This is a list of new features in Enswitch 3.11. It is not exhaustive; some enhancements are too minor to list. A full list of features is also available.

  • The Enswitch client for Android has a new user interface, and an integrated SIP softphone which automatically provisions from the server.
  • The AJAX control panel shows all PBX features in the customer. Calls can be dragged and dropped onto them to route calls to them.
  • A plain English description of a call's flow can be produced from the call history.
  • Bulk dialer destinations can be retried up to four weeks later.
  • Bulk dialer calls can be billed at a different rate then normal calls.
  • The Enswitch voicemail can play either oldest or newest message first. This can be set per mailbox.
  • The Enswitch voicemail can toggle the temporary greeting on or off.
  • Faxes sent from the web interface can include a cover page.
  • Telephone lines can have separate internal and external caller names.
  • Individual telephone lines can have their daily spending limited.
  • The number of concurrent calls to each peer can be limited.
  • Alerts can be created for any number of concurrent calls in a customer.
  • Web users can be automatically logged out after a period of inactivity.
  • Classes of service (both customer and rate plan) can be controlled by time of day, day of week, etc.
  • Customers can have a temporary credit limit controlled by time of day, day of week, etc.
  • The system owner can record sounds and make them available to all customers on the system.
  • System owner and reseller users can switch between customers faster.
  • The call routing and peer pricing tables can be loaded from an external web URL rather than the system's own database.
  • Asterisk's ConfBridge() can optionally be used for conferencing rather than Meetme(), removing the need for kernel modules to be loaded.
  • Asterisk 11 is used by default for new installations.
  • Kamailio 4 is used by default for new installations.
  • Kamailio automatically rejects SIP packets from known scanners.
  • Outbound SIP calls can include the In-Reply-To header, making it easier to audit and debug call flows using external tools such as VoIPmonitor.
  • The enswitch_routed daemon can be restarted without dropping calls in progress.
  • The Asterisk Manager Interface (AMI) events produced by enswitch_routed are more logical and consistent.
  • Enswitch daemons can use multiple database servers, with automatic failover between them.