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freeswitch/debian/changelog
Anthony Minessale b4ba1a2255 changelog 
git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@9164 d0543943-73ff-0310-b7d9-9358b9ac24b2
2008-07-24 11:08:03 +00:00

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freeswitch (1.0.1-1) unstable; urgency=low
* 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
-- Mike Jerris <mike@jerris.com> Thu, 24 Jul 2008 07:00:00 -0500
freeswitch (1.0.1~trunk) unstable; urgency=low
* Updated revision number
* Fixed init problem reported by Jay Binks (FSSCRIPTS-1)
* Added a patch to the debian build system add more features (thanks to Hadley Rich) (FSBUILD-45)
- Added en-us-callie sounds and music on hold packages
- Added recommends and suggests
- Added mod_say_es and mod_say_nl
- Updated descriptions
- Added mod_cdr_csv
* Fixed typos and some errors in the previus patch.
* Modified monit script. Now it should work.
* The debian build system now bootstrap automagically if it's necessary and all scripts are in place.
-- Massimo Cetra <devel@navynet.it> Sun, 6 Jul 2008 16:30:00 +0100
freeswitch (1.0.0-1) unstable; urgency=low
* 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
-- Michael Jerris <mike@jerris.com> Tue, 27 May 2008 01:30:00 -0400
freeswitch (1.0~rc6-1) unstable; urgency=low
* 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
-- Michael Jerris <mike@jerris.com> Tue, 13 May 2008 02:01:00 -0400
freeswitch (1.0~rc5-1) unstable; urgency=low
* 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
-- Michael Jerris <mike@jerris.com> Tue, 13 May 2008 02:01:00 -0400
freeswitch (1.0~8327) unstable; urgency=low
* Adding perl and lua separate packages
* Adding mod_voipcodecs
-- root <root@fs.navynet.it> Tue, 6 May 2008 09:46:26 +0000
freeswitch (1.0~rc4-1) unstable; urgency=low
* 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"
-- Michael Jerris <mike@jerris.com> Wed, 23 Apr 2008 12:58:00 -0400
freeswitch (1.0~rc3-1) unstable; urgency=low
* 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
-- Michael Jerris <mike@jerris.com> Wed, 9 Apr 2008 12:58:22 -0400
freeswitch (1.0~rc2-1) unstable; urgency=low
* 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
-- Michael Jerris <mike@jerris.com> Wed, 9 Apr 2008 12:58:22 -0400
freeswitch (1.0~rc1-1) unstable; urgency=low
* loads of fixes
new cdr-csv module
new spidermonkey-curl module
-- Michal Bielicki <michal.bielicki@voiceworks.pl> Mon, 14 Jan 2008 23:37:04 +0100
freeswitch (1.0~beta3-1) unstable; urgency=low
* Additional scripts for changing the user to freeswitch
Added Startup Scripts
Monit integration
Settings file for integration into init
init.d file
added user freeswitch to own and run all off freeswitch
cleaned up config file control
new upstream release
split off codec pakcages
split off spidermonkey packages
-- Michal Bielicki <michal.bielicki@voiceworks.pl> Tue, 27 Nov 2007 13:20:21 +0100
freeswitch (1.0~beta2-1) unstable; urgency=low
* New upstream release
-- Paul van Genderen <paulvg@member.fsf.org> Wed, 17 Oct 2007 19:32:09 +0200
freeswitch (1.0~beta1-1) unstable; urgency=low
* New packages.
-- Robert McQueen <robot101@debian.org> Sun, 12 Nov 2006 17:32:23 -0500
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