Release notes

Version 1.41.0 - General deployment
  • Limitation: Hiding your number doesn't work in combination with the module Asterisk-1.2x (M6503)
  • Feature: Added CLIR mode in order to be compliant with IMS standard (M5276)

Version 1.40.2 - General deployment
  • Bugfix: Removing of the the parenthesis into the variable LastUserExt at the line 48 (M7049)

Version 1.40.1 - General deployment
  • Bugfix: A deprecated application was used which caused bad execution on the Communication Server module

Version 1.40 - General deployment
  • Improvement: Compatibility with the Communication Server module

Version 1.38 - General deployment
  • Improvement: Renamed a number of options to make them more understandable
  • Improvement: Allow setting of Sip Privacy header by setting ${SipPrivacyHeader} global variable

Version 1.37 - General deployment
  • Bugfix: Better implementation of RFC3325 on implementation of CLIR on SIP

Version 1.36 - General deployment
  • Feature: Added possibility to hide the outgoing number

Version 1.35 - General deployment
  • Bugfix: SipPrivateAssertedIdentityNumber bug fix

Version 1.34 - General deployment
  • Bugfix: Bad release cause propagated in case of congested or unavailable interface (M0001704)

Version 1.33 - General deployment
  • Bugfix: CallerId is sometimes hidden while it should not

Version 1.32 - General deployment
  • Feature: Add the possibility of DTMF based supplementary services

Version 1.31 - General deployment
  • Bugfix: "Amount of possible digits to strip" > 7 was not possible

Version 1.30 - General deployment
  • Fix: Play a busy tone after the 4th interface
  • Feature: Increased the amount of possible digits to strip
  • Feature: Add the option to clear the RDNIS
  • Feature: If [PrivateAssertedIdentityNumber is defined, use P-Asserted-Identity privacy header with the given number.

Version 1.29 - General deployment
  • Handle DIALSTATUS busy correctly
  • number of digit to strip increased
  • set maximum call duration

Version 1.28 - General deployment
  • Handle DIALSTATUS busy correctly

Version 1.27 - General deployment
  • add ENUM support
  • add the possibility to see the Redirecting Number when local channel are created during a phone call forward. (Needs STARTDYNAMIC Application 4.00)
  • add the possibility to change the sender of the ringtone between the network/equipment and the PBX

Version 1.26 - General deployment
  • ENUM support added
  • Potential update impact level 2 DONE: in the event this update contains a bug, it might have critical impact. Respect dependencies and retest your most important callflows and applicative integrations. possibility added to see the Redirecting Number when local channels are created during a phone call forward. (Needs STARTDYNAMIC Application 4.00)

Version 1.25 - General deployment
  • increase of the amount of front digits to strip

Version 1.24 - General deployment

Version 1.23 - General deployment
  • fix BRI fallback

Version 1.22 - General deployment
  • add default CLI parameters
  • add Caller ID transparency policy

Version 1.20 - General deployment
  • possibility to add a new prefix per interface.

Version 1.10 - General deployment
  • possibility to strip and add front digits

Version 1.00 - General deployment

Version 0.9 - Deprecated

Version proxy - Deprecated
  • add ENUM support
  • add the possibility to see the Redirecting Number when local channel are created during a phone call forward. (Needs STARTDYNAMIC Application 4.00)
  • add the possibility to change the sender of the ringtone between the ISDN network/equipment and the PBX

Action parameters

  • Description: Free description to describe the goal of the action instance in the routing
  • Default Outgoing Interface : First interface to call
  • Fallback interfaces : Alternative interface in case of congestion of the first interface
  • Number of front digit to strip : Remove leading digit(s) of the number which must be dialed. By default 1 digit is removed in order to remove the external access code.
  • Prefix applied to the Default Interface : Prefix which is be added to the number which must be dialed after the front digit are stripped for the default interface. This is often used for example to select an alternative provider.
  • Prefix applied to Fallback Interface(s) : Similar prefix for fallback interfaces.
  • Caller ID policy:
    • Translate (previously "As set in Global Parameter"): Update the caller id using the information from Outgoing Number Mapping (callerid_[caller id] global parameters). If the calleID is null, the CLIR flag and the sreening indicator flag will be set.
    • Transparent: Do not change the caller id
    • Clear RDNIS: RDNIS (Redirected Dialed Number Information Service) is mainly used on ISDN interfaces. The normal way is that when a call is redirected through a phone set call forwarding, the RDNIS will be the initial caller id. Fore some reasons, some provider does not support RDNIS (example: BASE mobile provider). This feature enables to clear it. (available on version 1.30 or higher)
    • CLIR: CLIR stands for Calling Line Restriction. On an ISDN line this will set the CLIR flag and the screening indicator flag in the outgoing SETUP in order to ask the network to keep the Caller ID private, except for the emergency services and billing services. In SIP, it will add the Privacy header in order to indicate that the provided caller ID must remain private. When used together with the 'Translate' mode, this will make sure that a proper caller ID is always provided to the remote network. If the callerid_[caller id] is empty, the PrivateAssertedIdentityNumber will be used instead in order to set the Caller ID.
  • Options:
    • Network ring tone: keep the network ringing tone
    • PBX ring tone: force the generation of the ringing tone by the PBX
    • DTMF call control: enable DTMF based call transfer


  • Global Parameters:
    • SipPrivateAssertedIdentityNumber (extension, adminconst) (deprecated): If this variable is set when the action is called, the Goto.Interface action will use this number to send the privacy header in SIP invite message when the CLI must be masked. This can be requested by specific SIP providers (Example: Belgacom NGN SIP trunk). (Only used in version 1.30 or higher). See notes bellow for more information.
    • SipPrivacyHeader (string, adminconst) (deprecated): When SipPrivateAssertedIdentityNumber is set you also have to set the SipPrivacyHeader variable. For more information, visit the SipProviderSettings page. (Only used in version 1.38 or higher). See notes bellow for more information.
    • ForceCLIR: If set to 1, the CallerID will considered as hidden. This parameter is to be used on asterisk 1.2 only. On the communication Server the Caller Presentation indocator is used instead.
    • MaximumCallDuration : If this variable is set, the call duration will have the corresponding maximum duration. (Only used in version 1.29 or higher)


  • In replacement of the SipPrivateAssertedIdentityNumber and SipPrivacyHeader you can set in the SIP trunk, the Caller ID method to P-Asserted-Id method (outgoing). The action CallInterface will take care to add the header based on the proper Caller ID as defined in the Caller ID policy in the CallInterface parameters. With the Communication Server, the Caller ID presentation indicator will also be used in order to decide to add or not the Privacy header and to remove the caller ID from the From header. In asterisk-1.2, this can be achieved by set the parameter _ForcleCLIR to 1.
  • For a very customized behaviour, the action SIpSetHeader can be used in order to have a full control of the SIP headers. In that case set the Caller Id Method in the SIP trunk to Use From Header.

Phone based call forwarding

Version 1.42.0 removed a legacy feature which is no longer supported. If you were setting a phone based call forward to an external number and you were using an outgoing callflow to the PSTN, you have to modify the outgoing callflow to maintain this legacy behaviour. You have to add a few actions to your callflow:
If($[$["${CHANNEL:0:5}" = "Local"] & $["${LastUserExt}"!=""]])
 | True: SetVar(CALLERID(num)=${LastUserExt})
 |  |
 |  | Redirect (<extension of false>)
 | False : <Rest of the previous callflow>
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