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This document provides the release notes for INCA 2.2 with incremental bug fix releases. The release notes for INCA 2.2 describe new features, improvements and issues fixed after INCA 2.1.

IP-station 02.02.3.1(2011-16-11)

Release: Official, available on request

IP Dual Display upgrade file:
Image: A100G802D0.02_02_3_1.bin
Checksum: E7151957

IP Station Main upgrade file:
Image: A100G80200.02_02_3_1.bin
Checksum: 9AFF3E64

IP Station Main supports all STENTOFON stations except IP Dual Display and IP ARIO

IP Dual Display Enhancement

Added "dots" in the DAK display

Added "dots" in the IP Dual Display DAK display to indicate which button belongs to a given line of text (same as for analog Dual Display).

Enabled Pulse/SIP mode

Enabled station modes Pulse and SIP for IP Dual Display

Support for flashing / inverse video

Added support for flashing / inverse video, see IND for more details

DIP Error Corrected

BZ 283 Display text bug with quote is fixed

BZ 605 Leading space in directory number should not be possible anymore

BZ 643 Echo on remote side when using handset is fixed

SIP/Pulse Error Corrected

Registration failure when backup server is down

Fixed a bug where the station would not register on Primary Server if it was registered on Backup Server and Backup Server 2 was empty.

Fixed memory leak during call forwarding

Station sending SIP requests with wrong CSeq

Fixed an error that made the station send SIP requests using a previously used CSeq. This could make the receiving station confuse it with a retransmission instead of a new SIP request.

The station will now handle 6xx responses on a register message

The display will now automatically update when the station is not registered

SIP/Pulse Enhancement

SIP MESSAGE audio control

(SIP only) SIP MESSAGE can now control audio direction if "Message Controlled Volume Override Mode" is enabled under Call Settings:

SIP MESSAGE ”Audio_receive_only”: Turns the microphone off and loudspeaker on
SIP MESSAGE ”Audio_send_only”: Turns microphone on and loudspeaker off
SIP MESSAGE ”Audio_send_receive”: Turns both microphone and loudspeaker on

Forced Push To Talk mode

Added option to force station to be in Push To Talk mode under Station Configuration -> Audio Settings.

The station is now busy while dialing or using handset

Call queue

Calls are now put in a queue when the station is busy. The first call in the queue is shown when the previous call is hanged up. A call is in the queue for a maximum of 20 seconds.

Language settings

It is now possible to choose between different languages in the web menu under SIP Configuration -> Language Settings. It is also possible to upload your own translation or download existing translations.

Improved ringlist

The ringlist can now setup multiple calls at once and the station ends up in conversation with the first station that answer, while all other calls are cancelled.

Display user menu

The user may now configure call forwarding, auto answer and summer time from the menu without requiring password.

Call history

Missed calls, Outgoing Calls and Incoming calls will now only show the last call attempt from/to a station (instead of adding a new entry for each call attempt).

New SNMP traps

Added new SNMP traps: (Input) Button Pressed, (Input) Button Released, Relay Activated, Relay Deactivated

New options for input buttons during call

Added options for Input Buttons in call: End Call / Do Nothing / Send DTMF / Send Text.

DTMF 0 resets audio direction if Remote Controlled Volume Override is enabled

Sending DTMF 0 now resets audio to open duplex mode if Remoted Controlled Volume Override is enabled.

A tone is now played in handset when idle

A tone is now played on call accepted

Status LED now shows when the station is registered / got IP / not registered

Renamed "update" to "refresh" in the webserver

Added "Press M to Transfer" when transferring

Added "Press M to Answer" when the station is ringing

Pulse Error Corrected

Pulse station would display Backup Servers in the web and display

Sound during high load

The sound should not lag during high load anymore.

Pulse Enhancement

Push to talk is now always enabled for Pulse stations

Busy override

Calling stations may now do busy override if a remote station is busy and the profile is allowed to use busy override. Busy override puts the calling station first in the queue and hangs up the previous call.

Group call

It is possible to use up to 4 group calls. Using group call requires that you configure who are allowed to start a group call in Server Management -> Profiles settings, then add stations in Group Call settings. The group calls can have 4 priorites: LOW, NORMAL, HIGH, EMERGENCY. The sound in a group call with LOW/NORMAL priority are not play during normal calls. HIGH priority group call is mixed with normal calls. EMERGENCY priority group calls are played out with max volume and normal calls are muted.

Outgoing group calls now show nickname

Server ringlist configuration

It is now possible to configure the ringlist of all Pulse stations at the same time from Server Management -> Ringlist.

Directory display menu shows allowed group calls

Known Issues

BZ 494 Multicast does not work without a default gateway defined

Workaround is to use the Alphacom IP address or Pulse Server IP address as gateway. Note that group calls require multicast in Pulse and (depending on configuration) Alhpacom mode.

BZ 663 Webcall does not work when "Select Line" is configured in DAK settings

BZ 665 Wrong configuration in Advanced configuration->Updates reboots the station

BZ 665 Removing a number in the IP address field under Advanced configuration-->Updates then clicking save will reboot the station each time that webpage is loaded.

BZ 675 Show no arrow parameter is not supported in IP Dual Display

Changing to a different language requires reboot of the stations

RFC2833 does not work for stations in Pulse mode

Workaround is to use SIP INFO or to set the station in SIP mode.

INCA 2.2 - Release Notes

From Zenitel Wiki

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