Firmware Updates
Note: When using firmware versions 2.7 and above (PCM support) with SIP over UDP (the default), we recommend enabling Force Codec Choice and disabling Enable Video Stream in System Settings to improve compatibility with networks which do not support fragmented packets.
Changelog
- Add global DNS configuration
- Add new panel to web interface Network Settings page to configure servers
- Add Icecast streaming support
Changelog
- Add SIP redundant multi-streaming (RFC 7198) (licensed feature)
- Add enable switch to System Settings
- Add per-interface network stats to web interface Status page
- Add CallMe Hub support
- Allows web-based remote monitoring and control, including firmware updates
- Add USB status light support
- Add support for PCM at 22,050Hz and 11,025Hz
- Add PCM support to RTSP
- Fix reading of wrong sample rate from RTSP session description
- Allow spaces in quick-dial friendly names
- Add icons to license add / remove buttons on web interface
- Prevent SIP account changes when calls are active
- DNS improvements for more reliable multi-interface operation
- Remove Ethernet static DNS settings
- Override DHCP-acquired Ethernet DNS settings
- Internal SIP registration improvements
- Prevent hang and failure to re-REGISTER when changing SIP account settings
- Prevent failure to process SIP messages after re-REGISTER
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