View source for INCA 2.3 - Release Notes

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This document provides the release notes for INCA 2.3 with incremental bug fix releases. The release notes for INCA 2.3 describe new features, improvements and issues fixed after INCA 2.2.

=IP-station 02.03.3.3 (2013-07-18)=
 Release: Official, available on request

'''IP Station Main upgrade file:'''<br />
Image: A100G80200.02_03_3_3.bin<br />
Checksum: ACC51113<br />

==Bug Fixes==

===Fixed a bug with IPARIO DAK led handling (led handling didnt exist in the 02.03 versions).=== 
===Fixed a potential error with Pulse server which could cause infinite loop of messages taking up alot of system resources.=== 
===Fixed a problem in Pulse mode where call could end up without echo cancellation in certain combinations with group call.=== 
===Fixed a problem with registration towards some SIP servers.===

=IP-station 02.03.3.2 (2013-07-08)=
 Release: Official, available on request

'''IP Station Main upgrade file:'''<br />
Image: A100G80200.02_03_3_2.bin<br />
Checksum: 81E1BA97<br />

==Bug Fixes==

===Fixed a bug where handset/normal volume could get mixed up.=== 
===Fixed a bug where disabling relay activation would cause remote controlled volume override to stop working.=== 
===Fixed a bug in Pulse Server where it would use wrong port in the SIP messages. It could cause conversations to be set up wrong if the SIP device was using non-standard SIP port (anything other than 5060).===
===Fixed a bug with provisioning of dak/input in call settings.===

=IP-station 02.03.3.1 (2013-02-28)=
 Release: Official, available on request

'''IP Dual Display upgrade file:'''<br />
Image: A100G802D0.02_03_3_1.bin<br />
Checksum: 7A0A9B8F<br />

'''IP Station Main upgrade file:'''<br />
Image: A100G80200.02_03_3_1.bin<br />
Checksum: 84FAF0CD<br />

''IP Station Main supports all STENTOFON stations except IP Dual Display and Turbine''

==General Enhancement==

===IPARIO now uses the same image as IP Desktop / Master / Sub===
Images for IP Desktop / Master / Sub now also supports IPARIO. 

===Can now choose IGMP version===
It is possible to choose between IGMP version 2 or 3, or put it on "default" for highest available version. Recommended is "default".

===Now possible to do factory reset from web===
Factory reset can be performed from Station Administration -> Reboot.

===Automatic redirection in web===
When choosing static IP settings and clicking save, the browser should now redirect to the correct IP address when the station has rebooted.

===DHCP request option 12 and 81 is now used===
DHCP request option 12 and 81 is set to "zenitel" + the last 6 digits of the mac address by default, it is also configureable.

===Now showing more network information in the station information screen===
Station information shows discovered DNS servers.

===The stations will now remember their last DHCP IP Address and request it during DHCP negotiation===
If the stations restarts it will always ask to get the same IP Address as the last time. This is always enabled.

===Stations can now continue to use last DHCP IP Address when the DHCP server fails to respond===
This can be turned on in Main Settings by enabling the "Use Last IP On DHCP Failure" checkbox.

==SIP Enhancement / Bug fixes==

===SIP should now handle IP address changes better===
If the station was on DHCP and then received a new one, it would still use the old ip address in the SIP messages. This has been fixed.

===Fixed an error in the ACK message which caused conversations to be hung up after 30 seconds===
An error in the ACK message caused calls to time out. This has been fixed.

===Fixed an error with URI in Authorization header which caused problems against Samsung SIP Server===

===Added more SNMP GET information===
Added SNMP GET information for tone test status, software version and button hanging status. 

===Added support for 3 parallell registrations and configurable seriel registration===
Default is now parallell registration. Note: Parallell registration can cause problems in some environments for example with Cisco. Use seriel registration instead if parallell causes problems.

===Added support for DTMF Flash on Input Button===
This can be used when using "Send DTMF" during calls.

===Fixed some problems that could cause calls to not be set up properly when using proxies / authentication===

===Fixed error with "Stop call" from web interface===
The station could end up in private mode.

===Completely changed the relay functionality, added alot of new options===
Can configure several DTMF events to change the state of the relay. Can also configure station state events to change the state of the relay. This can be done in SIP Configuration -> Relay Settings.

===Added option for increasing tone volume===
This is configureable in SIP Configuration -> Audio Settings.

===Added option to configure delay for setting up call (press input/dak button for X seconds)===
This can be configured in SIP Configuration -> SIP Settings with "Delay Call Setup".

===TFTP provisioning configuration has been changed alot===
Configuration of ringlist and dak/input buttons has been changed (example to configure input 1 is now input1_value=1003 instead of speeddial_1_c1). See updated wiki about SIP Provisioning.
Can now configure tone test, new relay configuration, noise reduction, echo parameter.

===Fixed some issues with RFC2833 when using other payload type than 101 in the RTP packets===
The station will now correctly receive RFC2833 with other payloads than 101.

===Fixed an error which caused hostnames not to be resolved if the DNS server responded later than 5ms===
This was only a problem if a domain name was configured as sip domain.

===Fixed an error with DSCP on VoIP packets===
Only packets sent to port 61000:61150 had DSCP.

===Increased SIP receive buffer size to 5000 bytes (from 3000 bytes)===
Some SIP servers sent messages above 3000 bytes, which caused problems.

==DIP Enchancement / Bug Fixes==

===Added support for using old handset in new 8023 hardware at the headset input===
New 8023 hardware does not support using old handset in the handset slot. Use the headset slot instead of and enable the flag under Advanced Alphacom -> Audio called "Handset In Headset Slot".

===Added support for tonetest in IP only environments===

===Fixed several errors where sound could dissappear in conferences===
Using 2 audio channels could cause sound to dissappear (for example talking in conference). Using normal multicast was bugged (relayed multicast was working).

===Fixed an error which caused multicast to dissappear when using IGMP snooping===
This caused problems for functions using groupcall / conference.

===Added more options for MVO (mask volume override)===

[[Category: Release notes]]