iSymphony 3.2 Beta 1 ChangeLog

Last modified by superadmin on 2021/09/09 20:55

The following is a complete list of changes made between iSymphony 3.2 Alpha 1 and iSymphony 3.2 Beta 1. If you are upgrading from iSymphony 3.0.x, please see the changelog for iSymphony 3.1.4 as well.

Features

  • A new open option (Always) has been added to the CRM Widget. When the CRM Widget is set to use this option, it will keep the specified URL open at all times.
  • A new Live API has been added to the REST interface, that provides realtime information concerning the state of the server, along with facilities to execute actions.
    • The server now accepts HTTP REST requests on the URL http://<ip or hostname of the server>:58080/communication_manager/rest/live.
    • The server now accepts HTTP REST requests on the URL https://<ip or hostname of the server>:55050/communication_manager/rest/live.
    • Resources for the following objects have been added to the REST interface:
      • Core Servers
      • Users
      • User Logins
      • User Status
      • Extensions
      • Queues
      • Agents
      • Calls
      • Statistics
    • Resources for the following actions have been added to the REST interface:
      • Setting the status of a user
      • Setting the DND state of an extension
      • Originating calls
      • Transferring calls
      • Logging agents in and out of queues
      • Pausing and Unpausing Agents
      • Setting the penalty of an agent
  • A new WebSocket Event API has been added, that provides realtime events concerning changes to the state of the server:
    • The server now accepts WebSocket connections on the URL path ws://<ip or hostname of the server>:58080/communication_manager/ws/event.
    • The server now accepts WebSocket connections on the URL path wss://<ip or hostname of the server>:55050/communication_manager/ws/event.
    • The ability to provide filters, specifying which events the WebSocket connection wants to receive, has been added.
    • The following events are now fired by the WebSocketEvent API:
      • Agent Login Event
      • Agent Logout Event
      • Agent Pause Event
      • Agent Unpause Event
      • Agent Statistic Event
      • Dial Begin Event
      • Dial End Event
      • DND Disable Event
      • DND Enable Event
      • Hangup Event
      • Link Begin Event
      • Link End Event
      • Queue Call Enter Event
      • Queue Call Leave Event
      • Queue Statistic Event
      • User Login Event
      • User Logout Event
      • User Status Event

Resolved Bugs

  • Resolved an issue that would cause client sessions to leak memory. These leaks could lead to JVM heap exhaustion and trigger OutOfMemoryErrors, causing the panel to become unstable and possibly trigger high CPU usage.
  • Resolved an issue that would cause the system to deadlock, and hang, when communication with the Asterisk server was lost.
  • Resolved an issue that would cause the Asterisk server connection socket to lock up indefinitely, when the network connection was severed unexpectedly. This would prevent any re-connection attempts to the Asterisk server, and would cause call state in the client to hang.
  • Resolved an issue that would cause the chat system to stop functioning and prevent users from receiving chat messages.
  • Resolved an issue that would cause errors to be thrown when sorting by Call Status, in the Users Widget.
  • Resolved an issue that would cause the message "Loading Recordings" to appear indefinitely in the Recording Widget, if no recordings were available on the system.
  • Resolved an issue that would prevent white space from being removed on entered username and password values, before executing login requests for the client or administrator. 
  • Resolved an issue that would cause a delay when performing attended transfers or parking calls in the panel.
  • Resolved an issue that would prevent updating of Asterisk channel variables and throw UnsupportedOperationExceptions, when new channels were added in Asterisk.
  • Resolved an issue that would cause excessive log entries when issues occurred parsing recording file names or performing CDR lookups for non-existent channel IDs. 
  • Resolved an issue that would prevent proper sorting of queue calls by position, if the number of calls in queue exceeded 9.
  • Resolved an issue that would prevent license activation errors from reporting properly in the administrator.
  • Resolved an issue that would prevent the license activation area, in the administrator, from re-enabling on a successful activation.
  • Resolved an issue that would cause an error to be thrown when logging out of the panel, if an item was selected in the Recording Widget.
  • Resolved an issue where the pause status of an agent would not update when using the FreePBX agent pause feature code.
  • Resolved an issue where updates to an agent's last call time, and number of calls taken, would be delayed.
  • Resolved an issue where the max property of the queue would report the number of total calls seen by the queue, instead of the maximum number of calls allowed by the queue. 

Improvements

  • The client and administrator login forms now trim any whitespace surrounding the entered username and password.
  • The following URL variables have been added for use in the CRM Widget:
    • ${USER_ID}: Specifies the id of the current logged in user.
    • ${USER_LOGIN_ID}: Specifies the id of the user's current login session.
  • The following CRM Widget open options have been renamed:
    • Never -> On Demand
    • Ring -> On Ring
    • Answer -> On Answer

Regressions

  • Due to the new method of setting SIP headers for PJSIP endpoints, iSymphony will no longer auto answer any calls that are stolen from another extension, or transferred from a parking lot, if the destination of the call is a PJSIP endpoint. Auto answer, in these situations, will still fiction as before, if the endpoint is managed by the chan SIP driver. Auto answer, on origination from a PJSIP endpoint, will function correctly.
   
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