The World's First Cross-Platform Scalable FREE Multi-Protocol Soft Switch

FreeSWITCH is a scalable open source cross-platform telephony platform designed to route and interconnect popular communication protocols using audio, video, text or any other form of media.  It was created in 2006 to fill the void left by proprietary commercial solutions.  FreeSWITCH also provides a stable telephony platform on which many applications can be developed using a wide range of free tools. FreeSWITCH was originally designed and implemented by Anthony Minessale II with the help of Brian West and Michael Jerris.  All 3 are former developers of the popular Asterisk open source PBX.  The project was initiated to focus on several design goals including modularity, cross-platform support, scalability and stability.  Today, many more developers and users contribute to the project on a daily basis. We support various communication technologies such as SIP, H.323 and WebRTC making it easy to interface with other open source PBX systems such as sipXecs, Call Weaver, Bayonne, YATE or Asterisk. FreeSWITCH can perform full video transcoding and MCU functionality using its conferencing module. FreeSWITCH supports many advanced SIP features such as presence/BLF/SLA as well as TCP TLS and sRTP.  It also can be used as a transparent proxy with and without media in the path to act as a SBC (session border controller) and proxy T.38 and other end to end protocols. FreeSWITCH supports both wide and narrow band codecs making it an ideal solution to bridge legacy devices to the future.  The voice channels and the conference bridge module all can operate at 8, 12, 16, 24, 32 or 48 kilohertz in mono or stereo and can bridge channels of different rates.  The G.729 codec is also available under a commercial license. FreeSWITCH builds natively and runs standalone on several operating systems including Windows, Mac OS X, Linux, BSD and Solaris on both 32 and 64 bit platforms. FreeSWITCH supports FAX, both over audio and T.38, and can gateway between the two. Our developers are heavily involved in open source and have donated code and other resources to other telephony projects including openSER, sipXecs, The Asterisk Open Source PBX and Call Weaver. a Spec Sheet is available on our Wiki.


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FreeSWITCH 1.6.12 released!

October 11 Releases no comments

The FreeSWITCH 1.6.12 release is here!

This is also a routine maintenance release. Change Log and source tarball information below.

Release files are located here:

New features that were added:

  • FS-9242 [mod_verto] Convert to adapter.js
  • FS-8955 [verto_communicator] Adding DTMF shortcuts and handling DTMF history on DTMF widget
  • FS-9601 [mod_opus] Make adjustable bitrate mutually exclusive with FEC enforcing on the decreasing trend, add step calculation for bitrate adjustment, fix bug on context settings
  • FS-8644 [mod_opus] OPUS_SET_BITRATE(), codec control and estimators for packet loss and RTT (with Kalman filters) to detect a slow or congested link. Feature enabled with “adjust-bitrate” in opus.conf.xml – it’s a feedback loop with incoming RTCP.

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

  • FS-8623 [build] Fixed sun studio build errors building libvpx
  • FS-9553 [core] Refactor video-on-hold
  • FS-9616 [libvpx] Update libvpx to latest upstream
  • FS-9618 [libyuv] Update libyuv to latest upstream

The following bugs were squashed:

  • FS-9574 [mod_verto] We shouldn’t print data set on the buffer because of the potential security issues
  • FS-9508 [verto_communicator] Adding AGC option on settings, enabled by default
  • FS-7876 [verto_communicator] Adding hold button for video calls
  • FS-9242 [verto.js] Fixed screen share for chrome to work in VC with additional camera
  • FS-9586 [mod_local_stream] Fixed the local_stream video queue sticking when not being read from
  • FS-9610 [core] Video keyframe requests not being propagated properly
  • FS-9612 [core] RTCP-MUX wrongly enabled in cases where answer contains RTCP but offer didn’t / remote
    address not obtained in UDPTL mode
  • FS-9580 [core] Add auto adjust for RTCP separate from RTP for sync NAT
  • FS-9548 [core] Crash on Invite due to bad config for sip profile
  • FS-9498 [mod_conference] Fixed a regression with 100% cpu

FreeSWITCH Week in Review (Master Branch) September 24th – October 1st

October 4 Week In Review no comments

This week mod_opus got a cool new feature that allows the detection of a slow or contaminated link. And verto is converting to adapter.js as well as getting DTMF shortcuts. The FreeSWITCH configuration audit is ongoing with initial minor commits and will continue throughout the year. If you are looking to volunteer to help with that or would like more information email brian@freeswitch.org or join the Bug Hunt on Tuesdays at 12:00pm Central Time.

Join us Wednesdays at 12:00 CT for some more FreeSWITCH fun! And, head over to freeswitch.com to learn more about FreeSWITCH support.

New features that were added:

  • FS-8644 [mod_opus] OPUS_SET_BITRATE(), codec control and estimators for packet loss and RTT (with Kalman filters) to detect a slow or congested link. Feature enabled with “adjust-bitrate” in opus.conf.xml – it’s a feedback loop with incoming RTCP.
  • FS-9242  [mod_verto] Convert to adapter.js
  • FS-8955  [verto_communicator] Adding DTMF shortcuts and handling DTMF history on DTMF widget

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

  • FS-8623  [build] Fixed sun studio build errors building libvpx
  • FS-9553  [core] Refactor video-on-hold

The following bugs were squashed:

  • FS-9548  [core] Crash on Invite due to bad config for sip profile
  • FS-9580  [core] Add auto adjust for RTCP separate from RTP for sync NAT
  • FS-9574  [mod_verto] We shouldn’t print data sent on the buffer because of the potential security issues
  • FS-9508  [verto_communicator] Adding AGC option on settings, enabled by default
  • FS-7876  [verto_communicator] Adding hold button for video calls
  • FS-9242  [verto.js] Fixed screen share for chrome to work in VC with additional camera
  • FS-9586  [mod_local_stream] Fixed the local_stream video queue sticking when not being read from

FreeSWITCH Week in Review (Master Branch) September 17th – September 24th

September 30 Week In Review no comments

This week mod_conference and mod_verto saw the most action with added sounds and user variables, whitelisting, and syncing outbound calls with the user directory respectively. The FreeSWITCH configuration audit is ongoing with initial minor commits and will continue throughout the year. If you are looking to volunteer to help with that or would like more information email brian@freeswitch.org or join the Bug Hunt on Tuesdays at 12:00pm Central Time.

Join us Wednesdays at 12:00 CT for some more FreeSWITCH fun! And, head over to freeswitch.com to learn more about FreeSWITCH support.

New features that were added:

  • FS-9533 [mod_conference] Add member-enter-sound
  • FS-9526 [mod_conference] Add deaf sounds
  • FS-9543 [core] Add pre-exec state change hooks to core
  • FS-9549 [mod_verto] Add userVariables to DMTF and INFO messages
  • FS-9524 [mod_verto] Enable whitelisting of Verto connections by IP using FS ACL
  • FS-9550 [mod_verto] Set user on outbound verto calls to sync with user directory
  • FS-9551 [switch_ivr – json cdr] Adding app-stamp to app_log

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

  • FS-9527 [mod_avmd] Fix MAP_POPULATE on FreeBSD

The following bugs were squashed:

  • FS-9538 [core] Fixed a segfault while reading local ringback file
  • FS-9548 [core] Return with an error on wrong RTP IP given from configuration
  • FS-9536 [core] Fixed the return value
  • FS-9552 [mod_conference] Added ‘deaf’ to the json status per member
  • FS-9435 [mod_av] Fixed an issue with receiving PLI requests once per second
  • FS-9525 [mod_av] Fixed an issue with client initiated RE-INVITE with a different audio codec calling into conference causing choppy audio
  • FS-9557 [rtp] Eating AV in proxy media mode causing a few seconds of black screen
  • FS-9498 [mod_conference] Fixed a regression with 100% cpu

The FreeSWITCH 1.6.11 release is here!

September 28 Releases no comments

The FreeSWITCH 1.6.11 release is here!

This is also a routine maintenance release. Change Log and source tarball information below.

Release files are located here:

New features that were added:

  • FS-9138 [mod_avmd] Add config to vanilla folder
  • FS-9480 [mod_kazoo] Add api enhancements
  • FS-9457 [mod_http_cache] Allow GET and PUT from Azure Blob Service
  • FS-9551 [switch_ivr – json cdr] Adding app-stamp to app_log
  • FS-9487 [core] Add CBR param to video file recording params
  • FS-9543 [core] Add pre-exec state change hooks to core
  • FS-9472 [core] Add originate_retry_timeout, a maximum duration in seconds to allow retry
    attempts to occur, and originate_retry_min_period_ms, a minimum spacing between retry attempts
  • FS-9495 [mod_conference] Add conference_join_energy_level variable
  • FS-9533 [mod_conference] Add member-enter-sound
  • FS-9526 [mod_conference] Add deaf sounds
  • FS-9474 [mod_conference] Add variables to set initial volume on mod_conference
  • FS-9498 [mod_conference] Fixed a regression with 100% cpu
  • FS-9549 [mod_verto] Add userVariables to DMTF and INFO messages
  • FS-9524 [mod_verto] Enable whitelisting of Verto connections by IP using FS ACL
  • FS-9550 [mod_verto] Set user on outbound verto calls to sync with user directory
  • FS-9242 [mod_verto] Convert to adapter.js
  • FS-8955 [verto_communicator] Adding DTMF shortcuts and handling DTMF history on DTMF widget
  • FS-9469 [verto.js] Added onGrant/onDenied callbacks in onStream/onError of RTCCallbacks so that the client code gets notified of a webrtc permission change immediately
  • FS-8644 [mod_opus] OPUS_SET_BITRATE(), codec control and estimators for packet loss and
    RTT (with Kalman filters) to detect a slow or congested link. Feature enabled with “adjust-bitrate” in opus.conf.xml – it’s a feedback loop with incoming RTCP.

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

  • FS-9466 [build] Use system MD5 if available
  • FS-8623 [build] Fixed sun studio build errors building libvpx
  • FS-9553 [core] Refactor video-on-hold
  • FS-9465 [ubuntu packages] Add xenial instrumentation to Debian/utils.sh script
  • FS-9471 [verto_communicator] Updating In Call display after receiving display update mes\
    sage from mod_verto.
  • FS-9475 [mod_verto] Video bandwidth not conveyed in SDP for verto
  • FS-9551 [mod_sofia] Compare session before setting TFLAG_SKIP_EARLY
  • FS-9488 [mod_http_cache] Fixed a compile error
  • FS-9498 [mod_conference] Try to make video writing thread more efficient
  • FS-9527 [mod_avmd] Fix MAP_POPULATE on FreeBSD
  • FS-9616 [libvpx] Update to the latest upstream version

The following bugs were squashed:

  • FS-9538 [core] Fixed a segfault while reading local ringback file
  • FS-9548 [core] Return with an error on wrong RTP IP given from configuration and fixed a crash on Invite due to bad configuration for sip profile
  • FS-9536 [core] Fixed an issue with the return value
  • FS-9580 [core] Add auto adjust for RTCP separate from RTP for sync NAT
  • FS-9482 [core] Fixed a segfault on the second attempt to use uuid_media_3p
  • FS-9484 [core] Fixed a variable type format spec
  • FS-6954 [core] Use channel flags to check for proxy media or bypass media
  • FS-9483 [mod_conference] Fixed a missing keyframe after re-invite
  • FS-9493 [mod_conference] Fixed a possible crash when changing from normal to personal canvas on the fly
  • FS-9494 [mod_conference] Fixed issues with video avatar switching when video starts/stops
  • FS-9552 [mod_conference] Added ‘deaf’ to the json status per member
  • FS-9435 [mod_av] Fixed an issue with receiving PLI requests once per second
  • FS-9525 [mod_av] Fixed an issue with client initiated RE-INVITE with a different audio codec calling into conference causing choppy audio
  • FS-9497 [mod_av] Fixed an AV sync record issue
  • FS-9557 [rtp] Eating AV in proxy media mode causing a few seconds of black screen
  • FS-9574 [mod_verto] We shouldn’t print data sent on the buffer because of the potential security issues
  • FS-9475 [mod_verto] Video bandwidth not conveyed in SDP for verto
  • FS-9471 [verto_communicator] Updating In Call display after receiving display update message from mod_verto.
  • FS-9346 [verto_communicator] Add DTMF icon while on a video call, fixing conferences with pin number
  • FS-9508 [verto_communicator] Adding AGC option on settings, enabled by default
  • FS-7876 [verto_communicator] Adding hold button for video calls
  • FS-9242 [verto.js] Fixed screen share for chrome to work in VC with additional camera
  • FS-9586 [mod_local_stream] Fixed the local_stream video queue sticking when not being read from
  • FS-9486 [mod_sofia] Fixed an issue with uuid_drop_dtmf switching between tone replace and digit
  • FS-9458 [mod_avmd] Set channel variable before BEEP event is fired

FreeSWITCH Week in Review (Master Branch) September 3rd – September 10th

September 12 Week In Review no comments

Mod_kazoo had some API enhancements, mod_http_cache has GET and PUT from Azure Blob services, and mod_conference added a variable called conference_join_energy_level. The FreeSWITCH configuration audit is ongoing with initial minor commits and will continue throughout the year. If you are looking to volunteer to help with that or would like more information email brian@freeswitch.org or join the Bug Hunt on Tuesdays at 12:00pm Central Time.

Join us Wednesdays at 12:00 CT for some more FreeSWITCH fun! And, head over to freeswitch.com to learn more about FreeSWITCH support.

New features that were added:

  • FS-9480 [mod_kazoo] Add API enhancements
  • FS-9457 [mod_http_cache] Allow GET and PUT from Azure Blob Service
  • FS-9487 [core] Add CBR param to video file recording params
  • FS-9495 [mod_conference] Add conference_join_energy_level variable

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

  • FS-9551 [mod_sofia] Compare session before setting TFLAG_SKIP_EARLY
  • FS-9488 [mod_http_cache] Fixed a compile error
  • FS-9498 [mod_conference] Try to make video writing thread more efficient

The following bugs were squashed:

  • FS-9482 [core] Fixed a segfault on the second attempt to use uuid_media_3p
  • FS-9483 [mod_conference] Fixed a missing keyframe after re-invite
  • FS-9484 [core] Fixed a variable type format spec
  • FS-9493 [mod_conference] Fixed a possible crash when changing from normal to personal canvas on the fly
  • FS-9494 [mod_conference] Fixed issues with video avatar switching when video starts/stops
  • FS-9486 [mod_sofia] Fixed an issue with uuid_drop_dtmf switching between tone replace and digit
  • FS-9458 [mod_avmd] Set channel variable before BEEP event is fired
  • FS-6954 [core] Use channel flags to check for proxy media or bypass media
  • FS-9346 [verto_communicator] Add DTMF icon while on a video call, fixing conferences with pin number
  • FS-9497 [mod_av] Fixed an AV sync record issue

FreeSWITCH Week in Review (Master Branch) August 20th – August 27th

August 30 Week In Review no comments

It was a quiet week in the code with some minor build updates and improvements. The FreeSWITCH configuration audit has begun with initial minor commits and will continue throughout the year. If you are looking to volunteer to help with that or would like more information email brian@freeswitch.org or join the Bug Hunt on Tuesdays at 12:00pm Central Time.

Join us Wednesdays at 12:00 CT for some more FreeSWITCH fun! And, head over to freeswitch.com to learn more about FreeSWITCH support.

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

  • FS-9442 [Debian] Tweak the packages to properly install the debug symbols via freeswitch-all-dbg and freeswitch-meta-all-dbg
  • FS-8608 [configuration] Beginning the default configuration and a first step is establishing that parameters should have dashes

The following bugs were squashed:

  • FS-9443 [core] Fixed a segfault caused by SDP in a verto.invite with missing ICE candidates
  • FS-9447 [mod_avmd] Increased the number of samples to skip to avoid false beep detection on some voicemails for Windows
  • FS-9452 [libsofia] Fixed the true/false logic for using destination flag
  • FS-7706 [mod_callcenter] Hangup agent channel if we failed to bridge it with member channel

The FreeSWITCH 1.6.10 release is here!

August 25 Releases no comments

The FreeSWITCH 1.6.10 release is here!

This is also a routine maintenance release. Change Log and source tarball information below.

Release files are located here:

New features that were added:

  • FS-9192 [mod_sofia] Added renegotiate-codec-on-hold parameter for proxy hold when proxy media and proxy mode are disabled; it’s similar to proxy-refer
  • FS-9276 [mod_sofia] Added proxy in-dialog calls sip notify and info similar to proxy hold
  • FS-9440 [mod_sofia] Add transfer_destination
  • FS-9154 [libsofia] Add & remove video on re-invites
  • FS-9009 [mod_avmd] Add amplitude estimation
  • FS-9069 [mod_avmd] Add detection time to beep event
  • FS-9287 [mod_spandsp] Add channel variable to make spandsp_start_tone_detect easier to use from dialplan/embedded scripts.
  • FS-9310 [mod_sms_flowroute] Added native support for Flowroute SMS API over HTTP(S)
  • FS-9264  [mod_dptools] Add detect_audio and detect_silence API calls
  • FS-9230 [mod_conference] Customize video muted banner
  • FS-9441 [mod_conference] Add the option to not beep between dials in outcall
  • FS-9281 [mod_verto] Add support for QQVGA resolution in Verto
  • FS-9256 [mod_v8] Add DB.Finalize() in order to close statements.
  • FS-9419 [mod_commands] Add event_channel_broadcast api

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

  • FS-9263 [build] Attempting to find the proper lua5.2 version on openbsd
  • FS-9260 [build] Fixed make detection to not fail on openbsed,  fixed libtoolize detection to attempt to find libtoolize the same version as specified libtool, and added -ltermcap for openbsd so it can correctly link to libedit
  • FS-9403 [build] Add timestamp for when user was pushed into queue that lives with the channel
  • FS-9449 [build] Enable clock calibration and clock_realtime on Solaris
  • FS-9424 [build] Define byte order correctly on Solaris/SPARC
  • FS-9317 [configuration] Added screen share examples to the vanilla configurations
  • FS-9350 [configuration] Add mod_av commented to modules.conf.xml
  • FS-9320 [mod_local_stream] When the entity playing the local_stream video file has a greater or equal frame rate, reduce the buffering
  • FS-9315 [mod_http_cache] Added support for video file formats
  • FS-9386 [mod_snmp] Use net-snmp-config for SNMP libs if available
  • FS-9385 [mod_conference] Check for ghosts before destroying a conference
  • FS-9161 [mod_verto] Added example Verto settings to example configs
  • FS-9362 [mod_sofia] Fixed sofia compile error on newer clang included in new osx
  • FS-9447 [mod_avmd] Increase default value of samples to skip to increase robustness against false detections
  • FS-9439 [mod_loopback] Check chained loopback for loopback_bowout
  • FS-9447 [mod_avmd] Fixed PRId64 on Windows
  • FS-7534 [Ubuntu] Added sysvinit-utils dependancy for ubuntu to debian bootstrap.sh
  • FS-9442 [Debian] Tweak the packages to properly install the debug symbols via freeswitch-all-dbg and freeswitch-meta-all-dbg
  • FS-9373 [Debian] Added mod-verto and mod-rtc to freeswitch-meta-all package

The following bugs were squashed:

  • FS-9244 [core] Fixed debug lines
  • FS-9265 [core] Fixed an issue with receiving INCOMPATIBLE_DESTINATION when there is no RTCP
  • FS-9292 [core] Fixed a core dump while playing videos or showing images usually with a high number of callers
  • FS-9312 [core] Fixed and unreachable code block in switch_core_media
  • FS-9328 [core] Fixed switch_jb_peek_frame bug where it uses the len of the whole packet and does not subtract the len of the rtp header when copying and returning the size of the packet read.
  • FS-9337 [core] Fixed invalid sdp generated with soa disabled
  • FS-9352 [core] Fixed overzealous ptime adjust issues on opus
  • FS-9355 [core] Fixed a segfault in case of null frame
  • FS-9356 [core] Fixed an issue with DTMF not recognized when coming from a Cisco SIP trunk
  • FS-9289 [core] Fixed a MOH issue with b side hold causing silence for the a leg
  • FS-9365 [core] Fixed the SDP format on reply to RE-INVITE to be RFC-4566 compliant
  • FS-9357 [core] Handle packet loss and reset decoder on memory error
  • FS-9382 [core] Fixed an issue with video broken between two users in verto
  • FS-9390 [core] Fixed a ‘Segmentation fault’ during call setup
  • FS-9369 [core] Added the variable add_ice_candidates=true to enable inserting ice candidates in outgoing sdp
  • FS-9436 [core] Fixed RTCP PLI media source SSRC not updating after re-INVITE
  • FS-9422 [core] Fixed an issue with Freeswitch Exit/Crash on SDP negotiation by removing renegotiate-codec-on-hold and renegotiate-codec-on-reinvite
  • FS-9410 [core] Fixed PLI Missing Media Source SSRC
  • FS-9423 [core] Handle null value in ACL list name
  • FS-9434 [core] Fixed the SDP parser in sofia not recognizing UDP/TLS/RTP/SAVP
  • FS-9380 [core] Fixed a problem with ext-rtp-ip not being used when originating
  • FS-9443 [core] Fixed a segfault caused by SDP in a verto.invite with missing ICE candidates
  • FS-9401 [core][mod_amqp] Fixed a leak in usage of hash itterator
  • FS-9357 [verto communicator] Fixed an issue with VP9 codec screensharing on mod_conference (mux/transcode) not working
  • FS-9342 [verto_communicator] Fixed a problem with settings not being saved when closing the settings panel
  • FS-8761 [libsofia][mod_verto] Fixed a memory leak
  • FS-9452 [libsofia] Fixed true/false logic for using dst flag
  • FS-9375 [mod_sofia] Fixed DTMF not working on OPUS after call transfer
  • FS-9423 [mod_sofia] Handle null value in ACL list name
  • FS-9183 [mod_sofia] Handle 415 Unsupported Media Type as 488
  • FS-9376 [mod_sofia] Fixed a hold negotiation problem on a call received from a Cisco Session Manager
  • FS-9381 [mod_sofia] Fixed a leak in sofia_presence_chat_send
  • FS-9368 [mod_sofia] Fixed a problem with errant duplicate video frames causing video recording issues
  • FS-9241 [mod_sofia] Use tls_public_url instead of tls_url in INVITE Contact when NAT is detected
  • FS-9316 [mod_sofia] Fixed an issue caused by INVITE with empty SDP from Cisco VCS not setting up video
  • FS-9333 [mod_sofia] Disable video refresh by sip INFO by default because this method is outdated
  • FS-9297 [mod_sofia] Fixed multiple crashes from passing invalid null values in sofia.conf
  • FS-9301 [mod_sofia] Handled a race condition on startup of mod_sofia with error conditons causing segfault
  • FS-9271 [mod_conference] Fixed a segfault trying to record a canvas that does not exist
  • FS-9221 [mod_conference] Add inactive support for calls to prevent termination if just the video stream is removed
  • FS-9303 [mod_conference] Removed unnecessary checks as the video flag is not sent to file open unless using transcode mode, you can record mp4 but it will only contain the audio if in passthru mode
  • FS-9305 [mod_conference] Fix for fs_cli crashing due to vid-logo-img incorrectly being set to nothing after originally setting it to a bad image
  • FS-9307 [mod_conference] Fixed a race condition caused by trying to use a closed file handle when playing a video file after closing files before video threads are done
  • FS-9314 [mod_conference] Fixed a crash when starting conference in mux mode while specifying or defaulting to a layout group that does not exist. We will now fall back to transcode mode in this case
  • FS-9353 [mod_conference] Fixed a problem with clear-vid-floor producing an error while working
  • FS-9425 [mod_conference] Fixed a copy and paste error where the height was not set properly
  • FS-9437 [mod_conference] Delete avatar if video is enabled mid-call
  • FS-9267 [mod_cv] Fixed an issue where the VPX codec returns the same image to the core when doing repeated decoding. Updates to that image match the updates to the stream so if a media bug modifies the image between key frames it messes up the picture until the next key frame is received.
  • FS-9283 [mod_hiredis] Fixed an issue with using hiredis_raw on channels without media such as an originate
  • FS-9302 [mod_mongo] Fixed mongo_find_one and mongo_find_n to return -ERR when the connection to the database fails
  • FS-9313 [mod_opus] Fixed sprop_stereo interpretation causing bad audio
  • FS-9345 [mod_httapi] Fixed an issue with HTTAPI truncating a string when responses span multiple packets
  • FS-9296 [mod_httapi] Fixed video support
  • FS-9343 [mod_smpp] Fixed a problem with failing to send a message via Nexmo
  • FS-9259 [mod_spandsp] Fixed a missing “m=image 0” when replying to INVITE with disable image line
  • FS-8783 [libsrtp] Fix alignment issue
  • FS-9394 [mod_av] Fixed the h263 leak
  • FS-9409 [mod_av] Wait for avformat reader thread before reading to prevent spurious EOF and skipped playback
  • FS-8783 [libsrtp] Fix alignment issue
  • FS-9415 [mod_spy] Increasing loop so we can also look for variable_verto_user and variable_verto_host to fix mod_spy not reacting on verto channels
  • FS-7706 [mod_callcenter] Hangup agent channel if we failed to bridge it with member channel.