PBX, Unified Communications

PortSIP PBX v16.0.1 is Ready

Posted on by Administrator
PortSIP PBX
30
Mar

PortSIP PBX provides a fully integrated modern cloud business communications system alongside advanced unified communications (UC) features including messaging, video meetings, sharing, WebRTC, Teams Direct Routing, and phone in one reliable, easy to use solution.

Marrch 30, 2023 – PortSIP, the developer of next-generation unified communications resolution, announced the launch of PortSIP PBX v16.0.1 for both Windows and Linux.

The v16.0.1 is a minor release that focuses on fixing bugs.

Changes for Release v16.0.1

The following changes are included in this release:

  • If the “Set agent to Ready automatically” option of the queue is disabled, once an agent completes a non-queue call, their status will be restored to their previous status instead of “Wrap up”.
  • Display only the last part of the license key and mask other parts with *.
  • A new REST API GET /api/sessions/directly has been added which allows launching calls by URL.
  • The REST API POST /api/users/{id}/call has been removed.
  • When a caller presses a DTMF that is not predefined, a prompt will play.
  • The MOH file of the queue is no longer mandatory. The queue will use the default MOH file if the user has not uploaded a MOH file for a queue.
  • If the user is offline and push notifications are enabled, their status will be displayed as “PushOnline” in the Web portal.
  • The Web portal will display the language as the browser language.
  • The WebRTC client URL port has been changed to 10443, don't forget to create the firewall rule for port 10443 on TCP.
  • Passwords will be auto-generated when creating a user.
  • Custom menus can be used to link to external websites.
  • Display the IP Address when viewing the details of an online user.
  • If a user signs in to the Web portal with an incorrect password or if the user does not exist, the detailed reason will not be returned.
  • The issue where auto-provisioning for GrandStremm IP Phones the URL is wrong has been fixed.
  • The issue where there is maybe no voice after completing an attended transfer has been fixed.
  • The issue where extension call forwarding is not affected if the call comes from Virtual Receptionist or Queue has been fixed.
  • The issue where the outbound caller ID is set as starting with + and it won’t set to the INVITE message when making a call to the trunk has been fixed.
  • The issue where if a user registers to PBX from multiple devices and launches a call from REST API with that user and the call actually answered but the CDR will display as not answered has been fixed.
  • The issue where if you launch a call by REST API, one party may have no voice has been fixed.
  • In the Webhook and WSI messages, the ID has been changed from in64_t to string in order to be compatible with JS.
  • The billing issue is where if the call duration is multiples of the billing circle, don’t plus more 1 billing round; in the previous version, if the duration is 10 seconds and the billing circle is 5 seconds, we will charge 3 rounds, now just charge 2 rounds.

Please follow the guide to upgrading the PBX and SBC if you are running the PortSIP PBX v16.0.0 and PortSIP SBC 10.0.0.

Upgrading PortSIP PBX to New Versions

Upgrading PortSIP SBC to New Versions

For more details, please download the free edition here.

Administrator