card (15x25mm).

SIM card placement


After the SIM card is inserted, you can check that the gateway is successfully registered to the GSM network in Mobile > Status.

Some parameters to look for are:

  • Operator = The telecom provider
  • Signal Quality = Shows the signal strength. The signal should be in the range of 10 - 31.
Gateway successfully registered to the GSM network


If these parameters does not show, your gateway is not registered to the GSM network. Check that SIM card is properly inserted, and that the PIN code is correctly entered.

TIP: Try the SIM card in a regular mobile phone.

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 ICX or AlphaCom
SIP Settings


Status will show Not Registered.

Configure outgoing calls to the GSM network

The configuration is done in Route > Lan To Mobile Settings.

Two-stage dialing

This is the default-setting after a factory-reset.

The call progress is like this:

  1. The caller dials a prefix
  2. A connection will be established to the GSM Gateway, and a second dial-tone is generated from the GSM-gateway
  3. The caller dials the destination phone number
Two-stage Lan to Mobile


Additional information:
As soon as the prefix is dialed, a voice connection is established between the ICX/AlphaCom and the GSM gateway. The GSM gateway will respond with the SIP message OK. Further dialing is signaled as DTMF signals to the GSM gateway.

One-stage dialing

In this mode the outgoing number can either be configured directly in the GSM gateway or be sent to the GSM-gateway in the SIP invite from the ICX/AlphaCom.

Phone number defined in the GSM gateway:
In this mode the call is set up to the GSM gateway and the defined number is dialed automatically.

One-stage Lan to Mobile


Phone number received in SIP-INVITE from ICX/AlphaCom:
In this mode the entire number is sent from the ICX/AlphaCom.

N.B. If using MV-370 together with an ATLB card and analogue stations this is the recommended method.

One-stage lan to Mobile


Additional information:
When One-Stage Dialing is used, the GSM gateway will respond with the SIP message RINGING while the call is being established. The caller will hear the ringing tone generated by the ICX/AlphaCom. When the remote party answers the call, the GSM Gateway will send SIP message OK, and the voice communication will be established between the ICX/AlphaCom and the GSM gateway.

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 dialing 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 = *

Automatic dialing

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


Select Route > Mobile to LAN Settings, and set:

  • CID = *
  • URL = 101


Routing Mobile to LAN - Automatic dialing



DTMF Input from outside line

During a conversation between a intercom station and a telephone, the telephone operator can activate an output (e.g. trigger a relay for door opening) in the ICX/AlphaCom 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 ICX/AlphaCom

Create a SIP trunk node in the ICX/AlphaCom. Follow the instructions in this article: SIP trunk node

Two-stage dialing

This section is valid when using two-stage dialing for outgoing calls.

  1. To access the GSM Gateway from the ICX/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


2. In order to automatically call a fixed telephone number from a DAK key, from a Call Button, or via Call Forwarding or AutoSearch, set "SIP dial delay" = 20 (2 sec.) in Exchange & System -> System -> VoIP:
Set "SIP Dial Delay" when using automatic call to fixed phone number


DAK key configuration:

DAK 1 will dial 0 + 73456823


One-stage dialing

This section is valid when using One-stage dialing.

  • To collect the phone number in ICX/AlphaCom, a directory number with feature 81/<SIP node> must be created in the Directory & Features window of AlphaPro. Parameter 2 defines the max amount of digits in the phone number.
  • If less digits than "max" is dialed, you can press M-key to send the number, or wait for timeout (3 sec by default).
Example: Directory number "0" is used to start collecting numbers in ICX/AlphaCom. Parameter 2 defines the maximum amount of numbers


Features

Outgoing calls from ICX/AlphaCom:

  • Selective: Prefix + phone number: Yes
  • Dialed 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 signaling (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 dialing - second dial tone: Yes
  • Two-step (selective) inward dialing - 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 transferred (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

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