Difference between revisions of "Substations calling external telephone(s)"

Latest revision as of 12:28, 24 February 2023

Calling external phone numbers in sequence

This article describes how substations in the ICX-AlphaCom can be configured to call external telephone number(s).

Multiple phone numbers can be called in sequence. If the first phone doesn't answer within a preset time, the call can be routed to a second (and a third...) phone number.
If all lines in the SIP gateway are busy, the call will be queued. When the line becomes free, the call will automatically progress.

The ICX-AlphaCom and SIP Gateway must be configured to use One Stage (Enbloc) dialing to achieve sequential calls and call queuing

Prerequisites

The AlphaCom and SIP Gateway must be configured as described in relevant articles.

Configure the Call Button of the substation

The call button can be configured in two different ways to dial a preconfigured phone number.

Let's assume that the prefix to the external phone line is "0", and the phone number to call is 987654321.

Alternative 1:

Call Button configuration

Start the string with a "I", followed by the prefix number "0". Then "P", followed by the phone number.

Alternative 2:

Call Button configuration

Start the string with a "T", followed by the prefix number "0". Then "w" or "W", followed by the phone number. "w" introduces a 0.5 second pause, while "W" introduces a 1.0 second pause. See Play DAK Feature for further details.

Configure the Prefix Number

One Stage dialing ("enbloc dialing") must be used. After the prefix is dialed, the succeeding digits are collected in the ICX-AlphaCom and sent to the SIP gateway in the INVITE message. The prefix number must be programmed in the AlphaCom directory table with Feature 81 and Node = SIP Trunk node number. In the "Parameter 2" field you must enter the maximum number of digits in a phone number.

Prefix number is 0, SIP trunk node is 100, collect up to 16 digits

When the prefix is dialed, the ICX-AlphaCom will collect the number of digits specified in the “Parameter 2" before sending the INVITE. If fewer digits are dialed, the ICX-AlphaCom will send the INVITE after a predefined timeout. The timeout is by default 3 seconds, and can be configured in Exchange & System > System VoIP: SIP digit collection timeout. The call setup time will be faster if this timer is set to a low value.

Digit collection timeout set to 1.0 seconds

Set SIP Gateway in One Stage dialing mode

GSM Gateway MV370

Set the GSM Gateway to One Stage dialing mode by selecting Route > LAN to Mobile Settings. Set "URL" = * and "Call Num" = #:

GSM Gateway set to One Stage dialing

AudioCodes MP114/118

Set the MP114 to One Stage dialing mode by selecting the FXO Settings page, enable 'One-Stage' dialing, 'Wait for dialtone' and 'Answer Supervision'.

Configuration tab > VoIP menu > GW and IP to IP submenu > Analog Gateway submenu > FXO Settings page item

If the FXO lines are assigned to a "Telephone Profile ID", you need to modify that particular "Telephone Profile ID". Check in the Endpoint Phone Number Table page if the FXO line is assigned to a "Telephone Profile ID":

Configuration tab > VoIP menu > GW and IP to IP submenu > Hunt Group submenu > EndPoint Phone Number page item

In the Tel Profile Settings page, select the Profile ID number, and set Dialing Mode to 'One-Stage'.

Configuration tab > VoIP menu > Coders and Profiles submenu > Tel Profile Settings page item

Make sure that Early Media is disabled in SIP Definitions > General Parameters, and in Coders And Profiles > Tel Profile Settings. If Early Media is enabled, the Gateway will respond with "183 Session Progress" instead of the required "180 Ringing" message, and the call will not be properly processed

Call queuing

If all lines in the SIP Gateway is in use, the call can be queued in the AlphaCom, and processed when a line becomes free.

In order to provide this functionality, one have to define how many lines (VoIP channels) the SIP Gateway is allowed to use. This is done from AlphaPro, Exchange & System > NetRouting, Advanced Settings tab:

Define the number of lines used by the SIP gateway

For the GSM gateway, enter the number "1", as this gateway has one channel only.
For the MP114/118, enter the number of analog FXO lines connected to the gateway.

When the call button is pressed and the gateway has no free lines, the call will be placed in a "Camp on Busy" queue and the caller will receive a busy signal. As soon as a line becomes free, the call will progress.

The Camp on Busy Time is by default 30 seconds, but can be changed in Exchange & System > System > Timers: Camp on Busy Time.

Calling multiple numbers in sequence

If the first number does not answer the call, it is possible to call a second number. If the second number doesn't answer, one can call a third number and so on. One can also route the call back to the first number, causing the calls to loop until answered. The number can be an external phone number or an internal intercom number, or a combination.

The event handler is used to detect if a call is not answered, and to set up the next call. The event 33 - Private Ringing Outgoing is reported OFF with subevent = 0 when a call in private ringing mode times out without being answered.

Example: A substation calls phone number 987654321. If no answer within the Private Ringing timeout, the call be routed to a second phone number 123456789:

Event 33 is used to manage the sequence of calls

Action commands:

IF %op(%2.dir,=,987654321)
pause
pause
pause
pause
$PD L%1.dir "0w123456789"
ENDIF

The four PAUSE commands creates a 400ms delay, which is needed for the substation to return to idle before calling the next number.

The ringing time between each number is configured in Exchange & System > System > Timers > Private Call Time. Default is 30.0 seconds. This is a global timer that applies to all private ringing calls in the system.

When the call is routed to an external mobile phone:

Activate Call waiting feature. This will ensure that the call will continue to ring also if the called phone is busy, and the call will loop to next number if not answered in due time
Disable voice mail and any other answering services, as the call will be considered as answered when connected to a voice service, and the call won't loop any further
Activate Call Forwarding to another number when the mobile is switched off or out of coverage

Example 1 - Call two different phone numbers in sequence

Call phone A. If no answer, call phone B

Example 2 - Call intercom station, then two different phone numbers in sequence.

The method described in this article work both for internal intercom numbers as well as for external phone numbers.

Call intercom station A. If no answer, call phone B, then phone C.

The intercom must be called in Private Ringing mode. Other call features like Ringing Group or Call Request are not supported

Example 3 - Call three different phone numbers in sequence. Loop back if nobody answers

Call phone A. If no answer, call B, then C. Then start over again calling phone A

@@ Line 1: / Line 1: @@
-{{A}}
+{{AI}}
-This article describes how substations in the AlphaCom can be configured to call external telephone number(s).
+[[image:Call Phone Sequence.png|thumb|right|600px|Calling external phone numbers in sequence]]
-* '''Multiple phone numbers''' can be defined. If the first phone doesn't answer within a preset time, the call can be routed to a second (and a third...) phone number.
+This article describes how substations in the ICX-AlphaCom can be configured to call external telephone number(s).
+* '''Multiple phone numbers''' can be called in sequence. If the first phone doesn't answer within a preset time, the call can be routed to a second (and a third...) phone number.
 * If all lines in the SIP gateway are busy, the '''call will be queued'''. When the line becomes free, the call will automatically progress.
-{{Note|The AlphaCom and SIP Gateway must be configured to use '''One Stage''' (Enbloc) dialing to achive the described functionality}}
+{{Note|The ICX-AlphaCom and SIP Gateway must be configured to use '''One Stage''' (Enbloc) dialing to achieve sequential calls and call queuing}}
 == Prerequisites ==
 The AlphaCom and SIP Gateway must be configured as described in relevant articles.
+* [[IP Telephony Service (ICX-AlphaCom)]]
 * [[AudioCodes MP114/118, v6.2 and higher|AudioCodes MP114/118]]
 * [[GSM gateway (Mobile VoIP)|GSM gateway (MV370)]]
-<br>
 ==Configure the Call Button of the substation==
@@ Line 30: / Line 30: @@
 Start the string with a "T", followed by the prefix number "0". Then "w" or "W", followed by the phone number. "w" introduces a 0.5 second pause, while "W" introduces a 1.0 second pause. See [[Play DAK Feature]] for further details.
-<br>
 ==Configure the Prefix Number ==
-One Stage dialing ("enbloc dialing") must be used. After the prefix is dialed, the succeeding digits are collected in the AlphaCom and sent to the SIP gateway in the INVITE message.
+One Stage dialing ("enbloc dialing") must be used. After the prefix is dialed, the succeeding digits are collected in the ICX-AlphaCom and sent to the SIP gateway in the INVITE message.
 The prefix number must be programmed in the [[Directory_%26_Features_(AlphaPro)|AlphaCom directory table]] with '''Feature 81''' and '''Node''' = SIP Trunk node number. In the "Parameter 2" field you must enter the maximum number of digits in a phone number.
@@ Line 39: / Line 38: @@
 <br style="clear:both;" />
-When the prefix is dialed, the AlphaCom will collect the number of digits specified in the “Parameter 2" before sending the INVITE. If fewer digits are dialed, the AlphaCom will send the INVITE after a predefined timeout.
+When the prefix is dialed, the ICX-AlphaCom will collect the number of digits specified in the “Parameter 2" before sending the INVITE. If fewer digits are dialed, the ICX-AlphaCom will send the INVITE after a predefined timeout.
 The timeout is by default 3 seconds, and can be configured in [[Exchange_%26_System_(AlphaPro)#VoIP|Exchange & System > System VoIP: '''SIP digit collection timeout''']]. The call setup time will be faster if this timer is set to a low value.
 [[File:SIPDigitTimeout.PNG|left|thumb|500px|Digit collection timeout set to 1.0 seconds]]
 <br style="clear:both;" />
-<br>
 == Set SIP Gateway in One Stage dialing mode ==
@@ Line 68: / Line 66: @@
 <br style="clear:both;" />
-{{Note|Make sure that '''Early Media''' is disabled in '''SIP Definitions''' > '''General Parameters''', and in '''Coders And Profiles''' > '''Tel Profile Settings'''. If Early Media is enabled, the Gateway will respond with "183 Session Progress" instead of the required "180 Ringing" message}}
+{{Note|Make sure that '''Early Media''' is disabled in '''SIP Definitions''' > '''General Parameters''', and in '''Coders And Profiles''' > '''Tel Profile Settings'''. If Early Media is enabled, the Gateway will respond with "183 Session Progress" instead of the required "180 Ringing" message, and the call will not be properly processed}}
@@ Line 79: / Line 77: @@
 <br style="clear:both;" />
-* For the GSM gateway, enter the number "1", as this gatway has one channel only.
+* For the GSM gateway, enter the number "1", as this gateway has one channel only.
 * For the MP114/118, enter the number of analog FXO lines connected to the gateway.
@@ Line 86: / Line 84: @@
 The Camp on Busy Time is by default 30 seconds, but can be changed in [[Exchange_%26_System_(AlphaPro)#Timers|Exchange & System > System > Timers: '''Camp on Busy Time''']].
-<br>
 == Calling multiple numbers in sequence ==
@@ Line 93: / Line 90: @@
 The event handler is used to detect if a call is not answered, and to set up the next call. The event [[Private Ringing Outgoing (Event Type)|33 - Private Ringing Outgoing]] is reported OFF with subevent = 0 when a call in private ringing mode times out without being answered.
-Example: A substation calls phone number 987654321. If no answer within the Private Ringing timeout, the call be routed to a second phone number 1234567789:
+Example: A substation calls phone number 987654321. If no answer within the Private Ringing timeout, the call be routed to a second phone number 123456789:
 [[File:Dial Sequence.PNG|left|thumb|500px|Event 33 is used to manage the sequence of calls]]
 <br style="clear:both;" />
-Action Commands:
+{{Code2|
+  IF %op(%2.dir,<nowiki>=</nowiki>,987654321)
-  IF %op(%2.dir,=,987654321)
   pause
   pause
@@ Line 107: / Line 103: @@
   $PD L%1.dir "0w123456789"
   ENDIF
+}}
-<br>
 The four PAUSE commands creates a 400ms delay, which is needed for the substation to return to idle before calling the next number.
 The ringing time between each number is configured in [[Exchange_%26_System_(AlphaPro)#Timers|Exchange & System > System > Timers > '''Private Call Time''']]. Default is 30.0 seconds. This is a global timer that applies to all private ringing calls in the system.
+{{note| When the call is routed to an external '''mobile phone''':
+* ''Activate '''Call waiting''' feature. This will ensure that the call will continue to ring also if the called phone is busy, and the call will loop to next number if not answered in due time''
+* '''''Disable voice mail''' and any other answering services, as the call will be considered as answered when connected to a voice service, and the call won't loop any further''
+* ''Activate '''Call Forwarding''' to another number when the mobile is switched off or out of coverage'' }}
+<br>
 === Example 1 - Call two different phone numbers in sequence===
-[[File:PhoneSequence1.png|left|thumb|700px|Call two different phone numbers in sequence]]
+[[File:PhoneSequence1.png|left|thumb|800px|Call phone A. If no answer, call phone B]]
+<br style="clear:both;" />
+<br>
+=== Example 2 - Call intercom station, then two different phone numbers in sequence. ===
+The method described in this article work both for internal intercom numbers as well as for external phone numbers.
+[[File:PhoneSequence3.png|left|thumb|800px|Call intercom station A. If no answer, call phone B, then phone C.]]
+<br style="clear:both;" />
+{{note|The intercom must be called in [[Private mode|Private Ringing mode]]. Other call features like [[Ringing Group Feature|Ringing Group]] or [[Call Request]] are not supported }}
+<br>
+=== Example 3 - Call three different phone numbers in sequence. Loop back if nobody answers===
+[[File:PhoneSequence2.png|left|thumb|800px|Call phone A. If no answer, call B, then C. Then start over again calling phone A]]
 <br style="clear:both;" />
+[[Category: ICX-AlphaCom - SIP Integration]]
+[[Category: AlphaCom - SIP Integration]]
+[[Category: AlphaCom Applications]]
+[[Category: ICX-AlphaCom Applications]]