Release Number | AIX | Linux | Solaris | Windows |
---|---|---|---|---|
2.2.120.08 [04/10/18] – General | X |
This release note applies to all 2.2.12 releases of 420HD IP Phone Firmware (part of the Genesys 400 series of IP Phones).
This is a General release for this product. Version 2.2.12 includes firmware builds 2.2.12.172 and 2.2.12.126. This section describes new features that were introduced in this release.
Version 2.2.12 with the firmware build 2.2.12.172 offers the following features:
Technician's digit key code. Technicians installing phones at customer sites no longer need to connect laptops to phones to provision them. After connecting phones to the network, technicians now enter a specific digit key code which changes the phones' provisioning URL to the server's URL. If the code that the technician enters matches, the phones are automatically provisioned from that server. The feature requires software customization.
Configurable OPUS dynamic payload type. Ability to configure the OPUS dynamic payload type. Previously, the OPUS dynamic payload type could not be modified.
Canadian French (Français Canadien) language
Version 2.2.12 with the firmware build 2.2.12.126 offers the following features:
Capability to handle multiple calls - N Concurrent calls (NCC) The phone is capable of managing up to 8 concurrent calls per line, for example, of holding multiple calls and switching between them. The feature is most relevant to the enterprise front desk.
Electronic Hook Switch (EHS) DHSG. Answering calls and changing volume level with EHS-capable headsets is now supported. This newly supported capability can be enabled by setting the configuration file parameter 'voip/services/electronic_hook_switch/enabled' to 1.
The feature was verified using the following headsets:
The headset's base unit connects to the phone's headphone port. The Audio connector connects to the headphone's port. The management connector connects to the Auxiliary port using a DHSG cable which can be ordered from AudioCodes.
A beep can be played to headsets when a call comes in, instead of ringing. The beep is heard even if 'Auto answer' is configured to 0. Two new configuration file parameters were added:
Enhanced quality of experience (QoE). Reports (SIP PUBLISH) were improved. Fixes to QoE-related issues were implemented.
New capability to provide a provisioning path via DHCP for VLAN configuration. VLAN can be configured using (1) Link Layer Discovery Protocol (LLDP) (2) Cisco Discovery Protocol (CDP) (3) manually. If (1) is unsuccessful, (2) is attempted, etc. The new capability provides another VLAN configuration option.
New method to refresh an existing call: SIP UPDATE. A SIP UPDATE message is now used instead of a SIP Re-Invite message in order to refresh an existing call.
New ring tones. Three new ring tones were added:
OPUS configuration management for enhanced voice quality despite poor network conditions. The feature allows the OPUS audio codec's configuration to be changed on the fly when poor conditions such as packet loss or jitter are detected in the network. The OPUS functions at a lower channel bit rate and consumes less bandwidth, delivering better voice quality in spite of the poor network conditions.
The phone plays a fast busy tone when it is automatically disconnected on the remote side. When the phone is automatically disconnected from the remote side, it not only displays a 'Disconnected' message for three seconds (default) but also plays a fast busy tone that can be configured with parameter 'voip/dialing/automatic_disconnect_delay_timer'. When the parameter 'enable_remote_disconnect_warningTone' is configured to 1 and the phone accepts an incoming call, if the remote side automatically ends the call (disconnects) the phone plays a fast busy tone.
A new timeout parameter 'Interdigit Short Timeout' has been added to the Web interface. Configuration > Voice over IP > Dialing) below the parameter 'Dialing Timeout'. The new parameter is shorter than 'Dialing Timeout'. Default: 3 seconds. It was implemented as 0S for the Dial Map. If a user wants to make an international call by dialing 00 and wants to dial the secretary/operator by dialing 0, the user can do both by adding 0S to the Dial Map. For example, if the digit map string= *xx|[2-9]11|0S|[2-9]xxxxxxxxx|1xxx[2-9]xxxxxx, it has 0S in it. When the user dials 0, 0 will match 0S and will therefore start the 'Interdigit Short Timeout' timer. After this timeout, 0 is dialed out. User can dial 00 or 0123 within the 'Interdigit Short Timeout'. After the 'Dialing Timeout', the string is dialed out.
The phone's mute key can be disabled with a new configuration file parameter. A new configuration file parameter voip/block_mute_key allows network administrators to configure enabling or disabling the mute key.
Improved Japanese language phone version
The registration expired time is now configurable. The registration expired time is that time that lapses before the refresh registration message is sent. A new 'register_before_expires_percent' parameter has been added to the configuration file. Default (in percentage): 15%. Non-percentage values are 5-85. These represent the time that must lapse before the new registration message is sent, for example, 15% means that if the expiration time is 100 seconds, the registration refresh message will be sent after 85 seconds. In previous releases, it was 33%.
This release includes the following corrections and modifications:
For Build 2.2.12.172
Pressing a Programmable Key to return from hold may cause the held call to be disconnected and the phone's VoIP application to restart.
In ACD mode, the phone screen doesn't display the current agent's status.
In some environments, when the operator attempts to leave a 3PCC call it may drop the original call.
Redialing sometimes does not function flawlessly.
Voice VLAN may not be configured correctly.
DHCP replies with a 'Destination Unreachable' when an ACK is received for an INFORM message because the 'Client ID' header is missing in the INFORM message.
Establishing a local conference is not possible when the phone is configured with OPUS and one of the remote parties doesn't support OPUS.
A local conference cannot be established using the OPUS vocoder. The conference is initiated using the G.711 vocoder instead.
The phone gets stuck on 'Acquiring IP' if it receives a DHCP Option message longer than 308 chars.
A codec negotiation issue occurs when using the G722 vocoder.
Conference involving OPUS and SRTP: A local conference results in no voice.
A call may not be established via TLS when the SIP proxy is configured with an IP address rather than configured with a domain.
For Build 2.2.12.126
Telnet access is sometimes denied after disconnecting and then reconnecting the network cable.
The phone's volume resets to the default value when rebooting.
The phone publishes an incorrect DHCP Option 12 (hostname). The DHCP Option 12 value changes from
The phone gets a data VLAN instead of a voice VLAN from some L2 switches, due to incorrect device ID parsing.
The phone's default ToS value is incorrect.
Calls from an environment with SRTP to an environment with RTP fails as the phone rejects the call with a SIP 488 'Not acceptable here' message.
A SIP ACK message is sent over UDP instead of over TLS and the call drops.
Japanese Language: The incorrect date is displayed (one month ahead).
In a Genesys environment: Consultative Transfer fails when working with a soft client (in auto-answer mode).
The Redundant Proxy cannot be set to a value of more than 32 characters.
Attended Transfer when using Speed Dial while another call already exists is not working.
The phone occasionally doesn't display a name in the Call Log if the call is unanswered (and the name is saved in the Personal Directory).
The phone gets stuck if the LDAP is set to 'Enabled' and there is no LDAP server.
The PC connected behind the phone is unable to perform EAL-TLS authentication.
This section provides the latest information on known issues and recommendations associated with this product.
For Build 2.2.12.172
The Multicast Group Paging feature doesn't function correctly in this version.
It's recommended to configure 802.1x EAP-TLS with the configuration file rather than from the phone screen.
802.1x EAP-TLS is disabled by default. For environments which require 802.1x EAP-TLS, a special version can be provided.
HTTPS provisioning is unsupported when server-side authentication (mutual authentication) is enabled.
Reporting Quality of Service events:
The SIP PUBLISH message doesn't function correctly in a conference call (conference holder or remote parties).
The SIP PUBLISH message doesn't function correctly when two concurrent calls exist.
A DNS query is sent instead of an SRV query with priorities ignored.
The Jitter Buffer increases when Music On Hold is played.
CDP Enhanced functions well but publishes incorrect values. The value is always 02 01, which means 10M Half-Duplex.
RFC 2833 functions well only with the default payload type value (101). Changing the payload type using the configuration file is not recommended.
Firmware cannot be updated manually from Chrome when accessing the Web interface over HTTPS.
When the phone is set to Off-Hook dialing (which allows dialing all digits in idle mode until pressing 'Dial'), the phone collects the digits but does not display them.
Contact Center: Fails to log in the ACD when SIP Transport Protocol is set to TCP.
XSI - 'Reject' incoming call is not functioning.
In a Genesys environment: Transfer a call from an existing 3-way conference - in order to 'drop' and leave the two remote parties in the call - may fail.
When the phone is set to static IP address and provisioning is static, the phone does not perform provisioning after a reset.
TLS does not initiate a handshake when a static IP address is configured.
[SIP] the Transfer softkey appears when starting a conference.
In a Genesys environment: There is no voice when the call is made to an off-hooked line.
For Build 2.2.12.126
The Multicast Group Paging feature doesn't function correctly in this version.
It's recommended to configure 802.1x EAP-TLS with the configuration file rather than from the phone screen.
802.1x EAP-TLS is disabled by default. For environments which require 802.1x EAP-TLS, a special version can be provided.
HTTPS provisioning is unsupported when server-side authentication (mutual authentication) is enabled.
Reporting Quality of Service events:
The SIP PUBLISH message doesn't function correctly in a conference call (conference holder or remote parties).
The SIP PUBLISH message doesn't function correctly when two concurrent calls exist.
A DNS query is sent instead of an SRV query with priorities ignored.
The Jitter Buffer increases when Music On Hold is played.
CDP Enhanced functions well but publishes incorrect values. The value is always 02 01, which means 10M Half-Duplex.
RFC 2833 functions well only with the default payload type value (101). Changing the payload type using the configuration file is not recommended.
Firmware cannot be updated manually from Chrome when accessing the Web interface over HTTPS.
When the phone is set to Off-Hook dialing (which allows dialing all digits in idle mode until pressing 'Dial'), the phone collects the digits but does not display them.
Contact Center: Fails to log in the ACD when SIP Transport Protocol is set to TCP.
XSI - 'Reject' incoming call is not functioning.
In a Genesys environment: Transfer a call from an existing 3-way conference - in order to 'drop' and leave the two remote parties in the call - may fail.
When the phone is set to static IP address and provisioning is static, the phone does not perform provisioning after a reset.
TLS does not initiate a handshake when a static IP address is configured.
[SIP] the Transfer softkey appears when starting a conference.
In a Genesys environment: There is no voice when the call is made to an off-hooked line.
The constraints are resolved; the solution will be part of the next official release.
This section documents features that are no longer supported in this software. This cumulative list is in release-number order with the most recently discontinued features at the top of the list.
There are no discontinued support items for this product.
Information in this section is included for international customers.
There are no known internationalization issues for this product.
Additional information on Genesys Telecommunications Laboratories, Inc. is available on our Customer Care website. The following documentation also contains information about this software.
Product documentation is provided on the Customer Care website, the Genesys Documentation website, and the Documentation Library DVD.