| Release Number | AIX | Linux | Solaris | Windows |
|---|---|---|---|---|
| 2.2.164.28 [01/10/21] – Hot Fix | X | |||
| 2.2.162.51 [07/11/19] – Hot Fix | X | |||
| 2.2.160.16 [01/31/19] – General | X | |||
| 2.2.120.08 [04/10/18] – General | X |
This release note applies to all 2.2.1 releases of 405 IP Phone Firmware (part of the Genesys 400 series of IP Phones).
This is a hot fix for this product. Version 2.2.16 includes firmware builds 2.2.16.428, 2.2.16.408, 2.2.16.400, and 2.2.16.376. This section describes new features that were introduced in this release.
Version 2.2.16 with the firmware build 2.2.16.428 offers the following features:
The phone supports an option to switch from ACW (After Call Work) to ‘Not Ready’ state.
A new option ‘Auto Answer incoming calls’ has been added to the phone's Settings screen (MENU > Settings). The option allows users to disable/enable the auto-answer feature.
Option 57 in the phone’s DHCP request. Option 57 is used to define the maximum-length DHCP message that a client (phone) will accept. The minimum DHCP message size supported is 576 bytes. The maximum DHCP message size supported is 1000 bytes (DHCP Option 57).
‘Auto Answer incoming calls’ can now be disabled or enabled using the phone’s programmable keys. A new ‘personal_settings/menu/callautoanswering/enabled’ configuration file parameter has been added. Default: True. The setting allows network administrators to control the Call AutoAnswer option appearance in MENU > Settings.
A random mechanism has been added for the REST_API Keepalive that’s sent to the Device Manager. The motivation is to prevent an overload on the server if all phones go up at once, for example, after electric power is interrupted.
A random mechanism has been added for configuration file provisioning (when the timer is set to check ‘every x minutes’ or ‘hourly’). The motivation is to prevent an overload on the server if all phones go up at once, for example, after electric power is interrupted.
Call Park is supported on the MetaSwitch application server.
3CX’s ‘Let's Encrypt’ CA has been added to the phone’s trusted CA store.
BLF Subscribe is now allowed when the phone is configured for a generic application server. IGS-2260
AudioCodes’ new corporate logo is now displayed on the 405 and 405HD phones. The logo is also displayed in these phones’ management interfaces.
The VLAN interface can now be changed ‘on the fly’ during regular phone operation.
OpenSSL has been upgraded to version 1.0.2p.
lighttpd has been upgraded to version 1.4.49.
Support for RTP/SRTP capability negotiation according to a subset of RFC 5939, in compliance with Genesys environment requirements. The phone sends an SDP offer with RTP and SRTP capabilities according to a new configuration parameter ‘voip/media/srtp /NegotiationMode’. Configurable parameter values:
A new option has been added to remove the ‘lifetime’ parameter from the SRTP Crypto line in SDP. According to RFC 4568, an optional ‘lifetime’ parameter such as "2^31" must be added to the a=crypto line. A new parameter ‘voip/media/srtp/use_lifetime’ has been added to allow the removal of the lifetime in all phone crypto lines in SDP. Configurable parameter values:
DIGICert has been added as ROOT-CA.
The preloaded well-known RootCA section for BroadCloud server has been updated.
A notification has been added to notify users and network administrators that the phone has entered Recovery mode. Previously, this was supported only on the 405HD phone; now, it’s supported on all phone models.
‘USB Headset Type’ has been added to the REST API status keep alive message sent to the AudioCodes Device Manager management interface. The management interface now features new displays in the Devices Status page:
The configuration of the provisioning time parameter has been updated to allow checking for updated files every five and 15 minutes. Previously, the minimum time was one hour.
Network administrators can control the DTMF tones level through a new configuration file parameter ‘voip/audio/gain/dtmf_rtp_event_signal_level’ that has been added.
This release includes the following corrections and modifications:
For Build 2.2.16 with Firmware Build 2.2.16.428
Support required for ‘Not Ready’ state from ACW.
In some environments, the phone identifies a VLAN priority and therefore changes to Native VLAN.
The phone fails to register to the Secondary Proxy if using the same IP address as the Primary Proxy but with a different port.
For an incoming call the phone doesn't beep, neither on the speaker nor on the headset.
The phone doesn't use SRV records to complete registration to the SIP proxy.
The phone screen doesn't show numbers of 10 digits or more.
The phone doesn't send OPTIONS/Register to the Primary Poxy after getting a 404 from the Redundant Proxy.
The Call Forward feature doesn't function correctly if the phone has multiple lines.
QoE does not function if the SIP protocol is different to the ‘QoE Publish’ protocol.
The phone doesn't unmute automatically when it gets a second ongoing call.
The phone does not display the name of a dialed contact if the caller name info arrives as part of the SIP OK ‘To’ header.
The phone can't return to the first call after the second call is disconnected by the remote side.
A loud, continuous beep is played after a call is held for 40 seconds.
The INVITE is followed by a BYE message when call disconnect is in hold state.
This is a hot fix for this product. Version 2.2.16 includes firmware builds 2.2.16.251, 2.2.16.142, 2.2.16.92, 2.2.14.26, and 2.2.12.224. This section describes new features that were introduced in this release.
Version 2.2.16 with the firmware build 2.2.16.251 offers the following features:
Support for RTP/SRTP capability negotiation according to a subset of RFC 5939, in compliance with Genesys / Avaya environment requirements. The phone sends an SDP offer with RTP and SRTP capabilities according to a new configuration parameter ‘voip/media/srtp /NegotiationMode’. Configurable parameter values:
The phone features new capability to send UNENCRYPTED_SRTCP packets to comply with Avaya environment requirements.
A new option has been added to remove the ‘lifetime’ parameter from the SRTP Crypto line in SDP. According to RFC 4568, an optional ‘lifetime’ parameter such as "2^31" must be added to the a=crypto line. A new parameter ‘voip/media/srtp/use_lifetime’ has been added to allow the removal of the lifetime in all phone crypto lines in SDP. Configurable parameter values:
The Ribbon Communications (formerly GENBAND) softswitch solution Kandy Business Solutions (KBS) has been added as a new application server.
DIGICert has been added as ROOT-CA.
The preloaded well-known RootCA section for BroadCloud server has been updated.
A notification has been added to notify users and network administrators that the phone has entered Recovery mode. Previously, this was supported only on the 405HD phone; now, it’s supported on all phone models.
‘USB Headset Type’ has been added to the REST API status keep alive message sent to the AudioCodes Device Manager management interface. The management interface now features new displays in the Devices Status page:
The configuration of the provisioning time parameter has been updated to allow checking for updated files every five and 15 minutes. Previously, the minimum time was one hour.
Network administrators can control the DTMF tones level through a new configuration file parameter ‘voip/audio/gain/dtmf_rtp_event_signal_level’ that has been added.
This release includes the following corrections and modifications:
For Build 2.2.16.251
[Genesys environment] A ‘HOLD’ action that is performed from the phone takes too long. Conversely, there’s no delay when a ‘HOLD’ action is performed via a soft client.
On some occasions, the Link Layer Discovery Protocol daemon (lldpd) causes the phone to respond sluggishly due to a memory leak.
The phone does not feature capability to disable the ‘Resume’ softkey using a configuration file parameter.
SIP user ID is shown in the DND (Do Not Disturb), CFD (Call Forward) and Voice Mail screens instead of Display Name.
A SIP UPDATE message is sent even in instances where it should not be used (referred to in SIP as not allowed).
One-way audio may occur when using phones that do not support OPUS but are mistakenly configured with an OPUS configuration.
The phone may be displayed as disconnected in the Device Manager even though it’s alive and registered.
The phone encounters issues when registering to a SIP Proxy when it’s behind SAS during survivability mode.
SSL2 connections are open even if disabled in the configuration file.
The phone is not processing the 200 OK to Register method due to a port scanner tool.
[420HD] In some environments, an outgoing call from the phone shows name only and the phone number does not appear.
[Genband proxy] Call transfer fails.
Birthday attacks against TLS ciphers with 64 bit block size vulnerability (Sweet32).
Vulnerability to Web Cross-site Scripting (XSS) attacks on mainform.cgi.
The Message Waiting Indicator (MWI) port is ignored in the SUBSCRIBE message.
A Core Dump is not generated correctly when the phone VoIP application is reloaded.
The phone occasionally incorrectly displays a ‘Duplicate IP address’ message.
Using the Web interface to trigger ‘Restore Defaults’ periodically crashes the phone.
[Genband environment] The phone unsuccessfully drops a participant from a conference.
[4xxHD] ‘Forward’ is incorrectly translated into Portuguese.
[Brazilian Portuguese] The BXfer softkey is not translated correctly.
The phone rejects REST API messages from the Device Manager (previously called the IP Phone Manager) if the given username parameter differs in upper or lower case from the phone’s local parameter.
VLAN priority (802.1p) is absent when VLAN settings are discovered via LLDP.
[420HD/405/405HD] The phone crashes when configuring a speed dial softkey from the menu.
[BSFT] The phone’s screen displays name and number instead of user ID.
[BSFT] A damaged PUBLISH occurs if there’s a Local Conference on OPUS.
The phone’s screen does not display an incoming caller’s number in the second displayed line.
The SIP message 200OK reply over TLS changes to UDP.
There is no support for HTTPS Security headers.
One-way voice is heard on the phone after the user retrieves the first call from a Consultative Transfer call.
Configuring a beep to play towards a headset upon auto-answer does not function.
French translation issues.
The phone displays ‘Transferred X to Y’ instead of ‘Transferring X to Y’ after a ‘202 Accepted’ arrives.
In a Semi-Attended Transfer scenario, the phone is missing a CWRR (Call Waiting Reminder Ringtone) indicating that the call was put on hold.
The phone performs restarts after making changes in the Web interface to the Programmable Keys.
There are no restrictions for this release. Version 2.2.16 includes firmware builds 2.2.16.142, 2.2.16.92, 2.2.14.26, and 2.2.12.224. This section describes new features that were introduced in this release.
Version 2.2.16 with the firmware build 2.2.16.142 offers the following features:
Audible indication played after a call has been on hold for a long time. After a call has been on hold for a long (configurable) time, a reminder tone is played every 10 seconds until the call is taken off hold.
Two new configuration file parameters have been added to support the feature:
New configuration file parameter enables the P-Asserted Identity header to be added to �18x� and �200� responses. Parameter ‘voip/signalling/sip/PAI_On_Replay/enable’ was added to enable the header to be added to these responses, configurable as follows:
Version 2.2.16 with the firmware build 2.2.16.92 offers the following features:
USB headsets are now officially supported by Genesys' generic SIP IP phones.
New voice dialing capabilities from the phone to any user in the corporate directory [Beta Version]. Genesys' 400HD Series of IP Phones is now directly integrated with Genesys' VocaNOM service to allow voice dialing to any other user in same corporate directory. To enable the service, the user must add a VocaNOM key, and IT must configure the VocaNOM IP address service on the phone. The caller hears a voice prompt requesting the callee's first and last name. When the service identifies the callee, the phone dials the callee's number just as it does in a regular call. Later, the user can press the REDIAL hard key on the phone and view the call logged in the phone's 'Dialed Calls' just like with any other call. The service is currently available in English and German only.
SRTP negotiation Support for Secure Real-Time Transport Protocol has been changed to allow a new SRTP Negotiation option. The configuration file parameter 'voip/media/srtp/mode' replaced the legacy configuration file parameter 'voip/media/srtp/enabled' to support new encryption levels. Three levels of encryption are now supported:
Phone hard keys and softkeys can be disabled using the configuration file [released as part of a post Version 2.2.12 release]. Hard keys that can be disabled include speaker, headset, voicemail, REDIAL, CONTACTS, MENU, TRANSFER, HOLD, VOL and mute. The feature is motivated by the requirement on the part of some enterprises to control the setting remotely to comply with company policy.
New configuration file parameters network administrators can use to disable phone hard keys and softkeys include:
/personal_settings/soft_keys/display_idle_screen_keys_when_dialing/enabled
/personal_settings/key/speaker_device/enabled
/personal_settings/key/headset_device/enabled
/personal_settings/key/voice_mail/enabled
/personal_settings/key/redial/enabled
/personal_settings/key/contacts/enabled
/personal_settings/key/menu/enabled
/personal_settings/key/hold/enabled
/personal_settings/key/volume/enabled
/personal_settings/key/mute/enabled
/personal_settings/menu/call_log/enabled
/personal_settings/menu/directory/enabled
/personal_settings/menu/keys_configuration/enabled
/personal_settings/menu/keys_configuration/speed_dial_keys/enabled
/personal_settings/menu/keys_configuration/soft_keys/enabled
/personal_settings/menu/keys_configuration/navigation_keys/enabled
/personal_settings/menu/settings/enabled
/personal_settings/menu/language/enabled
/personal_settings/menu/ring_tone/enabled
/personal_settings/menu/callwaiting/enabled
/personal_settings/menu/date_and_time/enabled
/personal_settings/menu/lcd_contrast/enabled
/personal_settings/menu/backlight_timeout/enabled
/personal_settings/menu/answer_device/enabled
/personal_settings/menu/restart/enabled
/personal_settings/menu/status/enabled
/personal_settings/menu/administration/enabled
/personal_settings/menu/languages/english/enabled
/personal_settings/menu/languages/spanish/enabled
/personal_settings/menu/languages/russian/enabled
/personal_settings/menu/languages/portuguese/enabled
/personal_settings/menu/languages/portuguesebrazilian/enabled
/personal_settings/menu/languages/german/enabled
/personal_settings/menu/languages/ukrainian/enabled
/personal_settings/menu/languages/french/enabled
/personal_settings/menu/languages/frenchcanadian/enabled
/personal_settings/menu/languages/italian/enabled
/personal_settings/menu/languages/hebrew/enabled
/personal_settings/menu/languages/polish/enabled
/personal_settings/menu/languages/korean/enabled
/personal_settings/menu/languages/finnish/enabled
/personal_settings/menu/languages/chinese/enabled
/personal_settings/menu/languages/chinesetraditional/enabled
/personal_settings/menu/languages/turkish/enabled
/personal_settings/menu/languages/japanese/enabled
/personal_settings/menu/languages/slovak/enabled
/personal_settings/menu/languages/czech/enabled
/personal_settings/speed_dial_programming/enabled
/personal_settings/new_call_screen/call_log_soft_key/enabled
/personal_settings/new_call_screen/directory_soft_key/enabled
personal_settings/soft_keys/incoming_call/sk_reject/enable=0
The phone screen's backlight timeout can be changed using the configuration file [released as part of a post Version 2.2.12 release]. A new configuration file parameter system/lcd/backlight/timeout has been added to allow changing the phone's backlight timeout using the configuration file. Range: 0-6. Previously, the screen's backlight timeout could only be changed on the phone.
Second dial tone to receive an external line has been enhanced [released as part of a post Version 2.2.12 release]. Until this version, when using a second dial tone, for example, when pressing 9 to get an external dial tone, the second dial tone was identical to the main dial tone, with a short break. In the current version, a different dial tone (not configurable) is played as the second dial tone.
Applicable configuration file parameters are:
voip/dialing/secondary_dial_tone/enabled=1
voip/dialing/secondary_dial_tone/key_sequence=<key sequence>
<Key sequence>can be one of the digits 1-9
Version 2.2.14 with the firmware build 2.2.14.26 offers the following features:
Paging a group of phones. Live announcements can be made (paged) from a phone to a group of phones, to notify a team (for example) that a meeting is about to commence. The paged announcement is multicast via a designated group IP address, in real time, on all idle phones in the group, without requiring listeners to pick up their receivers. The name of the group is displayed on phone screens when the paging call comes in. Before the user can configure a functional key for paging, the feature must be enabled in the Web interface by the network administrator. When it's disabled (default) and the user is in a regular call when a paging call comes in, they're prompted in the phone screen to accept or reject the paging call. If accepted, the regular call is put on hold and the paging call is heard.
The barge-in feature is only relevant if the paged user is in a regular call. If they're not in a regular call, the paged call is heard irrespective of whether the barge-in feature is enabled or disabled.
Version 2.2.12 with the firmware build 2.2.12.224 offers the following features:
Phone hard keys and softkeys can be disabled using the configuration file. Hard keys that can be disabled include speaker, headset, voicemail, REDIAL, CONTACTS, MENU, TRANSFER, HOLD, VOL and mute. The feature is motivated by the requirement on the part of some enterprises to control the setting remotely to comply with company policy.
New configuration file parameters network administrators can use to disable phone hard keys and softkeys include:
/personal_settings/soft_keys/display_idle_screen_keys_when_dialing/enabledThe phone screen's backlight timeout can be changed using the configuration file. A new configuration file parameter system/lcd/backlight/timeout has been added to allow changing the phone's backlight timeout using the configuration file. Range: 0-6. Previously, the screen's backlight timeout could only be changed on the phone.
Second dial tone to receive an external line has been enhanced . Until this version, when using a second dial tone, for example, when pressing 9 to get an external dial tone, the second dial tone was identical to the main dial tone, with a short break. In the current version, a different dial tone (not configurable) is played as the second dial tone.
Applicable configuration file parameters are:
voip/dialing/secondary_dial_tone/enabled=1
voip/dialing/secondary_dial_tone/key_sequence=<key sequence>
<Key sequence>can be one of the digits 1-9
This release includes the following corrections and modifications:
For Build 2.2.16
The mute icon indication goes missing from the phone’s screen if any key on the keypad is pressed after a call is muted.
The phone sends a SIP Invite message without a port number in the �Request line� and �contact� fields.
After toggling between multiple existing calls, a held call cannot be terminated by putting the handset back in the cradle (on-hooking).
The P-Asserted Identity header is not added to �18x� and �200� responses.
The SIP RE-INVITE message does not include a crypto line; this results in a �488 not Acceptable here� reply from the server.
The phone fails to complete a Dial Plan rule when the Dial Plan starts with '0'.
[Voice Dialing] The voice dialing number rather than the target user’s name is saved in the Call Log in the case of call and regret.
[Voice Dialing] The user does not receive a warning notification in the phone's screen if a call to the voice dialing number fails due to an incorrect IP configuration.
N-Way Conference (Remote Conference) in a BroadSoft environment may fail. In such a case, after pressing the Conf softkey, the next participant cannot be dialed, the phone gets stuck (the conference line cannot be disengaged from) and occasionally, the phone then reboots.
Some Web interface pages are accessible without user authentication.
The phone accepts HTTP and HTTPS messages sent by a REST API interface even though the connection is configured to use HTTPS only.
When the configuration file parameter ‘voip/voice_quality/mode’ is set to a value other than Disabled , Music on Hold (MoH) does not function – even though it should.
Using the configuration file parameter to configure removal of the 'Missed Calls' functionality does not work.
Disabling 'handset_mode' (by setting the ‘voip/handset_mode/enabled’ configuration file parameter to 0), still allows a call to be answered by picking up the handset.
Incoming calls result in a delay of up to ~1 second compared to the previous Version 2.2.12 release.
The phone still shows the ‘Missed Calls’ icon even though the keys that allow viewing the calls history list are blocked.
When the phone uses SIP over TLS and is configured to use a redundant Proxy, the SIP Invite to the redundant Proxy is sent without TLS.
For Build 2.2.16.92
When the remote party terminates a call, the phone should play a fast busy tone for a few seconds (preconfigured to 3 seconds) before reverting to idle mode. Instead, the phone disconnects and immediately reverts to idle mode.
A local conference cannot be established when one party offers the OPUS vocoder and the other party does not support OPUS.
The phone's audio device doesn't switch back to the handset after turning off the speaker.
In some configurations, QoE SIP PUBLISH is sent to the Secondary Proxy instead of to the SEM server's address.
Enabling EHS (Electronic Hook Switch) blocks upgrading to the Skype for Business version.
The phones do not support the �Privacy:ID� header for anonymous calls.
The parameter 'Block Caller ID on Outgoing Calls' creates a non-RFC compliant P-Asserted-Identity (PAI) header. The PAI header does not include ‘sip:'.
Chinese characters need improvement.
For Build 2.2.14
A very short tone is heard when answering an incoming call.
'Gateway Name' does not function correctly for an outgoing INVITE message.
Volume controls still affect the speaker even though it was disabled using parameter voip/hands_free_mode/enabled=0.
For Build 2.2.12
Chinese language improvements
Portuguese language improvements
Portuguese language requires a special license key to be used
[3-Way Conference] Transfer from an existing local conference initiated by the phone (the originator leaves the conference leaving the two remote parties continuing talking) cannot be initiated by the phone. It can only be initiated from the Genesys soft client.
[3-Way Conference] The phone's VoIP application is restarted during a local conference when both remote parties close the call before the originator closes it.
[Managed OPUS] Managed OPUS does not function correctly when the phone's configuration file parameter 'voip/voice_quality/mode' is set to Enable_RTCP.
A crackling noise can be heard during calls when the phone is set to automatic answer and when the phone is set to play a beep to the user’s headset when a call comes in.
The phone is unable to dial to numbers with spaces taken from the LDAP server.
In some scenarios, when the phone gets SIP message 403 as an answer to SIP Registration to Redundant Proxy, the phone tries to switch back to the Primary Proxy.
System/Password can't be generated using the phone’s proprietary Encryption Tool.
[QoE] QoE SIP PUBLISH messages may be sent by the phone to the Proxy and not to the QoE server.
Issue encountered when using the 'Redial' functionality (related to VI 107281). In some environments, the proxy replies with a �Remote-Party-ID� header which previously wasn’t supported on the phone and as a result, prevents the redial call to complete successfully.
When using SIP over TLS, an ACK SIP Message is sent over UDP instead of TLS. This can lead to the call being dropped.
The phone re-registers after an SBC High-Availability (HA) switchover. In some cases (mostly during calls), the switchover doesn’t complete successfully; the phone uses the incorrect destination proxy address.
In some environments, the phone cannot get VLAN via LLDP.
Volume controls are enabled even when speaker is disabled via the configuration file parameter voip/hands_free_mode/enabled=0
When using automatic provisioning to change the timezone, the phone does not update the time.
Gateway Name does not function correctly. The phone's configuration file parameter 'voip/signalling/sip/sip_outbound_proxy/addr' has been fixed to support Name and IP address to be used in the �From� header.
There are no restrictions for this release. 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 (Fran�ais 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).
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).
[405 only] DTMF is not sent in Early Media state.
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 2.2.16 with Firmware Build 22.164.28
In some environments, the phone takes the IP address from the wrong VLAN.
In some environments, the phone changes IP address after restarting.
The BLF ‘Call Pickup’ feature does not function.
Time Zone: GMT -09:00 is missing.
In the Device Manager, the phone is displayed with the status of ‘Started’ in the case of incorrect credentials.
In some environments, a call using the OPUS vocoder fails due to a missing semi-colon at the end of the OPUS a=fmtp: line in SDP.
Calls are sporadically not saved in the Call Log.
When using HTTPs and in the case of a very long provisioning URL, the phone fails to send HTTP POST to the Device Manager.
Media streaming / SRTP is incorrectly displayed in the Web interface page.
The first generated SRTP packets are encrypted with a ‘zero’ key.
The Supervisor Listen feature isn’t functioning.
On rare occasions, the phone stops sending a keep-alive message. After a prolonged period, this causes the Device Manager to display the status as disconnected.
The user is unable to change the ringtone if the OPUS codec is configured and used.
Incorrect ‘Chinese traditional’ is displayed during a call transfer scenario.
When the configuration file parameter ‘system/dnd/show_softkey’ is disabled, DnD remains displayed in the phone’s screen.
The phone doesn't set VLAN priority according to the policy priority header.
The phone doesn't answer incoming calls immediately if the value configured for the configuration file parameter ‘voip/advanced_auto_answer/timeout’ is 0.
The audio device changes unexpectedly from headset to speaker in the middle of a call.
If the phone is left off-hook after a call is disconnected and if another call then comes in, the ringer volume differs from the volume configured.
The beep reminder played when a call is put on hold for a prolonged period is enabled by default.
Even though the phone receives a CANCEL almost immediately after the initial INVITE, the headset continues to ring.
For Build 2.2.16
[BroadSoft environment] When 'SIP Proxy' and 'Default Gateway' are configured with an IP address instead of a Hostname, the Blind Transfer feature does not function correctly.
[BroadSoft environment] Feature Key Synchronization - the Call Forward and Do Not Disturb functionalities can be configured from the BroadSoft Server Web Interface or from the phone's screen but they cannot be configured via the phone's Web interface.
[BroadSoft environment] Shared Call Appearance - the user cannot toggle between two incoming calls using Programmable Key 1 and Programmable Key 2. Toggling can only be performed using the navigation key.
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.
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 phones are set to static IP address and provisioning is static, the phones do not perform provisioning after a reset.
[SCA] After several SCA scenarios with barge-in, the sidecar records display empty even if there are active calls in the phone's main screen.
[SCA] After a barge-in to a second call, the Index appearance LED is incorrect.
[EHS] When pressing mute on the headset, the mute LED on the phone doesn't light up.
BLF doesn't work if the SIP Proxy port that is used is not standard.
TLS does not initiate a handshake when a static IP address is configured.
[SIP] The Transfer softkey appears when starting a conference.
[QoE] A SIP PUBLISH message isn't created when there is a SIP BYE message.
There is no voice when a call is made to an off-hooked line.
Multiple lines: The busy screen is corrupted.
A call cannot be established between two phones with different settings of the 'voice_quality/mode' parameter. This can happen when one phone is set to use ‘Managed OPUS’ while the other phone isn’t.
[Jabra Evolve USB headset] Using the volume keys on this headset is not recommended as they send an unexpected unmute command. It’s advisable to use the phone’s mute key and volume up/down keys instead.
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.
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.
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.
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.
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. See the following documentation site for information about this software.
Product documentation is provided on the Customer Care website, the Genesys Documentation website, and the Documentation Library DVD.