v1.5 [Aug 24, 2016]
New features that were added:
FS-9079 [mod_callcenter] Add ring-progressively strategy which is a way to ring every agent similarly to a top-down strategy but without cancelling the previous calls.
FS-9248 [mod_callcenter] Adding truncate-tiers-on-load and truncate-agents-on-load options
FS-9216 [mod_sofia] Add Cisco SPA30X and Grandstream GXP user agents to send UPDATE
FS-9225 [mod_sofia] Allow to force SIP REGISTER Expires: to be within configured range instead of specific value
FS-9188 [mod_sofia] Added a channel variable to suppress auto-answer notify
FS-8652 [mod_sofia] Add a optional parameter “early-only” to replaces header parsing and only intercept the call if it is not bridged if this parameter is set to true
FS-9124 [mod_avmd] Extend XML config
FS-9142 [mod_avmd] Dynamic settings addition of checking of per session settings with locking synced on avmd session mutex
FS-9207 [core] Add ignore_sdp_ice=true to ignore ICE when parsing an SDP
FS-9157 [verto] Added the possibility to create dedicated audio/video tags for each dialog in verto
FS-9249 [verto_communicator] Close the settings panel if the user clicks outside the element
FS-9184 [mod_commands] Allow show calls to be filtered by accountcode
FS-8979 [mod_imagick] Added “lazy load” functionality to speed up the rendering of the first page of a PDF while continuing to load the following pages in the background
FS-9199 [scripts] Small change to make memory allocation tracing of ALL allocations easier and a script to analyze logs
Improvements in build system, cross platform support, and packaging:
FS-9070 [configuration] Fix build on 64-bit arm
FS-5936 [Debian] Add libesl-perl package containing and associated perl ESL bindings
FS-9075 [Debian] Additional tweaks to help ease upgrading freeswitch-all
FS-8788 [Debian] Fixed systemd error on Debian Jessie causing non enforcement of stack size limitation
FS-9174 [Debian] Fix installation of mod_png when installing via the -all packages
FS-8623 [build] Fix libvpx Solaris Studio build
FS-9158 [build] Add include for Solaris to changes to build
FS-9185 [build] Fixed the format of ifdefs for Solaris SPARC
FS-9152 [mod_avmd] Fixed warnings on FreeBSD
FS-9254 [mod_avmd] Fixed the windows build
FS-9155 [Centos] Fixed lang_es and lang_pt package to have the right language module
FS-9238 [mod_osp] Updated for OSP Toolkit 4.11.3.
FS-9134 [core] Tweaked fscore_pb to use new pastebin API
FS-9132 [mod_kazoo] Add more variables to default filter
FS-9164 [core] Add Session-Per-Sec-Last to heartbeat event
FS-9136 [core] Allow multiple instances of same video codec with different fmtp
FS-9106 [mod_vpx] Improve efficiency when using dedicated encoder mode in conference with vpx codecs
The following bugs were squashed:
FS-9131 [core] Improve validation of ice candidates to properly handle malformed candidates
FS-9135 [core] Handle incorrect uses of switch_core_media_set_sdp_codec_string function passing null sdp gracefully
FS-7783 [core] Properly handle NULL var_name for switch_play_and_get_digits
FS-9222 [core] Added a small tweak to freeswitch console to strip leading spaces from commands and added a fix for FreeSWITCH not sending binding response to VoIP client causing a one way audio call
FS-9235 [core] Fix sending RTCP in switch_core_media
FS-9219 [core] Fixed an issue with Re-INVITE with no SDP by using bypass_media_after_bridge_oldschool=true to cause bypass_media_after_bridge to use a standard RE-INVITE with SDP, instead of the more reliable method of using 3pcc RE-INVITE
FS-9246 [core] Fixed an issue with no audio after transferring a call
FS-9244 [core] Fixed an issue where RFC2833 payload_type offered is ignored
FS-9115 [mod_av] Initial work toward support for audio only mp4 recording
FS-9151 [mod_av] Fixed playback a mp4 file on a session without video not ending
FS-8795 [mod_png] Fixed an issue with audio only call
FS-8584 [mod_callcenter] Request agents and tiers when reloading queue
FS-9153 [mod_commands][mod_event_socket] Fixed a uuid_bridge issue on ESL
FS-9034 [mod_sofia] Fixed register processing in a new thread
FS-9160 [mod_sofia] Tweak sip_invite_failure_* chan vars for properly reporting last outbound call failure when there are multiple bridge attempts on a single call
FS-9214 [mod_sofia] Fixed 3pcc behavior and callflow issues with 3pcc=true and 3pcc=proxy and interactions with sip_wait_for_aleg_ack removes passthrough of 183 on 3pcc=proxy (that was previously not functioning)
FS-9227 [sofia-sip] Fixed wrong byte order in HEP packet for source and destination ports
FS-9167 [mod_conference] Fixed an issue where playing a file when all video feeds are vmuted does not show file
FS-9150 [mod_conference] Force the video-bridge-first-two only function when there are only 2 members who can watch video to prevent flipping between video feeds when video muting
v1.4 [Jun 25, 2014]
Fixed bugs:
- a regression in re-invite parsing, a few more issues uncovered by the continuing Coverity scans of the code base, addressing some build issues mod_perl, and addressing a WebRTC issue with Chrome where it now requires a longer DTLS Password.
