forked from Mirrors/freeswitch
acd1216ac2
git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@10992 d0543943-73ff-0310-b7d9-9358b9ac24b2
562 lines
32 KiB
Plaintext
562 lines
32 KiB
Plaintext
freeswitch (1.0.2)
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all: don't add module interfaces before returning from error conditions in module load functions (MDXMLINT-36)
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all: fixed multiple memory leaks
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all: improved module unloading/reloading support
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build: add support for --switchconfdir (FSBUILD-84)
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build: fixed netbsd build
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build: make freeswitch stop graceflly with /etc/init.d/freeswitch stop on debian add working dir to start-stop-dir so freeswitch dumps core in workdir
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build: multiple packaging fixes
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build: user freeswitch not added to audio group on deb install (FSBUILD-95)
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Configuration: many updates to default configuration
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core: Add ability to choose uuid from originate string, originate_uuid var (use at your own risk)
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core: add bridge_generate_comfort_noise option for bridge to generate comfort noise to the A leg when there is no audio on the B leg
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core: add chan vars to param event during hangup hook
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core: add exec directive to xml preprocessor (not available on windows)
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core: add force_transfer_dialplan and force_transfer_context variables
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core: add hashing to event header lookup
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core: add hits to tone_detect
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core: add last_dtmf_duration variable
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core: add msleep function to swigged languages
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core: add park_after_bridge variable
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core: add per leg timeouts and specific cause codes in reject_on_single_fail
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core: add runtime selection of the module dir (FSCORE-198)
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core: add scheduler support for heartbeat
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core: add session heartbeat feature
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core: add session.destroy psuedo method to sort of destroy a session at least for the sake of FS
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core: add session.unsetInputCallback
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core: add strftime format string validation for user supplied values
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core: add vars param to switch_ivr_originate for recursion (MODAPP-175)
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core: added a "group" concept to the user directory
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core: added ability to do dns lookup to find ip with host: like stun: (FSCORE-219)
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core: added better locking for codec changes during a call
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core: added current_application and current_application_data variables
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core: added error/ magic endpoint so modules can return error causes in situations like sofia_contact
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core: added read_result channel variable
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core: added support for "F" to indicate flash in dtmf (FSCORE-213)
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core: allow calls to be stolen from originate
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core: allow you to get the privacy bits in the caller_profile
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core: change dso code to load symbols local
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core: changes core flags to be array based so we have more
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core: eavesdrop causes the people being eavesdropped on to not hear ach other (MODAPP-140)
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core: expose time table to variable interface via caller field lookup
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core: fix 100% cpu when sending parked call to moh (FSCORE-234)
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core: fix bridge app to make sure both channels are ready for media when one is only in ringing state
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core: fix buffer overflow (FSCORE-188)
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core: fix conference dial by allowing multiple braces in originate, fix bad pointer op (FSCORE-208)
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core: fix double detection of DTMF in IVR (FSCORE-221)
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core: fix hangup_after_bridge is false on a bridge started with the intercept app
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core: fix issue where pid file is accidentally truncated
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core: fix ivr timeout (FSCORE-181)
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core: fix memory leak in alias tab completion code
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core: fix min digits in read app
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core: fix out-of-bounds pointer in variable expansion (FSCORE-171)
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core: fix segfault in media bugs when in bypass media (FSCORE-193)
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core: fix segfault on gtalk to sip calls (FSCORE-212)
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core: fix segfault on reloadxml (FSCORE-176)
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core: fix segfault on trasfering eavesdopping party (FSCORE-210)
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core: fix segfault using switch_system function (FSCORE-196)
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core: fix session.bridge
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core: fix setting effective_caller_id_name / effective_caller_id_number on bridge dialstring (MODAPP-122)
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core: fix stream_raw_write (MODAPP-145)
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core: fix using resampling on ringback file
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core: fixed performance bottleneck in sqlite db's
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core: fixed race in reloadxml
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core: increment app before execute in case it returns to execute it will go to the next item in the list and not the same
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core: ivr menu max_failures and max_timeouts now default to 3 if not specified or invalid (less than 1) values are specified (FSCORE-244)
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core: ivr_menu max-timeouts option, result in ivr_menu_status var (FSCORE-183)
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core: let b legs use park_after_bridge too
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core: make events less verbose unless verbose_events is set
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core: parse private events during originate
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core: pass pdd data to a leg on oubound calls using bridge
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core: prevent crash in crazy situation with xml interface lookup failures (FSCORE-169)
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core: reduce cpu requirement for generated comfort noise
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core: remove interface names from core db on unload
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core: reworked timing resulting in significant performance increase and better rtp timing
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core: rewrite switch_play_and_get_digits (MODAPP-166)
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core: session.recordFile never terminates (MODLANG-79)
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core: session.transfer make dialplan and context optional
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core: set_user app now sets domain vars as well as user vars
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core: tone_detect not triggering app after tone detection (MODAPP-182)
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core: unprivileged user setting bigger stack for switch_system thread failure (FSCORE-197)
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core: user_exists returns false when fetching a user from XML Curl or other xml interfaces
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libesl: added c event socket library and fs_cli
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libsndfile: fix autoconf 2.62 support (LBSNDF-5)
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mod commands: add "all" modifier to "break" command
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mod_celt: added new module
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mod_commands: Add support for more than 2 variables to uuid_setvar_multi (MODAPP-171)
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mod_commands: Add support for passing the cause of hangup to the uuid_kill command (FSCORE-217)
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mod_commands: add attr lookup to user_data
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mod_commands: add domain_exists fsapi command
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mod_commands: add eval fsapi command
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mod_commands: add flush_dtmf app and uuid_flush_dtmf api command
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mod_commands: add fsctl send_sighup, fsctl shutdown asap, unsched_api commands
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mod_commands: add fsctl shutdown [elegant|restart|cancel]
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mod_commands: add new syntax to uuid_setvar to allow you to unset a var. <uuid> <var> [value] (MODAPP-167)
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mod_commands: add reload fsapi command to reload a module
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mod_commands: add system fsapi and application (MODAPP-138)
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mod_commands: added hupall fsapi command
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mod_commands: added strftime_tz api command
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mod_commands: break all now stops broadcast too
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mod_commands: fix api command sent through sched_api was getting the last char lopped off
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mod_commands: fix race on transfer with -both
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mod_commands: fix system dialplan app problems (MODAPP-86)
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mod_commands: only send content-type on status when it really is http.
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mod_conference: add fsapi to stop async playback too
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mod_conference: add video caps to mod_conference with video follow audio
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mod_conference: better sound prefix handling when using say: and allow say: on kick sounds.
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mod_conference: fix race in record
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mod_conference: fix runaway thread when floor holder has no video and other people do have video
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mod_conference: fix seg when kicking many members quickly (MODAPP-129)
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mod_conference: fix segfault on invalid chat event
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mod_conference: perpetual sound does not auto-mute, you can do that yourself if you want it
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mod_dialplan_xml: add Hunt- vars in dialplan lookup after transfer
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mod_dialplan_xml: fail call on extensions with nested conditions
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mod_dingaling: (LBDING-7) fix segfault on os x
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mod_dingaling: end call on ice timeout
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mod_dingaling: fix presence on jabber to be less protocol ambiguous
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mod_dingaling: fix segfault (LBDING-10)
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mod_dingaling: update to support latest client from google
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mod_dptools: add a mechanism to tell if a file played from sendmsg over event socket
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mod_dptools: add playback_terminator support to phrase and say app
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mod_dptools: add playback_terminator_used variable (MODAPP-132)
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mod_dptools: add presence application
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mod_dptools: fix originate api not parsing users properly (FSCORE-246)
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mod_dptools: fix record and record_session to create directory if it does not exist (FSCORE-250)
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mod_dptools: fixed limit and + parsing bug in record_session app (MODAPP-148)
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mod_dptools: remove_bugs added to remove all media bugs on a session
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mod_erlang_event: add new module
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mod_event_socket: missing : after Content-Length in event socket (MODEVENT-33)
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mod_event_socket: add event socket listener filters
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mod_event_socket: add stateful listener fsapi commands for ajax-y type event interface over http
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mod_event_socket: fix arg parsing errors (MODEVENT-34)
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mod_event_socket: fix shutdown segfault race (MODEVENT-32)
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mod_event_socket: inbound connection to event_socket can now take over an existing channel with 'myevents <uuid>' to take on the behaviour of an outbound socket
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mod_event_socket: let any channel get messages
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mod_event_socket: make event socket wait for hangup on outbound mode and send disconnect message
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mod_expr: fix endless loop
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mod_fax: new module
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mod_fifo: add fifo_consumer_wrapup_time var (MODAPP-117)
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mod_fifo: added callback agents
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mod_fifo: honor keyword silence (MODAPP-118)
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mod_flite: added windows build
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mod_fsv: fix in a windows enviroment opening the record file in text mode. (MODAPP-169)
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mod_http: added new module
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mod_java: updated to new module api to support read/write locks on interface
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mod_limit: accept dialplan context for transfer (MODAPP-161)
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mod_limit: added hashtable based limit functions
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mod_limit: prevent empty error log message (MODAPP-134)
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mod_local_stream: add start_local_stream and stop_local_stream fsapi commands to start/stop dynamically (MODFORM-13)
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mod_local_stream: fix leak and improve error checking
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mod_local_stream: fix seg when no timer name specified in config file. (MODFORM-16)
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mod_loopback: add new module
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mod_lua: add local scripts directory support (MODLANG-86)
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mod_lua: don't eval blank string
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mod_lua: fix originate
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mod_lua: fix segfault (MODLANG-77)
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mod_lua: update to lua 5.1.4 (MODLANG-87)
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mod_lumenvox: removed
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mod_managed: new module replaces mod_mono now supports native .net runtime on windows as well
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mod_opal: added to trunk (still very beta)
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mod_perl: fix segfault (MODLANG-77)
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mod_pocketsphinx: fix rpm build
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mod_portaudio: fix cpu race on inbound call to pa when no ring file is set
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mod_radius_cdr: dictionary update for cause code changes (MODEVENT-27)
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mod_radius_cdr: fix unload (MODEVENT-29)
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mod_shout: add stereo recording broadcast support
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mod_shout: added windows build
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mod_shout: fix segfault when recording mp3's (MODFORM-12)
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mod_shout: improved stability of mp3 decoding
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mod_siren: added new module
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mod_sndfile added support to record 16bit for the various rates including 48kHz
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mod_sofia: Add filter to "sofia status profile XXX" (MODENDP-138)
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mod_sofia: Add force-register-db-domain which works in conjunction with force-register-domain.
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mod_sofia: Add optional <variables> and <params> tag to <gateway> tag.
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mod_sofia: Challenge the right realm when to_host is outside the users domain. (MODENDP-136)
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mod_sofia: Improve notify messages through a proxy (MODENDP-147)
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mod_sofia: MWI for multiple domains (MODAPP-126)
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mod_sofia: Move "a=sendrecv" from session to media section of SDP (MODENDP-148)
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mod_sofia: add 200 OK re-invite without sdp
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mod_sofia: add custom sofia::gateway_state event (MODENDP-112)
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mod_sofia: add fire events for the refer SIP NOTIFY event package (MODENDP-152)
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mod_sofia: add more params for xml_curl directory lookup
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mod_sofia: add new auto vals for challenge-realm param <param name="challenge-realm" value="auto_from|auto_to|<hardcoded_val>"/>
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mod_sofia: add option to turn of auto_restart of sofia profiles on ip change
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mod_sofia: add params to use sip callid as uuid on inbound calls and uuid as sip callid on outbound calls
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mod_sofia: add parsing of Privacy header for privacy info (MODENDP-133)
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mod_sofia: add proto_specific_hangup_cause to both legs
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mod_sofia: add proxy 3pcc mode
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mod_sofia: add redirect variable to channel as well as partner channe (MODENDP-135)
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mod_sofia: add sip-forbid-register to user params to refuse to let a certian user register
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mod_sofia: add sip: into register-proxy when it's not specified
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mod_sofia: add sip_history_info var for inbound invites.
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mod_sofia: add sip_via_protocol variable
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mod_sofia: add sofia xmlstatus (MODENDP-156)
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mod_sofia: add support for params other than Replaces in Refer-To (MODENDP-143)
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mod_sofia: add support for profiles sharing databases so that you can have a domain that uses multiple profiles for split dns type setups
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mod_sofia: add support for refer transfer involving multiple machines
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mod_sofia: add support to send a notify in the invite dialog by specifying the uuid of the call. (SFSIP-92)
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mod_sofia: add suppress_from_cidname var to not have display name in from header (MODENDP-153)
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mod_sofia: added sip_hangup_disposition variable
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mod_sofia: allow send_message and notify events to send a message/notify without a body if needed.
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mod_sofia: append -1 .. -N postfix after any X-headers as vars that have the same name
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mod_sofia: cache auth_gateway_name in sofia for challenged bye
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mod_sofia: cancel proxy or no-media mode if you purposely answer or pre_answer
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mod_sofia: correct result code mapping for Unallocated Number (MODENDP-124)
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mod_sofia: disable 100rel by default
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mod_sofia: don't accept crypto in the RTP/AVP (MODENDP-126)
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mod_sofia: don't put CN in sdp answer if it was not in the offer.
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mod_sofia: fix Incorrect IP address shows up in SDP "o" field when multiple external IPs available and FS not bound to first (MODENDP-132)
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mod_sofia: fix Wrong RTP media port destination after reinvite/UNHOLD (SFSIP-82)
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mod_sofia: fix bug on linksys where they lie about the ptime and handle linksys transfer problem
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mod_sofia: fix chat (send an IM) assumes that the user's profile is the same as their domain, which isn't necessarily so (SFSIP-83)
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mod_sofia: fix dtmf handling of broken info dtmf endpoints
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mod_sofia: fix eyebeam presence to be RFC compliant (MODENDP-144)
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mod_sofia: fix ip change detection when in proxy mode
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mod_sofia: fix register_proxy ignoring the paramaters (MODENDP-121)
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mod_sofia: fix remote session refresh triggers request glare (MODENDP-131)
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mod_sofia: fix rtp auto adjust running when it should not
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mod_sofia: fix rtp sent to wrong port after some re-INVITE scenarios (MODENDP-141)
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mod_sofia: fix sending of cn packets across bridge when we shouldn't
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mod_sofia: fix sqlite issue with select of the sip contact
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mod_sofia: fixed segfault on invalid presence payload
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mod_sofia: gateway ping needs to look for 501 (SFSIP-78)
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mod_sofia: handle multi contact register responses and register timeout better
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mod_sofia: improve gateway resilience
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mod_sofia: log ip and port you get reply to invite from
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mod_sofia: make multiple-registations=true use the contact method and call-id option to do it the old way
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mod_sofia: make proxy mode pull the port from m=image as well
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mod_sofia: make register-proxy preserve the url composed from proxy but target the packets to desired address (MODENDP-121)
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mod_sofia: many fixes for sonus rtp issues silence_when_idle=400 chanvar to send generated silence duing sleeps etc
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mod_sofia: many fixes in presence handling
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mod_sofia: passthrough t.38 fixes
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mod_sofia: pick ipv4 or ipv6 based on sipip instead of having mixed in sdp
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mod_sofia: send NOTIFY on TCP/UDP depending on the SUBSCRIBE (SFSIP-104)
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mod_sofia: setting profile option multiple-registrations=contact key multi reg off the contact string
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mod_sofia: wait for a reply on refer
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mod_soundtouch: fixes and improvements, many options changed (MODAPP-149)
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mod_soundtouch: updated to new module api
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mod_spidermonkey: Segmentation fault in check_hangup_hook at mod_spidermonkey.c:1589 (MODLANG-74)
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mod_spidermonkey: fix bug in apiExecute
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mod_spidermonkey: fix memory pool handling and leaks
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mod_spidermonkey: limit recursion busting through the stack (FSCORE-202)
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mod_spidermonkey: make session.getVariable return blank string not the word false
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mod_spidermonkey_curl: add optional content-type arg
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mod_spidermonkey_odbc: fix numRows and add numCols
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mod_spidermonkey_odbc: fix segfault (MODLANG-75)
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mod_stress: new module for voice stress analysis
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mod_syslog: don't log blank lines (FSCORE-163)
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mod_tone_stream: let silence_stream://0 indicate perpetual silence
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mod_vmd: add new module to detect voicemail "beep"
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mod_voicemail: Add vm_alternate_greet_id param to directory entry (MODAPP-174)
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mod_voicemail: Patch to add voicemail preference controlling date announcement new param 'play-date-announcement' to values 'first' 'last' or 'never' defaults to first to retain previous behavior (MODAPP-121)
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mod_voicemail: Update mwi light after delete vm via web. (MODAPP-124)
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mod_voicemail: add ability to get to options without listening to every saved message (MODAPP-115)
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mod_voicemail: add ability to skip greeting when leaving a voicemail. (MODAPP-181)
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mod_voicemail: add change-pass-key config file option
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mod_voicemail: add forwarding support
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mod_voicemail: add local dtmf driven alternat vm pass
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mod_voicemail: add proper notification of a vm message being too short
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mod_voicemail: add support for auth via a1-hash
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mod_voicemail: add the "storage-dir" parameter to be set on a per-user basis (MODAPP-133)
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mod_voicemail: add voicemail_greeting_path variable
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mod_voicemail: added voicemail_alternate_greet_id variable
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mod_voicemail: allow changing of password from voicemail to update user directory if using non-static config (MODAPP-156)
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mod_voicemail: created email date (int overflow) (MODAPP-125)
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mod_voicemail: don't try to deliver vm when no file was recorded. (MODAPP-133)
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mod_voicemail: fix MWI with xml_curl used for directory (MODAPP-176)
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mod_voicemail: fix Voicemail messages occasionally lost / stranded (MODAPP-178)
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mod_voicemail: fix invalid event after message deleted (MODAPP-170)
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mod_voicemail: fix mwi for phones with multiple registrations problem (MODAPP-153)
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mod_voicemail: fix voicemail segfault on incorrect password (FSCORE-187)
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mod_voicemail: fix voicemail_inject error handling (MODAPP-133)
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mod_voicemail: fix voicemail_inject usage api call
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mod_voicemail: improve error checking (MODAPP-142)
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mod_voicemail: localize notification emails (MODAPP-139)
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mod_voicemail: make more multi-domain friendly (MODAPP-162)
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mod_voicemail: make playback created file macros optional (MODAPP-150)
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mod_voicemail: recognize operator key in more places (MODAPP-159)
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mod_voicemail: web interface displays incorrect created / last heard dates (MODAPP-123)
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mod_wanpipe: removed
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mod_xml_cdr: add https support
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mod_xml_cdr: add optional a-leg prefix to xml cdr filenames (MDXMLINT-39)
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mod_xml_cdr: add support for fallback webserver for cdr posting (FSCORE-238)
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mod_xml_curl: Allow specification of HTTP method, and dynamic expansion of variables in URI. (MDXMLINT-41)
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mod_xml_curl: added redirect following (max 10)
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mod_xml_ldap: almost a complete rewrite of this module
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mod_xml_rpc: allow setting of global realm without a global user (MDXMLINT-45)
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mod_xml_rpc: fix multiple segfaults
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mod_xml_rpc: fix segfault on originate via http
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sofia-sip: updated to 1.12.10 (plus a few patches)
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freeswitch (1.0.1)
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FIX: prevent intercept race condition that can also be solved with continue_on_fail=originator_cancel
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FIX: NULL dereference detected by klockwork (www.klockwork.com)
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FIX: don't open failed local stream (MODFORM-9)
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FIX: instability in mod_local_stream in failure scenarios
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FIX: xmlrpc-c build on OS X 10.4 (FSBUILD-47)
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ENHANCEMENT: Added tab completion on many api commands in console
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ENHANCEMENT: polycom BLF support
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FIX: many sip NAT related fixes in mod_sofia
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FIX: support sip unregister with Contact: *
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FIX: multiple segfaults in xmlrpc-c
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FIX: sip unregister event being skipped
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FIX: hangup properly on malformed sip 3pcc calls being used as a way to ping
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ADD: enable-3pcc sofia profile param, it is now disabled by default.
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ADD: presence events to sip proxy mode
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ADD: legs param to cdr_csv
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ADD: support for perl as an embedded lanugage
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ENHANCEMENT: many new api's and functions to the embedded languages including api support, xml interface support, auto start scripts, and many new objects
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CHANGE: python embedded language api changed to match perl, lua, java
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FIX: many stability fixes in embedded langauges perl, lua, java, python
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ADD: failed_xml_cdr magic channel variable
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FIX: access free memory error in mod_sofia when using respond app
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ENHNACEMENT: make global_setvar only have 2 fields so you can set foo=bar=blah w/o quotes
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FIX: mod_spidermonkey keep hangup hook in the session thread
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ENHANCEMENT: mod_ldap added sasl support and search filters
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ADD: answered, waitForAnswer and mediaReady methods to embedded language Session object
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ENHANCEMENT: mod_voicemail param change to allow notification emails using templates
|
|
ADD: per user acl in sofia
|
|
FIX: deadlock in mod_portaudio
|
|
ENHANCEMENT: blank username in sip will trigger a lookup for the user "nobody"
|
|
ADD: import variable to import variables from a peer channel at time of originate
|
|
FIX: api type fix for c++ modules when incorrectly using enums
|
|
FIX: eliminate need for escaped , in [] on originate
|
|
ADD: NDLB-force-rport option to force behavior as if rport was sent in the via
|
|
ENHANCEMENT: honor execute_on_answer on outbound legs too
|
|
ADD: execute_on_ring variable
|
|
FIX: Seg fault in CoreSession() class destructor
|
|
ADD: per channel caller id in originate
|
|
ADD: sip_outgoing_call_id variable
|
|
FIX: multiple memory leaks in mod_sofia
|
|
FIX: find_local_ip IPv6 support
|
|
ADD: variable expansion to on execute vars.(FSCORE-114)
|
|
ADD: count optional arg to show calls and show channels (MODAPP-103)
|
|
FIX: MODEVENT-25 (WSAWOULDBLOCK error on socket send in windows) in event socket
|
|
FIX: multiple fixes to the logic in mod_say_zh
|
|
ADD: inter digit timeout to swigged embedded languages getDigits method. (MODLANG-65)
|
|
ADD: Linksys P-RTP-Stat SIP header values (SFSIP-66)
|
|
FIX: small leak in core
|
|
ADD: progress_timeout var to originate
|
|
UPDATE: portaudio library
|
|
FIX: added timeout to iax read
|
|
ADD: 'pa rescan' to portaudio to look for new devices
|
|
FIX: wait for broadcast to start when starting async hold to avoid race
|
|
FIX: mod_rss, don't always play the first news feed
|
|
FIX: mod_rss inverval to use the session inteval (audio problems on 30ms channels)
|
|
ADD: Path: support in mod_sofia on register
|
|
FIX: mod_shout record stream
|
|
ENHANCEMENT: mod_voicemail support for effective_caller_id_name/number
|
|
ADD: url encode/decode api calls
|
|
FIX: "nua()" in debug information in sofia instead of the real function name
|
|
FIX: better handling of sips: uris
|
|
FIX: don't seg when using more than SWITCH_MAX_CODECS and bump SWITCH_MAX_CODECS to 50 (we have more than 30 in tree) (MODFORM-10)
|
|
ADD: mod_yaml
|
|
FIX: segfault on freeswitch startup if installed directories are removed
|
|
FIX: segfault when intercept with inbound_late_negotiation=true set
|
|
FIX: dont flood logs with eavesdrop messages (MODAPP-101)
|
|
FIX: don't destroy a codec that has not been created (MODAPP-101)
|
|
ENHANCEMENT: allows the "eavesdrop_group" variable to contain several groups, comma separated. (MODAPP-101)
|
|
FIX: cross compile (FSBUILD-53)
|
|
FIX: add header that Nuaunce considers mandatory (MODASRTTS-5)
|
|
ADD: write locks to the core and a function to unregister event bindings (adds better ability to unload modules)
|
|
ENHANCEMENT: make modules unbind events and un-reserve subclasses on module unload
|
|
ADD: removable xml hook bindings
|
|
ADD: EventConsumer object to embedded languages so you can make event handlers
|
|
FIX: sending CN with supress-cng true
|
|
FIX: segfault in the event system when trying to remove NULL event
|
|
ADD: flags to turn off srtp auth and rtp auto adj (FSCORE-149 && MODENDP-115)
|
|
FIX: use lighter math and avoid infinite loop in port allocator (FSCORE-148)
|
|
ENHANCEMENT: let conference pin entry start during prompt (MODAPP-111)
|
|
ADD: mod_pocketsphinx
|
|
FIX: Misuse of SQLRowCount, issues with MSSQL (MODAPP-105)
|
|
FIX: segfaults in mod_python with dtmf callback
|
|
ENHANCEMENT: mod_conference auto-record parameter (MODAPP-112)
|
|
ENHANCEMENT: reload support to many modules
|
|
FIX: mod_sofia add replaces to supported header
|
|
ENHANCEMENT: add args callback to sleep so you can process dtmf and events while "sleeping"
|
|
ADD: mod_flite
|
|
ENHANCEMENT: switch_xml converted back to c code and support double globs on windows
|
|
ENHANCEMENT: mod_sofia support for adding and removing gateways without restarting profiles
|
|
ADD: extract contact header info into A channel when unhandled 3xx response is received (MODENDP-116)
|
|
FIX: outbound event_socket + late negotiation
|
|
ADD: copy_xml_cdr variable
|
|
ADD: silence_stream (like tone_stream but silent)
|
|
ADD: module_exists api call
|
|
ADD: emailer implementation for windows
|
|
ADD: wait_for_silence application
|
|
FIX: no error message generated if OS is unable to load a module ( due to dependency/installation issues )
|
|
FIX: segfault in media bugs
|
|
FIX: acl lists not correctly matching all ip adresses
|
|
FIX: mod_spidermonkey exit() does not stop script when called from the hangup callback (return "exit" from the callback)
|
|
FIX: mod_syslog works again
|
|
FIX: crash on terminal resize
|
|
FIX: audio problems on big endian
|
|
ENHANCEMENT: Disable multiple registrations on a per-device basis (MODENDP-117)
|
|
ADD: fifo_consumer_exit_key variable (MODAPP-100)
|
|
ADD: cidr based user auth in mod_sofia
|
|
ADD: uuid_send_dtmf fsapi command (MODAPP-114)
|
|
ADD: server registration fiels to sip_registration database (MODENDP-118)
|
|
FIX: use a variable, realm or to host to find gateway when it's not obvious (handles challenged REFER)
|
|
ADD: timeout to curl run in javascript
|
|
ADD: voicemail_inject fsapi command
|
|
ADD: reboot option for sip phones to flush_inboud_reg sofia profile api command
|
|
FIX: add small padding to end of mp3 to avoid cut off mp3 recording
|
|
FIX: patch multiple SDP connection lines in sdp for proxy media mode (MODENDP-109)
|
|
FIX: don't parse ringback varable in proxy situations
|
|
ADD: per call vm recording ext with vm_message_ext variable
|
|
ADD: sip_bye_h prefix to add headers to bye
|
|
ENHANCEMENT: more interfaces available in show fsapi command
|
|
FIX: don't leak in buffers on realloc fail
|
|
FIX: fail out of a conference call if write fails
|
|
ADD: auto ip-change detection
|
|
ADD: mod_snom
|
|
FIX: mod_sofia don't send sipfrag on transfer to cisco so they don't hang up the call
|
|
|
|
|
|
|
|
freeswitch (1.0.0)
|
|
|
|
Enhanced sofia sip nat handling
|
|
Many fixes found by Klockwork (www.klocwork.com)
|
|
Added disable_app_log variable
|
|
Fixed mod_local_stream with rates on windows
|
|
Fixed finding of files in rate dirs on windows
|
|
Fixed memory corruption from sofia_contact function
|
|
Added sofia profile param NDLB-received-in-nat-reg-contact
|
|
Added sofia profile param aggressive-nat-detection
|
|
Fixed video sip calls in proxy media mode
|
|
Added bridge_terminate_key var
|
|
Update xmlrpc-c lib to trunk revision from upstream, fix windows xmlrpc
|
|
Enhanced nat handling in proxy media mode in sip
|
|
Add progress media to timetable so you can calculate pdd
|
|
Fixed seg when using unicast on socket when call has no read_codec
|
|
Fixed missed log events on busy box
|
|
Added -bleg to intercept
|
|
Enhance configure detection of python
|
|
Fixed build on solaris and freebsd for several modules
|
|
Added param "vm-email-only" to make voicemail sent by email only (previously default behavior)
|
|
Added param "vm-mailto-notify" to allow sending a notification email
|
|
Fixed mod_java build
|
|
Fixed mwi failures for some devices that don't subscribe
|
|
Removed fsapi functions (killchan, transfer, session_displace, reject)
|
|
Removed fsapi functions (session_record, broadcast, hold, media)
|
|
Many updates to sofia-sip library including over 100 fixes
|
|
|
|
freeswitch (1.0.rc6)
|
|
|
|
Changed to not allow pass_2833 on transcoded calls (it never worked, now it will tell you)
|
|
Enhanced sofia sip nat handling
|
|
Fix libedit build on solaris
|
|
Fix session timers in mod_sofia
|
|
Fix conference fire-call
|
|
Change: add var_event down into the endpoints so chans with no parents can still pass options
|
|
Added enable-post-var param to xml_rpc
|
|
Fix mod_lua build on solaris
|
|
Many fixes found by Klockwork (www.klocwork.com)
|
|
Add unregister event in mod_sofia
|
|
Enhance python configure detection
|
|
Add vm_boxcount api func
|
|
Fixed att_xfer issue
|
|
Fix sip now includes the Allow-Events header in more places
|
|
|
|
freeswitch (1.0.rc5)
|
|
|
|
Changed internal state names to avoid confusion
|
|
Fixed video negotiation
|
|
Enhanced accuracy of windows timer
|
|
Fixed mod_ldap build
|
|
Added dialplan and context to sql table for channels
|
|
Multiple fixes to mod_lua and mod_perl
|
|
Fixed logic bug in fifo causing segfault
|
|
internal changes to sip stack so we can remove a hash redundant to the stack
|
|
Fixed multiple memory leaks in mod_sofia
|
|
Fixed event fetch segfault on sip subscribe
|
|
Fixed segfault on timer rollover in sofia on 64bit
|
|
Fixed audio timing issues in mod_portaudio
|
|
Changed names of sip profiles in default config to avoid confusion
|
|
Fixed memory usage leak-like behavior when playing files requiring resampling
|
|
Removed some unused api's
|
|
Fix rtp timeout when playing moh
|
|
Removed some un-needed libraries and files from tree
|
|
Fixed multiple issues in sip stack including multiple segfaults
|
|
Added support for sip transfers on bypass_media and proxy_media calls
|
|
Added say application
|
|
Fixed --disable-debug configure option
|
|
Enhanced switch_cpp wrapper (and perl, python, lua, java)
|
|
Fixed segfault on inavalid stun response
|
|
Fixed configure help output
|
|
Fixed segfault on mp3 playback
|
|
Fixed assert on invalid sdp (missing m= line)
|
|
Added configurable windows service name
|
|
Fixed proxy mode call transition to non proxy call
|
|
Fixed solaris build of voipcodecs
|
|
Fixed sofia seg when call failure edge case
|
|
|
|
freeswitch (1.0.rc4)
|
|
|
|
Add tab completion in cli
|
|
Add "inline" dialplan
|
|
Fixed segfault in enum
|
|
Enhance enum to fork dial equal priority entries
|
|
Added auto-reload to enum
|
|
Fixed odbc bug is mod_sofia presence handling
|
|
Add presence for conference and dial an eavesdrop
|
|
Fix stack overflow segfault when recursively parking calls
|
|
Fixed race is sofia registration handling
|
|
Enhance sofia registration, unregister on keep-alive OPTIONS failure
|
|
Added internal routing loop detection/avoidance
|
|
Fixed race in bgapi in event socket
|
|
Fixed vars to execute apps before bridge "bridge_pre_execute_aleg_app" and "bridge_pre_execute_bleg_app"
|
|
Fixed re-setting sound prefix to no prefix after a pharse
|
|
Enhanced setting of bracket vars from originate so they show in the CHANNEL_ORIGINATE event
|
|
Add "enable-timer" and "enable-100rel" options to turn off default behaviors in sofia
|
|
Add originate_timeout to originate vars
|
|
Fixed hanging channels in mod_portaudio
|
|
Added auto time sync on vps migration to different hardware
|
|
Fixed seg on transfer when both legs are not sip
|
|
Added configurable dtmf duration defaults
|
|
Enhanced voicemail, allow interruption of hello message
|
|
Fixed voicemail to not light up light on saved messages
|
|
Enhance mod_amr honor disable dtx in fmtp (MODCODEC-3)
|
|
Fixed bootstrap to install automake dependencies so you can use tarball without same version of automake installed
|
|
Fixed MODLANG-56 (bad audio on originate and javascript streamFile)
|
|
Added hold/unhold dialplan apps
|
|
Enhanced sofia error checking to outlaw 0.0.0.0 in sofia ip params
|
|
Backport fixes from sofia-sip tree
|
|
Fixed MSVC build
|
|
Fixed segfault on sip SUBSCRIBE with Expires: 0
|
|
Added mod_say_zh
|
|
Added --with-pyton and --with-pyton-config configure options
|
|
Added mod_lua
|
|
Enhanced switch_cpp wrapper in core and normalized interfaces for perl, python, lua, and java
|
|
Fixed multiple issues in cpp wrapper and the languages perl, python, lua and java
|
|
Added back mod_perl
|
|
Added sofia gateway option ping to adjust options ping frequency
|
|
Added .net event socket lib to contrib
|
|
Fixed passing of exact response codes of sip across a bridge
|
|
Added mod_reference, reference endpoint module
|
|
Enhanced build so you can now make commented out modules using "make mod_name"
|
|
|
|
freeswitch (1.0.rc3)
|
|
|
|
Enhance xml menu system
|
|
Fixes upstream from sofia-sip library
|
|
Enhance mod_fifo
|
|
Added close method to ODBC spidermonkey class
|
|
Fix multiple bugs in the cpp wrapper used in mod_java and mod_python
|
|
Fix hung sip channel issue using respond app or on re-invite with bypass media after 1xx or 2xx responses
|
|
|
|
freeswitch (1.0.rc2)
|
|
|
|
Fixed speex protocol negotiation issues (8k vs 16k)
|
|
Fixed mod_iax race conditions
|
|
Fixed ptime negotiation issues when re-packetizing
|
|
Added ip based acl lists
|
|
|
|
freeswitch (1.0.rc1)
|
|
|
|
|
|
|