Firmware Release Note

Contents

Introduction

This release note applies to all 2.2.1 releases of 405HD IP Phone Firmware (part of the Genesys 400 series of IP Phones).

• Links in the Contents section enable you to access information regarding a specific release. Releases are listed by version number order. Except when otherwise noted in the information for a specific release, each release includes all of the features and corrections that were introduced in earlier versions.
• The Known Issues and Recommendations section is a cumulative list. It includes information on when individual items were found and, if applicable, corrected.
• The Discontinued Support section is cumulative. It lists functionality that is no longer supported and the release number of the version in which support was discontinued.

Supported Operating Systems

New in This Release

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:

• Basic (default) [RTP/SRTP negotiation according to the document IMTC Best Practices for SIP Security].
• RFC 5939 [RTP/SRTP capability negotiation "a=tcap", "a=acap" and �a=pcfg� attributes as described in RFC 5939]

• 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:

• [0] = the lifetime is removed
• [1] = the lifetime is retained (default)

• DIGICert has been added as ROOT-CA.

• The preloaded well-known RootCA section for BroadCloud server has been updated.

• ‘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:

• Column ‘USB Headset Type’ displays a headset connected to a phone’s USB port
• Column ‘IPP Model’ displays the USB icon

• 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.

• 0 db (Minimum)
• 31 db (Maximum)

Corrections and Modifications

This release includes the following corrections and modifications:

For Build 2.2.16 with Firmware Build 2.2.16.428

• The display name doesn’t appear on the 405HD phone’s screen when the LDAP-based Corporate Directory is configured for displaying name manipulation.

• 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.

• 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.

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Supported Operating Systems

New in This Release

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:

• Basic (default) [RTP/SRTP negotiation according to the document IMTC Best Practices for SIP Security. This mode is supported by Broadsoft, Microsoft and many other vendors]
• RFC 5939 [RTP/SRTP capability negotiation "a=tcap", "a=acap" and “a=pcfg” attributes as described in RFC 5939]

• 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:

• [0] = the lifetime is removed
• [1] = the lifetime is retained (default)

• The Ribbon Communications (formerly GENBAND) softswitch solution Kandy Business Solutions (KBS) has been added as a new application server.

• Users can Park Calls to other users’ DNs (Directory Numbers) in a Ribbon Communications environment. For more information, see Ribbon Communications’ Kandy Business Solutions Feature Description Guide.
• A new Programmable Key type allowing users to Park / Retrieve calls, including BLF (presence), has been added to the 440HD phone in a Ribbon Communications environment.
• Caller ID has been added to the 440HD phone’s sidecar display for parking calls in a in a Ribbon Communications environment.

• 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:

• Column 'USB Headset Type' displays a headset connected to a phone’s USB port
• Column 'IPP Model' displays the USB icon

• 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.

• 0 db (Minimum)
• 31 db (Maximum)

Corrections and Modifications

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.

• [420HDG only] Packet Loss occurs during calls when the phone is deployed behind certain routers and is receiving voice.

• 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.

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Supported Operating Systems

New in This Release

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:

• voip/lhcwrr_enabled
• [1] (Default) Enabled. After the length of time configured for configuration file parameter voip/lhcwrr_wait_time lapses (see the parameter below), a reminder tone (beep) is played every 10 seconds until the call is taken off hold.
• [0] Disabled. No reminder tone (beep) is played, regardless of how long the call is on hold.
• voip/lhcwrr_wait_time (Default: 120 seconds) Defines the length of time that must lapse before a reminder tone (beep) is played. The tone will then be played every 10 seconds until the call is taken off hold.

• 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:

• [0] (Default) P-Asserted Identity header is not added to �18x� and �200� responses
• [1] P-Asserted Identity header is added to �18x� and �200� responses

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:

• DoNotSupportEncryption [Default] [SRTP is disabled]
• SupportEncryption [Negotiation]
• RequireEncryption [SRTP is enabled; both sides must support encryption]
Note regarding backward compatibility: Since the legacy parameter ‘'voip/media/srtp/enabled’ has become obsolete in this release, the network administrator must configure phones set to 'voip/media/srtp/enabled=1' in previous releases, to 'voip/media/srtp/mode=RequireEncryption’ in this release, to keep forcing SRTP.

• 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.

• Example 1: To disable the phone's REDIAL hard key, the configuration file parameter personal_settings/key/redial/enabled can be set to 0.
• Example 2: To disable the option to restart the phone, the configuration file parameter personal_settings/menu/restart/enabled can be set to 0.

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.

• 0 = Always On
• 1 = 10 seconds (default)
• 2 = 20 seconds
• 3 = 30 seconds
• 4 = 40 seconds
• 5 = 50 seconds
• 6 = 60 seconds

• 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.

• Barge-in
• If the network administrator has enabled the barge-in feature and the user is in a regular call when a paged call comes in, the paged call is heard and the regular call is placed on hold. The user can opt to revert to the regular call and ignore the paged call.


• If the network administrator has disabled the barge-in feature and the user is in a regular call when the paged call comes in, the user is prompted to accept or reject the paged call. If accepted, the regular call is placed on hold while the paged call is heard. The user can opt to return to the regular call. If rejected, the paged call is not heard and the regular call continues.

  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.

  • Example 1: To disable the phone's REDIAL hard key, the configuration file parameter personal_settings/key/redial/enabled can be set to 0.
  
  
  • Example 2: To disable the option to restart the phone, the configuration file parameter personal_settings/menu/restart/enabled can be set to 0.

  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. 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.

  • 0 = Always On
  • 1 = 10 seconds (default)
  • 2 = 20 seconds
  • 3 = 30 seconds
  • 4 = 40 seconds
  • 5 = 50 seconds
  • 6 = 60 seconds

• 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

Corrections and Modifications

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 phone displays an incorrect number during a remote conference.

• 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 a certain switch is deployed in a customer's network to enable MAC authentication, the switch checks the phone’s presence by ARP request every 120 seconds. The ARP reply is problematic because its target MAC address is the phone when it should be the MAC address of the switch.

• 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 phone does not accept RFC 2833 format's payload type number 100. The phone fails to change the payload type.

• 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.

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Known Issues and Recommendations

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.

• QoE does not function if the SIP protocol is different to the ‘QoE Publish’ protocol.

• 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.

• A delay of approximately 0.5 of a second occurs on the headset when beep and auto-answer are enabled.

• 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.

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Discontinued Support

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.

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Internationalization

Information in this section is included for international customers.

There are no known internationalization issues for this product.

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Additional Information

Additional information on Genesys Telecommunications Laboratories, Inc. is available on our Customer Care website. See the following documentation site for information about this software.

Genesys 400 Series IP Phones

Product documentation is provided on the Customer Care website, the Genesys Documentation website, and the Documentation Library DVD.

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