Version history for MicroSIP
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Changes for v3.19.15 - v3.19.18
- - added default Calls/Contacts action in Settings
- - added Dialing Prefix in Account
- - added Merge All / Separate All in conference
- - added more LPCM codec rates
- - added support of Yealink users directory
- - added Call Ended status in Calls
- - added scrolling in Calls and Contacts
- - shortcuts optimization
- - fixed recording after call hold
- - fixed redial in extended mode
- - fixed sorting by date in Calls
- - misc fixes and improvements
- - pjsip update 2.9
Changes for v3.19.14 - v3.19.15
- - auto unmute on a new call
- - ALT to set focus on the menu button
- - show blocked incoming calls on the call page
- - show original failed call message
- - fixed VP8 bitrate setting
Changes for v3.19.10 - v3.19.11
- - settings are divided into 2 columns
- - reject call with busy command (decline in the past)
- - added ability to specify multiple STUN or NS
- - hide voicemail button
- - pjsip update 5931
- - fixed call recording path
- - other improvements
Changes for v3.19.8 - v3.19.10
- - added call recording (all calls or manual recording)
- - added contacts CSV import/export
- - added command events: cmdOutgoingCall, cmdCallRing, cmdCallBusy
- - added number column in contacts list
- - fixed command events execution sequence
- - fixed codepage in audio and video devices names
- - other fixes
Changes for v3.19.7 - v3.19.8
- - added DNS SRV option
- - added support of UTF8 and UTF16 in Google CSV import
- - added tooltips for bottom buttons
- - added bugfix for CANCEL outgoing call
- - updated VP8 codec
Changes for v3.19.5 - v3.19.7
- - audio codecs adapted for screen readers, manage with Space and Delete keys
- - changed Dialer resizing
- - fixed Opus codec
- - fixed several possible crashes
- - pjsip update 5861
Changes for v3.19.3 - v3.19.5
- - missed call tray icon
- - AA and DND buttons changes
- - changed crash report
- - pjsip update 5851
Changes for v3.18.5 - v3.19.3
- - missed call tray icon
- - AA and DND buttons changes
- - changed crash report
- - pjsip update 5851
- - new layout
- - subscribe text info for contact (new column)
- - on-the-phone presnece status
- - changelog in update dialog
- - system fixe
Changes for v3.18.3 - v3.18.5
- - import contacts (Google CSV format)
- - AMR codec parameters changed to be compatible with Android SIP dialer
- - small fixes
Changes for v3.18.2 - v3.18.3
- - added RTP port range setting
- - added SIP source port setting
- - added "rport" option
- - added possibility to make call with media button
- - fixed window focus at startup
Changes for v3.17.8 - v3.18.2
- - improved compatibility in SDP negotiation
- - increased maximum SIP packet length
- 3.18.1 [MicroSIP-3.18.1.exe | portable] (1431 downloads), [MicroSIP-Lite-3.18.1.exe | portable] (296 downloads)
- - high quality WebRTC echo canceler instead of Speex
- - echo canceler enabled by default
- - Public Address setting now affects also on Via and Contact headers
- - rejecting an offered stream in SDP according RFC 3264
- - minor fixes
Changes for v3.17.3 - v3.17.5
- - UI changes
- - fixed multi instance management in Wine
Changes for v3.16.9 - v3.17.3
- - added option "Bring to Front on Incoming Call"
- - project has moved in new IDE and built with modern compiler
- - update library SDL 2.0.7
- - update library VPX 1.7.0
- - update library ffmpeg 3.4.1
- - update library x264 0.152
- - new G.729 codec implementation (annexes A and B supported)
- - software video render for RDP connection
- - other fixes
Changes for v3.16.7 - v3.16.9
- - fixed media buttons
- - fixed broken info in crash report
- - settings dialog internal edits
- - added AMR-WB codec
- - added option: "Handle Media Buttons"
- - improved multi instance management
- - exploit protection (enabled SafeSEH, removed shared memory block)
Changes for v3.16.1 - v3.16.4
- - added redial last number button
- - added G.723 codec (no licence, limited usage)
- - added possibility to pass DTMF automatically
- - added auto answer after by timeout "Call-Info: answer-after=5"
- - fixed application crash
- - fixed H.264 video
- - fixed save bitrate
- - fixed make active option in menu
- - code optimization
- - update openssl 1.1.0f
- - update pjsip 2.7.1
Changes for v3.15.10 - v3.16.1
- - pjsip update 2.7
- - openssl update 1.1.0f
- - pass DTMF commands after call established (number,DTMFsequence1,DTMFsequence1,,,DTMFsequence3), one comma means pause in one second
- - auto answer after timeout "Call-Info: answer-after=5"
- - reverted make active option in menu
- - small fixes
Changes for v3.15.9 - v3.15.10
- - shortcuts feature (configurable buttons)
Changes for v3.15.7 - v3.15.9
- - adjusting speakers volume only for calls
- - HW/SW level microphone adjustment option
- - microphone amplification option
- - mute improvement
- - improved VU-meters
- - fixed VU-meters for WinXP
- - fixed VU-meters for conference call
- - fixed VU-meters for Wine (Linux)
- - other fixes
Changes for v3.15.6 - v3.15.7
- - VU meters and volume buttons
- - added ini settings: autoHangUpTime, maxConcurrentCalls, noResize
- - fixed hiding voicemail icon
- - fixed call ending notification sound
Changes for v3.15.5 - v3.15.6
- - voicemail feature
- - disable session timers option
- - fixed date/time display in Calls list
Changes for v3.15.4 - v3.15.5
- - save position of incoming call window
- - port knocker feature (UDP-ping SIP server before register, see help for details)
- - added possibility to dial numbers with parameters and headers. Example: sip:user@host:port;param=pval?header=hval
- - STUN server moved into Settings
- - small fixes
Changes for v3.15.3 - v3.15.4
- - bug fixes
Changes for v3.15.1 - v3.15.3
- - DTMF method option (Auto, In-band, RCF2833, SIP-INFO)
- - audible remote DTMF signals (RCF2833, SIP-INFO)
- - mute microphone while sending in-band DTMF
- - fixed bug in dialing from command line
- 3.15.2
- - pjsip update 2.6
- - small edits
