Goto.INTERFACE

Release notes

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

Dependencies

• 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)

Notes

• 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>