Firmware Updates
Changelog
- Add PCM support
- Add TLS transport support
- Add support for RTL8152/RTL8153-based and RNDIS-based USB-Ethernet and cellular devices, including:
- Anker USB 3.0 to RJ45 Gigabit Ethernet Adapter
- EE Mini Hub Halo, EE71
- 5GEE WiFi [QTAD52E]
- Netgear Nighthawk M5 [MR5200-100EUS]
- Netgear Nighthawk M2 [MR2100-100EUS]
- Add minimum / maximum jitter buffer size setting
- Fix long periods of decoder packet loss / total stream loss on some decoders after brief Ethernet disconnection
- Increase RTSP receive buffer to improve stream stability for sources with high jitter
- Improve recording, ordering and retention of log messages
- Add charset to Content-Type header on web interface to ensure Unicode characters are displayed correctly
- Improve Contact header handling for NAT traversal
- Fix minor problem with web interface not updating when changing cellular APN from another web client
- Minor internal fixes
Changelog
- Add transport (Auto/UDP/TCP) settings for registration and calls (defaults to registration setting)
- Add support for sending DTMF (RTP (RFC 4733) and SIP INFO)
- Add initial support for USB cellular modems and APN setting to web interface
- Add support for USB Networking licence
- Add automatic private addressing in the absence of a DHCP server on Ethernet port
- Add internal event logging
- Use SHA256 for self-signed web interface certificates
- ICE improvements in situations where STUN fails, and where remote device is behind a SIP ALG
- ICE enhancements for particular combinations of NAT behaviours
- Allow call to proceed when STUN binding times out
- Fix auto-reconnect behaviour
- Fix asymmetrical bitrate in non-64k Opus calls (bitrate from caller was always 64k)
- Fix bug with duplicated dynamic RTP payload types
- Fix for DHCP IP changing on firmware update and Ethernet priority when upgrading from older firmware
- Fix for possibility of Ethernet IP address changing after firmware update immediately after factory reboot
- Fix for RØDECaster Pro running beta firmware 2.1.2
Changelog
- Add video stream support for Zoom compatibility
- Add support for receiving RTSP streams (same codec support as SIP calls) by specifying an rtsp:// URL in the destination field
- Prevent IP address changing on firmware update when configured for DHCP
- Include device name as "display name" in SIP headers
- Fix missing closing tag in web interface
Changelog
- Add user-selectable option for strict username matching for incoming calls
- Improve default audio levels for BOYA BY-PM700
- Fix failure to register without a reboot in certain circumstances, e.g. when first enabling registration
- Fix web interface bug where configured Auth Username was not displayed, which could result in it being erased when saving SIP settings
- Fix race condition when IP address changes which could leave unit unregistered and unable to call
Changelog
- Add support for quick-dials to CallMe-TS
- Add quick-dial mode for USB number pads (new switch in System Settings)
- Improve layout of Quick-Dials tab
- Display device model name in System Settings
- Improve default audio levels for RØDE NT-USB Mini
- Fix race condition when changing account settings which could leave unit unregistered and unable to call
- Improve reliability when operating behind SIP Application Layer Gateways
- Fix ICE bug where remote-candidates attribute would be included even when not in controlling role
- Internal bug fixes
Changelog
- Fix startup problems and USB keyboard issues on CallMe-TS
- USB device sharing support (beta)
2.0
Download - REPLACED BY 2.1 ABOVE
Changelog
- SmartStream
- Advanced Audio Routing
- New Audio Routing drop-down in call dialog and quick-dial configuration to specify Stereo, Left Only or Right Only
- New Incoming Call Audio Routing drop-down in System Settings to specify Stereo, Left Only, Right Only or Alternating
- Use a mono codec where relevant (Opus) when a mono routing is specified
- Multiple Simultaneous Calls (licensed feature)
- Allow up to 8 simultaneous calls, each with individual audio routing
- Status tab displays a table of all active calls with individual routing indication (L / R speaker icons), metering and drop button
- New Drop All button hangs up all active calls
- Expanded RTP port range (any port forwarding on firewalls / routers must be updated): 15004-15515
- USB keyboard support - allows dialling using a keyboard or number pad:
- Type a username (when registered to a SIP server) or full SIP URI and press enter to dial
- Backspace key can be used to delete previous characters
- By default, audio routing will be stereo; to override, when making the call, press and hold enter, press / on the number pad for left-only or * on the number pad for right-only, then release enter
- Escape and number pad . keys behave in the same way as front-panel DROP button on CallMe-TR: press and hold for 1s to drop all calls; press and hold for >=20s to factory reset (particularly useful on CallMe-TS)
- New "individual drop" mode for front-panel DROP button to drop a single call when several are active: a brief press of DROP activates the mode for 5s, after which a CALL button, or DROP + a CALL button (exactly as for dialling), can be pressed to drop the corresponding quick dial
- Full ICE support: new NAT Traversal drop-down on SIP Settings tab offers None, STUN or ICE
- New Register Keep-alive option on SIP Settings tab to maintain NAT binding for SIP port (improves compatibility with some SIP servers where incoming calls would intermittently fail)
- Improved front-panel LED behaviour. LEDs now illuminate correctly when quick-dial specified by username, without sip: prefix etc..
- Display caller name for incoming calls where available (in addition to SIP URI)
- New Friendly Name field in call dialog and quick-dial configuration to specify a name for the connection which will be displayed on the Status tab for outgoing calls
- Show CONNECTING status on web interface to indicate when a call attempt is in progress
- New Force Codec Choice option on System Settings tab to offer only the selected codec for a call (rather than just making it the preferred choice)
- Detect connection loss between web interface and codec; disable web interface when this occurs and automatically reconnect
- Improved call auto-reconnect behaviour: only reconnect automatically if the call actually connected successfully, and only if it wasn't actively dropped by either end. Continue to retry even on local network failure.
- New Licences panel on System Settings tab displays licensed features and allows add / remove of licences
- New MAC address display on System Settings screen
- Support for RØDECaster Pro USB audio
- Reduced jitter with USB audio devices
- Improved robustness of settings to power failures
- Fix to prevent IPv6-based spam calls and fix outgoing call failures after reboot when routable IPv6 address assigned by DHCP
- Fix to prevent the indication of "phantom" calls after automatically trying to reconnect after local network failure
- Fix for faster re-registrations to non-conformant SIP servers
- Internal performance improvements and memory leak fix
- Experimental wifi hotspot mode:
- When a compatible USB wifi adapter is connected, CallMe-T will create an open hotspot with the same name as the Device Name
- Connect to this network from a phone, tablet or PC (choose to "Stay connected" when prompted by Android despite no internet)
- Browse to http://10.0.0.1 or http://c.mt to access the normal web interface
- The interface currently doesn't work well on small screens, but should be good enough to configure IP settings and quick dials
- May not work on all browsers / devices; for best results, use Chrome
- New LED modes to reflect multiple calls:
- CALL buttons
- When the main or shift QD assigned to a button is connecting: flash 200ms on, 200ms off; otherwise:
- When only the main QD assigned to a button is connected: on solid
- When only the shift QD assigned to a button is connected: flash 500ms on, 500ms off
- When both main and shift QDs assigned to a button are connected: flash 2000ms on, 500ms off
- DROP button
- When "individual drop" mode is active: flash 100ms on, 100ms off; otherwise:
- When at least one call is connected: flash 500ms on, 500ms off; otherwise:
- If the unit is registered to a SIP server: on; otherwise: off
Changelog
- Initial support for USB-enabled hardware and USB audio devices
- Significantly reduced jitter
- Double number of quick-dials on CallMe-TR using DROP as shift
- Add call buttons to quick-dial configuration on CallMe-TR
- Add factory reset procedure on CallMe-TR (20s press of DROP button)
- Automatic reconnect on connection timeout
- Add SIP User-Agent header and non-empty SDP s= field
- Add Vendor= TXT record to mDNS-SD to allow filtering in Vortex Device Discovery Tool
- Fix network settings to allow empty default gateway with static IP
- Fix firmware file selection in recent versions of Chrome
- Various internal bug fixes
Changelog
- Add codec and bitrate selection for outgoing calls and quick-dials
- Add option to disable non-HTTPS web interface
- Fix bug in firmware update process via HTTPS web interface
- Fix PPM calibration
Changelog
- Add G.722 and G.711 (A-law and u-law)
- Improve registration status reporting
Changelog
- Add login and password change function to web interface (default password is MAC address)
- Add HTTPS support (currently self-signed certificate only so must accept browser security warning; connection is still encrypted)
- Add "please wait" dialog for network setting, password setting and reboot
- Show registration status on DROP LED when not connected
- Improve network address change / startup behaviour and reliability of SIP registration
- Improve NAT router support to ensure connections to external destinations can be dropped from remote end
- Drop connection if no audio received for 60s
Changelog
- Remove requirement to specify both primary and secondary DNS servers when setting a static IP
- Remove redundant internal network adapter which caused problems when setting a static IP in the range 192.168.11.x