Category Releases / Page 3

The FreeSWITCH 1.6.5 release is here!

Releases -

The FreeSWITCH 1.6.5 release is here! This release contains everything since version 1.6.2. This is a pretty big release for the 1.6 branch so upgrading now is a really good idea. This is a routine maintenance and security release and the resources are located here:

Release files are located here:

Security issues:

A bug allowing for a remotely exploited DoS attack through custom crafted network traffic via cJSON has been fixed. We classify this issue as High Severity. A patch was added by Anthony Minessale in commit 4bdca81 to resolve this issue. All versions from 1.4.4 through the previous release are vulnerable. We highly recommend updating to the current release version as soon as possible.
https://web.nvd.nist.gov/view/vuln/detail?vulnId=CVE-2015-7392

New features that were added:

  • FS-8243 [mod_opus] Improve the way FEC info is detected within frames by adding support for ptimes higher than 20 ms for FEC detection
  • FS-8161 [mod_opus] Keep FEC enabled only if loss > 10 ( otherwise PLC is supposed to be better)
  • FS-8179 [mod_opus] Improvement on new jitter buffer debugging (debug lookahead FEC)
  • FS-8313 [mod_opus] Introduced new configuration setting ‘decoder-stats’ to show decoder stats at end of call (how many times it did PLC or FEC)
  • FS-8254 [verto_communicator] Create a source map file
  • FS-8263 [verto_communicator] Created the reset banner action, floor and presenter badges, and lock icon in floorLocked status
  • FS-8288 [verto_communicator] Added an About screen with version information and links to FS.org and added a link to Confluence with documentation for VC
  • FS-8289 [verto_communicator] Make mute/unmute audio/video clickable
  • FS-8290 [verto_communicator] Automatically mark dedicated encoder if out/in bandwith isn’t set to ‘Server default’ and adding help text on how to enable dedicated remote encoder
  • FS-8030 [verto_communicator] Added ngSanitize as a dependency, vertoFilters module and picturify filter and changed chat image display behavior (break line before rendering).
  • FS-8293 [verto_communicator] Added built in speed test feature which gives feedback of available bandwidth and customizes call settings based on bandwidth available
  • FS-8401 [verto_communicator] Added Speaker selection in settings modal and video page and refactor the sinkid function into verto lib
  • FS-8545 [verto_communicator] Fixed a read lock regression and do not allow video floor on a member with a reservation id set
  • FS-8195 [core] Compatibility with Solaris 11 process privileges
  • FS-8547 [core] Add error log into stats to log when quality impacting events begin and end
  • FS-8321 [core] Add variable media_mix_inbound_outbound_codecs to mix inbound and outbound codecs. BEHAVIOR CHANGE
  • FS-8281 [core] Expose SRTP and SRTCP crypto keys as channel variables to aid with debugging
  • FS-8287 [mod_local_stream] Refactor local_stream API to be more consistent and add auto complete
  • FS-8375 [mod_conference] Add the field conferenceMemberID to the event broadcasted to inform a verto client about joining a conference.
  • FS-8543 [mod_conference] Improve mute handling on conference and WebRTC
  • FS-8545 [mod_conference][verto_communicator] Improve controls for screen share
  • FS-8546 [mod_conference][mod_verto] Make original video demo back-compatible with livearray-json-status
  • FS-8529 [mod_conference] Added video-floor to personal canvas mode
  • FS-8377 [mod_hiredis] Adding expanded support for limit_* functionality and fixed the handling of hiredis limit release when using an interval. The expectation for interval is to NOT decrement the limit.
  • FS-8380 [mod_av] Improve the handling of vw and vh core file parameters to avoid video cropping and crashing
  • FS-8415 [mod_sofia] Added support for early media with 180 using early_use_180=true
  • FS-8416 [mod_xml_radius] Added the ability to format the variable in the param field
  • FS-8534 [rtcp] Added calculated RTT average (RTCP SR) value to help with detecting congested network links
  • FS-8549 [mod_http_cache] Add support for AWS_ACCESS_KEY_ID and AWS_SECRET_ACCESS_KEY environment variables in S3 profiles
  • FS-8559 [mod_shout] Should have "mpga" in it's list of supported extensions

Improvements in build system, cross platform support, and packaging:

  • FS-8236 [build] Fixed building without libyuv on compilers that throw an error on unused static function and fixed ifdefs for building without libyuv
  • FS-8350 [build] Fix Windows build errors.
  • FS-8389 [build] Fixed msvc 2015 build warnings
  • FS-8316 [build][Debian] Fixed new build warning from latest clang and resolved the build warnings in the modules too
  • FS-8255 [Debian] Fixed codename changes since Jessie was released as stable
  • FS-8271 [Debian] Simplify package building for the default case
  • FS-8270 [Debian] Fix for package installation failing if /etc/freeswitch/tls is missing
  • FS-8285 [Debian] Removed heart attack inducing warning message when updating deb packages
  • FS-7817 Removed use of _NONSTD for Windows builds of spandsp, so (hopefully) eliminate compatibility problem
  • FS-8271 [Debian] Adding some logging, and more cautious handling of spaces in parameters. Now the default will build packages with the upstream FS package repos. This is a change in the default behavior of the Debian packaging system with the justification that in 1.6 it is now required to use the FS public repo for dependencies because system dependencies have been removed from the FS codebase which used to be included. And defaulting to automatically download the binary dependencies because without major changes to package building in cowbuilder(which is the primary supported method of building FS packages), you can’t access the network to build the binary packages from the source package. If using system apt repo list, then include the supplementary ones too
  • FS-7928 FS-7618 [Debian] Systemd and package build improvements
  • FS-8362 [Debian] Now if you install with freeswitch-all you will get the default fonts too
  • FS-8426 [Debian] Put freeswitch.pm into /usr/share/perl5 so it can be found on both Wheezy and Jessie
  • FS-8333 [build][Debian] Added mod_hiredis.deb
  • ESL-111 [python] Fixed esl/python/Makefile to create install directory
  • FS-8233 [automation] In order to clean up build dependencies for the automated tests, convert the tests/*/Makefile.am into an include file for the top level Makefile.am. This will greatly simplify dependency tracking, and allow tests to be rerun easily on FS source code changes.
  • FS-7820 [automation] Use a more appropriate function for printing diagnostics
  • FS-8194 FS-7910 FS-7937 Various systemd service improvements
  • FS-8298 [mod_gsmopen] Fixed a build error
  • FS-8398 [Ubuntu] Added event_handlers/mod_amqp to avoided modules for Ubuntu 14.04 Trusty
  • FS-8239 [mod_av] Fixed the default value to avoid failed build on CentOS 7
  • FS-8427 [build][mod_av] Fixed an incompatible type for %ld in prinrtf compiler error
  • FS-8248 [mod_event_socket] Moved python binaries into site arch path to match standards

The following bugs were squashed:

  • FS-8221 [verto_communicator] Fix number in call history
  • FS-8223 [verto_communicator] Fixing members list layout when callerid is too long
  • FS-8225 [verto_communicator] Avoid duplicate members when recovering calls
  • FS-8214 [verto_communicator] Better handling calls in VC, answering them respecting useVideo param
  • FS-8291 [verto_communicator] Fixed contributors url
  • FS-8229 [verto_communicator] Changing moderator actions bullet menu color to #333
  • FS-8219 [verto_communicator] Fix for camera not deactivating after init or after hangup
  • FS-8245 [verto_communicator] Fix for Video Resolutions available in “Video Quality” drop down not always correct
  • FS-8251 [verto_communicator] Factory reset now clears all local storage
  • FS-8257 [verto_communicator] Fixed configuration provision url because configuration doesn’t work with `grunt serve` and non pathname urls
  • FS-8273 [verto] [verto_communicator] Clear the CF_RECOVERING flag in a spot that was missed
  • FS-8260 [verto_communicator] Prompt for banner text
  • FS-8067 [verto_communicator] When no email is present make sure mm is the default avatar in the circle this way the talk indicator works on PSTN and SIP callers.
  • FS-8247 [verto_communicator] When websocket disconnects go to splash screen to wait for the reconnect
  • FS-8300 [verto_communicator] Fixing reload bug so reloading twice is no longer needed
  • FS-8331 [verto_communicator] Do not show reconnect splash when user has clicked logout
  • FS-8365 [verto_communicator] Fixed a bug with the chat notifications not going away unless you exited and came back to it
  • FS-8336 [verto_communicator] Updating mic and video overlay controls upon receiving member update from live array and use conferenceMemberID when checking if the updated member is the local user
  • FS-8222 [verto_communicator] Updated getScreenId.js in order to detect plugin issues and attached an ‘ended’ event to screenshare stream in order to detect ‘stop sharing’ click
  • FS-8542 [verto_communicator] Fixed the tooltips of video controls
  • FS-8556 [mod_verto] Screen shares are not recoverable so do not try
  • FS-8293 [mod_verto] Fixed some regressions where speed test caused excessive downlink bandwidth
  • FS-8232 [mod_conference] Conference sending too many video refresh requests
  • FS-8241 [mod_conference] Fix for conference stops playing video when local_stream changes source
  • FS-8261 [mod_conference] Fixed the conference segfaulting when trying to reset the banner
  • FS-8297 [mod_conference] A fix for auto STUN switching IPs quickly and WebRTC video not working
  • FS-8130 [mod_conference] Fix for micro cut-offs and unstable voice issues and fixed a regression causing excessive mark bit detection in some cases
  • FS-8317 [mod_conference] Fix for playing multiple files at once to stack them for immediate playback, sometimes breaking and the floor layer becoming unusable for the rest of the conference.
  • FS-8328 [mod_conference] Fixed missing ‘else’ keyword
  • FS-8307 [mod_conference] Fixed an issue with the order of codecs causing loss of RTP stream
  • FS-8280 [mod_conference] Fixed an issue with FS sending redundant stop-recording event notifications
  • FS-8384 [mod_conference] Fixed some locking contention issues between external commands and the video engine
  • FS-8527 [mod_conference] Do not send the video of last_video_floor_holder to video_floor_holder if the videos are related
  • FS-8053 [mod_conference][mod_sofia] Fix for WebRTC's SDP containing a=sendonly for video, but the client still receiving the video stream
  • FS-8220 [core] Fix for DTMF not working between telephone-event/48000 A leg and telephone-event/8000 B leg
  • FS-8166 [core] Mute/unmute while shout is playing audio fails because the channel “has a media bug, hard mute not allowed”
  • FS-8252 [core] Fixed a crash in rtp stack on dtls pointer
  • FS-8283 [core] Handle RTP Contributing Source Identifiers (CSRC)
  • FS-8275 [core] Fix for broken DTMF
  • FS-8282 [core] Fix for sleep is not allowing interruption by uuid_transfer
  • FS-8315 [core] Fix for rtp_media_timeout not working
  • FS-8304 [core] Fix for choppy audio during calls
  • FS-8320 [core] Fixed broken ZRTP not responding to HELLO packet
  • FS-8338 [core] Fix for ringback not working correctly on stereo channels. Also fixed an issue when setting the ringback variable with an outbound call via the bridge app, if the inbound leg is stereo the ringback tone is still rendered as mono causing the resulting ringback to be higher pitched and incorrect.
  • FS-8366 [core] Fixed a segfault in rxfax
  • FS-8275 [core] Fixed an issue with broken RFC2833 DTMF
  • FS-8368 [core] Reduce logging for audio/video sync because some call lines were repeating too often for callers in a conference
  • FS-8372 [core] Fixed a no media bug caused by sofia sending the wrong response to RTP/SAVPF without DTLS
  • FS-8381 [core] Reset jitter buffer if period loss is too high
  • FS-8382 [core] Fixed a segfault with inbound-proxy-media enabled
  • FS-8397 [core] Fixed a race condition incrementing the event-sequence number
  • FS-8154 [core] Fixed a segmentation fault occurring while eavesdropping on video call
  • FS-8391 [core] Fixed a SDP parsing error for rtcp-fb
  • FS-8414 [core] Fixed ptime not updating on codec renegotiation causing audio issues between two legs of a call
  • FS-8417 [core] Fixed SIP offering a=sendonly sometimes replying with a=inactive
  • FS-8404 [core] Media engine will default to PCMU/PCMA if you don’t specify any codecs
  • FS-8411 [core] Replace ping_frame with video_ping_frame in a couple places that were missed causing issues like being unable to record just one side of a video call
  • FS-8425 [core] Fix for DTMF sometimes missed on PSTN call
  • FS-8240 [mod_local_stream] Fixed a/v getting out of sync when running in the background and added video profile parameter for recording 264 and made it default
  • FS-8287 [mod_local_stream] Fixed a segfault from refactor
  • FS-8216 [mod_av] Fixed a regression in hup_local_stream from last commit
  • FS-8274 [mod_av] Fixed a memory leak caused by images not being freed in video_thread_run
  • FS-8318 [mod_av] Fix for recording being out of sync when video from chrome has packet loss
  • FS-8392 [mod_av] Fixed rtpmap to allow both H263 and H263+ codecs to be offered
  • FS-8373 [mod_av] Fix for bad recording quality when using fast encoding
  • FS-8256 [mod_opus] More FMTP cleanup
  • FS-8284 [mod_opus] Use use-dtx setting from config in request to callee.
  • FS-8234 [mod_opus] Send correct (configured) fmtp ptime,minptime,maxptime when originating call
  • FS-8243 [mod_opus] Adding back the missing part removed in 8b088c2 so FEC works in most surroundings
  • FS-8295 [mod_opus] FMTP fixes to continue the cleanup of FEC
  • FS-8302 [mod_opus] Fix some printing/logging because switch_opus_show_audio_bandwidth() was not returning TRUE/FALSE as expected
  • FS-8130 FS-8305 [mod_opus] Fix some warnings and errors caused by dtx and/or jittery webrtc, refactor of last patch, and add suppression of scary harmless message about opus FEC
  • FS-8296 [mod_opus] Improve the way Opus is initialized when a call comes in
  • FS-8179 [mod_opus] Fixed a regression setting fec_decode breaking output on stereo calls
  • FS-8287 [mod_opus] Fixed a segfault from refactor
  • FS-8319 [mod_opus] Fixed and cleaned up switch_opus_has_fec() and switch_opus_info() to avoid FALSE positives for packets with FEC at high frame sizes.
  • FS-8344 [mod_opus] Toggle FEC ON only on the last frame which is to be packed
  • FS-7929 [mod_sofia] Fixed an issue when processing SIP messages while using camp-on
  • FS-6833 [mod_sofia] Add content-type header to ack with sdp
  • FS-6834 [mod_sofia] Found and fixed a few missing content-types in requests/responses with SDP that were outside the norm
  • FS-7834 [mod_sofia] Fixed MOH not working with inbound-bypass-media and resume-media-on-hold
  • FS-8115 [mod_sofia] Disabled unnecessary session timer re-invites for webrtc
  • FS-8536 [mod_sofia] Update to send Keyframe when getting SIP INFO with picture_fast_update
  • FS-7989 [fixbug.pl] Escape double quotes from summary and added more debugging data
  • FS-8246 [mod_json_cdr] Use seconds as default value for delay parameter
  • FS-8308 [mod_format_cdr] Fix to double encode if urlencoding json that is already encoded
  • FS-8311 [mod_voicemail] Fix for leave-message event not containing verbose data for a forwarded voicemail
  • FS-8306 [mod_amqp] If the exchange doesn’t exist, then create it, else fail. This resolves several error cases. And now command queues can specify the queue to subscribe to. This enables very interesting use cases that would involve single job queue, and multiple consumers.
  • FS-8335 [mod_easyroute] Fixed a small error check that results in error message not being displayed
  • FS-8378 [mod_esf] [core] Fixed a crash when using esf_page over loopback when transcoding and added tests for esf over loopback. Also refactor a bit to clarify code and get better debug in gdb
  • FS-8370 [mod_rayo] Fixed another place in where a message was freed after being queued for delivery. This resulted in a freed object being serialized, crashing FS
  • FS-8413 [mod_lua] Fixed a segfault calling session:getVariable(nil) in lua script.
  • FS-8537 [mod_lua] Fixed a segfault caused by passing nil to various lua functions
  • OPENZAP-240 [mod_freetdm] Fixed a failure to parse caused by using incorrect length when parsing AT responses
  • OPENZAP-238 [mod_freetdm] Several core and gsm improvements including fixing signaling status reporting, a small memory leak, fixing caller id and dnis on inbound calls, span stop functionality, and compilation errors in gcc
  • FS-8553 [configuration] Include verto_contact into the dial-string in the samples
  • FS-8363 [configuration] Don’t register gateways from directory because it registers over what appears to be ipv6 but doesn’t work correctly

The FreeSWITCH 1.4.26 release is here!

Releases -

The FreeSWITCH 1.4.26 release is here! This release contains everything since version 1.4.23. And this is a pretty big release and one of the final routine maintenance releases for the 1.4 branch  so upgrading now is a really good idea.

The FreeSWITCH 1.4 branch is reaching end of life and the FreeSWITCH Team highly recommends beginning your migration to the 1.6 branch.

This is a routine maintenance and security release and the resources are located here:

Security issues:

A bug allowing for a remotely exploited DoS attack through custom crafted network traffic via cJSON has been fixed. We classify this issue as High Severity. A patch was added by Anthony Minessale in commit 4bdca81 to resolve this issue. All versions from 1.4.4 through the previous release are vulnerable. We highly recommend updating to the current release version as soon as possible.
https://web.nvd.nist.gov/view/vuln/detail?vulnId=CVE-2015-7392

Improvements in build system, cross platform support, and packaging:

  • FS-8269 [mod_sms] Fixed a build issue
  • FS-8244 [mod_dptools] Fixed a compilation issue

The following bugs were squashed:

  • FS-8246 [mod_json_cdr] Use seconds as default value for delay parameter
  • FS-8282 [core] Fix for sleep is not allowing interruption by uuid_transfer
  • FS-8166 [core] Mute/unmute while shout is playing audio fails because the channel “has a media bug, hard mute not allowed”
  • FS-8338 [core] Fix for ringback not working correctly on stereo channels and an issue when setting the ringback variable with an outbound call via the bridge app, if the inbound leg is stereo the ringback tone is still rendered as mono causing the resulting ringback to be higher pitched and incorrect.
  • FS-8215 Fixed the accuracy of MacOSX nanosleep
  • FS-7673 [mod_v8] ODBC NULL value incorrectly evaluated
  • FS-8190 [mod_event_socket] When using nixevent, freeswitch stops sending us certain custom event that were NOT part of the nixevent command
  • stereo the ringback tone is still rendered as mono causing the resulting ringback to be higher pitched and incorrect.
  • FS-8354 [mod_conference] Reverted a back ported patch for rate change detection because it introduced a regression that caused an audio issue
  • FS-8335 [mod_easyroute] Fixed a small error check that results in error message not being displayed
  • FS-8370 [mod_rayo] Fixed another place in where a message was freed after being queued for delivery. This resulted in a freed object being serialized, crashing FS
  • FS-8378 [mod_esf] [core] Fixed a crash when using esf_page over loopback when transcoding and added tests for esf over loopback. Also refactor a bit to clarify code and get better debug in gdb
  • FS-8308 [mod_format_cdr] Fix to double encode if urlencoding json that is already encoded
  • FS-8413 [mod_lua] Fixed a segfault calling session:getVariable(nil) in lua script.

The FreeSWITCH 1.4.23 release is here!

Releases -

The FreeSWITCH 1.4.23 release is here! This is a routine maintenance release and the resources are located here:

New features that were added:

  • FS-7752 [mod_rayo] Increase maximum number of elements from 30 to 1024 to allow adhearsion to create large grammars to navigate IVR menus.

Improvements in build system, cross platform support, and packaging:

  • FS-8055 [build] Add confdir variable to freeswitch.pc

The following bugs were fixed:

  • FS-7135 [mod_sofia] Fixed response to re-invite with duplicate sdp (such as we get from session refresh) when soa is disabled to include an sdp. Fixed t.38 fax failure on session refresh
  • FS-7903 [proxy_media] Fix Codec PROXY Exists but not at the desired implementation. 0hz 0ms 1ch error when using proxy media.
  • FS-8056 [mod_voicemail] Fixed a segfault on vm_inject, regression from FS-7968
  • FS-7968 [mod_voicemail] Fixed verbose events
  • FS-8110 [mod_rayo] Prompt IQ error reply was being deleted after being sent for delivery. This is incorrect since message delivery thread will clean up the message
  • FS-8082 [mod_rayo] Do not remove items from hash while iterating
  • FS-8103 [mod_rayo] Handle where finishes unexpectedly before start event is received
  • FS-8143 [mod_rayo] Fixed a crash caused by client disconnecting from mod_rayo while a message is being delivered to that client. This is caused by the XMPP context’s JID -> XMPP stream mapping not being cleaned up on XMPP stream destruction.
  • FS-8127 [mod_conference] Fixed an audio issue caused by the codec not updating often enough when detecting rate change
  • FS-8099 [mod_lua] Restored LUA dialplan ACTIONS functionality
  • FS-8142 Fixed thread cache thread-safety and caching
  • FS-8185 [core] Allow xml preprocessor to expand variables where the resulting value is much longer than the original size
  • FS-8167 [mod_lua] Fixed a segfault caused by using api:execute or session:execute and not quoting the first argument like api:execute(log, “Second argument”) instead of api:execute(“log”, “Second argument”)
  • FS-8169 Fixed uuid_displace on stereo channels can lead to memory corruption causing a crash
  • FS-8114 Fixed opus and telephone event payload types colliding on REFER

The FreeSWITCH 1.6.2 release is here!

Releases -

The FreeSWITCH 1.6.2 release is here! This is the release of the Video/MCU branch!

Release files are located here:

And we’re dropping support in packaging for anything older than Debian 8.0 and anything older than Centos 7.0 due to a number of dependency issues on older platforms.

New features that were added:

  • FS-8094 [verto_communicator] Added googEchoCancellation, googNoiseSuppression, and googHighpassFilter settings to the UI enabled by default
  • FS-8107 [switch_ivr_menu] Adding CUSTOM esl events menu::enter and menu::exit when a call enter and exit an ivr menu
  • FS-6833 Allow Freeswitch to initiate late offer calls
  • FS-8075 [mod_hiredis] Add conf file to spec file too and updates for limit release case
  • FS-8053 Handle a=sendonly, a=sendrecv, a=recvonly to change who is sending video during a call

Improvements in build system, cross platform support, and packaging:

  • FS-7966 [Windows] Build now uses visual studio 2015 and builds successfully, but is still missing video functionality
  • FS-8124 [verto_communicator] Adding option –debug to grunt build, dist app will be generated without minified code.
  • FS-7168 [Debian] Update packages so that FS core libraries are setup as runtime dependencies
  • FS-7697 Chown the /etc/freeswitch/tls directory so that the freeswitch user can have read/write for TLS certificate generation
  • FS-7966 [Windows] Explore use of nuget for wix build dependency
  • FS-7967 [build] SmartOS compatibility
  • FS-8072 [Debian] Fixed a missed space

The following bugs were squashed:

  • FS-8099 [mod_lua] Restored LUA dialplan ACTIONS functionality
  • FS-7988 [filebug.pl] Moved -askall to -terse so old -askall behavior is default and old default is now -terse
  • FS-8029 [jitterbuffer] Fixed robotic sound when using jitterbuffer
  • FS-8103 [mod_rayo] Fixed handling of malformed requests
  • FS-8108 [mod_lua] Removed legacy mod_lua, the regular mod_lua works with system lua now
  • FS-8082 [mod_rayo] Fixed a segfault
  • FS-8110 [mod_rayo] Prompt IQ error reply was being deleted after being sent for delivery. This is incorrect since message delivery thread will clean up the message
  • FS-8116 [verto] Fixed device enumeration hanging on init
  • FS-8118 [verto] Fixed calls not properly rejecting video when video is offered but only audio is accepted
  • FS-7994 [verto_communicator] Using factory for group calls in history
  • FS-8117 [verto_communicator] Calling verto.iceServers upon useSTUN changing on ModalSettings, correctly check for STUN setting in localStorage, and fixed ignoring useSTUN settings
  • FS-8127 [mod_conference] Fixed an audio issue caused by the codec not updating often enough when detecting rate change
  • FS-8088 [verto_communicator] Fixed members list being destroyed after switching conferences and ending the first conference
  • FS-8130  Port video buffer to also support audio and remove original STFU jitter buffer, add some more resilience to video packet loss, add codec control mechanism for both call-specific debug and codec/call specific parameters, make opus function better in packet loss and latent situations, use new codec control prams to make jitter buffer lookahead FEC optionally enabled or disabled mid-call, and add a parameter to allow jitter buffer lookahead to be enabled.
  • FS-8131 [mod_voicemail] Fixed issues with allowing an empty password change and then locking out the user
  • FS-8136 [mod_h26x] Do not load passthru video codecs by default
  • FS-8140 [mod_sofia] Fixed a user_name typo in sofia_handle_sip_i_invite
  • FS-8142 Fixed  a thread cache thread-safety and caching
  • FS-8075 [mod_hiredis] Fix for failover when you pull power on redis, while redis clients under load test
  • FS-8144 [mod_opus] Readability and code formatting cleanup
  • FS-8126 [switch_core] Fixed the pruning of a media bug causing all media bugs on a session to be pruned
  • FS-8143 [mod_rayo] Fixed a crash caused by client disconnecting from mod_rayo while a message is being delivered to that client. This is caused by the XMPP context’s JID -> XMPP stream mapping not being cleaned up on XMPP stream destruction.
  • FS-8147 [mod_erlang_event] Fixed the process spawning segfault
  • FS-8149 [mod_xml_cdr] Fixed curl dependency in makefile
  • FS-1772 [mod_voicemail] Fixed the reset of voicemail greeting to default to allow entering 0 to restore the default greeting

The FreeSWITCH 1.6.0 release is here!

Releases -

The FreeSWITCH 1.6.0 release is here! This is the release of the Video/MCU branch!

Release files are located here:

And we're dropping support in packaging for anything older than Debian 8.0 and anything older than Centos 7.0 due to a number of dependency issues on older platforms.

New features that were added:

  • FS-7337 [mod_sofia] Add support for Remote-Party-ID header in UPDATE request.
  • FS-7561 [mod_sofia] Add Perfect Forward Secrecy (DHE PFS)
  • FS-7560 [mod_nibblebill] Added new options to nibble bill for minimum charges and rounding
  • FS-7587 FS-7602 FS-7499 [mod_verto] Add ipv6 support to Verto / Websockets and additional support ice/dtls ipv6 functionality
  • FS-6801 [mod_sofia] Add sip_watched_headers variable to launch events when a SIP message contains a given SIP header
  • FS-7564 [mod_rayo] Added new algorithms for offering calls to clients
  • FS-7436 FS-7601 [mod_opus] Added FEC support
  • FS-7603 [mod_event_socket] Failover for socket application in dialplan
  • FS-7585 [mod_rtmp] Increased AMF buffer for larger video and add bandwidth settings to flash video
  • FS-7311 [mod_sofia] Updating display name is disabled when caller_id equal "_undef_"
  • FS-7513 [mod_conference] Add video-auto-floor-msec param to control how long a member must have the audio floor before also taking the video floor and make sure user does not have auto avatar when not visible
  • FS-7620 [ftmod_libpri] Correctly set calling number presentation and screening fields
  • FS-7138 [mod_callcenter] Added a new reserve-agents param
  • FS-7436 FS-7601 [mod_opus] FEC support
  • FS-7623 [mod_amqp] Allow for custom exchange name and type for producers and fixed param name ordering bug caused by exposing these params
  • FS-7638 Allow ipv4 mapped ipv6 address to pass ipv4 ACLs properly
  • FS-7643 [mod_opus] Added interpretation of maxplaybackrate and sprop-maxcapturerate
  • FS-7641 Added video support to eavesdrop
  • FS-7656 [mod_localstream] Added mod_local_stream video support, and make mod_conference move the video in and out of a layer when the stream has video or not, scan for relative file in art/eg.wav.png and display it as video when playing audio files, put video banner up if artist or title is set, and fixed a/v sync on first connection
  • FS-7629[mod_conference] Added member status in json format to the conference live array, add livearray-json-status to conference-flags to enable
  • FS-7517 FS-7519 [mod_av] [mod_openh264] Added H264 STAP-A packeting support so it would work with FireFox
  • FS-7664 [mod_verto] Set ICE candidate timeout to wait for only 1 second to fix media delays
  • FS-7660[mod_opus] Enabled with new API command “opus_debug” to show information about Opus payload for debugging.
  • FS-7519 [mod_av] Fixed bitrate and added some presets
  • FS-7693 [mod_conference] Lower the default energy level in sample configs to improve voice quality
  • FS-7720 Improve play_and_detect_speech to set current_application_response channel variable as follows: "USAGE ERROR": bad application arguments', "GRAMMAR ERROR": speech recognizer failed to load grammar, "ASR INIT ERROR": speech recognizer failed to allocate a session, and "ERROR": any other errors
  • FS-7732 Continue recording with uuid_transfer
  • FS-7752 [mod_rayo] Increase maximum number of elements from 30 to 1024 to allow adhearsion to create large grammars to navigate IVR menus.
  • FS-7750 [mod_commands] Allow for uuid_setvar to handle arrays
  • FS-7758 [mod_loopback] Emit an event if a loopback bowout occurs
  • FS-7759 [mod_sofia] Added the channel variable ignore_completed_elsewhere to suppress setting the completed elsewhere cause
  • FS-7771 Set a channel variable if the recording is terminated due to silence hits
  • FS-7760 Added xml fetch for channels to externally support nightmare transfer depends on channel-xml-fetch-on-nightmare-transfer profile param (default is disabled)
  • FS-7730 [mod_smpp] Added mod_smpp as an event handler module
    and fixed the default configs to provided sample load option for mod_sms and mod_smpp
  • FS-7774 Add mod_kazoo
  • FS-7780 Add new channel variable max_session_transfers. If set, this variable is used to count the number of session transfers allowed instead of the max_forwards variable. If not set, the existing behavior is preserved.
  • FS-7783 Add channel variable for capturing DTMF input when using play_and_get_digits when the response does not match
  • FS-7772 [mod_opus] Add functionality to keep FEC enabled on the encoder by modifying the bitrate if packet loss changes (Opus codec specific behaviour).
  • FS-7799 [mod_png] Add API command uuid_write_png
  • FS-7801 [mod_opus] Added support to set CBR mode
  • FS-7685 [mod_say_nl] Fix Dutch numbers pronunciation
  • FS-7198 Add coma separated values and reverse ranges for time-of-day and day-of-week matches
  • FS-7809 [mod_opus] Added 60 ms ptime for Opus at 8 khz ( opus@8000h@60i )
  • FS-7405 [mod_dialplan_xml] Fix condition regex="all" to work with time conditions
  • FS-7819 [mod_opus] Restore bitrate (if there's no more packet loss) and added step for 60 ms
  • FS-7773 [mod_sofia] Adding additional transfer events when the fire-transfer-events=true profile parameter is set
  • FS-7820 FreeSWITCH automated unit test and micro benchmark framework
  • FS-7769 [mod_conference] Add new multi-canvas and telepresence features
  • FS-7847 [mod_conference] Add layers that do not match the aspect ration of conference by using the new hscale layer param for horizontal scale, and add zoom=true param to crop layer instead of letterbox, add grid-zoom layout group that demonstrates these layouts, and fix logo ratios and add borders too.
  • FS-7813 [mod_conference] Add vmute member flag.
  • FS-7846 [mod_dptools] Add eavesdrop_whisper_aleg=true and eavesdrop_whisper_bleg=true channel variables to allow you to start eavesdrop in whisper mode of specific call leg
  • FS-7760 [mod_sofia] Revise channel fetch on nightmare transfer and add dial-prefix and absolute-dial-string to the nightmare xml
  • FS-7829 [mod_opus] Add sprop-stereo fmtp param to specify if a sender is likely to send stereo or not so the receiver can safely downmix to mono to avoid wasting receiver resources
  • FS-7830 [mod_opus] Added use-dtx param in config file (enables DTX on the encoder, announces in fmtp)
  • FS-7824 [mod_png] Add functionality for capturing screenshots from both legs to uuid_write_png
  • FS-7549 [mod_ladspa] Added an API for removing an active ladspa effect on a channel. For conformance reasons, the uuid_ladspa command now accepts 'stop' and 'start', while the previous functionality (without any verb) which will simply add ladspa remains intact.
  • FS-7848 [mod_opus] Add support for 80 ms, 100 ms, 120 ms packetization
  • FS-7519 FS-7677 [mod_av] Add H.263 codec support
  • FS-7885 Add getcputime to retrieve FreeSWITCH process CPU usage
  • FS-7889 [mod_conference] Move conference chat to use an event channel so messages only go to the right 'room' for the conference and move conference chat functionality to use event_channel.
  • FS-7900 [mod_png] Allow snapshot of single legged calls
  • FS-7912 [mod_lua] Added session UUID to lua error logs, if known and added session UUID to embedded language (lua, javascript, etc) logs when session sanity check fails
  • FS-7760 [mod_sofia] Improved the xml fetch lookup for channels on nightmare transfer
  • FS-7922 [mod_commands] Added uuid_redirect API command. This provides the equivalent functionality of the dptools "redirect" application as an api command.
  • FS-7806 FS-7803 [mod_amqp] Added new properties to amqp configuration, fixed the usage for enable_fallback_format_fields, and added amqp_util_encode to fix a routing key issue
  • FS-7972 [verto communicator] Creating Verto Communicator
  • FS-7988 Add a perl script to help file bugs from the command line and add fixbug.pl to tree
  • FS-8009 [verto communicator] Create a grunt project with livereload support. Documentation can be found here.
  • FS-8010 [verto communicator] Add options for googAutoGainControl, googNoiseSuppression, and googHighpassFilter
  • FS-7855 [verto communicator] Pass userVariables back to the live array to allow for displaying the Gravatar associated with a member's email address
  • FS-8075 [mod_hiredis] Add mod_hiredis including support for redis limits and added support for raw redis commands. Added deprecation notices to mod_redis
  • FS-8049 [mod_commands] Add getenv FSAPI
  • FS-8036 [verto.js] Add chatCallback to \$.verto.conf

Improvements in build system, cross platform support, and packaging:

  • FS-7610 Fixed a gcc5 compilation issue
  • FS-7499 Fixed a build error on 32bit platforms
  • FS-7570 Fixed a compilation issue w/ zrtp enabled
  • FS-7426 Only disable mod_amqp on Debian Squeeze and Wheezy
  • FS-7635 Removed msvc 2005, 2008, and 2010 non working build systems
  • FS-7373 Expose the custom repo and key path to the build-all command too
  • FS-7648 Foundation for QA testing config , adding leave/check videomail test cases, adding videomail voicemail profile, adding video record/ playback test cases, adding set video on hold, force pre-answer prefix, and adding an eavesdrop test case.
  • FS-7338 Removed mod_shout dep libs to system libs to continue cleaning up the libs for the 1.6 build process and added Debian packaging for several new modules, as well as handle system lib change for a handful of modules
  • FS-7653 Sample build system for a stand alone(out of tree) FreeSWITCH module
  • FS-7601 [mod_opus] [mod_silk] Removed a bounds check that can never be true in opus fec code and modify jitterbuffer usage to match the api change
  • FS-7648 More work toward setting up a QA testing configuration, add condition testing for regex all and xor cases, adding profile-variable for testing cases , add lipsync tests for playback and local stream, add stereo, and configuration for mcu test
  • FS-7338 Fixed bug in Debian packaging when trying to build against custom repo
  • FS-7609 Enable building of mod_sangoma_codec for Debian Wheezy/Jessie
  • FS-7667 [mod_java] Fixed include directory detection when using Debian java packages and use detected directory
  • FS-7655 Make libvpx and libyuv optional (none of the video features will work without them) The following modules require these libraries to be installed still: mod_av mod_cv mod_fsv mod_mp4v2 mod_openh264 mod_vpx mod_imagick mod_vpx mod_yuv mod_png mod_vlc, fix build issue w/ strict prototypes, and fix a few functions that need to be disabled without YUV
  • FS-7605 Fixed default configuration directory in Debian packages and fixed Debian packaging dependencies on libyuv and libvpx
  • FS-7669 When installing from Debian packaging if you don't have the /etc/freeswitch directory, we will install the default packages for you. If you already have this directory, we'll let you deal with your own configs.
  • FS-7297 [mod_com_g729] Updated the make target installer
  • FS-7644 Added a working windows build without video support for msvc 2013
  • FS-7666 [mod_managed] Fixed error building mod_managed on non windows platforms
  • FS-7707 Fix build error on CentOS7
  • FS-7655 Fixed a build error when we have PNG but not YUV
  • FS-7723 Change RPMs to use -ncwait instead of -nc. This will cause the initscript to pause and wait for FS to be ready before continuing.
  • FS-7648 Added a test cases for FS-7724 and FS-7687
  • FS-7726 Additional configurations for a QA test case
  • FS-7715 Updates to configure and spec files for next development branch and added images to spec file and fixed build/freeswitch.init.redhat since redhat likes to override settings in the script with TAGs in comments
  • OPENZAP-238 [freetdm] Fix some GSM compilation errors and do a bit of code cleanup
  • OPENZAP-237 [freetdm] Use __func__ instead of __FUNCTION__ to comply with c99 in gcc 5.1
  • FS-7628 [mod_erlang_event] Removed unused variables causing a compilation error
  • FS-7776 Add mod_kazoo to packaging
  • FS-7845 [mod_conference] Break up mod_conference into multiple source files to improve build performance
  • FS-7769 [mod_conference] Fixed a build issue
  • FS-7820 Fix build system typo. Don't assign the same variable twice.
  • FS-7043 Fixed apr1 unresolved symbols in libfreeswitch.so.1.0.0
  • FS-7130 Make /run/freeswitch persistent in the Debian packages, so it will start under systemd
  • FS-7860 Prevent a switch_rtp header conflict
  • FS-7130 Make /run/freeswitch persistent, so it will start under systemd
  • FS-7728 Fixed Windows build issues minus video features
  • FS-7965 [mod_conference] Fixed an error thrown when compiling with GCC
  • FS-7985 [mod_voicemail] Fixed a compilation error on 32-bit PCC platform
  • FS-8015 [mod_conference] Add project dir to include for mod_conference so it picks up mod_conference.h for Windows
  • FS-8061 [verto_communicator] Adding license to package.json
  • FS-8047 [build] Fixed build error in mod_basic, mod_rtmp, mod_oreka, and mod_sangoma_codec due to using __FUNCTION__ on newer compilers
  • FS-8054 [mod_rayo] Fixed a warning when building on Debian
  • FS-8055 [build] Add confdir variable to freeswitch.pc
  • FS-7966 [windows] Working msvc 2015 build support. Does not yet include video features.
  • FS-8019 [debian] Excluded few modules that fail to compile from debian/bootstrap.sh, fixed the handling of -T and -t and added debian/apt_sources.list with necessary dependencies to build master, and updated debian/README.source
  • FS-8058 [mod_vpx] Build correctly against libvpx that is not installed into default system library locations

The following bugs were squashed:

  • FS-7579 [mod_conference] Fixed a bug not allowing suppression of play-file-done
  • FS-7462 [mod_opus] Fix FMTP in the INVITE to use values from opus.conf.xml
  • FS-7593 [mod_skinny] Fixed a bug where skinny phones would stomp on each other in database when thundering herd occurs
  • FS-7597 [mod_codec2] Fixed encoded_data_len for MODE 2400, it should be 6 bytes. Also replaced 2550 bps bitrate (obsoleted operation mode) by 2400
  • FS-7604 [fs_cli] Fixed fs_cli tab completion concurrency issues on newer libedit
  • FS-7258 FS-7571 [mod_xml_cdr] Properly encode xml cdr for post to web server
  • FS-7584 More work on rtcp-mux interop issue with chrome canary causing video transport failure
  • FS-7586 [mod_conference] Change the default min-required-recording-participants option for mod_conference from 2 to 1 and silence the warning when the value is set to 1 in the configs
  • FS-7607 Update URLs to reflect https protocol on freeswitch.org websites and update additional URLs to avoid 301 redirects.
  • FS-7479 Fixed a crash caused by large RTP/PCMA packets and resampling
  • FS-7524 [mod_callcenter] Fixing tiers, level and position should default to 1 instead of 0
  • FS-7613 Fixed a crash in core text rendering
  • FS-7612 Fixed invalid json format for callflow key
  • FS-7609 [mod_sangoma_codec] Now that libsngtc-dev and libsngtc are in the FS debian repo, enable mod_sangoma_codec
  • FS-7621 [mod_shout] Fixed a slow interrupt
  • FS-7432 Fixed missing a=setup parameter from answering SDP
  • FS-7622 [mod_amqp] Make sure to close the connections on destroy. Currently the connection is malloc'd from the module pool, so there is nothing to destroy.
  • FS-7586 [mod_vlc] A fix for failing to encode audio during the recording of video calls
  • FS-7573 Fixed 80bit tag support for zrtp
  • FS-7636 Fixed an issue with transfer_after_bridge and park_after_bridge pre-empting transfers
  • FS-7654 Fixed an issue with eavesdrop audio not working correctly with a mixture of mono and stereo
  • FS-7641 Fixed a segfault in eavesdrop video support
  • FS-7649 [mod_verto] Fixed issue with h264 codec not being configured in verto.conf.xml
  • FS-7657 [mod_verto] Fixed a bug with TURN not being used. Note, you can pass an array of stun servers, including TURN, to the verto when you start it up. (see verto.js where iceServers is passed)
  • FS-7665 [mod_conference] Fixed a bug with the video floor settings not giving the video floor to the speaker
  • FS-7650 [mod_verto] Fixed crash when making a call from a verto user with profile-variables in their user profile
  • FS-7710 [mod_conference] Added the ability to set bandwidth to "auto" for conference config
  • FS-7432 Fixed dtls/srtp, use correct a=setup parameter on recovering channels
  • FS-7678 Fixed for fail_on_single_reject not working with | bridge
  • FS-7709 [mod_verto] Verto compatibility fixes for Firefox
  • FS-7689 [mod_lua] Fixed a bug with lua not loading directory configurations
  • FS-7694 [mod_av] Fixed for leaking file handles when the file is closed.
  • FS-7467 [mod_callcenter] Fixing stuck channels using uuid-standby agents
  • FS-7699 [mod_verto] Fixed for browser compatibility
  • FS-7722 Fixed an issue with record_session including params when creating path
  • FS-7489 [mod_unimrcp] Fixed a TTS Audio Queue Overflow
  • FS-7724 [mod_conference] Fixed a segfault when missing fonts when trying to render banner
  • FS-7519 [mod_av] Fixed a regression in the visual appearance of decode app output
  • FS-7703 Fixed a bug caused by answer_delay being set in the default configurations
  • FS-7679 [mod_verto] Fixed a bug causing one way audio on Chrome when video is enabled and when using a sip without video
  • FS-7729 [mod_verto] Fixed the formatting for IPv6 addresses
  • FS-7734 [mod_nibblebill] Fixed a deadlock
  • FS-7726 Fixed a bug with recording a video session on DTMF command
  • FS-7721 Fixed a segfault caused when using session:recordFile() and session:unsetInputCallback in a lua script
  • FS-7429 [mod_curl] Fixed to output valid json
  • FS-7746 [mod_verto] Fixed a device permission error in verto client
  • FS-7753 [mod_local_stream] Fixed some glitching and freezing video when using hold/unhold
  • FS-7761 [core] Fix shutdown races running api commands during shutdown
  • FS-7767 [mod_sofia] Fixed a segfault caused by invalid arguments to sip_dig
  • FS-7744 [mod_conference] Fixed a bug causing the first user's video stream to stop when another verto user calls the conference
  • FS-7486 [mod_sofia] Fixed the handling of queued requests
  • FS-7775 [mod_conference] Fix threading issue causing stuck worker threads
  • FS-7777 [mod_imagick] Fixed a regression causing a segfault when playing png & pdf in conference
  • FS-7778 [mod_sofia] Fixed a bug causing a SQL statement to fail because of a double quote instead of a single quote
  • FS-7754 [freetdm] Fixed a bug relating to single digit dial-regex with analog devices
  • FS-7785 [mod_opus] Fix for invalid ptime 30 ms for opus@8000h . Replaced 30 ms with 40 ms.
  • FS-7762 [mod_av] Handle buffer allocation failures of large buffers
  • FS-7849 [verto] Remove extra div breaking full screen in html
  • FS-7832 [mod_opus] Fixes when comparing local and remote fmtp params
  • FS-7731 [mod_xml_cdr] Fixed a curl default connection timeout
  • FS-7844 Fix packet loss fraction when calculating loss average
  • FS-7789 [mod_av] Fixed issue with audio dropping out partway through recordings
  • FS-7854 Add task_runtime to tasks table in core database
  • FS-7856 [mod_av] Fix some segfaults and leaks.
  • FS-7866 Fixed a crash when running incorrect var api expansion syntax "eval \${\${external_sip_ip}:4}"
  • FS-7861 FS-7862 [mod_conference] Fixed a crash and other issues caused by multi canvas feature
  • FS-7681 [mod_conference] Factor out conference->canvas and allow per canvas record and play
  • FS-7869 [mod_conference] Fixed a deadlock on shutdown after playing video file that will not display video
  • FS-7654 Fixed regressions on eavesdropping on channels playing a file and on channels with unlike rates
  • FS-7872 [mod_verto] Gracefully fail attempting to transfer 1 legged call
  • FS-7874 [mod_conference] Fixed incorrect layout group count
  • FS-7870 [mod_conference] Allow jsonapi commands to pass the string id field to pass special ID's like "last"
  • FS-7882 [mod_conference] Allow JSON API commands to send third arg for muting
  • FS-7888 [mod_verto] Fixed namespacing problems in javascript library masked by global verto object
  • FS-7811 Use more common format CIF for blank image
  • FS-7902 [mod_local_stream] Fix for queue filling up when you have a mix of video and non video files
  • FS-7891 [mod_spandsp] Allow spandsp dtmf detector to work on rates other than 8k
  • FS-7839 Correct firefox > 38 DTLS behavior to match new EC requirements
  • FS-7769 [mod_conference] Fixed vmute on personal canvas and fixed changing layouts on personal canvas
  • FS-7893 [mod_conference] Fixed a bug causing muxing write thread to occasionally not close on shutdown
  • FS-7904 Fixed alpha image patching
  • FS-7906 [mod_av] Correct crash from multi-threaded opening or closing of multiple files at the same time
  • FS-7913 [mod_conference] Fixed miscast variable
  • FS-7918 [mod_kazoo] Small fixes in mod_kazoo
  • FS-7917 [mod_sofia] Fixed default config, we really shouldn't be setting ext-*-ip settings for ipv6 profiles
  • FS-7908 FS-7092 Fixed the generated sdp including telephone-event for the rates of video codecs (90000) when it should only be audio codec rates
  • FS-7927 Fixed a typo in variable name: eavesdrop_annnounce_macro
  • FS-7940 [mod_conference] Fixed an issue where the video image does not appear on the new canvas when switching
  • FS-7930 [mod_conference] Correct termination of conference when the last member with endconf left.
  • FS-7953 [verto communicator] Fixed dialing when typing extension using the keyboard.
  • FS-7958 [mod_conference] Fixed a race condition causing crash in conference video MCU
  • FS-7951 [mod_rayo] Completely clean up mod_rayo if it fails to load
  • FS-7955 [mod_sofia] Fixed a crash caused by invalid contact when using event to send a notify message
  • FS-7970 Fixed crash in video_bug_thread caused by double free
  • FS-7971 [mod_opus] Fixed a rate mismatch and correctly advertise telephone-event and CN rates based on the advertised rates of offered codecs
  • FS-7960 Fixed check_ice routine in switch_core_media.c to not use dropped candidates
  • FS-7975 [mod_voicemail] Fix record-greeting event missing VM-Greeting-Path
  • FS-7969 Fixed a segfault due to pthread_setschedparam() on a thread that has exited
  • FS-7962 Fixed sporadic invite/replaces failure
  • FS-8004 Send keyframe on receiving nack with multiple consecutive packets
  • FS-8005 [mod_opus] Fix for rare decoder error when doing PLC, OPUS_GET_LAST_PACKET_DURATION might return 0
  • FS-8006 Changed the typedef of switch_core_video_thread_callback_func_t for consistency
  • FS-7932 [mod_verto] Removed the param from the getMute function in verto class, not needed on underlying method
  • FS-8008 [mod_verto] Separate verto default config to have sep v4 and v6 listeners
  • FS-8016 [mod_conference] Reduce buffering of video in conference mux
  • FS-7977 [verto communicator] Fixing default resolution and cleaning code
  • FS-7992 [verto communicator] Fixed device list at settings
  • FS-8017 [verto communicator] Fixed uses of serialized verto in local storage
  • FS-7986 [verto communicator] Fix for devices not refreshing if system config changes
  • FS-7998 [verto communicator] Don't prompt when recovering call, just do it.
  • FS-8003 [verto communicator] Use audioInDevices instead of audioDevices to match verto plugin
  • FS-8027 [verto communicator] Added watchTask flag to browserSync and add proper regex for replacements
  • FS-8026 [verto_communicator] Added an auto-focus directive to both dial-pad and login so that enter will just work. On dial-pad useful to provide keyboard only input without the need to using the mouse
  • FS-7995 [verto_communicator] Upon call recovery, emit an event on \$rootScope so that controllers are able to properly clear states.
  • FS-7945 [verto communicator] Use angular-prompt to ask the user to enter a text for the banner. If cancelled, nothing is done.
  • FS-8045 [verto communicator] Make the folder structure compliant with AngularJS best practices and adjust build system.
  • FS-7957 [verto_communicator] Make console less chatty by commenting liveArray updates and get initial state of the conference on liveArray boot event.
  • FS-7979 [verto_communicator] Prompt for extension before transferring a conference member
  • FS-8001 [verto_communicator] For this to work, passing in the parameter was missing
  • FS-7979 [verto_communicator] Removed extra console.log and commented line
  • FS-8025 [verto_communicator] Restored the blue background on the video controls and making icons white again, looking better.
  • FS-8062 [verto_communicator] Fixed video controls tool-tips, now they are visible
  • FS-8048 [verto_communicator] Fixed infinite reconnect after changing hostname and websocket url
  • FS-8066 [verto communicator] Added encoded avatar url to userVariables so that mod_conference can use it when no video, or video mute
  • FS-8018 [verto_communicator] Separation of concerns. Get storage service to manage all settings instead of vertoService
  • FS-8043 [verto_communicator] Removed unnecessary calls to localStorage
  • FS-8040 [verto_communicator] Check if we have a valid resolution reported before calling camera routines and hide controls if none are found
  • FS-8092 [verto_communicator] If there is no data in localStorage, select best resolution for selected camera
  • FS-7840 [verto_communicator] Use chatChannel to send and receive messages from conferences
  • FS-8088 [verto_communicator] Call conference destroy upon hangup and on event destroy to properly unsubscribe from events
  • FS-8046 [verto] Fixed for library not passing device preferences to dialog properly
  • FS-8053 [verto] Don't receive video on screen share
  • FS-8059 [verto] Fixed typo when transferring party from conference
  • FS-8060 [verto] Conditionally set video tag src to null for FF and empty string for others
  • FS-8087 [verto] Fixed issue in camera selection on recovery, refactor to use localStorage, change res detection, reload on network change, pass resCheck callback into deviceParams, always make one basic call to getusermedia to ensure perms are ok, pass valid res to callback, make \$.FSRTC.validRes available globally, sync minified, fix depth issue in cached json, test for valid cache before setting vars
  • FS-8028 [mod_shout] Fixed random sockets being closed regression from FS-7621
  • FS-8029 [jitterbuffer] Fixed robotic sound when using jitterbuffer when buffer timestamps get behind that of the packet timestamps, such as when the source clock is out of sync with our clock
  • FS-8056 [mod_voicemail] Fixed a segfault on vm_inject, regression from FS-7968
  • FS-7968 [mod_voicemail] Fixed verbose events
  • FS-7942 [udptl] Fixed rare segfault on t.38 fax FS-8014 is a duplicate of this issue
  • FS-8031 [dtls] Fixed delayed DTLS media due to changing ICE candidates
  • FS-7903 [proxy_media] Fix Codec PROXY Exists but not at the desired implementation. 0hz 0ms 1ch error when using proxy media.
  • FS-7989 [fixbug.pl] Add –author option
  • FS-8037 [mod_sofia] Fixed so zrtp-passthru doesn't activate unless the zrtp-hash is in the SDP
  • FS-7135 [mod_sofia] Fixed response to re-invite with duplicate sdp (such as we get from session refresh) when soa is disabled to include an sdp. Fixed t.38 fax failure on session refresh
  • FS-8050 [mod_av] Fixed a crash when streaming rtmp of desktop share
  • FS-7640 [rtp] Fixed some comparisons in switch_rtp.c to be wraparound proof
  • FS-8057 [core] Fixed a segfault when doing video call when built against libyuv but not libvpx
  • FS-8069 [stun] Fixed ipv6 support missing in stun code
  • FS-8071 [rtp] remove unnecessary auto adjust remote address rules when in ICE mode
  • FS-8077 [mod_conference] Fix memory leak in record
  • FS-8091 [core] Added some missing message names to session message name list (would have caused missing information in some log messages)
  • FS-8093 [mod_silk] Remove giant stack allocation in switch_silk_decode