2.1-p3

Fixed a bug in VIP where trunks and providers would always use the IP of the primary NIC, even if they were on the secondary NIC. 

Added capability for "late media" negotiation which enhances compatibility with OnSIP/WebRTC servies and CUCM

VIP can not offer outgoing call ringback in certain CUCM scenarios. Implemented a "fake" ringback tone option for these instances
added SIP trace packet capture options

Better support in web-based interface for non-latin characters

Better support for gateways on secondary port

Fix/changes to correct bug where SIP trunks could not be re-started when settings were changed (applied).

Additional config options in SIP trunks to handle different types of incoming call identification.
	   "Trunk Incoming Match" and "Trunk Incoming Match Parameter" allow matching on a configurable originating network address,
	   or configurable destination number.  Default settings match originating network address to realm to ID the trunk.


2.0
Split Studio – allows customer to run two studios from one mainframe

Call transfer - transferring SIP calls to internal VoIP extensions or other SIP addresses now supported

Dual Ethernet ports are enabled – greater configuration flexibility

“Call End Reason” display – GUI now displays reason for call termination

Numerous bug fixes

Optional mode allows directing calls specifically to hybrid 1 or hybrid 2 ("user air mode")


Optional tone played to caller to alert them when they are put on the air


Mainframes may now be named, and this name shows in the GUI and in Device Manager


Improved network configuration options to offer more control over security and compatibility in different IT environments 


Improved SIP Gateway compatibility to support T1/E1/ISDN gateways 


1.2 p2
Change port used for Control Surfaces--Mainframe 1.2 now requires Control surface 1.2 or above

Fixed audio bug resulting in distorted audio over time

Changes required for support of 3Com NBX SIP PBX

Changes to support VoIP delivered by Frontier (and hopefully other similar SIP trunking providers) New option is called "SIP Trunk"


What's new- Version 1.1 includes enhanced compatibility with SIP providers, PBXs and Gateways, adds "Big Time" clock and timer feature, and supports logging and extraction of caller info.

1.1
Adds Caller Ducking function and new version of AGC (see manual)

Fixes unintentional noise gate issue with on-air calls.

Adds configuration settings for CID name and CID number sent for outgoing calls.  This impacts SIP providers only.

Fixes "00000000" caller ID info when making outgoing calls.  It now properly displays the number dialed.

1.1-test2
Audio dropouts/clicks/pops with G.711 and gateway calls,delay buffers tolerance increased.  

Hybrid ordering option fixed with certain audio I/O configurations.

System crash after factory reset fixed.

1.1-test1 - Big Time, event logging, outgoing call fixes 

NEW FEATURE: Big Time - a large time display with a shared timer feature.  
Additionally, STAC VIP now supports NTP time synchronization, so STAC VIP time may be synced to internet time references or to standalone synchronization devices (e.g., GPS-based).

NEW FEATURE: Event Logging - STAC VIP now logs events internally, and an external program or script can request and download this log file periodically.  This exports all major system events as well as actions taken on calls.

Bug fixes and improvements made to outgoing call behavior. Occasionally outgoing calls could not be dropped properly. Fixes now include showing a new "outgoing ringing" state for certain SIP calls, to indicate the remote end is ringing.  Certain calls still behave as they did before, where the ringback is heard on handset or on air, however.

1.0-p8 - hold audio fixes, crash recovery, earpiece volume boost
Improvements made to hold audio behavior. Also adds some additional robustness to the system and allows recovery from certain crashes.

   
1.0-p7-test4 - adds Cisco UCM (CUCM) support
Adds SIP provider options which allow the STAC VIP to interoperate properly with a Cisco Unified Call Manager VoIP PBX system.
The two options appear under "advanced" settings in a generic SIP provider.  "CUCM Compatibility Mode" must be enabled to deal with the CUCM's behavior of sending no SDP in the SIP INVITE.  Additionally, the Codec Priority must be adjusted to exclude ISAC - new options have been added which include all codecs but ISAC, and G.722 only.

1.0-p7-test3 - adds dialing prefix to sip gw
Adds a dialing prefix option to SIP Gateway providers.  The original need is to support selecting the outgoing line when using a GrandStream GXW4108, though it may find other uses.


1.0-p7-test2 - DTMF during active call
Adds feature to allow sending DTMF from a control surface during an active call.  This allows using services such as Webex conferencing, which require a code to be entered.

Minor changes to audio feedback in earpiece when using the number keypad:

           - When dialing a call, 0-9 play a beep and "send" plays a chime
           - When an active call is in progress, 0-9*# play their respective
             touch tone codes.  There is a known slight delay between
             keypress and hearing the tones.
           - When using the AA menu, no tone feedback is used for button
             presses.  The voice prompts provde the feedback.

1.0-p7-test1 - bug fixes


With a bogus/unconfigured provider in the provider list, it would break incoming calls on providers listed after that bad provider by falsely returning USER_BUSY state and not accepting the call.
Deleting providers from the list would usually result in the config file not being written out, thus when rebooting the mainframe the deleted providers would re-appear.
Auto-attendant enable state not preserved during a configuration restart - system would show AA being enabled after a restart, but would not answer calls unless AA cycled off/on.
Simulated providers would not properly show as "registered" after a configuration restart, they would perpetually be stuck in the initializing state until a system reboot.

1.0-p6 - bug fixes

Mainly fixes line limiting per provider, should now properly return USER_BUSY if all assigned lines for a provider are full.

1.0-p6-test1 - bug fix

Fixes erroneous broadcast address setting under certain circumstances when using DHCP.

1.0-p5-test1 - bug fixes

Fixes "[blank] has exited chat" messages in GUI
	   
Now properly limits number of callers based on the line assignments in the configuration.  Callers will receive USER_BUSY SIP message if all assigned lines for a given provider are occupied.  This also fixes a bug where if the entire STAC VIP was full of active calls, that incoming calls with AA turned on would still be answered and placed on hold, but invisible to the STAC VIP GUI/control surfaces.

2012-11-14 version 1.0-p4 - fixes for blocking socket bug

Fixes an issue under heavy loads, where if clients did not cleanly disconnect, the socket buffers would fill up and the vip application would block until the socket timed out.  This caused what appeared to be a temporary system lockup, after which the system would appear to recover.

Also necessitated an upgrade to the latest configuration GUI, so the look and feel of that has changed.

In addition, an "OnSIP" provider profile has been added to make it easier to configure.


1.0-p3 - changes for SIP gateways

bug fixes and improvements for behavior of POTS-to-SIP gateways specifically tested with AudioCodes and Sangoma Vega products


1.0-p1 - changes for crexendo.com SIP provider
	   
adds a new SIP provider type "crexendo.com" which sets some internal parameters so the Request-URI portion of the REGISTER was to their liking.

added two parameters to the generic SIP provider which would allow this change the manual way if necessary ("outbound proxy" and "register proxy")


1.0 - initial release