debian: start over

v1.2.stable
Travis Cross 11 years ago
parent 073e405642
commit 9181e8e51b
  1. 875
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  2. 1
      debian/compat
  3. 279
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  4. 538
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  5. 5
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  6. 1
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  7. 1
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  8. 1
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  9. 1
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  11. 6
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  43. 15
      debian/sounds/freeswitch-sounds-en-us-callie/debian/buildsounds.sh
  44. 20
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  45. 66
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  46. 1
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  51. 39
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875
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@ -1,875 +0,0 @@
freeswitch (1.1.beta2.1-1) unstable; urgency=low
* Fixing FS-3449 as well as a bit of cleanup in prep for 1.2 release
packaging. Also a handful of lintian errors/warnings are now fixed.
-- William King <quentusrex@gmail.com> Thu, 19 Apr 2012 19:23:18 -0700
freeswitch (1.0.head-git.master.20110530.1-1) unstable; urgency=low
* added mod_cdr_sqlite
-- Michal Bielicki <michal.bielicki@seventhsignal.de> Mon, 30 May 2011 16:02:02 +0200
freeswitch (1.0.head-git.master.20110402.1-1) unstable; urgency=low
* Added Hebrew lang package
-- Michal Bielicki <michal.bielicki@seventhsignal.de> Sat, 02 Apr 2011 03:12:02 +0200
freeswitch (1.0.head-git.master.20110330.1-1) unstable; urgency=low
* removed mod_file_string since it has been merged into dptoolsa
-- Michal Bielicki <michal.bielicki@seventhsignal.de> Wed, 30 Mar 2011 11:39:02 +0200
freeswitch (1.0.head~git.master.20101222.1-1) unstable; urgency=low
* cleaning work
-- Michal Bielicki <michal.bielicki@seventhsignal.de> Wed, 22 Dec 2010 22:48:02 +0200
freeswitch (1.0.head~git.master.20101015.1-1) unstable; urgency=low
* reintroduced mod_flite
* disabled the patching stuff introduced by Julien ... needs an overwork
* reintroduced mod_tts_commandline
* cleaned up rules file and module make rules
* more trivial changes and updates :)
-- Michal Bielicki <michal.bielicki@seventhsignal.de> Fri, 15 Oct 2010 13:14:02 +0200
freeswitch (1.0.head~git.master.20101014.1-1) unstable; urgency=low
* replaced mod_openzap with mod_freetdm
* added mod_theora
* added mod_codec2
* added mod_amrwb
* added mod_portaudio_stream
* cleaned up rules file and module make rules
* added patches from Julien Duqene (FS-369)
* Various trivial changes and updates :)
-- Michal Bielicki <michal.bielicki@seventhsignal.de> Fri, 15 Oct 2010 05:05:02 +0200
freeswitch (1.0.head~git.master.20100601.2-1) unstable; urgency=low
* Various trivial changes and updates.
* Change upstream package version numbering scheme for unreleased versions:
new format is major.minor.micro~git.branch.date.commits-1
* Change source format to 3.0 (quilt).
* Upgrade debhelper compatibility to version 7.
* Update maintainer data and copyright file; now includes the full text
of the MPL since it is not (yet?) available in /usr/share/common-licenses/
* Build and install mod_file_string.so (FSBUILD-247)
* Work around build failure caused by clean rules for openzap.
* Remove several libraries from explicit dependencies, there were no special
version requirements and they should be picked up by dh_shlibdeps.
* Add dh_makeshlibs and make dh_shlibsdeps work again.
* Move openssl to Suggests: as it is not a required package to install or
run freeswitch.
* Add upstream-convert rule to apply patches, generate orig tarball
and set package version.
* Add check to ensure debian patches are applied before attempting build.
* Fix clean rule to avoid unwanted tree change before building orig.tar
archives with git-buildpackage.
-- Julien Plissonneau Duquene <exp-end-2010-06-88a41947@aqiii.org> Tue, 01 Jun 2010 09:53:44 -0400
freeswitch (1.0.6-1ubuntu1) maverick; urgency=low
[ Gabriel Gunderson ]
* upgrade: Added mod_callcenter and pulled out Python into its own
package.
[ Mathieu Parent ]
* Updated Uploaders list
* Updated Standards-Version to 3.9.1
-- Mathieu Parent <sathieu@debian.org> Thu, 23 Sep 2010 15:34:00 +0200
freeswitch (1.0.4-1ubuntu2) karmic; urgency=low
* upgrade: Add more verbosity when building to make it easier to find build
errors.
* upgrade: Remove the requirement for EXACTLY automake1.9 and change it to
need at least automake 1.9
* upgrade: Add the modules (directory, cluechoo, and valet_parking) to the
build files. These are in the standard build, so they should be here too.
-- William King <quentusrex@gmail.com> Fri, 18 Dec 2009 14:27:42 -0800
freeswitch (1.0.4-1ubuntu1) karmic; urgency=low
* upgrade: Pulling out the sounds into separate source files for easier management.
-- William King <quentusrex@gmail.com> Sun, 15 Nov 2009 16:38:13 -0800
freeswitch (1.0.4-1) unstable; urgency=low
* new
-- Mike Jerris <mike@jerris.com> Mon, 18 Feb 2009 17:39:00 -0500
freeswitch (1.0.3-1) unstable; urgency=low
* build: add targets cd-sounds[-install] and cd-moh[-install] for 48k sounds (r:11151)
* build: autoconf detect odbc library (FSBUILD-8)
* build: fix sound install on windows build (r:11635,11638)
* build: fix configure --sysconfdir (FSBUILD-84)
* build: fix uclibc build (MODLANG-99)
* build: fix adduser in debian (FSBUILD-122, FSBUILD-102)
* core: fix buffering issues (r:11101,11145,11152-11157,11162,11191,11200)
* core: fix c leg no hangup when converting attended to blind transfer before b leg answers (MODENDP-165/r:11061)
* core: fix codec and media handling issues (r:11104)
* core: fix double close of file handles and add recording of native files (r:11108-11113,11482,11483)
* core: fix file resampling issue (r:11090)
* core: fix incorrect call progress timestamps in timetable (r:11186-11187/FSCORE-268)
* core: fix media handling issues (r:11079-11082)
* core: fix multiple 2833 dtmf handling issues (r:11149,11261,11262,11266,11293,11294,11338/FSCORE-266,FSCORE-273)
* core: send more event types verbos bridge,unbridge,park,unpark (r:11097-11098)
* core: Prevent media setup on failed originates (r:11462/FSCORE-279)
* core: fix recorded soundfiles had random data at end of file (r:11491/MODAPP-205)
* core: fix user for windows service (r:11538/FSCORE-277)
* core: modify variable expansion code to expand in more places and to fix potential security issue from injecting variables (r:11569,11570)
* core: look for soundfiles in more locations based on rate (r:11601/MODFORM-23)
* core: state machine veto behavior changed (r:11610)
* core: add enable_file_write_buffering variable (r:11677)
* core: fix garbled audio on media bug during bridge using stateful codecs (FSCORE-288)
* core: fix tone detect running multiple bugs when detecting multiple tones
* core: add {instant_ringback=true} to make ringback not wait for indication to generate ringback
* core: fix segfault from race condition on multiple reloadxml calls (MOODAPP-211)
* core: modify xml locking so phrases do not lock the xml for the duration of playing them
* core: replace resampler with the speexdsp resampler
* core: fix windows calling convention on threads launched that return a value to fix shutdown segfault (FSCORE-298)
* core: do not auto-export origination_caller_id_* to avoid confusion (r:12052)
* core: API visibility support (GCC/SUNCC) (FSCORE-264)
* core: fix leak in exposed event class serialize method (r:12068)
* core: add volume as possible return value from input callback on embedded languages (r:12114)
* core: add resampler to seech handles (r:12141)
* core: add api.getTime to embedded languages (r:12149)
* freeswitch: allow you to specify -htdocs dir at runtime. (r:11614)
* fs_cli: add "debug" command to change the esl debug level at runtime (r:11057)
* iksemel: update to 1.3 (r:11645)
* libesl: fix disconnect failure (r:11078,11083)
* libesl: fix solaris build (r:11067,11068)
* libesl: add c++ wrapper and swigged wrappers for multiple scripting languages
* libg722_1: fix dct4.h code generator to include the "f" (r:11188-11189,11367)
* libilbc: update to new library from Steve Underwood
* mod_amrwb: add amr wideband passthrough codec (r:11971)
* mod_cepstral: fix failure return code handling (MODASRTTS-9)
* mod_conference: add 'conference xml_list' and 'conference [conf_name] xml_list' (r:11062-11063)
* mod_conference: make conference verbose-events param to control if events have all the channel data or not (r:11073-11077)
* mod_conference: add MINTWO flag to end conference when down to 1 participant (r:11523)
* mod_conference: refactor conference record function (r:11626)
* mod_conference: add conference list summary command (MODAPP-197)
* mod_conference: fix Deadlock or coredump on conference commands play, transfer (MODAPP-209)
* mod_dahdi_codec: added (MODCODEC-7)
* mod_dialplan_xml: make previous auto hunt feature optional and off by defaule use auto_hunt=true session or global variable to enable (r:12144)
* mod_dptools: Add failure_causes channel variable (r:12058)
* mod_easyroute: add configuration file example for custom-query (r:11055)
* mod_easyroute: add custom-query configuration option (r:11054)
* mod_easyroute: fix build error when not configured for odbc (r:11478)
* mod_easyroute: fix memory leak (r:11611)
* mod_erlang_event: add ability to spawn a process (module/function) outbound on a specified node. (r:11460,11477)
* mod_erlang_event: Fix some issues with standing up a new outbound listener and cleaning up after a failed session (r:11479)
* mod_erlang_event: Fix setting up a listener for an outbound session if one doesn't already exist (r:11488)
* mod_erlang_event: add "erlang" fscli command (r:11488)
* mod_erlang_event: Monitor spawned outbound processes for premature exits (r:11489)
* mod_erlang_event: Allow the event encoding strategy to be configurable; choices are string or binary (r:11495)
* mod_erlang_event: Allow certain tuple elements to be binaries or strings, to reduce conversion requirements on the erlang side (r:11496)
* mod_erlang_event: Support sending a message to a registered process to request a pid (when spawning won't cut it) (r:11499)
* mod_erlang_event: Ensure events received while a pid session is being created aren't discarded but are queued instead (r:11500)
* mod_erlang_event: Add freeswitch.erl - An erlang module to make talking to mod_erlang_event more painless (r:11525)
* mod_erlang_event: use rpc:call instead of spawn and to make the registered process argument to handlecall optional (r:11542)
* mod_event_socket: add ability to use a comma sep list of events on event-sink create-listener (r:11056)
* mod_event_socket: add debug logging to event-sink (r:11060)
* mod_event_socket: fix race condition (r:11680,12146)
* mod_dptools: add all modifier to break command (r:11557,11558)
* mod_dptools: add sound_test application (r:11658)
* mod_fax: Dont hangup after sending/receiving faxes
* mod_fifo: pause media bugs while not in a bridge (r:11466,11490)
* mod_fifo: allow unpark during chime list playing (r:11555/MODAPP-206)
* mod_fifo: fix outbound fifos doesn't check if the consumer is in the fifo in question. (r:11561/MODAPP-207)
* mod_fifo: Fix segfault when no argument were supplied to fifo_member call (MODAPP-210)
* mod_lcr: added (r:11180,11184,11532,11609)
* mod_limit: fix memory corruption caused by race condition when using limit hash (r:11070-11071)
* mod_limit: Fix transfer bug, fix leak and make the channel hangup if the extension is \!hangup_cause (r:11604,11932)
* mod_limit: add write different channel variables per realm_id (r:11608)
* mod_limit: Make max argument optional on the limit app, set the limit_usage variable to current count after inserting call in the db (r:11955)
* mod_lua: Create empty argv table when no args are passed to a Lua script (r:11559)
* mod_lua: use dll for lua windows build (FSCORE-299)
* mod_openmrcp: removed (r:11176-11179)
* mod_opal: added
* mod_pocketsphinx: fix leak (r:11974)
* mod_portaudio: fix stuck channels on outbound calls (r:11160,11470,11471,11472,11475,11476,11485)
* mod_python: fix build when site dir is not /usr/lib/python2.4 (r:12070)
* mod_say_en: add short form date/time (MODAPP-180)
* mod_sofia: add auto-rtp-bugs profile option to make rtp bug compensation configurable (r :11146-11147)
* mod_sofia: add support in sdp for a=maxptime (r:11103)
* mod_sofia: fix codec change race condition (r:11143)
* mod_sofia: fix notify event wasn't allowing content-length 0 (r:11106/MODENDP-167)
* mod_sofia: fix sending extra sdp in 200 OK when using 100rel in violation of sdp o/a protocol draft-ietf-sipping-sip-offeranswer-10 (r:11088)
* mod_sofia: fix sip_auto_answer=true (r:11069)
* mod_sofia: improve outbound registration error message (r:11059)
* mod_sofia: reset media timeout on re-invite (r:11161)
* mod_sofia: fix segfault due to missing contact header in invite (r:11463/MODENDP-177)
* mod_sofia: allow <params> tag in gateways as well as <variables> with direction inbound/outbound (default both) and call counter (r:11468)
* mod_sofia: add support or SLA, works with Polycom and Snom(Sylantro mode). (r:11562/MODENDP-179)
* mod_sofia: tolerate missing user in the request uri (r:11636)
* mod_sofia: Add purpose=gateways and profile=[name] so xml_curl requests make sense (MDXMLINT-46)
* mod_sofia: Add disable-srv and disable-naptr params to sip profiles (default false) (MODENDP-183)
* mod_sofia: add outbound-proxy param (MODENDP-184)
* mod_sofia: fix segfault with stun-enabled=false (SFSIP-120)
* mod_sofia: Profile Name in Expire Event is incorrect (MODENDP-185)
* mod_sofia: add "scrooge" mode to "inbound-codec-negotiation" (r:11881)
* mod_sofia: Add context to reconfig_sofia (r:12080)
* mod_sofia: fix segfault when calling from a Cisco 7940 using bypass_media (FSCORE-301)
* mod_sofia: ilbc to default to 30 if no mode= fmtp is defined on the inbound (r:12110)
* mod_sofia: fix challenge-realm (r:12113)
* mod_sofia: Segmentation fault when running killgw command on sofia profile without specifying a gateway (MODENDP-189)
* mod_sofia: gateways will inherit the context from its parent unless manually provided (r:12138)
* mod_sndfile: Add IMA ADPCM support (MODFORM-22)
* mod_spidermonkey: fix loading of spidermonkey modules (r:11084-11085)
* mod_spidermonkey: block some unwanted behaviours in session.originate
* mod_spidermonkey_socket: fix gc blocking (MODLANG-97)
* mod_xml_rpc: fixed authentication using @domain syntax (r:11064)
* mod_xml_rpc: fix http content types sent in responses (r:11099,11148,11150)
* mod_voicemail: voicemail insert into the proper fields (MODAPP-190)
* mod_voipcodecs: add G.726 24k (r:12083)
* sofia-sip: update to current sofia-sip repository
* spandsp: sync to latest snapshot and fix windows build
* speex: updated to 1.2rc1
* sqlite: fix random assert on windows (FSCORE-292)
-- Mike Jerris <mike@jerris.com> Mon, 18 Feb 2009 17:39:00 -0500
freeswitch (1.0.2-1) unstable; urgency=low
* 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)
-- Mike Jerris <mike@jerris.com> Mon, 29 Dec 2008 14:46:00 -0500
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 variable 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 previous 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

1
debian/compat vendored

@ -1 +0,0 @@
7

279
debian/control vendored

@ -1,279 +0,0 @@
Source: freeswitch
Section: comm
Priority: extra
Maintainer: FreeSWITCH developers <freeswitch-dev@lists.freeswitch.org>
Uploaders: Michal Bielicki <michal.bielicki@seventhsignal.pl>, Gabriel Gunderson <gabe@gundy.org>, William King <quentusrex@gmail.com>, Mathieu Parent <sathieu@debian.org>
Build-Depends: debhelper (>= 7), wget, automake (>=1.9), autoconf, libtool,
unixodbc-dev, libasound2-dev, libcurl3-openssl-dev|libcurl4-openssl-dev,
libssl-dev, ncurses-dev, libogg-dev, libvorbis-dev, libperl-dev, libgdbm-dev,
libdb-dev, libgnutls-dev, libtiff4-dev, python-dev, libx11-dev, uuid-dev,
libc6-dev (>= 2), bison, gawk
Homepage: http://freeswitch.org/
Standards-Version: 3.9.3
Vcs-Git: git://git.freeswitch.org/freeswitch.git
Vcs-Browser: http://fisheye.freeswitch.org/browse/freeswitch.git
Package: freeswitch
Architecture: any
Depends: ${shlibs:Depends}, ${misc:Depends}
Recommends: freeswitch-lang-en
Suggests: freeswitch-spidermonkey, freeswitch-lua, freeswitch-perl,
freeswitch-sounds-music-8000, monit, openssl
Description: open source telephony platform
FreeSWITCH is an open source telephony platform designed to facilitate the
creation of voice and chat driven products scaling from a soft-phone up to a
soft-switch. It can be used as a simple switching engine, a PBX, a media
gateway or a media server to host IVR applications using simple scripts or XML
to control the callflow.
.
This is the main package that includes the FreeSWITCH daemon and most modules.
Package: freeswitch-dbg
Section: debug
Architecture: any
Depends: ${misc:Depends}, freeswitch (= ${binary:Version})
Description: debugging symbols for FreeSWITCH
FreeSWITCH is an open source telephony platform designed to facilitate the
creation of voice and chat driven products scaling from a soft-phone up to a
soft-switch. It can be used as a simple switching engine, a PBX, a media
gateway or a media server to host IVR applications using simple scripts or XML
to control the callflow.
.
This package includes the debugging symbols useful for debugging FreeSWITCH.
The debugging symbols are used for execution tracing and core dump analysis.
Package: freeswitch-dev
Section: libdevel
Architecture: any
Depends: ${shlibs:Depends}, ${misc:Depends}, freeswitch
Description: development libraries and header files for FreeSWITCH
FreeSWITCH is an open source telephony platform designed to facilitate the
creation of voice and chat driven products scaling from a soft-phone up to a
soft-switch. It can be used as a simple switching engine, a PBX, a media
gateway or a media server to host IVR applications using simple scripts or XML
to control the callflow.
.
This package contains the include files used if you wish to compile a package
which require FreeSWITCH's source file headers.
Package: freeswitch-spidermonkey
Architecture: any
Depends: ${shlibs:Depends}, ${misc:Depends}, freeswitch
Description: Javascript engine for FreeSWITCH
FreeSWITCH is an open source telephony platform designed to facilitate the
creation of voice and chat driven products scaling from a soft-phone up to a
soft-switch. It can be used as a simple switching engine, a PBX, a media
gateway or a media server to host IVR applications using simple scripts or XML
to control the callflow.
.
This package contains the mod_spidermonkey language module.
Package: freeswitch-perl
Architecture: any
Depends: ${shlibs:Depends}, ${misc:Depends}, freeswitch
Description: Perl engine for FreeSWITCH
FreeSWITCH is an open source telephony platform designed to facilitate the
creation of voice and chat driven products scaling from a soft-phone up to a
soft-switch. It can be used as a simple switching engine, a PBX, a media
gateway or a media server to host IVR applications using simple scripts or XML
to control the callflow.
.
This package contains the mod_perl language module.
Package: freeswitch-lua
Architecture: any
Depends: ${shlibs:Depends}, ${misc:Depends}, freeswitch
Description: Lua engine for FreeSWITCH
FreeSWITCH is an open source telephony platform designed to facilitate the
creation of voice and chat driven products scaling from a soft-phone up to a
soft-switch. It can be used as a simple switching engine, a PBX, a media
gateway or a media server to host IVR applications using simple scripts or XML
to control the callflow.
.
This package contains the mod_lua language module.
Package: freeswitch-python
Architecture: any
Depends: ${shlibs:Depends}, ${misc:Depends}, freeswitch
Description: Python engine for FreeSWITCH
FreeSWITCH is an open source telephony platform designed to facilitate the
creation of voice and chat driven products scaling from a soft-phone up to a
soft-switch. It can be used as a simple switching engine, a PBX, a media
gateway or a media server to host IVR applications using simple scripts or XML
to control the callflow.
.
This package contains the mod_python language module.
Package: freeswitch-codec-passthru-g7231
Architecture: any
Depends: ${shlibs:Depends}, ${misc:Depends}, freeswitch
Description: pass through g723.1 codec support for FreeSWITCH
FreeSWITCH is an open source telephony platform designed to facilitate the
creation of voice and chat driven products scaling from a soft-phone up to a
soft-switch. It can be used as a simple switching engine, a PBX, a media
gateway or a media server to host IVR applications using simple scripts or XML
to control the callflow.
.
This package contains the mod_g723_1 codec module.
Package: freeswitch-codec-passthru-amr
Architecture: any
Depends: ${shlibs:Depends}, ${misc:Depends}, freeswitch
Description: pass through AMR codec support for FreeSWITCH
FreeSWITCH is an open source telephony platform designed to facilitate the
creation of voice and chat driven products scaling from a soft-phone up to a
soft-switch. It can be used as a simple switching engine, a PBX, a media
gateway or a media server to host IVR applications using simple scripts or XML
to control the callflow.
.
This package contains the mod_amr codec module.
Package: freeswitch-codec-passthru-amrwb
Architecture: any
Depends: ${shlibs:Depends}, ${misc:Depends}, freeswitch
Description: pass through AMRWB codec support for FreeSWITCH
FreeSWITCH is an open source telephony platform designed to facilitate the
creation of voice and chat driven products scaling from a soft-phone up to a
soft-switch. It can be used as a simple switching engine, a PBX, a media
gateway or a media server to host IVR applications using simple scripts or XML
to control the callflow.
.
This package contains the mod_amrwb codec module.
Package: freeswitch-codec-passthru-g729
Architecture: any
Depends: ${shlibs:Depends}, ${misc:Depends}, freeswitch
Description: pass through g729 codec support for FreeSWITCH
FreeSWITCH is an open source telephony platform designed to facilitate the
creation of voice and chat driven products scaling from a soft-phone up to a
soft-switch. It can be used as a simple switching engine, a PBX, a media
gateway or a media server to host IVR applications using simple scripts or XML
to control the callflow.
.
This package contains the mod_g729 codec module.
Package: freeswitch-lang-en
Architecture: any
Depends: ${shlibs:Depends}, ${misc:Depends}, freeswitch
Recommends: freeswitch-sounds-en-us-callie-8000
Suggests: freeswitch-sounds-en-us-callie-16000, freeswitch-sounds-en-us-callie-32000,
freeswitch-sounds-en-us-callie-48000
Description: English language files for FreeSWITCH
FreeSWITCH is an open source telephony platform designed to facilitate the
creation of voice and chat driven products scaling from a soft-phone up to a
soft-switch. It can be used as a simple switching engine, a PBX, a media
gateway or a media server to host IVR applications using simple scripts or XML
to control the callflow.
.
This package contains the mod_say_en module and available language
configuration files.
Package: freeswitch-lang-de
Architecture: any
Depends: ${shlibs:Depends}, ${misc:Depends}, freeswitch
Description: German language files for FreeSWITCH
FreeSWITCH is an open source telephony platform designed to facilitate the
creation of voice and chat driven products scaling from a soft-phone up to a
soft-switch. It can be used as a simple switching engine, a PBX, a media
gateway or a media server to host IVR applications using simple scripts or XML
to control the callflow.
.
This package contains the mod_say_de module and available language
configuration files.
Package: freeswitch-lang-fr
Architecture: any
Depends: ${shlibs:Depends}, ${misc:Depends}, freeswitch
Description: French language files for FreeSWITCH
FreeSWITCH is an open source telephony platform designed to facilitate the
creation of voice and chat driven products scaling from a soft-phone up to a
soft-switch. It can be used as a simple switching engine, a PBX, a media
gateway or a media server to host IVR applications using simple scripts or XML
to control the callflow.
.
This package contains the mod_say_fr module and available language
configuration files.
Package: freeswitch-lang-it
Architecture: any
Depends: ${shlibs:Depends}, ${misc:Depends}, freeswitch
Description: Italian language files for FreeSWITCH
FreeSWITCH is an open source telephony platform designed to facilitate the
creation of voice and chat driven products scaling from a soft-phone up to a
soft-switch. It can be used as a simple switching engine, a PBX, a media
gateway or a media server to host IVR applications using simple scripts or XML
to control the callflow.
.
This package contains the mod_say_it module and available language
configuration files.
Package: freeswitch-lang-es
Architecture: any
Depends: ${shlibs:Depends}, ${misc:Depends}, freeswitch
Description: Spanish language files for FreeSWITCH
FreeSWITCH is an open source telephony platform designed to facilitate the
creation of voice and chat driven products scaling from a soft-phone up to a
soft-switch. It can be used as a simple switching engine, a PBX, a media
gateway or a media server to host IVR applications using simple scripts or XML
to control the callflow.
.
This package contains the mod_say_es module and available language
configuration files.
Package: freeswitch-lang-nl
Architecture: any
Depends: ${shlibs:Depends}, ${misc:Depends}, freeswitch
Description: Dutch language files for FreeSWITCH
FreeSWITCH is an open source telephony platform designed to facilitate the
creation of voice and chat driven products scaling from a soft-phone up to a
soft-switch. It can be used as a simple switching engine, a PBX, a media
gateway or a media server to host IVR applications using simple scripts or XML
to control the callflow.
.
This package contains the mod_say_nl module and available language
configuration files.
Package: freeswitch-lang-ru
Architecture: any
Depends: ${shlibs:Depends}, ${misc:Depends}, freeswitch
Recommends: freeswitch-sounds-ru-ru-elena-8000
Suggests: freeswitch-sounds-ru-ru-elena-16000, freeswitch-sounds-ru-ru-elena-32000,
freeswitch-sounds-ru-ru-elena-48000
Description: Russian language files for FreeSWITCH
FreeSWITCH is an open source telephony platform designed to facilitate the
creation of voice and chat driven products scaling from a soft-phone up to a
soft-switch. It can be used as a simple switching engine, a PBX, a media
gateway or a media server to host IVR applications using simple scripts or XML
to control the callflow.
.
This package contains the mod_say_ru module and available language
configuration files.
Package: freeswitch-lang-he
Architecture: any
Depends: ${shlibs:Depends}, ${misc:Depends}, freeswitch
Description: Hebrew language files for FreeSWITCH
FreeSWITCH is an open source telephony platform designed to facilitate the
creation of voice and chat driven products scaling from a soft-phone up to a
soft-switch. It can be used as a simple switching engine, a PBX, a media
gateway or a media server to host IVR applications using simple scripts or XML
to control the callflow.
.
This package contains the mod_say_he module and available language
configuration files.
Package: freeswitch-freetdm
Architecture: any
Depends: ${shlibs:Depends}, ${misc:Depends}, freeswitch
Description: FreeTDM is a signaling and board API abstraction used mainly by the
FreeSWITCH project to place calls in TDM and analog telephony circuits. The library
was previously named "OpenZAP". Sangoma has worked along with the FreeSWITCH
developers in this library so Sangoma's customers can also use it to do custom
development. The library is still under heavy development but the overall API
does not change often. The intention of the library is to present a consistent
API for different telephony signaling stacks and board I/O APIs. FreeTDM can
either be used as a standalone API or along with FreeSWITCH as an endpoint
(mod_freetdm). If you want to use it as a part of FreeSWITCH remember following
the FreeSWITCH configuration section.
This package contains all the freetdm modules and libs and submodules

538
debian/copyright vendored

@ -1,538 +0,0 @@
This package was debianized by Michal Bielicki
<michal.bielicki@seventhsignal.de> on Nov 25, 2007.
The source was downloaded from http://www.freewitch.org/
Upstream maintainers:
Current: Michal Bielicki <michal.bielicki@seventhsignal.de>
Past: Nicholas Amorim <nicholas@montrealconsultoria.com.br>
See changelog.Debian.gz for a full list of contributors.
The Initial Developer of the Original Code is
Anthony Minessale II <anthm@freeswitch.org>
Portions created by the Initial Developer are Copyright (C)
the Initial Developer. All Rights Reserved.
The PRIMARY AUTHORS are (and/or have been):
Anthony Minessale II <anthm@freeswitch.org> - Primary developer of all core
components and many of the included modules. Much of freeswitch is based
on his work.
Michael Jerris <mike@jerris.com> - Windows porter and responsible for the
windows\msvc build system.
And here is an inevitably incomplete list of MUCH-APPRECIATED CONTRIBUTORS --
people who have submitted patches, reported bugs, and generally made Freeswitch
that much better:
Brian K. West - For countless hours of work on BSD and Mac support, finding
countless bugs, and moral support. Xcode project files.
Joshua Colp - For his help making mod_exosip possible (which we are now
getting rid of but oh well), and for just being a swell guy!
Michal "cypromis" Bielicki (michal.bielicki AT voiceworks.pl) - Solaris
porting, and autotools enhancements, debian, rpm and solaris packaging.
James Martelletti <james@nerdc0re.com> - All around cool guy (mod_syslog)
Johny Kadarisman <jkr888@gmail.com>
Yossi Neiman of Cartis Solutions, Inc. <freeswitch AT cartissolutions.com> -
implementation of mod_cdr (perldd, mysql, csv)
Stefan Knoblich - Sofia TLS, various patches and support. Thanks.
Justin Unger - <justinunger at gmail dot com> Lots of help with patches and
SIP testing. Thanks!
Paul D. Tinsley - Various patches and support. <pdt at jackhammer.org>
Ken Rice - <krice AT suspicious.org> - xmlcdr, sofia improvements, load
testing, 1 liners here and there.
Neal Horman <neal at wanlink dot com> - conference improvements, switch_ivr
menu additions and other tweaks.
Johny Kadarisman <jkr888 at gmail.com> - mod_python fixups.
Michael Murdock <mike at mmurdock dot org> - testing, documentation, bug
finding and usability enhancements.
Matt Klein <mklein@nmedia.net>
Jonas Gauffin <jonas at gauffin dot org> - mod_cdr_odbc,
mod_spidermonkey_socket, Bugfixes and additions in mod_spidermonkey_odbc and
mod_spidermonkey, .net event socket library.
Damjan Jovanovic <moctodliamgtavojtodnajmad backwards> - mod_java
Juan Jose Comellas <juanjo@comellas.org> - Patch to switch_utils for arg
parsing.
Dale Thatcher <freeswitch at dalethatcher dot com> - Additions to
mod_conference.
Simon Perreault & Marc Blanchet from Viagenie.ca - IPv6 Support.
A big THANK YOU goes to:
Justin Cassidy - Build related cleanups and automatic build setup.
Bret McDanel - Javascript Documentation, constant feedback and input, many
other things I am sure I am forgetting.
MOZILLA PUBLIC LICENSE
Version 1.1
---------------
1. Definitions.
1.0.1. "Commercial Use" means distribution or otherwise making the
Covered Code available to a third party.
1.1. "Contributor" means each entity that creates or contributes to
the creation of Modifications.
1.2. "Contributor Version" means the combination of the Original
Code, prior Modifications used by a Contributor, and the Modifications
made by that particular Contributor.
1.3. "Covered Code" means the Original Code or Modifications or the
combination of the Original Code and Modifications, in each case
including portions thereof.
1.4. "Electronic Distribution Mechanism" means a mechanism generally
accepted in the software development community for the electronic
transfer of data.
1.5. "Executable" means Covered Code in any form other than Source
Code.
1.6. "Initial Developer" means the individual or entity identified
as the Initial Developer in the Source Code notice required by Exhibit
A.
1.7. "Larger Work" means a work which combines Covered Code or
portions thereof with code not governed by the terms of this License.
1.8. "License" means this document.
1.8.1. "Licensable" means having the right to grant, to the maximum
extent possible, whether at the time of the initial grant or
subsequently acquired, any and all of the rights conveyed herein.
1.9. "Modifications" means any addition to or deletion from the
substance or structure of either the Original Code or any previous
Modifications. When Covered Code is released as a series of files, a
Modification is:
A. Any addition to or deletion from the contents of a file
containing Original Code or previous Modifications.
B. Any new file that contains any part of the Original Code or
previous Modifications.
1.10. "Original Code" means Source Code of computer software code
which is described in the Source Code notice required by Exhibit A as
Original Code, and which, at the time of its release under this
License is not already Covered Code governed by this License.
1.10.1. "Patent Claims" means any patent claim(s), now owned or
hereafter acquired, including without limitation, method, process,
and apparatus claims, in any patent Licensable by grantor.
1.11. "Source Code" means the preferred form of the Covered Code for
making modifications to it, including all modules it contains, plus
any associated interface definition files, scripts used to control
compilation and installation of an Executable, or source code
differential comparisons against either the Original Code or another
well known, available Covered Code of the Contributor's choice. The
Source Code can be in a compressed or archival form, provided the
appropriate decompression or de-archiving software is widely available
for no charge.
1.12. "You" (or "Your") means an individual or a legal entity
exercising rights under, and complying with all of the terms of, this
License or a future version of this License issued under Section 6.1.
For legal entities, "You" includes any entity which controls, is
controlled by, or is under common control with You. For purposes of
this definition, "control" means (a) the power, direct or indirect,
to cause the direction or management of such entity, whether by
contract or otherwise, or (b) ownership of more than fifty percent
(50%) of the outstanding shares or beneficial ownership of such
entity.
2. Source Code License.
2.1. The Initial Developer Grant.
The Initial Developer hereby grants You a world-wide, royalty-free,
non-exclusive license, subject to third party intellectual property
claims:
(a) under intellectual property rights (other than patent or
trademark) Licensable by Initial Developer to use, reproduce,
modify, display, perform, sublicense and distribute the Original
Code (or portions thereof) with or without Modifications, and/or
as part of a Larger Work; and
(b) under Patents Claims infringed by the making, using or
selling of Original Code, to make, have made, use, practice,
sell, and offer for sale, and/or otherwise dispose of the
Original Code (or portions thereof).
(c) the licenses granted in this Section 2.1(a) and (b) are
effective on the date Initial Developer first distributes
Original Code under the terms of this License.
(d) Notwithstanding Section 2.1(b) above, no patent license is
granted: 1) for code that You delete from the Original Code; 2)
separate from the Original Code; or 3) for infringements caused
by: i) the modification of the Original Code or ii) the
combination of the Original Code with other software or devices.
2.2. Contributor Grant.
Subject to third party intellectual property claims, each Contributor
hereby grants You a world-wide, royalty-free, non-exclusive license
(a) under intellectual property rights (other than patent or
trademark) Licensable by Contributor, to use, reproduce, modify,
display, perform, sublicense and distribute the Modifications
created by such Contributor (or portions thereof) either on an
unmodified basis, with other Modifications, as Covered Code
and/or as part of a Larger Work; and
(b) under Patent Claims infringed by the making, using, or
selling of Modifications made by that Contributor either alone
and/or in combination with its Contributor Version (or portions
of such combination), to make, use, sell, offer for sale, have
made, and/or otherwise dispose of: 1) Modifications made by that
Contributor (or portions thereof); and 2) the combination of
Modifications made by that Contributor with its Contributor
Version (or portions of such combination).
(c) the licenses granted in Sections 2.2(a) and 2.2(b) are
effective on the date Contributor first makes Commercial Use of
the Covered Code.
(d) Notwithstanding Section 2.2(b) above, no patent license is
granted: 1) for any code that Contributor has deleted from the
Contributor Version; 2) separate from the Contributor Version;
3) for infringements caused by: i) third party modifications of
Contributor Version or ii) the combination of Modifications made
by that Contributor with other software (except as part of the
Contributor Version) or other devices; or 4) under Patent Claims
infringed by Covered Code in the absence of Modifications made by
that Contributor.
3. Distribution Obligations.
3.1. Application of License.
The Modifications which You create or to which You contribute are
governed by the terms of this License, including without limitation
Section 2.2. The Source Code version of Covered Code may be
distributed only under the terms of this License or a future version
of this License released under Section 6.1, and You must include a
copy of this License with every copy of the Source Code You
distribute. You may not offer or impose any terms on any Source Code
version that alters or restricts the applicable version of this
License or the recipients' rights hereunder. However, You may include
an additional document offering the additional rights described in
Section 3.5.
3.2. Availability of Source Code.
Any Modification which You create or to which You contribute must be
made available in Source Code form under the terms of this License
either on the same media as an Executable version or via an accepted
Electronic Distribution Mechanism to anyone to whom you made an
Executable version available; and if made available via Electronic
Distribution Mechanism, must remain available for at least twelve (12)
months after the date it initially became available, or at least six
(6) months after a subsequent version of that particular Modification
has been made available to such recipients. You are responsible for
ensuring that the Source Code version remains available even if the
Electronic Distribution Mechanism is maintained by a third party.
3.3. Description of Modifications.
You must cause all Covered Code to which You contribute to contain a
file documenting the changes You made to create that Covered Code and
the date of any change. You must include a prominent statement that
the Modification is derived, directly or indirectly, from Original
Code provided by the Initial Developer and including the name of the
Initial Developer in (a) the Source Code, and (b) in any notice in an
Executable version or related documentation in which You describe the
origin or ownership of the Covered Code.
3.4. Intellectual Property Matters
(a) Third Party Claims.
If Contributor has knowledge that a license under a third party's
intellectual property rights is required to exercise the rights
granted by such Contributor under Sections 2.1 or 2.2,
Contributor must include a text file with the Source Code
distribution titled "LEGAL" which describes the claim and the
party making the claim in sufficient detail that a recipient will
know whom to contact. If Contributor obtains such knowledge after
the Modification is made available as described in Section 3.2,
Contributor shall promptly modify the LEGAL file in all copies
Contributor makes available thereafter and shall take other steps
(such as notifying appropriate mailing lists or newsgroups)
reasonably calculated to inform those who received the Covered
Code that new knowledge has been obtained.
(b) Contributor APIs.
If Contributor's Modifications include an application programming
interface and Contributor has knowledge of patent licenses which
are reasonably necessary to implement that API, Contributor must
also include this information in the LEGAL file.
(c) Representations.
Contributor represents that, except as disclosed pursuant to
Section 3.4(a) above, Contributor believes that Contributor's
Modifications are Contributor's original creation(s) and/or
Contributor has sufficient rights to grant the rights conveyed by
this License.
3.5. Required Notices.
You must duplicate the notice in Exhibit A in each file of the Source
Code. If it is not possible to put such notice in a particular Source
Code file due to its structure, then You must include such notice in a
location (such as a relevant directory) where a user would be likely
to look for such a notice. If You created one or more Modification(s)
You may add your name as a Contributor to the notice described in
Exhibit A. You must also duplicate this License in any documentation
for the Source Code where You describe recipients' rights or ownership
rights relating to Covered Code. You may choose to offer, and to
charge a fee for, warranty, support, indemnity or liability
obligations to one or more recipients of Covered Code. However, You
may do so only on Your own behalf, and not on behalf of the Initial
Developer or any Contributor. You must make it absolutely clear than
any such warranty, support, indemnity or liability obligation is
offered by You alone, and You hereby agree to indemnify the Initial
Developer and every Contributor for any liability incurred by the
Initial Developer or such Contributor as a result of warranty,
support, indemnity or liability terms You offer.
3.6. Distribution of Executable Versions.
You may distribute Covered Code in Executable form only if the
requirements of Section 3.1-3.5 have been met for that Covered Code,
and if You include a notice stating that the Source Code version of
the Covered Code is available under the terms of this License,
including a description of how and where You have fulfilled the
obligations of Section 3.2. The notice must be conspicuously included
in any notice in an Executable version, related documentation or
collateral in which You describe recipients' rights relating to the
Covered Code. You may distribute the Executable version of Covered
Code or ownership rights under a license of Your choice, which may
contain terms different from this License, provided that You are in
compliance with the terms of this License and that the license for the
Executable version does not attempt to limit or alter the recipient's
rights in the Source Code version from the rights set forth in this
License. If You distribute the Executable version under a different
license You must make it absolutely clear that any terms which differ
from this License are offered by You alone, not by the Initial
Developer or any Contributor. You hereby agree to indemnify the
Initial Developer and every Contributor for any liability incurred by
the Initial Developer or such Contributor as a result of any such
terms You offer.
3.7. Larger Works.
You may create a Larger Work by combining Covered Code with other code
not governed by the terms of this License and distribute the Larger
Work as a single product. In such a case, You must make sure the
requirements of this License are fulfilled for the Covered Code.
4. Inability to Comply Due to Statute or Regulation.
If it is impossible for You to comply with any of the terms of this
License with respect to some or all of the Covered Code due to
statute, judicial order, or regulation then You must: (a) comply with
the terms of this License to the maximum extent possible; and (b)
describe the limitations and the code they affect. Such description
must be included in the LEGAL file described in Section 3.4 and must
be included with all distributions of the Source Code. Except to the
extent prohibited by statute or regulation, such description must be
sufficiently detailed for a recipient of ordinary skill to be able to
understand it.
5. Application of this License.
This License applies to code to which the Initial Developer has
attached the notice in Exhibit A and to related Covered Code.
6. Versions of the License.
6.1. New Versions.
Netscape Communications Corporation ("Netscape") may publish revised
and/or new versions of the License from time to time. Each version
will be given a distinguishing version number.
6.2. Effect of New Versions.
Once Covered Code has been published under a particular version of the
License, You may always continue to use it under the terms of that
version. You may also choose to use such Covered Code under the terms
of any subsequent version of the License published by Netscape. No one
other than Netscape has the right to modify the terms applicable to
Covered Code created under this License.
6.3. Derivative Works.
If You create or use a modified version of this License (which you may
only do in order to apply it to code which is not already Covered Code
governed by this License), You must (a) rename Your license so that
the phrases "Mozilla", "MOZILLAPL", "MOZPL", "Netscape",
"MPL", "NPL" or any confusingly similar phrase do not appear in your
license (except to note that your license differs from this License)
and (b) otherwise make it clear that Your version of the license
contains terms which differ from the Mozilla Public License and
Netscape Public License. (Filling in the name of the Initial
Developer, Original Code or Contributor in the notice described in
Exhibit A shall not of themselves be deemed to be modifications of
this License.)
7. DISCLAIMER OF WARRANTY.
COVERED CODE IS PROVIDED UNDER THIS LICENSE ON AN "AS IS" BASIS,
WITHOUT WARRANTY OF ANY KIND, EITHER EXPRESSED OR IMPLIED, INCLUDING,
WITHOUT LIMITATION, WARRANTIES THAT THE COVERED CODE IS FREE OF
DEFECTS, MERCHANTABLE, FIT FOR A PARTICULAR PURPOSE OR NON-INFRINGING.
THE ENTIRE RISK AS TO THE QUALITY AND PERFORMANCE OF THE COVERED CODE
IS WITH YOU. SHOULD ANY COVERED CODE PROVE DEFECTIVE IN ANY RESPECT,
YOU (NOT THE INITIAL DEVELOPER OR ANY OTHER CONTRIBUTOR) ASSUME THE
COST OF ANY NECESSARY SERVICING, REPAIR OR CORRECTION. THIS DISCLAIMER
OF WARRANTY CONSTITUTES AN ESSENTIAL PART OF THIS LICENSE. NO USE OF
ANY COVERED CODE IS AUTHORIZED HEREUNDER EXCEPT UNDER THIS DISCLAIMER.
8. TERMINATION.
8.1. This License and the rights granted hereunder will terminate
automatically if You fail to comply with terms herein and fail to cure
such breach within 30 days of becoming aware of the breach. All
sublicenses to the Covered Code which are properly granted shall
survive any termination of this License. Provisions which, by their
nature, must remain in effect beyond the termination of this License
shall survive.
8.2. If You initiate litigation by asserting a patent infringement
claim (excluding declatory judgment actions) against Initial Developer
or a Contributor (the Initial Developer or Contributor against whom
You file such action is referred to as "Participant") alleging that:
(a) such Participant's Contributor Version directly or indirectly
infringes any patent, then any and all rights granted by such
Participant to You under Sections 2.1 and/or 2.2 of this License