PortSIP PBX v22.6: Enhanced Security, Smarter Call Routing, and Expanded Messaging

July 1, 2026

Date: July 1, 2026

PortSIP PBX v22.6 is a feature and maintenance release that introduces stronger security controls, improved IP phone provisioning, enhanced call routing, more flexible recording management, expanded messaging capabilities, and new contact center improvements.

This release adds voicemail PIN verification for auto-provisioning, Two-Factor Authentication and password reset support for System Administrators and Dealers, extension-level time zone support, enhanced Outbound Caller ID handling for SIP trunk transfers, WhatsApp Templates, MMS support, OpenAI-powered AI transcription, and improved Queue Wallboard reporting.

It also includes stability fixes and reliability improvements based on real-world production deployments, helping service providers and enterprises deliver UCaaS, CCaaS, and Cloud PBX services at scale.

New Features and Enhancements

Security and Authentication

Added Require Voicemail PIN Verification for Auto-Provisioning at both the system and tenant levels. When enabled, users must enter their voicemail PIN during IP phone auto-provisioning.
Added Provisioning PIN support for DECT phones and Hot Desking devices. After upgrading to v22.6, if Require Voicemail PIN Verification for Auto-Provisioning is enabled, administrators must configure a Provisioning PIN for each DECT phone and Hot Desking device. For details, refer to the user guide.
Added Two-Factor Authentication and password reset support for System Administrators and Dealers. Previously, these capabilities were available only for extension users.
Added password policy support for System Administrators and Dealers. Previously, password policies were supported only at the tenant level.
Made the Dealer email address mandatory. The email address is required for Two-Factor Authentication and password reset.

Recording and File Access

Added a tenant-level option to control whether call recording files are generated with public or private URLs. When private URLs are used, users must authenticate before accessing recording files.
Added more granular call recording controls. Administrators can now enable recording for specific Virtual Receptionists, Ring Groups, Queues, inbound rules, outbound rules, trunks, service types, and trunk calls.

API and System Behavior

Added an option to hide the PBX private IP address when PBX basic information is retrieved through the API.
Optimized error response codes for login and authentication REST APIs.
Added a new REST API to update an agent’s status across all queues the agent belongs to.
Split the caller field in voicemail and shared voicemail records into two separate fields: sender_name and sender_number.
Added a new tenant custom option, no_vm_sip_notification. When set to true, PortSIP PBX sends email notifications only for new voicemails and does not send SIP notifications.

Phone Provisioning and Device Management

Added support for specifying the IP address and DNS server for IP phones during auto-provisioning.
Added new columns on the Phones and DECT Phones pages to show online and offline status for IP phones and DECT phones.

Call Control and Routing

Added support for Selective Call Rejection and Selective Call Acceptance.
Added new Feature Access Codes to activate and deactivate Selective Call Rejection and Selective Call Acceptance.
Improved forwarding rule behavior when rules are activated or deactivated by Feature Access Code. When a forwarding rule is deactivated, it is now restored to its previous settings. In earlier versions, the rule was restored to the default settings.
Added time zone support for extensions. Call forwarding rules, routing logic, office hours, and holidays are now calculated based on the extension’s configured time zone. During upgrade, all extension time zones are set to the tenant time zone by default.
Updated the Outbound Caller ID selection logic for callee-side transfers to SIP trunks. This applies to scenarios such as forwarding rules, Blind REFER, exception handling, IVR timeout, IVR no-answer, IVR failure handling, and Night Mode transfers for Virtual Receptionists, Queues, and Ring Groups.
- For callee-side transfers to SIP trunks, PortSIP PBX now selects the Outbound Caller ID using the following priority:Callee Object Outbound Caller ID > Outbound Rule Outbound Caller ID > User Group Outbound Caller ID > Tenant-level Outbound Caller ID
- If the callee object does not have an Outbound Caller ID configured, PortSIP PBX does not fall back to the caller’s Outbound Caller ID. If no Outbound Caller ID is configured at any supported level, the caller ID is not replaced.
Added a Display Name field for Outbound Caller ID. When an Outbound Caller ID is selected for an outbound call, the configured Display Name is used as the display name in the From header of the SIP INVITE.
Updated outbound rule matching for calls initiated or forwarded by non-extension users, such as system extensions or trunks. By default, the User Group condition in outbound rules is still evaluated. If this condition should be ignored for these calls, the behavior can be controlled through custom options.
Improved call history display for Ring Group and Queue calls when simultaneous ringing is enabled. When multiple agents receive the same incoming call and one agent answers, the other agents no longer see the call as missed in the app call history. Instead, the call is treated as Call Completed Elsewhere for those agents.
Added support for selecting audio codecs for calls. Codecs that are not selected are removed from the SDP.
Added support for the SIP 603+ option at the system level.
Made the tenant website field mandatory. This website is used in SIP messages when calls are rejected with SIP 603+.

Virtual Receptionist, Voicemail, and Meetings

Enhanced the Virtual Receptionist URL Action to support capturing any DTMF input.
Increased the maximum voicemail greeting duration to 3 minutes.
Added support for uploading a custom meeting welcome prompt file.
Added automatic meeting cleanup. If no participant joins a meeting within 30 minutes after it is created, the meeting is closed automatically.
Improved prompt file format validation when prompt files are uploaded.

Messaging and WhatsApp

Added support for WhatsApp Templates, allowing businesses to send template messages to customers after the 24-hour customer service window has expired.
Added support for MMS messages.
Added support for sending images, audio, video, and files through WhatsApp.
Added support for customizing the WhatsApp message URL.

Contact Center and Reporting

Added support in Queue Wallboard to filter agent status by custom Not Ready Reason Code.
Added a new User Activities page and report.
Added a time zone filter for Call History in the Web Portal.

Administration and Notifications

Added new columns on the Tenants page to make tenant information easier to view and manage.
Added support for system-level email notification templates.
Optimized tenant-level email notification templates.
Added a new email notification when the pending webhook data size exceeds the configured threshold.
Added a Reset option for Microsoft 365 and Google Workspace integrations.

Integrations and Platform Updates

Integrated Sinch as a new provider for voice trunks and SMS.
Added OpenAI support for AI transcription.
Updated the HubSpot CRM integration to require the new crm.objects.companies.write permission, which is required to update company contacts.
Updated the Odoo CRM integration so it no longer uses the Company property.
Upgraded the Data Flow Server. ClickHouse has been upgraded to the latest LTS version.

Call Parking

Added a new Send park notification to the parker option in Park Global Settings. When a user parks a call to a group, the parker can also receive the notification and retrieve the parked call.
Added support for parking multiple calls on the same extension. In earlier versions, parking a new call to an extension failed if another call was already parked on that extension.

Fixed Issues

Fixed an issue where IP address blacklist and whitelist rules might not take effect.
Fixed an issue where configured SIP header forwarding did not take effect for trunk-to-extension calls when User calls was selected.
Fixed an issue where Music on Hold was not played when a queue call timed out and was transferred to another extension.
Fixed an issue where Queue Abandoned Call statistics could be calculated incorrectly.
Fixed an issue where the direction, cli, and Outbound Caller ID fields could be missing from target_xxx events.
Fixed an issue where an extension could remain in Other Call status after being added as a queue agent while already on a call.
Fixed a potential crash that could occur when an extension went offline and PortSIP PBX processed the registration timeout on the wrong thread.
Fixed a potential crash that could occur when users sent a large number of SIP PAGER messages.
Fixed a potential crash caused by an invalid destination value in CDR records.

SBC Updates

PortSIP SBC has been updated to v11.2.6.

This release includes the following updates:

Updated PortSIP ONE for WebRTC to v10.9.2, including the latest enhancements.
Fixed an issue where PortSIP SBC could crash in certain scenarios when processing invalid SDP.

App Updates

PortSIP ONE for Windows, iOS, Android, and macOS has been updated to v10.9.2, including the latest enhancements.

For details, see the PortSIP ONE Summary of Changes.

Upgrade Instructions

To upgrade to v22.6 from a previous v22.x version, follow the official upgrade guide:

Upgrade to the Latest Version Within v22.x

For more information or to download the latest release, visit www.portsip.com or contact the PortSIP support team.

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