update changelog for 1.0.1
git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@9163 d0543943-73ff-0310-b7d9-9358b9ac24b2
This commit is contained in:
Michael Jerris
parent
572d58de7c
commit
7ebef5b4ad
413
docs/ChangeLog
413
docs/ChangeLog
|
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@ -1,143 +1,270 @@
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freeswitch (1.0.0)
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||||
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Enhanced sofia sip nat handling
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Many fixes found by Klockwork (www.klocwork.com)
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Added disable_app_log variable
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Fixed mod_local_stream with rates on windows
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Fixed finding of files in rate dirs on windows
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Fixed memory corruption from sofia_contact function
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Added sofia profile param NDLB-received-in-nat-reg-contact
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Added sofia profile param aggressive-nat-detection
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Fixed video sip calls in proxy media mode
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Added bridge_terminate_key var
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Update xmlrpc-c lib to trunk revision from upstream, fix windows xmlrpc
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Enhanced nat handling in proxy media mode in sip
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Add progress media to timetable so you can calculate pdd
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Fixed seg when using unicast on socket when call has no read_codec
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Fixed missed log events on busy box
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Added -bleg to intercept
|
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Enhance configure detection of python
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Fixed build on solaris and freebsd for several modules
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Added param "vm-email-only" to make voicemail sent by email only (previously default behavior)
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Added param "vm-mailto-notify" to allow sending a notification email
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Fixed mod_java build
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Fixed mwi failures for some devices that don't subscribe
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Removed fsapi functions (killchan, transfer, session_displace, reject)
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Removed fsapi functions (session_record, broadcast, hold, media)
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Many updates to sofia-sip library including over 100 fixes
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freeswitch (1.0.rc6)
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Changed to not allow pass_2833 on transcoded calls (it never worked, now it will tell you)
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Enhanced sofia sip nat handling
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Fix libedit build on solaris
|
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Fix session timers in mod_sofia
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Fix conference fire-call
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Change: add var_event down into the endpoints so chans with no parents can still pass options
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Added enable-post-var param to xml_rpc
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Fix mod_lua build on solaris
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Many fixes found by Klockwork (www.klocwork.com)
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Add unregister event in mod_sofia
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Enhance python configure detection
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Add vm_boxcount api func
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Fixed att_xfer issue
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Fix sip now includes the Allow-Events header in more places
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freeswitch (1.0.rc5)
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Changed internal state names to avoid confusion
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Fixed video negotiation
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Enhanced accuracy of windows timer
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Fixed mod_ldap build
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Added dialplan and context to sql table for channels
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Multiple fixes to mod_lua and mod_perl
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Fixed logic bug in fifo causing segfault
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internal changes to sip stack so we can remove a hash redundant to the stack
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Fixed multiple memory leaks in mod_sofia
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Fixed event fetch segfault on sip subscribe
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Fixed segfault on timer rollover in sofia on 64bit
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Fixed audio timing issues in mod_portaudio
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Changed names of sip profiles in default config to avoid confusion
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Fixed memory usage leak-like behavior when playing files requiring resampling
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Removed some unused api's
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Fix rtp timeout when playing moh
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Removed some un-needed libraries and files from tree
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Fixed multiple issues in sip stack including multiple segfaults
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Added support for sip transfers on bypass_media and proxy_media calls
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Added say application
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Fixed --disable-debug configure option
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Enhanced switch_cpp wrapper (and perl, python, lua, java)
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Fixed segfault on inavalid stun response
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Fixed configure help output
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Fixed segfault on mp3 playback
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Fixed assert on invalid sdp (missing m= line)
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Added configurable windows service name
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Fixed proxy mode call transition to non proxy call
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Fixed solaris build of voipcodecs
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Fixed sofia seg when call failure edge case
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freeswitch (1.0.rc4)
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Add tab completion in cli
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Add "inline" dialplan
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Fixed segfault in enum
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Enhance enum to fork dial equal priority entries
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Added auto-reload to enum
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Fixed odbc bug is mod_sofia presence handling
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Add presence for conference and dial an eavesdrop
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Fix stack overflow segfault when recursively parking calls
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Fixed race is sofia registration handling
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Enhance sofia registration, unregister on keep-alive OPTIONS failure
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Added internal routing loop detection/avoidance
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Fixed race in bgapi in event socket
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Fixed vars to execute apps before bridge "bridge_pre_execute_aleg_app" and "bridge_pre_execute_bleg_app"
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Fixed re-setting sound prefix to no prefix after a pharse
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Enhanced setting of bracket vars from originate so they show in the CHANNEL_ORIGINATE event
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Add "enable-timer" and "enable-100rel" options to turn off default behaviors in sofia
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Add originate_timeout to originate vars
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Fixed hanging channels in mod_portaudio
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Added auto time sync on vps migration to different hardware
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Fixed seg on transfer when both legs are not sip
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Added configurable dtmf duration defaults
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Enhanced voicemail, allow interruption of hello message
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Fixed voicemail to not light up light on saved messages
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Enhance mod_amr honor disable dtx in fmtp (MODCODEC-3)
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Fixed bootstrap to install automake dependencies so you can use tarball without same version of automake installed
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Fixed MODLANG-56 (bad audio on originate and javascript streamFile)
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Added hold/unhold dialplan apps
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Enhanced sofia error checking to outlaw 0.0.0.0 in sofia ip params
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Backport fixes from sofia-sip tree
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Fixed MSVC build
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Fixed segfault on sip SUBSCRIBE with Expires: 0
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Added mod_say_zh
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Added --with-pyton and --with-pyton-config configure options
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Added mod_lua
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Enhanced switch_cpp wrapper in core and normalized interfaces for perl, python, lua, and java
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Fixed multiple issues in cpp wrapper and the languages perl, python, lua and java
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Added back mod_perl
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Added sofia gateway option ping to adjust options ping frequency
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Added .net event socket lib to contrib
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Fixed passing of exact response codes of sip across a bridge
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Added mod_reference, reference endpoint module
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Enhanced build so you can now make commented out modules using "make mod_name"
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freeswitch (1.0.rc3)
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Enhance xml menu system
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Fixes upstream from sofia-sip library
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Enhance mod_fifo
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Added close method to ODBC spidermonkey class
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Fix multiple bugs in the cpp wrapper used in mod_java and mod_python
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Fix hung sip channel issue using respond app or on re-invite with bypass media after 1xx or 2xx responses
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freeswitch (1.0.rc2)
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Fixed speex protocol negotiation issues (8k vs 16k)
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Fixed mod_iax race conditions
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Fixed ptime negotiation issues when re-packetizing
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Added ip based acl lists
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freeswitch (1.0.rc1)
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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
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ADD: per user acl in sofia
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FIX: deadlock in mod_portaudio
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ENHANCEMENT: blank username in sip will trigger a lookup for the user "nobody"
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ADD: import variable to import variables from a peer channel at time of originate
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FIX: api type fix for c++ modules when incorrectly using enums
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FIX: eliminate need for escaped , in [] on originate
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ADD: NDLB-force-rport option to force behavior as if rport was sent in the via
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ENHANCEMENT: honor execute_on_answer on outbound legs too
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ADD: execute_on_ring variable
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FIX: Seg fault in CoreSession() class destructor
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ADD: per channel caller id in originate
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ADD: sip_outgoing_call_id variable
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FIX: multiple memory leaks in mod_sofia
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FIX: find_local_ip IPv6 support
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ADD: variable expansion to on execute vars.(FSCORE-114)
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ADD: count optional arg to show calls and show channels (MODAPP-103)
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FIX: MODEVENT-25 (WSAWOULDBLOCK error on socket send in windows) in event socket
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FIX: multiple fixes to the logic in mod_say_zh
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ADD: inter digit timeout to swigged embedded languages getDigits method. (MODLANG-65)
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ADD: Linksys P-RTP-Stat SIP header values (SFSIP-66)
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FIX: small leak in core
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ADD: progress_timeout var to originate
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UPDATE: portaudio library
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FIX: added timeout to iax read
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ADD: 'pa rescan' to portaudio to look for new devices
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FIX: wait for broadcast to start when starting async hold to avoid race
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FIX: mod_rss, don't always play the first news feed
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FIX: mod_rss inverval to use the session inteval (audio problems on 30ms channels)
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ADD: Path: support in mod_sofia on register
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FIX: mod_shout record stream
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ENHANCEMENT: mod_voicemail support for effective_caller_id_name/number
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ADD: url encode/decode api calls
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FIX: "nua()" in debug information in sofia instead of the real function name
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FIX: better handling of sips: uris
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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)
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ADD: mod_yaml
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FIX: segfault on freeswitch startup if installed directories are removed
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FIX: segfault when intercept with inbound_late_negotiation=true set
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FIX: dont flood logs with eavesdrop messages (MODAPP-101)
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FIX: don't destroy a codec that has not been created (MODAPP-101)
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ENHANCEMENT: allows the "eavesdrop_group" variable to contain several groups, comma separated. (MODAPP-101)
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FIX: cross compile (FSBUILD-53)
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FIX: add header that Nuaunce considers mandatory (MODASRTTS-5)
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ADD: write locks to the core and a function to unregister event bindings (adds better ability to unload modules)
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ENHANCEMENT: make modules unbind events and un-reserve subclasses on module unload
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ADD: removable xml hook bindings
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ADD: EventConsumer object to embedded languages so you can make event handlers
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FIX: sending CN with supress-cng true
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FIX: segfault in the event system when trying to remove NULL event
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ADD: flags to turn off srtp auth and rtp auto adj (FSCORE-149 && MODENDP-115)
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FIX: use lighter math and avoid infinite loop in port allocator (FSCORE-148)
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ENHANCEMENT: let conference pin entry start during prompt (MODAPP-111)
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ADD: mod_pocketsphinx
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FIX: Misuse of SQLRowCount, issues with MSSQL (MODAPP-105)
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FIX: segfaults in mod_python with dtmf callback
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ENHANCEMENT: mod_conference auto-record parameter (MODAPP-112)
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ENHANCEMENT: reload support to many modules
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FIX: mod_sofia add replaces to supported header
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ENHANCEMENT: add args callback to sleep so you can process dtmf and events while "sleeping"
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ADD: mod_flite
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ENHANCEMENT: switch_xml converted back to c code and support double globs on windows
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ENHANCEMENT: mod_sofia support for adding and removing gateways without restarting profiles
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ADD: extract contact header info into A channel when unhandled 3xx response is received (MODENDP-116)
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FIX: outbound event_socket + late negotiation
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ADD: copy_xml_cdr variable
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ADD: silence_stream (like tone_stream but silent)
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ADD: module_exists api call
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ADD: emailer implementation for windows
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ADD: wait_for_silence application
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FIX: no error message generated if OS is unable to load a module ( due to dependency/installation issues )
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FIX: segfault in media bugs
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FIX: acl lists not correctly matching all ip adresses
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FIX: mod_spidermonkey exit() does not stop script when called from the hangup callback (return "exit" from the callback)
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FIX: mod_syslog works again
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FIX: crash on terminal resize
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FIX: audio problems on big endian
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ENHANCEMENT: Disable multiple registrations on a per-device basis (MODENDP-117)
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ADD: fifo_consumer_exit_key variable (MODAPP-100)
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ADD: cidr based user auth in mod_sofia
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ADD: uuid_send_dtmf fsapi command (MODAPP-114)
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ADD: server registration fiels to sip_registration database (MODENDP-118)
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FIX: use a variable, realm or to host to find gateway when it's not obvious (handles challenged REFER)
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ADD: timeout to curl run in javascript
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ADD: voicemail_inject fsapi command
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ADD: reboot option for sip phones to flush_inboud_reg sofia profile api command
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FIX: add small padding to end of mp3 to avoid cut off mp3 recording
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FIX: patch multiple SDP connection lines in sdp for proxy media mode (MODENDP-109)
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FIX: don't parse ringback varable in proxy situations
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ADD: per call vm recording ext with vm_message_ext variable
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ADD: sip_bye_h prefix to add headers to bye
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ENHANCEMENT: more interfaces available in show fsapi command
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FIX: don't leak in buffers on realloc fail
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FIX: fail out of a conference call if write fails
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ADD: auto ip-change detection
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ADD: mod_snom
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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)
|
||||
|
||||
|
||||
|
||||
|
|
|
|||
Loading…
Reference in New Issue