Tuesday, 16 October 2012


NEW - Provide SIP VOIP Extensions for use with USB headsets

NEW - Add ability to match SIP requests on To Header or Request Header

NEW - Add SIP Livewire music on hold

NEW - Allow SDP pass through when receiving calls from SIP provider

NEW - Add dial short code and dial short code withhold functionality

NEW - Add support for multiple Riedel interfaces

FIX - Show router output name when device routed to unknown destination for current client

FIX - Axia GPIO incorrect pin number being reported

FIX - Cisco AnyConnect preventing client loading correctly

FIX - Hide Close Services button when in mini-mode as appropriate

FIX - Allow SIP handset port to be specified

FIX - Check ACK from SIP Invite for new audio port

FIX - Improve efficiency of unparking to SIP handset

FIX - Improve efficiency of answering to SIP handset

FIX - Use IP and port when transferring between SIP handsets

FIX - Numerous fixes for Pathfinder implementation -  loading initial routes and route change notification

FIX - Numerous sync issues

FIX - Copy codec colour when copying device layout in web manager

FIX - Sorting on external inputs and outputs in web manager

FIX - Copy client editor setting when copying machine config in web manager

FIX - Vx service extension page in web manager

FIX - Validate dial service before saving extension in web manager

FIX - Chat messages in random order after client reconnect