INCA 2.3 - Release Notes

IP-station 02.03.3.3 (2013-07-18)

Release: Official, available on request

IP Station Main upgrade file:
Image: A100G80200.02_03_3_3.bin
Checksum: ACC51113

IP-station 02.03.3.2 (2013-07-08)

Release: Official, available on request

IP Station Main upgrade file:
Image: A100G80200.02_03_3_2.bin
Checksum: 81E1BA97

IP-station 02.03.3.1 (2013-02-28)

Release: Official, available on request

IP Dual Display upgrade file:
Image: A100G802D0.02_03_3_1.bin
Checksum: 7A0A9B8F

IP Station Main upgrade file:
Image: A100G80200.02_03_3_1.bin
Checksum: 84FAF0CD

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.

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 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".

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.