freeswitch/docs/ChangeLog

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freeswitch (1.0.2)
all: don't add module interfaces before returning from error conditions in module load functions (MDXMLINT-36)
all: fixed multiple memory leaks
all: improved module unloading/reloading support
build: add support for --switchconfdir (FSBUILD-84)
build: fixed netbsd build
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
build: multiple packaging fixes
build: user freeswitch not added to audio group on deb install (FSBUILD-95)
Configuration: many updates to default configuration
core: Add ability to choose uuid from originate string, originate_uuid var (use at your own risk)
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
core: add chan vars to param event during hangup hook
core: add exec directive to xml preprocessor (not available on windows)
core: add force_transfer_dialplan and force_transfer_context variables
core: add hashing to event header lookup
core: add hits to tone_detect
core: add last_dtmf_duration variable
core: add msleep function to swigged languages
core: add park_after_bridge variable
core: add per leg timeouts and specific cause codes in reject_on_single_fail
core: add runtime selection of the module dir (FSCORE-198)
core: add scheduler support for heartbeat
core: add session heartbeat feature
core: add session.destroy psuedo method to sort of destroy a session at least for the sake of FS
core: add session.unsetInputCallback
core: add strftime format string validation for user supplied values
core: add vars param to switch_ivr_originate for recursion (MODAPP-175)
core: added a "group" concept to the user directory
core: added ability to do dns lookup to find ip with host: like stun: (FSCORE-219)
core: added better locking for codec changes during a call
core: added current_application and current_application_data variables
core: added error/ magic endpoint so modules can return error causes in situations like sofia_contact
core: added read_result channel variable
core: added support for "F" to indicate flash in dtmf (FSCORE-213)
core: allow calls to be stolen from originate
core: allow you to get the privacy bits in the caller_profile
core: change dso code to load symbols local
core: changes core flags to be array based so we have more
core: eavesdrop causes the people being eavesdropped on to not hear ach other (MODAPP-140)
core: expose time table to variable interface via caller field lookup
core: fix 100% cpu when sending parked call to moh (FSCORE-234)
core: fix bridge app to make sure both channels are ready for media when one is only in ringing state
core: fix buffer overflow (FSCORE-188)
core: fix conference dial by allowing multiple braces in originate, fix bad pointer op (FSCORE-208)
core: fix double detection of DTMF in IVR (FSCORE-221)
core: fix hangup_after_bridge is false on a bridge started with the intercept app
core: fix issue where pid file is accidentally truncated
core: fix ivr timeout (FSCORE-181)
core: fix memory leak in alias tab completion code
core: fix min digits in read app
core: fix out-of-bounds pointer in variable expansion (FSCORE-171)
core: fix segfault in media bugs when in bypass media (FSCORE-193)
core: fix segfault on gtalk to sip calls (FSCORE-212)
core: fix segfault on reloadxml (FSCORE-176)
core: fix segfault on trasfering eavesdopping party (FSCORE-210)
core: fix segfault using switch_system function (FSCORE-196)
core: fix session.bridge
core: fix setting effective_caller_id_name / effective_caller_id_number on bridge dialstring (MODAPP-122)
core: fix stream_raw_write (MODAPP-145)
core: fix using resampling on ringback file
core: fixed performance bottleneck in sqlite db's
core: fixed race in reloadxml
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
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)
core: ivr_menu max-timeouts option, result in ivr_menu_status var (FSCORE-183)
core: let b legs use park_after_bridge too
core: make events less verbose unless verbose_events is set
core: parse private events during originate
core: pass pdd data to a leg on oubound calls using bridge
core: prevent crash in crazy situation with xml interface lookup failures (FSCORE-169)
core: reduce cpu requirement for generated comfort noise
core: remove interface names from core db on unload
core: reworked timing resulting in significant performance increase and better rtp timing
core: rewrite switch_play_and_get_digits (MODAPP-166)
core: session.recordFile never terminates (MODLANG-79)
core: session.transfer make dialplan and context optional
core: set_user app now sets domain vars as well as user vars
core: tone_detect not triggering app after tone detection (MODAPP-182)
core: unprivileged user setting bigger stack for switch_system thread failure (FSCORE-197)
core: user_exists returns false when fetching a user from XML Curl or other xml interfaces
libesl: added c event socket library and fs_cli
libsndfile: fix autoconf 2.62 support (LBSNDF-5)
mod commands: add "all" modifier to "break" command
mod_celt: added new module
mod_commands: Add support for more than 2 variables to uuid_setvar_multi (MODAPP-171)
mod_commands: Add support for passing the cause of hangup to the uuid_kill command (FSCORE-217)
mod_commands: add attr lookup to user_data
mod_commands: add domain_exists fsapi command
mod_commands: add eval fsapi command
mod_commands: add flush_dtmf app and uuid_flush_dtmf api command
mod_commands: add fsctl send_sighup, fsctl shutdown asap, unsched_api commands
mod_commands: add fsctl shutdown [elegant|restart|cancel]
mod_commands: add new syntax to uuid_setvar to allow you to unset a var. <uuid> <var> [value] (MODAPP-167)
mod_commands: add reload fsapi command to reload a module
mod_commands: add system fsapi and application (MODAPP-138)
mod_commands: added hupall fsapi command
mod_commands: added strftime_tz api command
mod_commands: break all now stops broadcast too
mod_commands: fix api command sent through sched_api was getting the last char lopped off
mod_commands: fix race on transfer with -both
mod_commands: fix system dialplan app problems (MODAPP-86)
mod_commands: only send content-type on status when it really is http.
mod_conference: add fsapi to stop async playback too
mod_conference: add video caps to mod_conference with video follow audio
mod_conference: better sound prefix handling when using say: and allow say: on kick sounds.
mod_conference: fix race in record
mod_conference: fix runaway thread when floor holder has no video and other people do have video
mod_conference: fix seg when kicking many members quickly (MODAPP-129)
mod_conference: fix segfault on invalid chat event
mod_conference: perpetual sound does not auto-mute, you can do that yourself if you want it
mod_dialplan_xml: add Hunt- vars in dialplan lookup after transfer
mod_dialplan_xml: fail call on extensions with nested conditions
mod_dingaling: (LBDING-7) fix segfault on os x
mod_dingaling: end call on ice timeout
mod_dingaling: fix presence on jabber to be less protocol ambiguous
mod_dingaling: fix segfault (LBDING-10)
mod_dingaling: update to support latest client from google
mod_dptools: add a mechanism to tell if a file played from sendmsg over event socket
mod_dptools: add playback_terminator support to phrase and say app
mod_dptools: add playback_terminator_used variable (MODAPP-132)
mod_dptools: add presence application
mod_dptools: fix originate api not parsing users properly (FSCORE-246)
mod_dptools: fix record and record_session to create directory if it does not exist (FSCORE-250)
mod_dptools: fixed limit and + parsing bug in record_session app (MODAPP-148)
mod_dptools: remove_bugs added to remove all media bugs on a session
mod_erlang_event: add new module
mod_event_socket: missing : after Content-Length in event socket (MODEVENT-33)
mod_event_socket: add event socket listener filters
mod_event_socket: add stateful listener fsapi commands for ajax-y type event interface over http
mod_event_socket: fix arg parsing errors (MODEVENT-34)
mod_event_socket: fix shutdown segfault race (MODEVENT-32)
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
mod_event_socket: let any channel get messages
mod_event_socket: make event socket wait for hangup on outbound mode and send disconnect message
mod_expr: fix endless loop
mod_fax: new module
mod_fifo: add fifo_consumer_wrapup_time var (MODAPP-117)
mod_fifo: added callback agents
mod_fifo: honor keyword silence (MODAPP-118)
mod_flite: added windows build
mod_fsv: fix in a windows enviroment opening the record file in text mode. (MODAPP-169)
mod_http: added new module
mod_java: updated to new module api to support read/write locks on interface
mod_limit: accept dialplan context for transfer (MODAPP-161)
mod_limit: added hashtable based limit functions
mod_limit: prevent empty error log message (MODAPP-134)
mod_local_stream: add start_local_stream and stop_local_stream fsapi commands to start/stop dynamically (MODFORM-13)
mod_local_stream: fix leak and improve error checking
mod_local_stream: fix seg when no timer name specified in config file. (MODFORM-16)
mod_loopback: add new module
mod_lua: add local scripts directory support (MODLANG-86)
mod_lua: don't eval blank string
mod_lua: fix originate
mod_lua: fix segfault (MODLANG-77)
mod_lua: update to lua 5.1.4 (MODLANG-87)
mod_lumenvox: removed
mod_managed: new module replaces mod_mono now supports native .net runtime on windows as well
mod_opal: added to trunk (still very beta)
mod_perl: fix segfault (MODLANG-77)
mod_pocketsphinx: fix rpm build
mod_portaudio: fix cpu race on inbound call to pa when no ring file is set
mod_radius_cdr: dictionary update for cause code changes (MODEVENT-27)
mod_radius_cdr: fix unload (MODEVENT-29)
mod_shout: add stereo recording broadcast support
mod_shout: added windows build
mod_shout: fix segfault when recording mp3's (MODFORM-12)
mod_shout: improved stability of mp3 decoding
mod_siren: added new module
mod_sndfile added support to record 16bit for the various rates including 48kHz
mod_sofia: Add filter to "sofia status profile XXX" (MODENDP-138)
mod_sofia: Add force-register-db-domain which works in conjunction with force-register-domain.
mod_sofia: Add optional <variables> and <params> tag to <gateway> tag.
mod_sofia: Challenge the right realm when to_host is outside the users domain. (MODENDP-136)
mod_sofia: Improve notify messages through a proxy (MODENDP-147)
mod_sofia: MWI for multiple domains (MODAPP-126)
mod_sofia: Move "a=sendrecv" from session to media section of SDP (MODENDP-148)
mod_sofia: add 200 OK re-invite without sdp
mod_sofia: add custom sofia::gateway_state event (MODENDP-112)
mod_sofia: add fire events for the refer SIP NOTIFY event package (MODENDP-152)
mod_sofia: add more params for xml_curl directory lookup
mod_sofia: add new auto vals for challenge-realm param <param name="challenge-realm" value="auto_from|auto_to|<hardcoded_val>"/>
mod_sofia: add option to turn of auto_restart of sofia profiles on ip change
mod_sofia: add params to use sip callid as uuid on inbound calls and uuid as sip callid on outbound calls
mod_sofia: add parsing of Privacy header for privacy info (MODENDP-133)
mod_sofia: add proto_specific_hangup_cause to both legs
mod_sofia: add proxy 3pcc mode
mod_sofia: add redirect variable to channel as well as partner channe (MODENDP-135)
mod_sofia: add sip-forbid-register to user params to refuse to let a certian user register
mod_sofia: add sip: into register-proxy when it's not specified
mod_sofia: add sip_history_info var for inbound invites.
mod_sofia: add sip_via_protocol variable
mod_sofia: add sofia xmlstatus (MODENDP-156)
mod_sofia: add support for params other than Replaces in Refer-To (MODENDP-143)
mod_sofia: add support for profiles sharing databases so that you can have a domain that uses multiple profiles for split dns type setups
mod_sofia: add support for refer transfer involving multiple machines
mod_sofia: add support to send a notify in the invite dialog by specifying the uuid of the call. (SFSIP-92)
mod_sofia: add suppress_from_cidname var to not have display name in from header (MODENDP-153)
mod_sofia: added sip_hangup_disposition variable
mod_sofia: allow send_message and notify events to send a message/notify without a body if needed.
mod_sofia: append -1 .. -N postfix after any X-headers as vars that have the same name
mod_sofia: cache auth_gateway_name in sofia for challenged bye
mod_sofia: cancel proxy or no-media mode if you purposely answer or pre_answer
mod_sofia: correct result code mapping for Unallocated Number (MODENDP-124)
mod_sofia: disable 100rel by default
mod_sofia: don't accept crypto in the RTP/AVP (MODENDP-126)
mod_sofia: don't put CN in sdp answer if it was not in the offer.
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)
mod_sofia: fix Wrong RTP media port destination after reinvite/UNHOLD (SFSIP-82)
mod_sofia: fix bug on linksys where they lie about the ptime and handle linksys transfer problem
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)
mod_sofia: fix dtmf handling of broken info dtmf endpoints
mod_sofia: fix eyebeam presence to be RFC compliant (MODENDP-144)
mod_sofia: fix ip change detection when in proxy mode
mod_sofia: fix register_proxy ignoring the paramaters (MODENDP-121)
mod_sofia: fix remote session refresh triggers request glare (MODENDP-131)
mod_sofia: fix rtp auto adjust running when it should not
mod_sofia: fix rtp sent to wrong port after some re-INVITE scenarios (MODENDP-141)
mod_sofia: fix sending of cn packets across bridge when we shouldn't
mod_sofia: fix sqlite issue with select of the sip contact
mod_sofia: fixed segfault on invalid presence payload
mod_sofia: gateway ping needs to look for 501 (SFSIP-78)
mod_sofia: handle multi contact register responses and register timeout better
mod_sofia: improve gateway resilience
mod_sofia: log ip and port you get reply to invite from
mod_sofia: make multiple-registations=true use the contact method and call-id option to do it the old way
mod_sofia: make proxy mode pull the port from m=image as well
mod_sofia: make register-proxy preserve the url composed from proxy but target the packets to desired address (MODENDP-121)
mod_sofia: many fixes for sonus rtp issues silence_when_idle=400 chanvar to send generated silence duing sleeps etc
mod_sofia: many fixes in presence handling
mod_sofia: passthrough t.38 fixes
mod_sofia: pick ipv4 or ipv6 based on sipip instead of having mixed in sdp
mod_sofia: send NOTIFY on TCP/UDP depending on the SUBSCRIBE (SFSIP-104)
mod_sofia: setting profile option multiple-registrations=contact key multi reg off the contact string
mod_sofia: wait for a reply on refer
mod_soundtouch: fixes and improvements, many options changed (MODAPP-149)
mod_soundtouch: updated to new module api
mod_spidermonkey: Segmentation fault in check_hangup_hook at mod_spidermonkey.c:1589 (MODLANG-74)
mod_spidermonkey: fix bug in apiExecute
mod_spidermonkey: fix memory pool handling and leaks
mod_spidermonkey: limit recursion busting through the stack (FSCORE-202)
mod_spidermonkey: make session.getVariable return blank string not the word false
mod_spidermonkey_curl: add optional content-type arg
mod_spidermonkey_odbc: fix numRows and add numCols
mod_spidermonkey_odbc: fix segfault (MODLANG-75)
mod_stress: new module for voice stress analysis
mod_syslog: don't log blank lines (FSCORE-163)
mod_tone_stream: let silence_stream://0 indicate perpetual silence
mod_vmd: add new module to detect voicemail "beep"
mod_voicemail: Add vm_alternate_greet_id param to directory entry (MODAPP-174)
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)
mod_voicemail: Update mwi light after delete vm via web. (MODAPP-124)
mod_voicemail: add ability to get to options without listening to every saved message (MODAPP-115)
mod_voicemail: add ability to skip greeting when leaving a voicemail. (MODAPP-181)
mod_voicemail: add change-pass-key config file option
mod_voicemail: add forwarding support
mod_voicemail: add local dtmf driven alternat vm pass
mod_voicemail: add proper notification of a vm message being too short
mod_voicemail: add support for auth via a1-hash
mod_voicemail: add the "storage-dir" parameter to be set on a per-user basis (MODAPP-133)
mod_voicemail: add voicemail_greeting_path variable
mod_voicemail: added voicemail_alternate_greet_id variable
mod_voicemail: allow changing of password from voicemail to update user directory if using non-static config (MODAPP-156)
mod_voicemail: created email date (int overflow) (MODAPP-125)
mod_voicemail: don't try to deliver vm when no file was recorded. (MODAPP-133)
mod_voicemail: fix MWI with xml_curl used for directory (MODAPP-176)
mod_voicemail: fix Voicemail messages occasionally lost / stranded (MODAPP-178)
mod_voicemail: fix invalid event after message deleted (MODAPP-170)
mod_voicemail: fix mwi for phones with multiple registrations problem (MODAPP-153)
mod_voicemail: fix voicemail segfault on incorrect password (FSCORE-187)
mod_voicemail: fix voicemail_inject error handling (MODAPP-133)
mod_voicemail: fix voicemail_inject usage api call
mod_voicemail: improve error checking (MODAPP-142)
mod_voicemail: localize notification emails (MODAPP-139)
mod_voicemail: make more multi-domain friendly (MODAPP-162)
mod_voicemail: make playback created file macros optional (MODAPP-150)
mod_voicemail: recognize operator key in more places (MODAPP-159)
mod_voicemail: web interface displays incorrect created / last heard dates (MODAPP-123)
mod_wanpipe: removed
mod_xml_cdr: add https support
mod_xml_cdr: add optional a-leg prefix to xml cdr filenames (MDXMLINT-39)
mod_xml_cdr: add support for fallback webserver for cdr posting (FSCORE-238)
mod_xml_curl: Allow specification of HTTP method, and dynamic expansion of variables in URI. (MDXMLINT-41)
mod_xml_curl: added redirect following (max 10)
mod_xml_ldap: almost a complete rewrite of this module
mod_xml_rpc: allow setting of global realm without a global user (MDXMLINT-45)
mod_xml_rpc: fix multiple segfaults
mod_xml_rpc: fix segfault on originate via http
sofia-sip: updated to 1.12.10 (plus a few patches)
freeswitch (1.0.1)
FIX: prevent intercept race condition that can also be solved with continue_on_fail=originator_cancel
FIX: NULL dereference detected by klockwork (www.klockwork.com)
FIX: don't open failed local stream (MODFORM-9)
FIX: instability in mod_local_stream in failure scenarios
FIX: xmlrpc-c build on OS X 10.4 (FSBUILD-47)
ENHANCEMENT: Added tab completion on many api commands in console
ENHANCEMENT: polycom BLF support
FIX: many sip NAT related fixes in mod_sofia
FIX: support sip unregister with Contact: *
FIX: multiple segfaults in xmlrpc-c
FIX: sip unregister event being skipped
FIX: hangup properly on malformed sip 3pcc calls being used as a way to ping
ADD: enable-3pcc sofia profile param, it is now disabled by default.
ADD: presence events to sip proxy mode
ADD: legs param to cdr_csv
ADD: support for perl as an embedded lanugage
ENHANCEMENT: many new api's and functions to the embedded languages including api support, xml interface support, auto start scripts, and many new objects
CHANGE: python embedded language api changed to match perl, lua, java
FIX: many stability fixes in embedded langauges perl, lua, java, python
ADD: failed_xml_cdr magic channel variable
FIX: access free memory error in mod_sofia when using respond app
ENHNACEMENT: make global_setvar only have 2 fields so you can set foo=bar=blah w/o quotes
FIX: mod_spidermonkey keep hangup hook in the session thread
ENHANCEMENT: mod_ldap added sasl support and search filters
ADD: answered, waitForAnswer and mediaReady methods to embedded language Session object
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)