Scope
This document covers version changes to VX Enterprise and VX Prime+ VoIP phone systems.
Latest update : November 11, 2021
Please note that the software versions described here could be in
the beta or stable stages of the release process at any given time.
Typically, Telos Alliance Support will provide this document and
recommend a specific software version to address a particular issue.
To access the latest stable release software instead, please visit
the appropriate product page at TelosAlliance.com, where you will find
the current stable release software as well as installation instructions.
v1.3.1
Use generated root passwords (#16457)
Allow sending kernel output to serial console (off by default)
Improve system stability (#16160)
Security improvements based on PEN testing (various tickets)
Improve logging during system restarts
Improve ramp-up time after connecting call audio (#16092)
Improve RTP codec renegotiation during an active call (#15554)
v1.3.0
Add local console UI for network configuration (#14873)
Add per-line SIP outbound proxy config file option (#14876/#14982)
Add display of interface MAC addresses in the web UI (#15071)
Improve SMPTE 2110-30 compatibility for AES67 stream timestamps (#10608)
Improve PTP clock sync recovery (#13679)
Improve handling of DNS configuration on systems with IPv6 (#15331)
Improve web UI security (#14874, #14875)
Improve RTP payload type negotiation for DTMF event packets (#14919)
Remove remaining Flash metering elements (#15281)
Resolve RTP sequence numbering and ordering issues for DTMF event packets (#14872)
Resolve FW upload and bank switch issues introduced in v1.2.0 (#15175, #15177)
v1.2.1
Resolve FW upload and bank switch issues introduced in v1.2.0 (#15175, #15177)
Add local console UI for network configuration (#14873)
v1.2.0
Enhance web UI security (#4350, #13675, #13852, #13853)
Enhance SIP security (#13664)
Improve default VLAN tag behavior for Live Stereo streams (#13632)
Filter ptp4l log messages to reduce write activity (#13814)
Improve studio AGC ramp-up time (#13907)
Prevent caller audio latency build-up in edge cases
Add support for Australian call progress tones (#14251)
v1.1.2
Allow handling of the "+" (and some other) characters in the SIP user field (#13574)
v1.1.1
Allow QoS settings changes via qos.cfg (#11613, #10652)
v1.1.0.3
Add interactive local console menu for viewing and changing network settings (#14873)
v1.1.0.2
Fix rare crash caused by repeated LWCP connections (#13062)
Prevent repeated log entries from DSP Engine when disconnected from network
v1.1.0
This version is identical to v1.0.11; only the version number was updated.
v1.0.11
Fix high-pitched voice after long call hold (#12324)
Fix crash on reboot
Mitigate TCP SACK PANIC vulnerabilities
Address wrong key used for address book lookup
v1.0.10
Changes to PTP are now written to persistent storage
Fix for 200 OK to incoming INVITE not being retransmitted
v1.0.9
Added SAP support for outbound AES67 streams
Fix VSet ringer disruptions when using PTP clock
Fix incorrect AES67 TX stream timestamps
v1.0.8
Log important RTP and DSP events, like underruns and sequence errors, as they happen
Log stream stats summary when line is deleted
Log min/max RTP transit delta when line is deleted
Log RTP jitter histogram when line is deleted
Further improvments to VoIP stream lock
v1.0.7
Limit the maximum rate of PTP delay requests to 1/second
Fix stream stats erroneously showing TX underruns on AES67/standard streams
Fix overruns on Stream Stats page not being counted
Fix VoIP streams unstable lock causing clicks
Fix low buffering on AES67
v1.0.6
Limit the maximum rate of PTP delay requests to 1/second (#11429)
Fix stream stats erroneously showing TX underruns on AES67/standard streams