VitalPBX 2.0.5

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  • Trunks / Dialing Manipulation Rules: A new section has been included in the trunks module named “Dialing Manipulation Rules”, that allows you to manipulate the dialed number depending of the trunk. E.g.: Suppose you have two providers, both have emergency calls service, but the number to dial is different for each one, for the first provider you must to dial 933 and for the second one you must to dial 944. So, you can configure in your outbound route the 911 and replace this number depending on the trunk on which the call is dialed through.

Dialing Mainipulation Rules

  • Trunks / DID and CID from Headers: Now is possible to define if the DID or the CID will be extracted from the SIP Headers. This only applies for SIP trunks

Get DID/CID from SIP Headers

  • Blacklist / Pattern and Destinations:  Now it is possible to define a blacklist number with a pattern and define a destination for it. Also, it is possible to disable/enable a blacklist item through the GUI. If no destination is defined for the blacklisted item a message will be played to the caller.


  • SIP Settings / NAT: Now it is possible to define the NAT setting as a global parameter. If you want to use the global value for SIP devices/trunks, you must set NAT parameter to the “Default” value for them

Global Nat Setting

  • Paging / MulticastRTP: Now it is possible to make paging through the MulticastRTP channel.  This new feature for paging can be used for: emergency broadcast service for fires, earthquakes, emergency broadcast services for hospitals, In a school setting, In the corporate setting; it could inexpensively replace costly PA systems requiring dedicated wiring, speakers, and amplifiers.

Paging Multicast RTP

  • Status Report: Now the Peers & Hints can be filtered easily

Status Report

  • Security: The default fail2ban(Intrusion Detection) settings have been updated to secure the PBX more efficiently
  • Language: Spanish and Russian translations have been updated


  • Inbound Routes: The incoming routes were not generated in the right way when the DID field was empty
  • Night Mode: It was not possible to change the night mode status or set the global mode status
  • Voicemail: The voicemail configurations were not generated in the right way
  • Destinations: The destinations for the “Terminated Call” option were not diverted to the defined location
  • CID Lookup: The incoming routes with a CID Lookup assigned were broken when the defined CID Lookup server was down




About Us

This project started with the objective of creating a system/interface for the administration of PBX systems based on Asterisk,  easy to use, totally adapted for different mobile devices (Fully Responsive Design), and with all the characteristics of an advanced telecommunications system; Combining the flexibility from Asterisk with concepts that have been satisfactorily used in traditional telephone systems, concepts that somehow were ignored by the new generations of IP telephony.

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