PortSIP PBX v22.8.0 is a major release that introduces Executive and Assistant call handling, Last Call Routing, more flexible call forwarding, expanded recording access controls, Salesforce CRM integration, and a redesigned System Administrator Web Portal.
It also expands support for AI transcription providers and IP phones, adds provisioning options, improves time zone handling, and lets a single server run separate Data Flow instances for multiple PBX deployments. Contact center and media fixes further improve reliability in production environments.
We recommend that all PortSIP PBX customers upgrade to v22.8.0.
New Features and Enhancements
Executive and Assistant
- Added Executive and Assistant functionality, allowing assistants to handle calls on behalf of executives.
- Added Feature Access Codes (FACs) for the Executive and Assistant feature.
Call Routing
- Added Last Call Routing, which associates external numbers with the extension that most recently communicated with them. Inbound calls can be routed to the matched extension, while outbound calls associate the dialed external number with the calling extension. If the matched extension is unavailable or declines the call, the call falls back to the configured Inbound Rule.
- Added support for configuring separate call forwarding rules for external calls received via SIP trunks.
- Optimized emergency call handling to comply with applicable regulatory requirements.
Contact Center
- Added the Allow Queue Calls to Busy Agents option.
- Optimized Queue Callback voice prompts.
Call Recording
- Added a pre-announcement prompt that notifies all participants that the call is being recorded once the call is established.
- Added User Group-based recording access control. User Group administrators can be granted access to recordings belonging to members of the groups they administer. They can access their own recordings and those of members in their administered groups, but not recordings of users outside those groups.
- Added the ability to filter and search call recordings by User Group.
- Added the ability to search call recordings and call history by caller or callee display name.
AI and Integrations
- Added Salesforce CRM integration.
- Added support for third-party OpenAI-compatible AI providers for transcription.
- Tenant AI transcription quotas are now calculated in minutes per month.
SIP Trunks
- Added support for configuring custom SIP headers to be added to, or relayed through, a specific SIP trunk.
IP Phone Provisioning
- Updated Fanvil IP phone provisioning templates to support RFC 5626 (SIP Outbound) configuration.
- Added support for the Snom M253 DECT Phone, D892M, and D810W IP phones.
- Added support for the Yealink T88W, T88W Pro, and T88V Pro IP phones.
- Added the ability to specify DNS servers and IP address settings when auto-provisioning IP phones.
- Added support for configuring the country-specific ringtone used when provisioning IP phones.
- Added support for the
original_brandkey in custom IP phone templates.
Web Portal and REST API
- Reimplemented the System Administrator Web Portal with a modernized UI. The Tenant Web Portal will be modernized in the next release.
- Standardized REST API response formats for improved consistency.
Time Zone Handling
- Extension email notifications now display dates and times in the extension’s configured time zone.
- When an extension uses a FAC to access voicemail or shared voicemail, voicemail dates and times are announced in the extension’s configured time zone.
Data Flow
- Added support for running multiple Data Flow Docker instances on a single server, allowing one server to provide Data Flow services for multiple PBX instances and reducing hardware requirements. Each Data Flow instance remains standalone; clustered Data Flow deployment is not supported.
Fixed Issues
- Fixed an issue where the caller heard mixed audio if a Ring Group or Queue agent replied with 183 and SDP during call setup.
- Fixed an issue where the agent heard mixed audio during a Queue Callback if the customer side replied with 183 and SDP.
- Fixed an issue where an IP whitelist or blacklist entry with a far-future expiration year, such as 3026, did not take effect.
- Fixed an issue where recording files could fail to be generated under a very high volume of concurrent calls.
- Fixed an issue where a Queue agent’s state did not automatically return to Ready if the agent declined a queue call and received another call before the declined call ended.
- Fixed an issue where a Queue agent’s state remained Not Ready after the agent declined a queue call that then overflowed back to the same agent, who answered and completed it.
- Fixed an issue where exporting extensions could produce duplicate entries or omit some extensions.
- Fixed an issue where the Company field was empty when the PBX synced individual-level contacts.
- Fixed an issue in certain scenarios that could cause the media server to send keepalive packets in a loop.
Changes
- Removed Play recording notifications from Inbound Rule settings. The new call recording pre-announcement feature supersedes this option.
For detailed information, please refer to the Summary of Changes.
SBC Updates
PortSIP SBC has been updated to v11.2.9.
This release includes the following update:
- Updated PortSIP ONE for WebRTC to v10.9.4, including the latest enhancements.
App Updates
PortSIP ONE for Windows, iOS, Android, and macOS has been updated to v10.9.4, including the latest enhancements.
For details, see the PortSIP ONE Summary of Changes.
Upgrade Instructions
To upgrade to v22.8.0 from a previous v22.x release, 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.
