Version 30.8.0.8

Released Date: July 5, 2018

New Feature

Multicast Paging
Multicast Paging allows you to easily and quickly broadcast instant audio announcements to phone users who
are listening to the configured multicast IP address and channel.

Note: The IP phone should support multicast paging feature.
Added support for monitoring status of dynamic agents
Set a BLF key on IP phone to monitor the status of dynamic agents. The BLF key value is *75{extension number}*{queue number}.
For example, to monitor the status of agent 1000 that is in queue 6700, set BLF key value to *751000*6700.
In the new version, dynamic agents have two methods to log in and log out a queue.
- Dial {queue number}* to log in a queue; dial {queue number}** to log out a queue.
- Dial *75{queue number} to log in a queue, dial *75{queue number} again to log out a queue.
Added email template for Voicemail to Email and Fax to Email
Go to Settings > System > Email > Email Template to modify the email template.

Security

Extension

Deleted the Name setting, added Caller ID name setting. Generally, set Caller ID name to the user's name.
Fixed the issue that All Busy Mode for SIP Forking setting didn't take effect if the extension had a DOD number.
Fixed the compatibility issue with Vtech IP phone: TLS extension could not be registered.

SIP Trunk

Added Outbound Parameters to configure SIP parameters for outbound calls.
- P Asserted Identify: Choose which "P Asserted Identify" value should be contained in the SIP INVITE headers.
- Diversion: Choose which "Diversion" value should be contained in the SIP INVITE headers.
- Remote Party ID: Choose which "Remote Party ID" value should be contained in the SIP INVITE headers.
Added Transfer Parameters to configure SIP parameters for transferred calls.
- From: Choose which "From" value should be contained in the SIP INVITE headers.
- Diversion: Choose which "Diversion" value should be contained in the SIP INVITE headers.
- Remote Party ID: Choose which "Remote Party ID" value should be contained in the SIP INVITE headers.
- P Asserted Identify: Choose which "P Asserted Identify" value should be contained in the SIP INVITE headers.
Supports up to 64 DID numbers.

Jitter Buffer

The path of Jitter Buffer settings are changed to Settings > PBX > General > Jitter Buffer.
Added support for choosing which extensions and trunks to apply jitter buffer settings.
Added support for configuring the Adaptive Adjustment Size of jitter buffer.

API Interfaces

Added support for sending API reports to an absolute path of the third-party software.
Added support for querying, downloading, and deleting voicemail files.
Optimized the Hotel Wake-up Service.
Optimized the DTMF output.
Fixed the issue that monitoring analog phones would fail.
Fixed the issue that the PBX didn't stop sending API reports when the API token expired.
Fixed the issue that API reports would be omitted.
Fixed the issue on the calls between extensions: The called extension would ring even though the caller didn't pick up the phone to trigger the call.
Fixed the issue that the extension status on the PBX Monitor page didn't refresh automatically.

Storage

Optimized Network Drive settings: If the configurations are incorrect, the system will prompt the cause of the error.
Optimized FTP sharing:
- When you enable FTP sharing, the system will prompt you to enable FTP service.
- When you disable FTP sharing, the system will prompt you that the FTP sharing is enabled.
Fixed the Network Drive issue: If a mounted network drive was disconnected then automatically mounted again, the PBX could not recognize the network drive.

Event Center

Firewall

Optimized Drop All setting:
- After enabling Drop All, the system will prompt you to add a firewall rule to accept DNS address.
- After enabling Drop All, the system will automatically add two firewall rules to allow the PBX to get new version of PBX firmware and Applications.

Operation Log

The operation of "Data Processing Agreement" will be recorded in operation logs.

Static Route

Prompt

CDR