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{{stub}}This article describes how a Call Request can be forwarded to an external telephone via a SIP Gateway. Two types of call forwarding are described:  
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{{AI}}
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[[File:ForwdCallReq.PNG|thumb|right|500px|Forwarding of Call Request to external telephone]]
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This article describes how a [[ Call Request]] can be forwarded to an external telephone via a SIP Gateway. Two types of call forwarding are described:  
 
* '''Manual Call Forwarding:''' The Call Forwarding is switched on and off by the operator by dialling a code or pressing a DAK key.
 
* '''Manual Call Forwarding:''' The Call Forwarding is switched on and off by the operator by dialling a code or pressing a DAK key.
 
* '''Automatic Call Forwarding''': If the Call Request is not answered within a programmable time, the calling station is connected to the telephone  
 
* '''Automatic Call Forwarding''': If the Call Request is not answered within a programmable time, the calling station is connected to the telephone  
Line 5: Line 7:
 
You can choose to use both types of forwarding, or only one of them.  
 
You can choose to use both types of forwarding, or only one of them.  
  
The call to the telephone will be activated only if there is a free telephone line. If all available lines are busy, the system will retry every 5 seconds until a line is free. The macro [[%tin]] (Trunk In Use) is used to check if there are any free lines.
+
The call to the telephone will be activated only if there is a free telephone line. If all available lines are busy, the system will retry every 5 seconds until a line is free. If there are several Call Requests at the same time, they will be queued in the ICX-AlphaCom and handled one by one. The macro [[%tin|%tin (Trunk In Use)]] is used to check if there are any free lines.
  
There will be no redial if the telephone subscriber is busy. Also there is no option for dialling of alternative telephone numbers.
+
There will be no redial if the telephone subscriber is busy. Also this article does not describe any option for dialing of alternative telephone numbers.
  
Software requirement: AMC 11.2.3.3 or newer.
+
The SIP Gateway can for instance be [[Configuration guide for AudioCodes MP114/118, v5.4 and higher|AudioCodes MP114/118]] for analogue phonelines or [[Configuration guide for Mobile VoIP|MV-370]] for GSM network. The SIP gateways should be configured as described in the relevant AlphaWiki articles.
  
=== Manual Call Forwarding ===
 
  
 +
See also [[Calling multiple external telephone numbers in sequence]]
 +
 +
 +
== Software requirement ==
 +
* AlphaCom: 11.2.3.3 or newer.
 +
* ICX-AlphaCom: any
 +
 +
== Manual Call Forwarding ==
 
The Call Forwarding is turned on and off from the Queuing Station by forwarding the Call Request to a dummy station. The dummy number can be any station directory number in the exchange that is not in use. In this article 5152 on physical 552 is used as an example.  
 
The Call Forwarding is turned on and off from the Queuing Station by forwarding the Call Request to a dummy station. The dummy number can be any station directory number in the exchange that is not in use. In this article 5152 on physical 552 is used as an example.  
 
*'''Call Forwarding On''': Dial 7870 + Dummy Number (e.g. dial 7870 + 5152, or press a DAK key with the programming: I 7870 I 5152).   
 
*'''Call Forwarding On''': Dial 7870 + Dummy Number (e.g. dial 7870 + 5152, or press a DAK key with the programming: I 7870 I 5152).   
 
*'''Call Forwarding Off''': Dial 70
 
*'''Call Forwarding Off''': Dial 70
 
  
 
In AlphaPro, [[Exchange_%26_System_%28AlphaPro%29#Events|Exchange & System -> Events]], press '''Insert''' to add a new event.
 
In AlphaPro, [[Exchange_%26_System_%28AlphaPro%29#Events|Exchange & System -> Events]], press '''Insert''' to add a new event.
  
 
'''Event 1:'''
 
'''Event 1:'''
This event will start a 1 second timer when the dummy station receives a call request.
+
Start a 1 second timer when the dummy station receives a call request.
 +
[[File:Application4a.png|500px|left|thumb|]]
 +
<br style="clear:both;" />
  
{|  
+
{{Code2|
|-
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$ST L%1.dir W10 %2.ref  
|width="100pt"|'''Event Owner''':
+
}}
|width="300pt"| The Dummy Station
 
|-
 
|'''Event type''': || 10 - Received Mail
 
|-
 
|'''Subevent''': || 0
 
|-
 
|'''When change to:''' || ON
 
|-
 
|'''When related to''': || All
 
|-
 
|'''Action''': || $ST L%1.dir W10 %2.ref  
 
|-
 
|}
 
<br>
 
  
 
'''Event 2:'''
 
'''Event 2:'''
When the timer expires, and if there is a free phone line, the calling station will be connected to the SIP gateway (i.e. dial "0") and the phone number 12345678 will be dialled.
+
When the timer expires, and if there is a free phone line, the calling station will be connected to the SIP gateway (i.e. dial "0") and the phone number 12345678 will be dialed.  
  
 
If there are no free lines the timer is restarted, and a new attempt will be made 5 seconds later.
 
If there are no free lines the timer is restarted, and a new attempt will be made 5 seconds later.
 +
[[File:Application4b.png|500px|left|thumb]]
 +
<br style="clear:both;" />
  
{|  
+
{{Code2|
|-
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[[%tin|IF %op(%tin(100),>=,1)]] ''(If the line is already in use on SIP Trunk node 100...)''
|width="100pt"|'''Event Owner''':
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[[SET TIMER|$ST L%1.dir W50 %2.ref]]  ''(...then try again in 5 seconds by restarting the timer)''
|width="300pt"| The Dummy Station
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STOP
|-
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ENDIF ''(If the line is free...)''
|'''Event type''': || 21 - Event Timeout
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[[PLAY DAK|$PD %2.ref "0W12345678"]] ''(...dial 0, then 1 sec pause, then dial 12345678)''
|-
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[[CANCEL MAIL|$CANM %2.ref L%1.dir]] ''(Delete the call request from the dummy station)''
|'''Subevent''': || 0
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}}
|-
 
|'''When change to:''' || ON
 
|-
 
|'''When related to''': || All
 
|-
 
|'''Action''': || IF %op(%tin(100),=,1)  ''(If 1 line is in use on SIP Trunk node 100...)''
 
|-
 
| &nbsp; || $ST L%1.dir W50 %2.ref
 
|-
 
| &nbsp; || STOP
 
|-
 
| &nbsp; || ENDIF
 
|-
 
| &nbsp; || $PD %2.ref "0W12345678"  ''(Dial 0, then 1 sec pause, then dial 12345678)''
 
|-
 
| &nbsp; || $CANM %2.ref L%1.dir ''(Delete the call request from the dummy station)''
 
|}
 
 
<br>
 
<br>
 +
 +
About the expression "'''IF %op(%tin(100),>=,1)'''":
 +
* 100 = the node number of the SIP Trunk. Change as required.
 +
* 1 = There is one phone line connected to the SIP Gateway. If you are using MP114/118 with multiple lines connected, you need to change this number accordingly.
 +
 +
 +
In the expression "'''$PD %2.ref "0W12345678"'''" substitute 12345678 with the wanted phone number.
  
 
== Automatic Call Forwarding ==
 
== Automatic Call Forwarding ==
  
The programming below will forward the call request to the external telephone id the call request is not answered within 30 seconds. The time can of course be adjusted to suit your requirements.
+
The programming below will forward the calling station to the external telephone if the call request is not answered within 30 seconds. The time can of course be adjusted to suit your requirements.
  
  
Line 81: Line 69:
  
 
'''Event 1:'''
 
'''Event 1:'''
When a Call Request is received at the queuing station, a 30 sec timer is started:
+
When a call request is received at the queuing station, a 30 sec timer is started. If the call request is deleted (i.e. the call request is answered), the timer is stopped:
 +
[[File:Application4c.png|500px|left|thumb]]
 +
<br style="clear:both;" />
  
{|  
+
{{Code2|
|-
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$ST L%1.dir W%chg(300,0) %2.ref
|width="100pt"|'''Event Owner''':
+
}}
|width="300pt"| The Queing Station
+
 
|-
+
In the expression "'''$ST L%1.dir W%chg(300,0) %2.ref '''" 300 means 30,0 seconds. You can change this parameter if you want to use a different time.
|'''Event type''': || 10 - Received Mail
 
|-
 
|'''Subevent''': || 0
 
|-
 
|'''When change to:''' || ON ''(When the call request is received)''
 
|-
 
|'''When related to''': || All
 
|-
 
|'''Action''': || $ST L%1.dir W300 %2.ref  
 
|-
 
|}
 
<br>
 
  
  
 
'''Event 2:'''
 
'''Event 2:'''
When the timer exipres, and there is a free line on the SIP Gateway, the calling styation is connected to the SIP gateway. Else the timer is restarted and a new attempt is made in 5 seconds:
+
When the timer expires, and there is a free line on the SIP Gateway, the calling station is connected to the SIP gateway. Else the timer is restarted and a new attempt is made 5 seconds later:
 +
[[File:Application4d.png|500px|left|thumb]]
 +
<br style="clear:both;" />
 +
 
 +
{{Code2|
 +
[[%tin|IF %op(%tin(100),>=,1)]]  ''(If the line is already in use on SIP Trunk node 100...)''
 +
[[SET TIMER|$ST L%1.dir W50 %2.ref]] ''(...then try again in 5 seconds by restarting the timer)''
 +
STOP
 +
ENDIF  ''(If the line is free...)''
 +
[[PLAY DAK|$PD %2.ref "0W12345678"]]  ''(...dial 0, then 1 sec pause, then dial 12345678)''
 +
[[CANCEL MAIL|$CANM %2.ref L%1.dir]] ''(Optional: Delete the call request from the queing station)''
 +
}}
 +
 
 +
 
 +
 
 +
About the expression "'''IF %op(%tin(100),>=,1)'''":
 +
* 100 = the node number of the SIP Trunk.
 +
* 1 = There is one phone line connected to the SIP Gateway. If you are using MP114/118 with multiple lines connected, you need to change this number accordingly.
  
{|
 
|-
 
|width="100pt"|'''Event Owner''':
 
|width="300pt"| The Queing Station
 
|-
 
|'''Event type''': || 21 - Event Timeout
 
|-
 
|'''Subevent''': || 0
 
|-
 
|'''When change to:''' || ON
 
|-
 
|'''When related to''': || All
 
|-
 
|'''Action''': || IF %op(%tin(102),=,1)  ''(If 1 line is in use on SIP Trunk node 100...)''
 
|-
 
| &nbsp; || $ST L%1.dir W50 %2.ref
 
|-
 
| &nbsp; || STOP
 
|-
 
| &nbsp; || ENDIF
 
|-
 
| &nbsp; || $PD %2.ref "0W12345678"  ''(Dial 0, then 1 sec pause, then dial 12345678)''
 
|-
 
| &nbsp; || $CANM %2.ref L%1.dir ''(Delete the call request from the dummy station)''
 
|}
 
<br>
 
  
 +
In the expression "'''$PD %2.ref "0W12345678"'''" substitute 12345678 with the wanted phone number.
  
 +
== Related articles ==
 +
* [[Calling multiple external telephone numbers in sequence]]
  
[[Category: SIP]]
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[[Category: AlphaCom Applications]]
[[Category: Applications]]
+
[[Category: ICX-AlphaCom Applications]]
 +
[[Category:Control Room Applications]]
 +
[[Category: ICX-AlphaCom - SIP Integration]]

Latest revision as of 15:12, 30 October 2024

AI.png
Forwarding of Call Request to external telephone

This article describes how a Call Request can be forwarded to an external telephone via a SIP Gateway. Two types of call forwarding are described:

  • Manual Call Forwarding: The Call Forwarding is switched on and off by the operator by dialling a code or pressing a DAK key.
  • Automatic Call Forwarding: If the Call Request is not answered within a programmable time, the calling station is connected to the telephone

You can choose to use both types of forwarding, or only one of them.

The call to the telephone will be activated only if there is a free telephone line. If all available lines are busy, the system will retry every 5 seconds until a line is free. If there are several Call Requests at the same time, they will be queued in the ICX-AlphaCom and handled one by one. The macro %tin (Trunk In Use) is used to check if there are any free lines.

There will be no redial if the telephone subscriber is busy. Also this article does not describe any option for dialing of alternative telephone numbers.

The SIP Gateway can for instance be AudioCodes MP114/118 for analogue phonelines or MV-370 for GSM network. The SIP gateways should be configured as described in the relevant AlphaWiki articles.


See also Calling multiple external telephone numbers in sequence


Software requirement

  • AlphaCom: 11.2.3.3 or newer.
  • ICX-AlphaCom: any

Manual Call Forwarding

The Call Forwarding is turned on and off from the Queuing Station by forwarding the Call Request to a dummy station. The dummy number can be any station directory number in the exchange that is not in use. In this article 5152 on physical 552 is used as an example.

  • Call Forwarding On: Dial 7870 + Dummy Number (e.g. dial 7870 + 5152, or press a DAK key with the programming: I 7870 I 5152).
  • Call Forwarding Off: Dial 70

In AlphaPro, Exchange & System -> Events, press Insert to add a new event.

Event 1: Start a 1 second timer when the dummy station receives a call request.

Application4a.png


Action commands:

$ST L%1.dir W10 %2.ref 


Event 2: When the timer expires, and if there is a free phone line, the calling station will be connected to the SIP gateway (i.e. dial "0") and the phone number 12345678 will be dialed.

If there are no free lines the timer is restarted, and a new attempt will be made 5 seconds later.

Application4b.png


Action commands:

IF %op(%tin(100),>=,1)  (If the line is already in use on SIP Trunk node 100...)
$ST L%1.dir W50 %2.ref  (...then try again in 5 seconds by restarting the timer)
STOP
ENDIF (If the line is free...)
$PD %2.ref "0W12345678"  (...dial 0, then 1 sec pause, then dial 12345678)
$CANM %2.ref L%1.dir (Delete the call request from the dummy station)



About the expression "IF %op(%tin(100),>=,1)":

  • 100 = the node number of the SIP Trunk. Change as required.
  • 1 = There is one phone line connected to the SIP Gateway. If you are using MP114/118 with multiple lines connected, you need to change this number accordingly.


In the expression "$PD %2.ref "0W12345678"" substitute 12345678 with the wanted phone number.

Automatic Call Forwarding

The programming below will forward the calling station to the external telephone if the call request is not answered within 30 seconds. The time can of course be adjusted to suit your requirements.


In AlphaPro, Exchange & System -> Events, press Insert to add a new event.

Event 1: When a call request is received at the queuing station, a 30 sec timer is started. If the call request is deleted (i.e. the call request is answered), the timer is stopped:

Application4c.png


Action commands:

$ST L%1.dir W%chg(300,0) %2.ref


In the expression "$ST L%1.dir W%chg(300,0) %2.ref " 300 means 30,0 seconds. You can change this parameter if you want to use a different time.


Event 2: When the timer expires, and there is a free line on the SIP Gateway, the calling station is connected to the SIP gateway. Else the timer is restarted and a new attempt is made 5 seconds later:

Application4d.png


Action commands:

IF %op(%tin(100),>=,1)  (If the line is already in use on SIP Trunk node 100...)
$ST L%1.dir W50 %2.ref (...then try again in 5 seconds by restarting the timer)
STOP
ENDIF  (If the line is free...)
$PD %2.ref "0W12345678"  (...dial 0, then 1 sec pause, then dial 12345678)
$CANM %2.ref L%1.dir (Optional: Delete the call request from the queing station)



About the expression "IF %op(%tin(100),>=,1)":

  • 100 = the node number of the SIP Trunk.
  • 1 = There is one phone line connected to the SIP Gateway. If you are using MP114/118 with multiple lines connected, you need to change this number accordingly.


In the expression "$PD %2.ref "0W12345678"" substitute 12345678 with the wanted phone number.

Related articles