-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)
-
-
-
+freeswitch (1.0.1)\r
+\r
+ FIX: prevent intercept race condition that can also be solved with continue_on_fail=originator_cancel\r
+ FIX: NULL dereference detected by klockwork (www.klockwork.com)\r
+ FIX: don't open failed local stream (MODFORM-9)\r
+ FIX: instability in mod_local_stream in failure scenarios\r
+ FIX: xmlrpc-c build on OS X 10.4 (FSBUILD-47)\r
+ ENHANCEMENT: Added tab completion on many api commands in console\r
+ ENHANCEMENT: polycom BLF support\r
+ FIX: many sip NAT related fixes in mod_sofia\r
+ FIX: support sip unregister with Contact: *\r
+ FIX: multiple segfaults in xmlrpc-c\r
+ FIX: sip unregister event being skipped\r
+ FIX: hangup properly on malformed sip 3pcc calls being used as a way to ping\r
+ ADD: enable-3pcc sofia profile param, it is now disabled by default.\r
+ ADD: presence events to sip proxy mode\r
+ ADD: legs param to cdr_csv \r
+ ADD: support for perl as an embedded lanugage\r
+ ENHANCEMENT: many new api's and functions to the embedded languages including api support, xml interface support, auto start scripts, and many new objects\r
+ CHANGE: python embedded language api changed to match perl, lua, java\r
+ FIX: many stability fixes in embedded langauges perl, lua, java, python\r
+ ADD: failed_xml_cdr magic channel variable\r
+ FIX: access free memory error in mod_sofia when using respond app\r
+ ENHNACEMENT: make global_setvar only have 2 fields so you can set foo=bar=blah w/o quotes\r
+ FIX: mod_spidermonkey keep hangup hook in the session thread\r
+ ENHANCEMENT: mod_ldap added sasl support and search filters\r
+ ADD: answered, waitForAnswer and mediaReady methods to embedded language Session object\r
+ ENHANCEMENT: mod_voicemail param change to allow notification emails using templates\r
+ ADD: per user acl in sofia\r
+ FIX: deadlock in mod_portaudio\r
+ ENHANCEMENT: blank username in sip will trigger a lookup for the user "nobody"\r
+ ADD: import variable to import variables from a peer channel at time of originate\r
+ FIX: api type fix for c++ modules when incorrectly using enums\r
+ FIX: eliminate need for escaped , in [] on originate\r
+ ADD: NDLB-force-rport option to force behavior as if rport was sent in the via\r
+ ENHANCEMENT: honor execute_on_answer on outbound legs too\r
+ ADD: execute_on_ring variable\r
+ FIX: Seg fault in CoreSession() class destructor\r
+ ADD: per channel caller id in originate\r
+ ADD: sip_outgoing_call_id variable\r
+ FIX: multiple memory leaks in mod_sofia\r
+ FIX: find_local_ip IPv6 support\r
+ ADD: variable expansion to on execute vars.(FSCORE-114)\r
+ ADD: count optional arg to show calls and show channels (MODAPP-103)\r
+ FIX: MODEVENT-25 (WSAWOULDBLOCK error on socket send in windows) in event socket\r
+ FIX: multiple fixes to the logic in mod_say_zh\r
+ ADD: inter digit timeout to swigged embedded languages getDigits method. (MODLANG-65)\r
+ ADD: Linksys P-RTP-Stat SIP header values (SFSIP-66)\r
+ FIX: small leak in core\r
+ ADD: progress_timeout var to originate\r
+ UPDATE: portaudio library\r
+ FIX: added timeout to iax read\r
+ ADD: 'pa rescan' to portaudio to look for new devices\r
+ FIX: wait for broadcast to start when starting async hold to avoid race\r
+ FIX: mod_rss, don't always play the first news feed\r
+ FIX: mod_rss inverval to use the session inteval (audio problems on 30ms channels)\r
+ ADD: Path: support in mod_sofia on register\r
+ FIX: mod_shout record stream\r
+ ENHANCEMENT: mod_voicemail support for effective_caller_id_name/number\r
+ ADD: url encode/decode api calls\r
+ FIX: "nua()" in debug information in sofia instead of the real function name\r
+ FIX: better handling of sips: uris\r
+ 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)\r
+ ADD: mod_yaml\r
+ FIX: segfault on freeswitch startup if installed directories are removed\r
+ FIX: segfault when intercept with inbound_late_negotiation=true set\r
+ FIX: dont flood logs with eavesdrop messages (MODAPP-101)\r
+ FIX: don't destroy a codec that has not been created (MODAPP-101)\r
+ ENHANCEMENT: allows the "eavesdrop_group" variable to contain several groups, comma separated. (MODAPP-101)\r
+ FIX: cross compile (FSBUILD-53)\r
+ FIX: add header that Nuaunce considers mandatory (MODASRTTS-5)\r
+ ADD: write locks to the core and a function to unregister event bindings (adds better ability to unload modules)\r
+ ENHANCEMENT: make modules unbind events and un-reserve subclasses on module unload\r
+ ADD: removable xml hook bindings\r
+ ADD: EventConsumer object to embedded languages so you can make event handlers\r
+ FIX: sending CN with supress-cng true\r
+ FIX: segfault in the event system when trying to remove NULL event\r
+ ADD: flags to turn off srtp auth and rtp auto adj (FSCORE-149 && MODENDP-115)\r
+ FIX: use lighter math and avoid infinite loop in port allocator (FSCORE-148)\r
+ ENHANCEMENT: let conference pin entry start during prompt (MODAPP-111)\r
+ ADD: mod_pocketsphinx\r
+ FIX: Misuse of SQLRowCount, issues with MSSQL (MODAPP-105)\r
+ FIX: segfaults in mod_python with dtmf callback\r
+ ENHANCEMENT: mod_conference auto-record parameter (MODAPP-112)\r
+ ENHANCEMENT: reload support to many modules\r
+ FIX: mod_sofia add replaces to supported header\r
+ ENHANCEMENT: add args callback to sleep so you can process dtmf and events while "sleeping"\r
+ ADD: mod_flite\r
+ ENHANCEMENT: switch_xml converted back to c code and support double globs on windows\r
+ ENHANCEMENT: mod_sofia support for adding and removing gateways without restarting profiles\r
+ ADD: extract contact header info into A channel when unhandled 3xx response is received (MODENDP-116)\r
+ FIX: outbound event_socket + late negotiation\r
+ ADD: copy_xml_cdr variable\r
+ ADD: silence_stream (like tone_stream but silent)\r
+ ADD: module_exists api call\r
+ ADD: emailer implementation for windows\r
+ ADD: wait_for_silence application\r
+ FIX: no error message generated if OS is unable to load a module ( due to dependency/installation issues )\r
+ FIX: segfault in media bugs\r
+ FIX: acl lists not correctly matching all ip adresses\r
+ FIX: mod_spidermonkey exit() does not stop script when called from the hangup callback (return "exit" from the callback)\r
+ FIX: mod_syslog works again\r
+ FIX: crash on terminal resize\r
+ FIX: audio problems on big endian\r
+ ENHANCEMENT: Disable multiple registrations on a per-device basis (MODENDP-117)\r
+ ADD: fifo_consumer_exit_key variable (MODAPP-100)\r
+ ADD: cidr based user auth in mod_sofia\r
+ ADD: uuid_send_dtmf fsapi command (MODAPP-114)\r
+ ADD: server registration fiels to sip_registration database (MODENDP-118)\r
+ FIX: use a variable, realm or to host to find gateway when it's not obvious (handles challenged REFER)\r
+ ADD: timeout to curl run in javascript\r
+ ADD: voicemail_inject fsapi command\r
+ ADD: reboot option for sip phones to flush_inboud_reg sofia profile api command\r
+ FIX: add small padding to end of mp3 to avoid cut off mp3 recording\r
+ FIX: patch multiple SDP connection lines in sdp for proxy media mode (MODENDP-109)\r
+ FIX: don't parse ringback varable in proxy situations\r
+ ADD: per call vm recording ext with vm_message_ext variable\r
+ ADD: sip_bye_h prefix to add headers to bye\r
+ ENHANCEMENT: more interfaces available in show fsapi command\r
+ FIX: don't leak in buffers on realloc fail\r
+ FIX: fail out of a conference call if write fails\r
+ ADD: auto ip-change detection\r
+ ADD: mod_snom\r
+ FIX: mod_sofia don't send sipfrag on transfer to cisco so they don't hang up the call\r
+\r
+\r
+\r
+freeswitch (1.0.0)\r
+\r
+ Enhanced sofia sip nat handling\r
+ Many fixes found by Klockwork (www.klocwork.com)\r
+ Added disable_app_log variable\r
+ Fixed mod_local_stream with rates on windows\r
+ Fixed finding of files in rate dirs on windows\r
+ Fixed memory corruption from sofia_contact function\r
+ Added sofia profile param NDLB-received-in-nat-reg-contact\r
+ Added sofia profile param aggressive-nat-detection\r
+ Fixed video sip calls in proxy media mode\r
+ Added bridge_terminate_key var\r
+ Update xmlrpc-c lib to trunk revision from upstream, fix windows xmlrpc\r
+ Enhanced nat handling in proxy media mode in sip\r
+ Add progress media to timetable so you can calculate pdd\r
+ Fixed seg when using unicast on socket when call has no read_codec\r
+ Fixed missed log events on busy box\r
+ Added -bleg to intercept\r
+ Enhance configure detection of python\r
+ Fixed build on solaris and freebsd for several modules\r
+ Added param "vm-email-only" to make voicemail sent by email only (previously default behavior)\r
+ Added param "vm-mailto-notify" to allow sending a notification email\r
+ Fixed mod_java build\r
+ Fixed mwi failures for some devices that don't subscribe\r
+ Removed fsapi functions (killchan, transfer, session_displace, reject)\r
+ Removed fsapi functions (session_record, broadcast, hold, media)\r
+ Many updates to sofia-sip library including over 100 fixes\r
+ \r
+freeswitch (1.0.rc6)\r
+\r
+ Changed to not allow pass_2833 on transcoded calls (it never worked, now it will tell you)\r
+ Enhanced sofia sip nat handling\r
+ Fix libedit build on solaris\r
+ Fix session timers in mod_sofia\r
+ Fix conference fire-call\r
+ Change: add var_event down into the endpoints so chans with no parents can still pass options\r
+ Added enable-post-var param to xml_rpc\r
+ Fix mod_lua build on solaris\r
+ Many fixes found by Klockwork (www.klocwork.com)\r
+ Add unregister event in mod_sofia\r
+ Enhance python configure detection\r
+ Add vm_boxcount api func\r
+ Fixed att_xfer issue\r
+ Fix sip now includes the Allow-Events header in more places\r
+\r
+freeswitch (1.0.rc5)\r
+\r
+ Changed internal state names to avoid confusion\r
+ Fixed video negotiation\r
+ Enhanced accuracy of windows timer\r
+ Fixed mod_ldap build\r
+ Added dialplan and context to sql table for channels\r
+ Multiple fixes to mod_lua and mod_perl\r
+ Fixed logic bug in fifo causing segfault\r
+ internal changes to sip stack so we can remove a hash redundant to the stack\r
+ Fixed multiple memory leaks in mod_sofia\r
+ Fixed event fetch segfault on sip subscribe\r
+ Fixed segfault on timer rollover in sofia on 64bit\r
+ Fixed audio timing issues in mod_portaudio\r
+ Changed names of sip profiles in default config to avoid confusion\r
+ Fixed memory usage leak-like behavior when playing files requiring resampling\r
+ Removed some unused api's\r
+ Fix rtp timeout when playing moh\r
+ Removed some un-needed libraries and files from tree\r
+ Fixed multiple issues in sip stack including multiple segfaults\r
+ Added support for sip transfers on bypass_media and proxy_media calls\r
+ Added say application\r
+ Fixed --disable-debug configure option\r
+ Enhanced switch_cpp wrapper (and perl, python, lua, java)\r
+ Fixed segfault on inavalid stun response\r
+ Fixed configure help output\r
+ Fixed segfault on mp3 playback\r
+ Fixed assert on invalid sdp (missing m= line)\r
+ Added configurable windows service name\r
+ Fixed proxy mode call transition to non proxy call\r
+ Fixed solaris build of voipcodecs\r
+ Fixed sofia seg when call failure edge case\r
+\r
+freeswitch (1.0.rc4)\r
+\r
+ Add tab completion in cli\r
+ Add "inline" dialplan\r
+ Fixed segfault in enum\r
+ Enhance enum to fork dial equal priority entries\r
+ Added auto-reload to enum\r
+ Fixed odbc bug is mod_sofia presence handling\r
+ Add presence for conference and dial an eavesdrop\r
+ Fix stack overflow segfault when recursively parking calls\r
+ Fixed race is sofia registration handling\r
+ Enhance sofia registration, unregister on keep-alive OPTIONS failure\r
+ Added internal routing loop detection/avoidance\r
+ Fixed race in bgapi in event socket\r
+ Fixed vars to execute apps before bridge "bridge_pre_execute_aleg_app" and "bridge_pre_execute_bleg_app"\r
+ Fixed re-setting sound prefix to no prefix after a pharse\r
+ Enhanced setting of bracket vars from originate so they show in the CHANNEL_ORIGINATE event\r
+ Add "enable-timer" and "enable-100rel" options to turn off default behaviors in sofia\r
+ Add originate_timeout to originate vars\r
+ Fixed hanging channels in mod_portaudio\r
+ Added auto time sync on vps migration to different hardware\r
+ Fixed seg on transfer when both legs are not sip\r
+ Added configurable dtmf duration defaults\r
+ Enhanced voicemail, allow interruption of hello message\r
+ Fixed voicemail to not light up light on saved messages\r
+ Enhance mod_amr honor disable dtx in fmtp (MODCODEC-3)\r
+ Fixed bootstrap to install automake dependencies so you can use tarball without same version of automake installed\r
+ Fixed MODLANG-56 (bad audio on originate and javascript streamFile)\r
+ Added hold/unhold dialplan apps\r
+ Enhanced sofia error checking to outlaw 0.0.0.0 in sofia ip params\r
+ Backport fixes from sofia-sip tree\r
+ Fixed MSVC build\r
+ Fixed segfault on sip SUBSCRIBE with Expires: 0\r
+ Added mod_say_zh\r
+ Added --with-pyton and --with-pyton-config configure options\r
+ Added mod_lua\r
+ Enhanced switch_cpp wrapper in core and normalized interfaces for perl, python, lua, and java\r
+ Fixed multiple issues in cpp wrapper and the languages perl, python, lua and java\r
+ Added back mod_perl\r
+ Added sofia gateway option ping to adjust options ping frequency\r
+ Added .net event socket lib to contrib\r
+ Fixed passing of exact response codes of sip across a bridge\r
+ Added mod_reference, reference endpoint module\r
+ Enhanced build so you can now make commented out modules using "make mod_name"\r
+ \r
+freeswitch (1.0.rc3)\r
+\r
+ Enhance xml menu system\r
+ Fixes upstream from sofia-sip library\r
+ Enhance mod_fifo\r
+ Added close method to ODBC spidermonkey class\r
+ Fix multiple bugs in the cpp wrapper used in mod_java and mod_python\r
+ Fix hung sip channel issue using respond app or on re-invite with bypass media after 1xx or 2xx responses\r
+\r
+freeswitch (1.0.rc2)\r
+\r
+ Fixed speex protocol negotiation issues (8k vs 16k)\r
+ Fixed mod_iax race conditions\r
+ Fixed ptime negotiation issues when re-packetizing\r
+ Added ip based acl lists\r
+\r
+freeswitch (1.0.rc1)\r
+\r
+\r
+ \r