Cloud Voice 84.23.0.24-beta1

This page summarizes what changed in the first beta build of firmware 84.23.0.24 (release version V24.1-Beta), released April 23, 2026. To install it, open your Cloud Voice admin portal and check for a firmware update. Updates to the Cloud Voice App are tracked in the app’s own release notes.

Caution

This is a pre-release beta build. Run it only where you are comfortable testing unfinished firmware, and avoid production systems until the general release.

New features

Onsite Proxy

Onsite Proxy lets you connect remote IP phones to your phone system without setting up port forwarding. You install the Onsite Proxy software on the same subnet as the remote phones, and it opens a secure channel back to the PBX (Private Branch Exchange, your phone system). From that point on, the phones can be auto-provisioned and reached over the internet as though they shared the local network with the PBX.

This release also adds the following capabilities around Onsite Proxy:

• Remote provisioning. Remote IP phones can now be auto-provisioned through Onsite Proxy.
• Error alerting. A new Onsite Proxy Error event notification (System > Event Notification > Event Type > System) fires when the link between the PBX and Onsite Proxy drops, or when Onsite Proxy’s resource usage crosses the configured thresholds.
• API support. New API (Application Programming Interface, for driving the PBX from your own tools) interfaces let you list Onsite Proxy instances, search for a specific instance, read the details of one or more instances, view or reset an instance’s secret key, and add, edit, or delete an instance.

Improvements and bug fixes

Extension

• The maximum extension-number length is now 11 digits, up from 8.

• You can copy the IP address of an extension’s online device with a single click.

  

• You can upload an avatar when you create or edit extensions, one at a time or in bulk, under Extension and Trunk > Extension > User > User Information.

  Note

  Avatars must be .jpg or .png and no larger than 1 MB.

  

• Call forwarding (Extension and Trunk > Extension > Presence > Call Forwarding) has two behavior changes:

  • A new Busy destination is available. When a forwarding condition is met and this destination is set, the PBX rejects the call and replies with the SIP (Session Initiation Protocol, the signaling that sets up and tears down calls) response 486 Busy Here.

    

  • For the When Busy condition with a Hang Up destination, the PBX now replies with SIP 480 Temporarily Unavailable instead of the previous 486 Busy Here.

• You can set a ringback tone per extension under Extension and Trunk > Extension > Features > Prompt. The tone plays to the caller while the extension rings, before the call is answered.

  

• An extension can now use more than one transport protocol (UDP, TCP, or TLS: the different ways a phone’s SIP messages travel to and from the PBX) at a time (Extension and Trunk > Extension > Advanced > VoIP Settings > Transport).

  

  Caution

  When you upgrade to this build, the system adjusts some settings automatically:

  • Ports. The SIP TCP port is enabled (PBX Settings > SIP Settings > General > Basic). If any existing extension uses TLS, the TLS port is enabled as well (PBX Settings > SIP Settings > TLS).
  • Extension protocols. Existing extensions gain TCP; new extensions default to both TCP and UDP.

  Note

  When several protocols are enabled, the PBX chooses one by priority:

  • Cloud Voice App and auto-provisioned IP phones: TLS, then TCP, then UDP.
  • Manually registered IP phones: the protocol the phone specifies, or the default priority (TLS, then TCP, then UDP) if none is specified.

• Fixed: with Accept calls from Ring Group disabled, an extension could not receive a ring group call that had been forwarded into a queue the extension belonged to.

• Fixed: after an extension’s mobile number was cleared to null through the API, member-selection lists for queues, ring groups, extension groups, outbound routes, and similar features showed no extensions.

Cloud Voice App Server

Extension users can now sign in to the Cloud Voice App with their extension number. Turn this on with the Client Login Mode setting under Extension and Trunk > Extension > Cloud Voice App Server > Basic.

Client Permissions

A new Upload Avatar permission (Extension and Trunk > Client Permission > Preference Settings > Configuration Item) controls whether the selected extension users can change their extension avatar from the Cloud Voice App.

Note

On mobile, users need Cloud Voice App 5.25.15 or later (iOS) or 5.25.14 or later (Android).

Trunk

• SIP trunks gain a Support SIP REFER setting (Extension and Trunk > Trunk > Advanced > VoIP Settings). When enabled, you choose which transfer destinations are allowed, Internal Number, External Number, or both, and the PBX honors REFER requests on that trunk by transferring the call accordingly.

  Note

  • For external numbers, the PBX prepends the configured prefix to the target number in the REFER request.
  • Only blind transfer is supported.

  Caution

  Allowing External Number as a REFER destination lets the far end of the trunk redirect a live call to any outside number, which can expose you to toll fraud. Turn on external transfers only for trunks you trust, and keep the allowed destination types as narrow as the setup requires.

  

• Fixed: on outbound calls over a SIP trunk with SRTP (Secure Real-time Transport Protocol, which encrypts the call audio) enabled, audio dropped to one-way after 15 minutes and the caller could no longer hear the callee.

Auto Provisioning

• A new Enable Yealink Phone Configuration File Encryption option (Auto Provisioning > Phones > Options) controls whether the PBX encrypts configuration files for Yealink phones during provisioning.

  

• You can copy an IP phone’s IP address from the phone list with one click.

  

• Auto-provisioned Mitel phones now support a Distinctive Ringtone setting (open the phone’s settings gear under Auto Provisioning > Phones), where you map alert-info values to ringtones.

  

• The PBX now handles the 48 kHz sampling rate of the Opus audio codec from IP phones, processing those RTP (Real-time Transport Protocol, the media stream that carries call audio) streams correctly for clearer audio.

  Note

  This applies both to auto-provisioned phones and to phones on manually registered extensions.

• Fixed: provisioning an Avaya phone with the language set to J169/J179 French failed to save.

Inbound Route

Fixed: a very long inbound route name caused calls through that route to fail.

Call Queue

• Queue manager permissions gain two settings for queue call logs (Call Features > Queue > Queue Panel Permissions > Manager):

  

  Setting	Description
  Queue Incoming Call Logs	Lets queue managers view the logs of all answered calls in the queue. Only logs created in version 84.21.0.117 or later are available; older logs are not.
  Delete Queue Incoming Call Logs	Lets queue managers delete the logs of answered calls in the queue. Deleted records disappear for all agents and managers but remain in the CDR (Call Detail Record, the master log the system keeps for every call).

• Queue agent permissions now offer finer control over which answered-call logs an agent can see (Call Features > Queue > Queue Panel Permissions > Agent):

  

  Option	Description
  Personal Only	Authorized agents see only the queue calls they answered themselves in the Cloud Voice App.
  All Agents	Authorized agents see the queue calls answered by any agent in the queue. Only logs created in version 84.21.0.117 or later are available.

• Fixed: the time shown on the Wallboard was one hour ahead of the PBX system time.

Voicemail

• Users can forward their voicemails to one or more extensions from the Cloud Voice App.

  Note

  This requires Cloud Voice App 1.22.4 or later on desktop, and 5.25.15 or later (iOS) or 5.25.14 or later (Android) on mobile.

• When a user listens to a forwarded voicemail through a feature code, the system now announces who forwarded it at the start of playback.

Call Recording

• The download size limit for recording files is now 2 GB, up from 600 MB.
• Fixed: a filename format that a third-party platform could not parse left multi-party conference recordings incompletely synced to that platform.

Call Disposition

The maximum number of call dispositions is now 200, up from 20.

Hot Desking

Fixed: users occasionally failed to log in or out when hot desking on Yealink IP phones.

Messaging

• SMS and WhatsApp channels gain an Allow the Creation of Duplicate Active Sessions setting (Messaging > Message Channel > SMS Channel/WhatsApp Channel > Messaging Settings).

  

  When it is enabled and a user starts a session in the Cloud Voice App that already exists for the same sender and receiver, the app prompts the user. If the user continues, they take over the existing session along with its full chat history.

  Note

  On mobile, this requires Cloud Voice App 5.25.15 or later (iOS) or 5.25.14 or later (Android).

  

• External chat log management improved (Reports and Recordings > External Chat Logs):

  • You can filter sessions by Current Session Handler.

    

  • A quick filter button surfaces chat sessions belonging to deleted extensions and queues.

    

  • You can transfer one or more chat sessions to a chosen destination.

    

PBX Settings

• Distinctive Caller ID Name gains a Display the Diversion SIP Header for Extension Forwarding option (PBX Settings > Preferences > Caller ID Name).

  

  When it is enabled and an incoming call is forwarded directly to a destination type listed in Extension Forwarding with Diversion SIP Header (PBX Settings > SIP Settings > Advanced > Other Options), the Caller ID shows the name and number of the extension the call was forwarded from.

  Note

  This appears on both IP phones and the Cloud Voice App. For it to show correctly in the app, users need Cloud Voice App 1.22.4 or later on desktop, and 5.25.15 or later (iOS) or 5.25.14 or later (Android) on mobile.

• Queue is now a valid destination type for Extension Forwarding with Diversion SIP Header (PBX Settings > SIP Settings > Advanced > Other Options). When an incoming call to an extension is forwarded to a queue, the INVITE carries a Diversion header identifying the extension the call came from.

  

• Fixed: with Calls initiated via “Open API” enabled, Masked Number did not hide the phone number in the caller-name field in the Cloud Voice App web experience for API-initiated calls.

Email Template

• Notification email templates now support Spanish (System > Email > Email Templates > Notification Email Language).
• The email signature has been removed from the bottom of every template.

Archive

Fixed: recording files failed to archive to Amazon S3.

Active Directory Integration

User synchronization improved:

• The Map section adds an Avatar field, so user avatars sync from Active Directory to the matching PBX extensions.

  

• You can now edit the extension details of synced users. If you turn off mapping for a field, that field no longer syncs and you can set it directly on the PBX.

  

Collaboration Integration

The user-matching logic changed when Auto associate Extensions with the Users that share the same email address is enabled:

Integration	How matching works
Microsoft Entra ID (Azure Active Directory)	Matches on the Entra ID user’s User principal name rather than the Email Address field configured in the Map section.
Google Workspace	Matches on the Workspace Primary email rather than the Email Address field configured in the Map section.

Note

For Active Directory and Red Hat SSO integrations, the PBX still matches on the Email Address field configured in the Map section. To keep matching consistent and secure, prevent users from changing their email addresses in the identity provider.

Custom CRM/Helpdesk Integration Template

OAuth 2.0 authorization now supports Authorization Code Flow with Proof Key for Code Exchange (PKCE). When you configure OAuth 2.0 in an integration template, set it up with the Authorization Mode and PKCE Verification Method items under Request Configuration > Authentication Method. PKCE authorizes using a challenge-verifier pair instead of a client secret, which guards against authorization-code interception.

Dynamics 365 CRM Integration

Fixed: even with Play Call Recording enabled, the recording field on Dynamics 365 phone-call activities stayed empty.

API

Several API interfaces were extended:

• Extension.
  • Manage avatars: a new extension/uploadtempavatarfile interface uploads an avatar image and returns a file ID, and a new avatar parameter on extension/get, extension/query, extension/create, and extension/update reads or sets the extension’s avatar by file ID.
  • A new send_busy value is available for the destination parameters in extension presence settings on extension/get, extension/query, extension/create, and extension/update. With send_busy, the PBX rejects incoming calls and returns SIP 486 Busy Here.
• Trunk. New refer_to_support, refer_to_mode_list, and refer_to_prefix parameters on trunk/get, trunk/query, trunk/create, and trunk/update read or set the SIP REFER options, whether the trunk processes REFER requests, which destination types are allowed, and the prefix for external transfers.
• Voicemail. A new vm/forward interface forwards a voicemail to one or more extensions, and a new vm_detailInfo parameter on vm/get and vm/query returns the forwarding source (name and number) for forwarded voicemails.
• Message Channel. A new enb_duplicate_active_session parameter on message_channel/get, message_channel/query, message_channel/create, and message_channel/update reads or sets whether duplicate sessions are allowed on the channel.

CDR

Fixed: a contact’s call records appeared in a CDR search, but the scheduled report downloaded for the same contact came back empty.

Call Report

Fixed: running the Queue Performance report failed with a “request timeout” error even though the actual request limit had not been reached.