diff --git a/tools/build-system/launchpad/sflphone-client-gnome/debian/changelog b/tools/build-system/launchpad/sflphone-client-gnome/debian/changelog
index 003c33d69cbe10ad5a83b1bebc81c43be2f76df3..c6bec4586d4b6391b32588c2eb5265f2bdbce8c6 100644
--- a/tools/build-system/launchpad/sflphone-client-gnome/debian/changelog
+++ b/tools/build-system/launchpad/sflphone-client-gnome/debian/changelog
@@ -1,4 +1,573 @@
-sflphone-client-gnome (0.9.6-SYSVER) SYSTEM; urgency=low
+sflphone-client-gnome (0.9.7~beta~ppa1~karmic) karmic; urgency=low
+
+ ** 0.9.7~beta~ppa1~karmic **
+
+ * [#1933] Cleanup debug
+ * [#1933] Clean up debug
+ * Fix mic
+ * [#1933] Set the IAx format earlier
+ * [#1933] Move IAX sendAudioFromMic outside if (call) statement
+ * [#1933] Fix startstream when offhold in iax and add debug concerning
+ codec neg.
+ * [#2371] sflphone_notify_voice_mail: minor gettext message formatting
+ cleanup
+ * [#2371] select_account_cb: properly gettextize status message
+ * [#2371] show_account_list_config_dialog: properly gettextize status
+ message
+ * INSTALL: Minor tidyup of core install guide
+ * Add /sflphone-client-gnome/src/icons/Makefile to .gitignore
+ * [#2181] Updated OpenSUSE files (tmp)
+ * [#1933] Add debug for codec negociation for iax
+ * [#1933] Get rid of getMicAvail and getMicData in audiolayer (not
+ used anymore)
+ * [#1933] Add "audio codec not determined" error in IAX
+ * [#1933] Test flush data
+ * [#1933] Do not need to start audio stream in iax anymore
+ * [#1933] Protecting pointer
+ * [#2284] Remove more compilation/execution warnings
+ * [#2284] Cleanup debug in client, use DEBUG instead of g_print
+ * [#2284] Clean up uimanager
+ * [#2370] Remove warnings
+ * [#2366] Clean up other debug
+ * [#2366] Clean up debug
+ * [#2366] Call pa_xfree explicitely in writeToSpeaker
+ * [#2284] Remove address book warnings
+ * [#2365] Fixes bad cast
+ * [#2352] Fix continuous ringing when peer hangup and call not yet
+ answered
+ * [#2181] Added version support
+ * [#2181] Fixed some minor issues
+ * [#2360] Moved MainBuffer from AudioLayer to ManagerImpl
+ * [#2352] Makes getMainBuffer() everywhere
+ * [#2352] Use 50 sec latency on pulseaudio stream creation
+ * [#2352] Add alsa debug
+ * [#2359] Update repository documentation
+ * [#2354] Move pulseaudio disconnectAudioStream after stopping main
+ loop
+ * [#2352] Adjust nb byte copied in pulseaudio according to
+ writeableSize
+ * [#2352] Specify pulseaudio tlength parameters using pa_usec_to_bytes
+ * [#2322] Convert italian translation to UTF-8
+ * [#2357] Fixes window size
+ * [#2357] Display only actionnable tool item
+ * [#2333] Update streams parameters
+ * [#2347] Use GNOME user settings for Menu and Toolbar appareance
+ * [#2349] Load/Save properly audio params
+ * [#2322] Update translations from Launchpad
+ * [#2181] Added Francois Marier script
+ * [#2350] Remove non-valid test
+ * [#2181] Updated launchpad packaging
+ * [#2333] Fix Pulseaudio Capture
+ * [#2333] Use pulseaudio ADJUST_LATENCY flag and ALSA RT-SCHEDULING
+ * [#2333] Pulseaudio Interpolate timing
+ * [#2333] Change (again) Pulseaudio settings to fit logiteck usb hdw
+ requirement
+ * [#2333] Adjust pulseaudio fragment size to 4096 (max sflphone's
+ frames per buffer)
+ * [#2284] Remove recurrent compilation warning (g++ linker problem)
+ * [#2333] Safer Audiostream parameters
+ * [#2333] Fix alsa playback to reduce underrun
+ * [#2333] Better audiostream parameters
+ * [#2181] Updated version management
+ * [#2333] Exclusive test in playback loop
+ * [#2181] Updated build system
+ * [#2333] Less underrun with these value
+ * [#2333] Update playback audiostream parameters
+ * [#2333] Lengthen the audio buffer reduce number of underrun in
+ pulseaudio
+ * [#2333] Add ALSA recovery functions for underrun (begin)
+ * [#2333] Add pa_stream_trigger in pulse audio underrun callabck
+ * [#2048] Reduce prebuffering in pulseaudio (which affect incomming
+ calls' plbck)
+ * [#2316] Do not display any icons to the right on the history tab
+ * [#2333] Comment pa_stream_trigger in pulseaudio underrun
+ * [#2333] Modify pulseaudio streams parameters
+ * [#2318] Fix transfer tool button double signal
+ * [#2181] Updated
+ * [#2333] Fix ALSA ringtone
+ * [#2333] Flush all main buffer before starting audio
+ * [#2333] Open/Close Alsa thread between calls while there is no audio
+ * [#2333] Add debug message and test condition on starting playback
+ and capture
+ * [#2181] Fixed gnome client makefile
+ * [#2181] Updated
+ * [#2308] Remove getTelephoneTone debug
+ * [#2308] Change plughw for default in ALSA
+ * [#2308] Oups, forgot to change function name in audiolayertest.cpp
+ * [#2308] Cleanup in pulseaudio code (debug, function name)
+ * [#2308] Fix pulseaudio stream closing assertion failure
+ * [#2308] Moved pulseaudio mainloop locking from AudioStream
+ disconnect stream
+ * [2308] Fix latency at the beginning of a call, when playing DTMF and
+ wehn starting tone
+ * [#2181] Updated karmic
+ * [#2317] [#2319] Fix address book toggle button contextual behaviour
+ * [#2308] Stop stream when refusing a call
+ * [#2308] Stop pulseaudio stream when peer hungup
+ * [#2308] Fix tone and ringtone
+ * [#2312] Display the STUN entry widget when opening the tab
+ * [#2308] Implement two different callbacks for capture/playback in
+ pulseaudio
+ * [#2309] Open/close pulseaudio connections in startStream/stopStream
+ * [2308] Leave pulseaudio stream running, do not cork/uncork them
+ anymore
+ * [#2295] Set gtk file chooser to None if nothing is set in
+ configuration
+ * [#1976] Add codec and conference documentation
+ * [#2209] Fix recording in regard of resamling
+ * [#2297] Update .gitignore
+ * [#2297] Update translation files
+ * [#2297] Add reference to our coding standards
+ * [#2297] Remove old docbook code
+ * [#2296] Reinit tls account settings after modification
+ * [#2253] Add DcBlocker class to remove capture's dc offset
+ * [#2034] Fixes for TLS transport to initialize
+ * [#2284] Add silent build rule + client clean warnings
+ * [#2274] Fix unserialize history items in cilent at startup
+ * [#2274] Complete display name parsing and displaying
+ * [#2274] Parse the Display Name in sip INVITE message
+ * [#2050] Fix capture volume control in ALSA
+ * [#1970] Volume controls disable when using pulseaudio
+ * [#1970] Disable volume controls when using pulseaudio
+ * [#2277] Fix direct ip2ip ZRTP enabling/disabling in ip2ip
+ preferences
+ * [#2181] Added launchpad debian files
+ * [#2181] Added spec files for OSC
+ * [#2274] Set display name for "Contact" sip header as the hostname
+ * [#2181] Fixed daemon issues
+ * [#2181] Fixed gnome client issues
+ * [#1976] Remove warnings - need to fix the transfer
+ * [#2006] Add init is_rec variable in ManagerImpl
+ * [#2006] Update codec display on call selection
+ * [#2006] Restore double click actions in history and contact calltree
+ (GTK)
+ * [#2176] use XDG_CACHE_HOME when initializing sfl.zid file
+ * [#1976] Fix calltree switching from history
+ * [#2209] (Re)Fix cache for zid
+ * [#2209] Clean up debug messages
+ * [#2209] Clean debug messages
+ * [#2209] Fix trasnfering a call during a conference
+ * [#2209] Speex decode must return the number of bytes
+ * [#2209] Change frameSize speex 32kHz
+ * [#2209] Fix speex codec framesize
+ * [#2209] Reinit converterSamplingRate in RTP sessions
+ * [#2209] Change speex ultra wide band framesize
+ * [#1747] Add pixmap data
+ * [#2252] Fix Receiving a server error 488 crashes the callee
+ * [#2209] Fix iax low rate packate sending
+ * [#2209] Clean up debug messages
+ * [#2209] Add resampling changes for IAX
+ * [#2209] Clean up resampling code
+ * [#2209] Fix latency introduced by pulseaudio
+ * [#2209] Fix initialization of mainbuffer's internal sampling rate
+ * [#2176] Fix upsampling buffer size in audiolayer
+ * [#2209] Add dynamic converter sampling rate in audiortp sessions
+ * [#1747] Fixes runtime warnings
+ * [#1747] Remove from repo
+ * [#1747] register our icons to be used as stock icons
+ * [#2209] Fix number of byte in alsa's write to speaker
+ * [#2209] Fix putting non-resampled data in RTP's mainbuffer
+ * [#2209] Add alsa resampler
+ * [#2209] Add a samplerate converter in PulseLayer
+ * [#2209] Add mainbuffer's internal sampling rate and flushall method
+ * [#2176] Add mainbuffer stateInfo debug method
+ * [#2209] Resampling is optimal using SRC_LINEAR not SRC_FASTEST
+ * [#2176] Remove debug recordings
+ * [#2176] Fix Holding a conference participant on new calls
+ * [#2224] Add confID in callable object
+ * [#2176] Fix putting onhold a call participating to a conference when
+ pressing new call
+ * [#2176] Reset auidio buffers when adding streams (rtp, audiolayer)
+ * [#1976] Use xml to describe toolbars - Add a naviguation toolbar
+ * [#2176] Remove conference default_id in joinParticipant
+ * [#2176] Display error message in alsa's snd_pcm_avail_update call
+ * [#2176] Alsa mic avail data debug
+ * [#2176] Add some debug message for mic loss problem
+ * [#2176] Flush mic ring buffer when offholding a call
+ * [#2176] Reset ringbuffers' readpointer when adding main participant
+ * [#2176] Fix getAvailData algorithm
+ * [#2176] Reset ringbuffer's readpointer when adding a new participant
+ to a conference
+ * [#1744] Regex object renamed to Pattern. Previous attempt at
+ providing
+ * [#2176] Fix detach main participant problem when adding new one
+ * [#1976] Use right domain to translate
+ * [#1976] Add xml menu description
+ * [#2176] Store a list of confernece participant in client
+ * [#2176] Fix add participant, joinparticipant methods
+ * [#2181] Do not install dbus-c++ headers + add return value
+ * [#2176] Fix minor call handling instabilities
+ * [#2174] Fix incoming IP call contact address
+ * [#2211] Add test to protect NULL pointer
+ * [#1163] Add Advanced account configuration section
+ * [#2176] Add some usefull comments and debugging info
+ * [#2176] Add conditions to display security icons in conference
+ * [#2176] Fix detaching one participant while keeping communication to
+ others
+ * [#2176] Reenable userActive.svg in call tree
+ * [#2176] Make user active blue (not red)
+ * [#2176] Fix user active picture
+ * [#2176] Fix "hidden" merge conflict in sipvoiplink
+ * [#2176] Remove iax audio stream on peer hungup
+ * [#2174] Multiple UDP transports functional (TESTED with 2 accounts
+ and 3 calls)
+ * [#2176] Fix fix audio stream binding in iax
+ * [#2174] Create a default UDP transport + use tp selector for dialogs
+ also
+ * [#2176] Register iax audio stream in mainbuffer
+ * [#2176] Fix getAudioCodecName in IAXvoipLink
+ * [#2176] Fix iax account init
+ * [#2176] Handle multiple account using the same sip transport
+ * [#2165] Add .png files
+ * [#2176] Small fixes concerning dtmf
+ * [#2176] Fix make uninstall in codecs
+ * [#2174] remove stund makefile generation
+ * [#2176] Add conference lock
+ * [#2174] Add transport selector for multiple accounts
+ * [#2176] Change userActive picture from red to blue
+ * [#2176] Fix security pixbuff in calltree
+ * [#2176] Replace sfl.zid in .cache/sflphone instead of .sflphone
+ * [#2176] Fix add call description
+ * [#2176] Remove detach button from toolbar
+ * [#2176] Fix calltree call description state and state code in
+ conferences
+ * [#2176] Fix pulse audio double free
+ * [#2176] Fix conference selection
+ * [#2174] Clean up - remove stun settings in client network
+ configuration panel
+ * [#2174] Remove voviva stun code
+ * [#2174] Rsolve STUN with pjsip - DO NOT WORK
+ * [#2165] Add user svg
+ * [#2165] Debugging sip call failed
+ * [#929] Link against uuid if installed
+ * Oops
+ * Fixed bugs related to libsexy (with GTK < 2.16)
+ * [#929] Remove uuid-dev dependency in the core
+ * [#2165] Debugging no negociated codecs at communicatio start
+ * [#2165] Fix calltree bug (gtktreestore instead of gtkliststore)
+ * [#2165] Fix several merge problems
+ * Updated opensuse packaging script
+ * [#1163] Add missing figures
+ * [#1163] Update INSTALL file
+ * [#2165] Fix IAX
+ * [#2165] Add recordabe interface
+ * [#2165] Finish recording refactoring for call (not for conference)
+ * [#2165] Enable speaker recording for two different calls
+ simultanously
+ * [#2165] Implement call recording using the Recordable interface
+ * [#2165] Add get and set to AudioLayer's audio recorder
+ * [#2165] Add class recordable from which inherit call and conference
+ * [#2006] Fix G722 and Speex 8khz codec conferencing
+ * [#2006] add recording of audio buffers
+ * [#1163] Add general settings section
+ * [#1163] Fixes makefile error
+ * [#2006] Fix some minor issues
+ * [#2006] Drag a conference call on another conference call
+ (difference conferences)
+ * [#2006] Fix dragging a conference on itself
+ * [#1744] Integrating some of the needed regular expression patterns
+ in order
+ * COmplete call features
+ * [#1744] Added support for named subgroup in the Regex object. Also,
+ new
+ * [#1744] Adds thread safety features, compile() and setPattern()
+ methods to the Regex class.
+ * [#1744] Fix inconsistency in the finditer method from the last
+ commit.
+ * [#1744] Added regex pattern object built on top of libpcre. To be
+ used
+ * [#1744] Initial commit towards implementing RFC4568. Unimplemented
+ in the
+ * [#2157] Hide "security" and "advanced" tabs for IAX under account
+ * [#1163] Add call features section
+ * [#2006] Add joinConference capabilities
+ * [#2006] Add dbus joinConference signal
+ * [#2006] Drag a conference call onto a conference to add it
+ * [#1163] Add addressbook section
+ * [#2006] Drag a conference call onto a single call to create a
+ conference
+ * [#2006] Expand rows automatically
+ * [#2006] Add minimal multiple conference handling
+ * [#2006] Add atached/detached conference icons
+ * [#2006] Add function processRemainingParticipant
+ * [#2006] Deep refactoring, fix hangup bug
+ * [#1163] Update documentation - Accounts part
+ * [#1976] Integrate user doc to gnome client build system
+ * [#2122] Remove double inclusion in dbus-c++/src/Makefile.am
+ * Remove pjproject version number
+ * [#2006] Fix peerHungup
+ * [#1976] Make Yelp accessible from the GNOME client (need to install
+ the sflphone.xml first)
+ * [#2006] Fix multiconferencing hangup
+ * [#2006] Fix hangup calls in a conference
+ * [#2150] Make IAx2 reappear
+ * [#2006] Fix detach participant on multiple call
+ * [#2006] Can remove rining call from a conference
+ * [#2006] Reinit confID when removing a participant
+ * [#2006] Remove get isCurrentCAll in hangup/peerhungup (SipVoipLink)
+ * [#2006] Fix refuse call
+ * [#2006] Fix answerring incoming call
+ * [#2006] Refactor conference's participant list
+ * [#2101] Re-integrate test compilation in main build system
+ * [#2101] Make the test directory compile
+ * [#2136] Restore history functionality
+ * [#2006] Fix binding main participant to himself
+ * [#2006] Fix add current/incoming/onHold participant to an existing
+ conference
+ * [#2006] Fix add incoming calls to an already created conference
+ * [#2006] Fix remove stream
+ * [#2006] Fix detachParticipant/removeParticipant switchCall ids
+ * [#2006] Fix adding a call in conference having state "CURRENT"
+ * [#2006] Remove/add main participant from conferences
+ * [#2006] Hold/unHold conference
+ * [#2006] Detach a partcipant from drag n drop
+ * [#2006] Hangup a conference
+ * [#2006] Add hold/unhold conference dbus messages
+ * [#2034] gtk-ui fix under the "basic" tab.
+ * [#2006] Fix dragging calls on conference calls
+ * [#2006] Fix detach participant from a conference
+ * [#2034] Added default message is status bar under the account config
+ dialog
+ * [#2112] Fix a crashed caused when a non-md5 password was sent to
+ pjsip.
+ * [#2006] Detach participant by ID
+ * [#2006] Fix addParticipant method in managerImpl to handle
+ incoming/answered calls
+ * [#2006] Add addParticipant method in managerimpl and related dbus
+ messages
+ * [#2111] Added the ability to configure zrtp on sip.sflphone.org from
+ * [#2106] Fixed problem in the account assistant under gtk-ui. Also,
+ assistant.c
+ * [#2006] Fix dragging a conference call on another conference call
+ (same conference)
+ * [#1904] Small UI fix. Assistant was moved from "Call" to "Edit"
+ menu.
+ * [#1904] Fix a wrong label under gtk-ui.
+ * [#2034] Renaming and source code splitting.
+ * [#2034] Status bar added to account window to better reflect the
+ registration
+ * [#2006] Make calltree_remove_call recursive (for GtkTreeStore)
+ * [#1110] Small gtk-UI fix in the account window (alignment).
+ * [#2006] Fix remove conference, display children which are still
+ active
+ * [#2006] Recursive function call in calltree_update_call
+ * [#2006] Add multilayered capabilities to calltree (GtkTreeStore)
+ * [#2006] Implement remove conference in calltree
+ * [#2034] Now useless as Direct Ip calls settings moved under
+ Preferences.
+ * [#2034] Edit/add buttons were set insensitive all the time under
+ gtk-ui.
+ * [#1887] Information about the state of the current SIP call is
+ displayed
+ * [#2006] Add call tree remove callback
+ * [#2006] Fix create_conference function
+ * [#2006] Update conference_added_cb to add new conference to the list
+ * [#812] Added new tab under GTK-ui Preferences. Moving Direct Ip
+ Calls from
+ * [#2121] Disable temporarily test compilation
+ * [#2006] Fix conferencelist to handle conference_obj_t instead of
+ gchar
+ * [#2006] Add conference_obj structure
+ * [#2121] Update version
+ * [#2006] Fix conference selection
+ * [#2101] Use the new source tree to fetch the right object files
+ * [#2006] Add conference in calltree
+ * [#2006] Add Dbus signal conference added/removed/changed
+ * [#2006] Add getConferenceDetails call on dbus
+ * [#1904] Registration expire now appears as a spin box under gtk-ui.
+ * [#812] Fixing a segmentation fault caused by a non-existing account
+ ID
+ * [#2006] Add getConfList method over dbus
+ * [#2006] Add a conferencelist data structure in client-gnome
+ * [#812] Defaults value are now sent if a non-existing account is
+ requested
+ * [#2006] Add sflphone action sflphone_join_participant
+ * [#2006] Fix buffer read pointer problem deletion
+ * [pjsip] Attempt at fixing via header incompatibility with
+ Freeswitch.
+ * [#1797] forget something
+ * [#2006] Add call new state conferencing in deamon
+ * [#2006] Remove addParticipant method for conference, use
+ joinParticipant only
+ * [#1163] Update INSTALL documentation
+ * [#812] Msec/sec values were not taken into account.
+ * [#1797] Make pjproject-1.4 compile
+ * [#2006] Add Detach participant method
+ * [#2006] Dragndrop fully functional with INCOMING and HOLD call
+ * [#1797] Add pjproject-1.4
+ * [#1797] Remove pjproject-1.0.3
+ * [#2006] Get call state in conference related function
+ * [#2006] Add joinParticipant (conference) method in ManagerImpl
+ * [#2006] Add joinConference DBUS message
+ * [#2006] Store the previously selected call_id on dragndrop
+ * [#2006] Fix GValue pointer unref in selection callback
+ * [#2006] Store dragged call_id
+ * [#2006] Update drag_data_received_cb callback to manipulate CallIDs
+ * [#2006] Add dragndrop signals
+ * [#2006] Set calltree reordable
+ * [#812] Adds the ability to create a TLS listener in case the user
+ requests
+ * [#812] Adds the ability to configure local/published address from
+ * [#1883] Move switchCall in onHoldCall function
+ * [#812] Deals with the published address/port problem when
+ integrating TLS.
+ * [#1883] Switch call id in managerimpl when peerHungUp
+ * [#1883] Switch call id before hangup
+ * [#1883] Add usefull and permanent debug info for conference
+ cretion/deletion
+ * [#812] Fix various segmentation faults related to Direct IP kind of
+ calls.
+ * [#1883] Fix deletion of std::map elements using iterators
+ * [#2014] Add libzrtpcpp build dependency
+ * [#1883] Still some for loop test ambiguity (while loop instead)
+ * [#1883] Fix for loop initial test ambiguity (use while loop instead)
+ * [#1883] We must discard data in urgent ring buffer if data is get in
+ mainbuf
+ * [#1883] Fix availForGet same id for ringbuffer and readpointer
+ * [#812] Match "sips" as a Direct IP Call when the user enter a sip
+ uri
+ * [#812] Fix segmentation fault related to SIP URI creation.
+ * [#812] Towards integrating multiple tls listeners at the same time.
+ This
+ * [#1883] Add debug messages in conference and fix mainbufferTest
+ * [#812] gkt-ui fix. Private key must be fed as a filename and not as-
+ is.
+ * [#812] TLS integration within sipvoiplink and pjsip. Also,
+ configure.ac
+ * [#1883] Fix Alsa/Pulse mallocation
+ * [#1883] Fix data corruption in AudioRtp's micData buffer
+ * [#812] Full dbus integration for all the tls related options under
+ gtk-ui.
+ * [#1883] Fix memory leaks in audiortp session
+ * [#1883] Fix mem leaks in audio rtp
+ * [#812] Fix setAccountDetails where TLS_ENABLE was set to the value
+ * [#812] Small gtk-ui fix.
+ * [#811][#812] Small gtk-ui fix.
+ * [#812] Introduced a mechanism for configuration files that makes
+ possible
+ * [#812] New dbus bindings added. Also, configuration compliance was
+ enforced
+ * [#1881] Remove default buffer from MainBuffer (update unit-tests)
+ * [#1881] Add ring buffer read pointer tests
+ * [#1883] Fix issues in ringbuffer reader pointers
+ * [#2034] Implementing a new configuration dialogue for TLS transport
+ settings
+ * [#1883] Add some usefull debug and safety checks
+ * [#2028] Notify the client with libnotify when the zrtp negotiation
+ failed.
+ * [#811] Harmless no to throw an exception, an makes the application
+ less
+ * [#2028] A minidialog is showed to the user under sflphone-client-
+ gnome
+ * Removed useless file.
+ * Ignoring Makefile in src/widget
+ * [#2027] Fix segmentation fault when showMessage callback is called
+ after
+ * [#2026] keyExchange was set to ZRTP instead of "1"
+ * [#2024] Fix the wrong summary at the end of the assistant.
+ * [#1883] Fix mnagerimpl conference map insertion
+ * [#1883] Add Mutexes in MainBuffer
+ * [#811] Gtk ui was not presenting the right information about zrtp
+ for
+ * [#2023] security icons were not installed in sflphone-client-gnome.
+ * [#2021] Fix a mistake in the readme from sflphone-common that gives
+ wrong
+ * [#811] The current SRTP mode was not properly displayed for the
+ IP2IP
+ * [#1743] Re-implementation of the "automatically remove error dialogs
+ [...]"
+ * [#2017] [#2019] Fix the inability to dial a number and place a
+ registered
+ * [#811] Final re-integration of ZRTP support in the main branch from
+ 0.9.6
+ * [#1883] Fix map insertion methods
+ * [#811] Combo box now is now set to the active key exchange method
+ * [#811] ZRTP options now configurable back again from the Gtk UI.
+ IP2IP
+ * Updated hostname for git clone
+ * [#1883] Add minimal functionalities to create a conference
+ * [#811] re-integration of all the methods and signals on dbus.
+ ManagerImpl
+ * [#811] Got out of a precarious position were nothing would compile.
+ * [#1976] Build documentation squeleton with docbook
+ * [#1883] Add sflphone-client "addParticipant" button for conference
+ * [#1994] Better organize the source directory structure. New
+ subdirectories
+ * [#1883] Add a simple Conference class
+ * [#1882] Use static audio buffer in Pulse and ALSA layer (instead of
+ malloc)
+ * [#811] First commit toward re-integration and refactoring of ZRTP
+ * [#1882] Flush RTP ring buffer before entering mainloop
+ * [#1882] Fixed MainBuffer::UnBinCallID() in case there is no
+ ringbuffer
+ * [#1882] Test (and fixe) high level conference and mixing
+ functionalities
+ * [#1772] Apply patch to compile on fedora (sent by Marcin
+ Zajączkowski <mszpak@wp.pl>)
+ * [#1882] Update Bind, unBind call_id in MainBuffer
+ * [#1959] This adds the ability to store password as an MD5 Hash in
+ the
+ * [#1538] Fixes rules compilation
+ * [#1930][#1931] Fixed a mistake (again) related to index and
+ credential count
+ * [#1753] Remove ILBC from pjproject - Hacks in pjsip
+ * [#1930][#1931] Credential was not selected properly using realm
+ * [#1882] Finilize multiple reading pointer in RingBuffer
+ * [#1538] Remove configure from autogen.sh to respect debian upstream
+ authors policy
+ * [#1773] Remove generated files from repo
+ * [#1791] Use XDG_CACHE_HOME to save pid file
+ * [#1791] Fixes path to save history
+ * [#1791] Fix debian installation scripts
+ * [#1930][#1931] Settings are now taken into account in the server.
+ * [#1882] Add ringbuffer default ring buffer pointer in methods
+ involving mStart
+ * [#1882] Add default ringbuffer pointer
+ * [#1882] Add RingBuffer multiple read pointer basic functionnalities
+ * [#1882] Fix MainBuffer flushData unit test
+ * [#1930][#1931] Ability to save and retreive the configuration from
+ * [#1882] Added Multiple CallID mapping to MainBuffer
+ * [#1791] Not much
+ * [#1791] If XDG env variables are not null but empty, use default
+ ones
+ * [#1791] Make XDG_CONFIG_HOME writable
+ * [#1930][#1931] Partial commit. Not working yet. Cannot delete
+ account
+ * [#1881] Fixed alsa capture latency problem
+ * [#1881] Fixed Alsa capture temporarily
+ * [#1930] [#1931] Partial unbroken commit providing the ability to
+ * [#1881] MainBuffer implemented in AudioLayer/AudioRTP
+ * [#1881] Add discard and flush unit-tests
+ * [#1881] Add discard and flush functionnalites to MainRingBuffer
+ * [#1881] Add availForGet in MainBuffer
+ * [#1881] Add availForPut function to MainBuffer
+ * [#1880] Remove AudioRTP* pointer from SipVoIP (reapered while
+ merging master)
+ * [#1881] Add a map between call id and coresponding ring buffer
+ * [#1855] Refresh pot file and upload on Launchpad
+ * [#1881] MainBuffe now robust to false ids on getData and putData
+ * [#1881] Fix big big big memory leak
+ * [#1881] Add getData and putData to mainBuffer
+ * [#1881] Unit-test basic ring buffer functionnaities
+ * [#1881] Add class MainBuffer and basic buffer creation unit-tests
+ * [#1880] Fix call transfer (step2) issues
+ * [#1880] Moved AudioRtp* pointer from SIPVoIPLink to SIPCall class
+ * [#1791] Add postinst script to keep user data when migrating
+ config/history file
+ * [#1797] Make pjsip compile
+ * [#1777] Code indentation
+ * [#1791] Use XDG_DATA_HOME and XDG_CONFIG_HOME for sflphonedrc and
+ history + unit tests
+ * [#1746] Useless space does not appear anymore when volume sliders
+ and
+ * [#1643] GtkCheckMenuItem is used instead of icons for elements in
+ the
+ * [#1110] [#1668] STUN parameters are now located in the preferences,
+ under
+
+ -- Julien Bonjean <julien.bonjean@savoirfairelinux.com> Fri, 06 Nov 2009 11:20:01 -0500
+
+sflphone-client-gnome (0.9.6-SYSVER) karmic; urgency=low
** 0.9.6 **
@@ -65,7 +634,7 @@ sflphone-client-gnome (0.9.6-SYSVER) SYSTEM; urgency=low
-- SFLphone Automatic Build System <team@sflphone.org> Mon, 27 Jul 2009 09:53:19 -0400
-sflphone-client-gnome (0.9.6~rc2-SYSVER) SYSTEM; urgency=low
+sflphone-client-gnome (0.9.6~rc2-SYSVER) karmic; urgency=low
** 0.9.6~rc2 **
@@ -120,7 +689,7 @@ sflphone-client-gnome (0.9.6~rc2-SYSVER) SYSTEM; urgency=low
-- SFLphone Automatic Build System <team@sflphone.org> Fri, 17 Jul 2009 19:12:58 -0400
-sflphone-client-gnome (0.9.6~rc1-SYSVER) SYSTEM; urgency=low
+sflphone-client-gnome (0.9.6~rc1-SYSVER) karmic; urgency=low
** 0.9.6~rc1 **
@@ -228,7 +797,7 @@ sflphone-client-gnome (0.9.6~rc1-SYSVER) SYSTEM; urgency=low
-- SFLphone Automatic Build System <team@sflphone.org> Fri, 10 Jul 2009 15:50:26 -0400
-sflphone-client-gnome (0.9.6~beta-SYSVER) SYSTEM; urgency=low
+sflphone-client-gnome (0.9.6~beta-SYSVER) karmic; urgency=low
** 0.9.6~beta **
@@ -523,7 +1092,7 @@ sflphone-client-gnome (0.9.6~beta-SYSVER) SYSTEM; urgency=low
-- SFLphone Automatic Build System <team@sflphone.org> Tue, 23 Jun 2009 11:13:42 -0400
-sflphone-client-gnome (0.9.5-SYSVER) SYSTEM; urgency=low
+sflphone-client-gnome (0.9.5-SYSVER) karmic; urgency=low
** 0.9.5 release **
@@ -554,7 +1123,7 @@ sflphone-client-gnome (0.9.5-SYSVER) SYSTEM; urgency=low
-- SFLphone Automatic Build System <team@sflphone.org> Mon, 25 May 2009 11:34:48 -0400
-sflphone-client-gnome (0.9.5-SYSVER~rc2) SYSTEM; urgency=low
+sflphone-client-gnome (0.9.5-SYSVER~rc2) karmic; urgency=low
** 0.9.5 rc2 **
@@ -608,7 +1177,7 @@ sflphone-client-gnome (0.9.5-SYSVER~rc2) SYSTEM; urgency=low
-- SFLphone Automatic Build System <team@sflphone.org> Tue, 19 May 2009 12:08:18 -0400
-sflphone-client-gnome (0.9.5-0ubuntu1~rc1) SYSTEM; urgency=low
+sflphone-client-gnome (0.9.5-0ubuntu1~rc1) karmic; urgency=low
[ SFLphone Project ]
* [#1262] Updated changelogs for version 0.9.5-0ubuntu1 Snapshot 2009-
@@ -637,7 +1206,7 @@ sflphone-client-gnome (0.9.5-0ubuntu1~rc1) SYSTEM; urgency=low
-- Sflphone Project <sflphone@mtl.savoirfairelinux.net> Tue, 05 May 2009 19:16:13 -0400
-sflphone-client-gnome (0.9.5-0ubuntu1~beta) SYSTEM; urgency=low
+sflphone-client-gnome (0.9.5-0ubuntu1~beta) karmic; urgency=low
[ Julien Bonjean ]
* Updated Eclipse stuff
@@ -859,7 +1428,7 @@ sflphone-client-gnome (0.9.5-0ubuntu1~beta) SYSTEM; urgency=low
-- Sflphone Project <sflphone@mtl.savoirfairelinux.net> Mon, 27 Apr 2009 17:00:03 -0400
-sflphone-client-gnome (0.9.4-0ubuntu2) SYSTEM; urgency=low
+sflphone-client-gnome (0.9.4-0ubuntu2) karmic; urgency=low
[ Alexandre Savard ]
* Restore speex and GSM detection
@@ -869,7 +1438,7 @@ sflphone-client-gnome (0.9.4-0ubuntu2) SYSTEM; urgency=low
-- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Wed, 8 Apr 2009 11:29:15 -0500
-sflphone (0.9.4-0ubuntu1) SYSTEM; urgency=low
+sflphone (0.9.4-0ubuntu1) karmic; urgency=low
[ Emmanuel Milou ]
* Integrate DBus-c++ and libiax2 in the main build system
@@ -894,7 +1463,7 @@ sflphone (0.9.4-0ubuntu1) SYSTEM; urgency=low
-- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 3 Apr 2009 18:29:15 -0500
-sflphone (0.9.4-rc1) SYSTEM; urgency=low
+sflphone (0.9.4-rc1) karmic; urgency=low
[ Emmanuel Milou ]
* Fix bug while trying to hold/unhold several simultaneous call
@@ -908,7 +1477,7 @@ sflphone (0.9.4-rc1) SYSTEM; urgency=low
-- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 27 Mar 2009 18:29:15 -0500
-sflphone (0.9.4-0beta1) SYSTEM; urgency=low
+sflphone (0.9.4-0beta1) karmic; urgency=low
[ Alexandre Savard ]
* Display codec used during conversation on the GUI
@@ -924,7 +1493,7 @@ sflphone (0.9.4-0beta1) SYSTEM; urgency=low
-- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 20 Mar 2009 18:29:15 -0500
-sflphone (0.9.3-0ubuntu3) SYSTEM; urgency=low
+sflphone (0.9.3-0ubuntu3) karmic; urgency=low
[ Alexandre Savard ]
* Both playback and record streams in PA_STREAM_CORKED (pulseaudio)
@@ -949,7 +1518,7 @@ sflphone (0.9.3-0ubuntu3) SYSTEM; urgency=low
-- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Sat, 14 Feb 2009 13:29:15 -0500
-sflphone (0.9.3-0ubuntu2) SYSTEM; urgency=low
+sflphone (0.9.3-0ubuntu2) karmic; urgency=low
[ Emmanuel Milou ]
* Add compilation note in README
@@ -1018,7 +1587,7 @@ sflphone (0.9.3-0ubuntu2) SYSTEM; urgency=low
-- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Sat, 14 Feb 2009 13:29:15 -0500
-sflphone (0.9.3-0ubuntu1) SYSTEM; urgency=low
+sflphone (0.9.3-0ubuntu1) karmic; urgency=low
* Remove debug
* Join thread before leaving
@@ -1031,7 +1600,7 @@ sflphone (0.9.3-0ubuntu1) SYSTEM; urgency=low
-- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 06 Feb 2009 19:17:32 -0500
-sflphone (0.9.2-2ubuntu9) SYSTEM; urgency=low
+sflphone (0.9.2-2ubuntu9) karmic; urgency=low
[ Alexandre Savard ]
* Speex audio codec preprocessing initialization
@@ -1059,7 +1628,7 @@ sflphone (0.9.2-2ubuntu9) SYSTEM; urgency=low
-- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Thu, 05 Feb 2009 18:27:53 -0500
-sflphone (0.9.2-2ubuntu8) SYSTEM; urgency=low
+sflphone (0.9.2-2ubuntu8) karmic; urgency=low
[ Emmanuel Milou ]
* Update changelogs
@@ -1103,7 +1672,7 @@ sflphone (0.9.2-2ubuntu8) SYSTEM; urgency=low
-- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Wed, 04 Feb 2009 14:00:30 -0500
-sflphone (0.9.2-2ubuntu7) SYSTEM; urgency=low
+sflphone (0.9.2-2ubuntu7) karmic; urgency=low
[ Emmanuel Milou ]
* Update changelog to 0.9.2-6
@@ -1125,7 +1694,7 @@ sflphone (0.9.2-2ubuntu7) SYSTEM; urgency=low
-- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 23 Jan 2009 17:48:13 -0500
-sflphone (0.9.2-2ubuntu6) SYSTEM; urgency=low
+sflphone (0.9.2-2ubuntu6) karmic; urgency=low
[ Emmanuel Milou ]
* Migrate STUN configuration to the main config window
@@ -1159,7 +1728,7 @@ sflphone (0.9.2-2ubuntu6) SYSTEM; urgency=low
-- Emmanuel Milou <manu@sulfur.inside.savoirfairelinux.net> Fri, 16 Jan 2009 18:19:05 -0500
-sflphone (0.9.2-2ubuntu5) SYSTEM; urgency=low
+sflphone (0.9.2-2ubuntu5) karmic; urgency=low
* Fix memory leak in the pulseaudio callback
* Update debian package generation script
@@ -1175,7 +1744,7 @@ sflphone (0.9.2-2ubuntu5) SYSTEM; urgency=low
-- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Wed, 14 Jan 2009 21:17:20 -0500
-sflphone (0.9.2-2ubuntu4) SYSTEM; urgency=low
+sflphone (0.9.2-2ubuntu4) karmic; urgency=low
[ Emmanuel Milou ]
* add german translation
@@ -1185,7 +1754,7 @@ sflphone (0.9.2-2ubuntu4) SYSTEM; urgency=low
-- Yun Liu <yun.liu@savoirfairelinux.com> Thu, 08 Jan 2009 13:08:51 -0500
-sflphone (0.9.2-2ubuntu3) SYSTEM; urgency=low
+sflphone (0.9.2-2ubuntu3) karmic; urgency=low
[ Emmanuel Milou ]
* The main thread synchronizes the ringtone thread
@@ -1197,13 +1766,13 @@ sflphone (0.9.2-2ubuntu3) SYSTEM; urgency=low
-- Yun Liu <yun.liu@savoirfairelinux.com> Tue, 06 Jan 2009 16:18:38 -0500
-sflphone (0.9.2-2ubuntu2) SYSTEM; urgency=low
+sflphone (0.9.2-2ubuntu2) karmic; urgency=low
* Fix bug ticket #129
-- Yun Liu <yun.liu@savoirfairelinux.com> Wed, 5 Jan 2009 15:54:53 -0500
-sflphone (0.9.2-2ubuntu1) SYSTEM; urgency=low
+sflphone (0.9.2-2ubuntu1) karmic; urgency=low
* Migrate from eXosip library to pjsip
* Add multiple SIP accounts support
diff --git a/tools/build-system/launchpad/sflphone-common/debian/changelog b/tools/build-system/launchpad/sflphone-common/debian/changelog
index 3d95374e2598dae7692cbf901ef52e4ddb27eed6..98e662737c05b07b4651f4ba6d0edfd830ead1f5 100644
--- a/tools/build-system/launchpad/sflphone-common/debian/changelog
+++ b/tools/build-system/launchpad/sflphone-common/debian/changelog
@@ -1,4 +1,573 @@
-sflphone-common (0.9.6-SYSVER) SYSTEM; urgency=low
+sflphone-common (0.9.7~beta~ppa1~karmic) karmic; urgency=low
+
+ ** 0.9.7~beta~ppa1~karmic **
+
+ * [#1933] Cleanup debug
+ * [#1933] Clean up debug
+ * Fix mic
+ * [#1933] Set the IAx format earlier
+ * [#1933] Move IAX sendAudioFromMic outside if (call) statement
+ * [#1933] Fix startstream when offhold in iax and add debug concerning
+ codec neg.
+ * [#2371] sflphone_notify_voice_mail: minor gettext message formatting
+ cleanup
+ * [#2371] select_account_cb: properly gettextize status message
+ * [#2371] show_account_list_config_dialog: properly gettextize status
+ message
+ * INSTALL: Minor tidyup of core install guide
+ * Add /sflphone-client-gnome/src/icons/Makefile to .gitignore
+ * [#2181] Updated OpenSUSE files (tmp)
+ * [#1933] Add debug for codec negociation for iax
+ * [#1933] Get rid of getMicAvail and getMicData in audiolayer (not
+ used anymore)
+ * [#1933] Add "audio codec not determined" error in IAX
+ * [#1933] Test flush data
+ * [#1933] Do not need to start audio stream in iax anymore
+ * [#1933] Protecting pointer
+ * [#2284] Remove more compilation/execution warnings
+ * [#2284] Cleanup debug in client, use DEBUG instead of g_print
+ * [#2284] Clean up uimanager
+ * [#2370] Remove warnings
+ * [#2366] Clean up other debug
+ * [#2366] Clean up debug
+ * [#2366] Call pa_xfree explicitely in writeToSpeaker
+ * [#2284] Remove address book warnings
+ * [#2365] Fixes bad cast
+ * [#2352] Fix continuous ringing when peer hangup and call not yet
+ answered
+ * [#2181] Added version support
+ * [#2181] Fixed some minor issues
+ * [#2360] Moved MainBuffer from AudioLayer to ManagerImpl
+ * [#2352] Makes getMainBuffer() everywhere
+ * [#2352] Use 50 sec latency on pulseaudio stream creation
+ * [#2352] Add alsa debug
+ * [#2359] Update repository documentation
+ * [#2354] Move pulseaudio disconnectAudioStream after stopping main
+ loop
+ * [#2352] Adjust nb byte copied in pulseaudio according to
+ writeableSize
+ * [#2352] Specify pulseaudio tlength parameters using pa_usec_to_bytes
+ * [#2322] Convert italian translation to UTF-8
+ * [#2357] Fixes window size
+ * [#2357] Display only actionnable tool item
+ * [#2333] Update streams parameters
+ * [#2347] Use GNOME user settings for Menu and Toolbar appareance
+ * [#2349] Load/Save properly audio params
+ * [#2322] Update translations from Launchpad
+ * [#2181] Added Francois Marier script
+ * [#2350] Remove non-valid test
+ * [#2181] Updated launchpad packaging
+ * [#2333] Fix Pulseaudio Capture
+ * [#2333] Use pulseaudio ADJUST_LATENCY flag and ALSA RT-SCHEDULING
+ * [#2333] Pulseaudio Interpolate timing
+ * [#2333] Change (again) Pulseaudio settings to fit logiteck usb hdw
+ requirement
+ * [#2333] Adjust pulseaudio fragment size to 4096 (max sflphone's
+ frames per buffer)
+ * [#2284] Remove recurrent compilation warning (g++ linker problem)
+ * [#2333] Safer Audiostream parameters
+ * [#2333] Fix alsa playback to reduce underrun
+ * [#2333] Better audiostream parameters
+ * [#2181] Updated version management
+ * [#2333] Exclusive test in playback loop
+ * [#2181] Updated build system
+ * [#2333] Less underrun with these value
+ * [#2333] Update playback audiostream parameters
+ * [#2333] Lengthen the audio buffer reduce number of underrun in
+ pulseaudio
+ * [#2333] Add ALSA recovery functions for underrun (begin)
+ * [#2333] Add pa_stream_trigger in pulse audio underrun callabck
+ * [#2048] Reduce prebuffering in pulseaudio (which affect incomming
+ calls' plbck)
+ * [#2316] Do not display any icons to the right on the history tab
+ * [#2333] Comment pa_stream_trigger in pulseaudio underrun
+ * [#2333] Modify pulseaudio streams parameters
+ * [#2318] Fix transfer tool button double signal
+ * [#2181] Updated
+ * [#2333] Fix ALSA ringtone
+ * [#2333] Flush all main buffer before starting audio
+ * [#2333] Open/Close Alsa thread between calls while there is no audio
+ * [#2333] Add debug message and test condition on starting playback
+ and capture
+ * [#2181] Fixed gnome client makefile
+ * [#2181] Updated
+ * [#2308] Remove getTelephoneTone debug
+ * [#2308] Change plughw for default in ALSA
+ * [#2308] Oups, forgot to change function name in audiolayertest.cpp
+ * [#2308] Cleanup in pulseaudio code (debug, function name)
+ * [#2308] Fix pulseaudio stream closing assertion failure
+ * [#2308] Moved pulseaudio mainloop locking from AudioStream
+ disconnect stream
+ * [2308] Fix latency at the beginning of a call, when playing DTMF and
+ wehn starting tone
+ * [#2181] Updated karmic
+ * [#2317] [#2319] Fix address book toggle button contextual behaviour
+ * [#2308] Stop stream when refusing a call
+ * [#2308] Stop pulseaudio stream when peer hungup
+ * [#2308] Fix tone and ringtone
+ * [#2312] Display the STUN entry widget when opening the tab
+ * [#2308] Implement two different callbacks for capture/playback in
+ pulseaudio
+ * [#2309] Open/close pulseaudio connections in startStream/stopStream
+ * [2308] Leave pulseaudio stream running, do not cork/uncork them
+ anymore
+ * [#2295] Set gtk file chooser to None if nothing is set in
+ configuration
+ * [#1976] Add codec and conference documentation
+ * [#2209] Fix recording in regard of resamling
+ * [#2297] Update .gitignore
+ * [#2297] Update translation files
+ * [#2297] Add reference to our coding standards
+ * [#2297] Remove old docbook code
+ * [#2296] Reinit tls account settings after modification
+ * [#2253] Add DcBlocker class to remove capture's dc offset
+ * [#2034] Fixes for TLS transport to initialize
+ * [#2284] Add silent build rule + client clean warnings
+ * [#2274] Fix unserialize history items in cilent at startup
+ * [#2274] Complete display name parsing and displaying
+ * [#2274] Parse the Display Name in sip INVITE message
+ * [#2050] Fix capture volume control in ALSA
+ * [#1970] Volume controls disable when using pulseaudio
+ * [#1970] Disable volume controls when using pulseaudio
+ * [#2277] Fix direct ip2ip ZRTP enabling/disabling in ip2ip
+ preferences
+ * [#2181] Added launchpad debian files
+ * [#2181] Added spec files for OSC
+ * [#2274] Set display name for "Contact" sip header as the hostname
+ * [#2181] Fixed daemon issues
+ * [#2181] Fixed gnome client issues
+ * [#1976] Remove warnings - need to fix the transfer
+ * [#2006] Add init is_rec variable in ManagerImpl
+ * [#2006] Update codec display on call selection
+ * [#2006] Restore double click actions in history and contact calltree
+ (GTK)
+ * [#2176] use XDG_CACHE_HOME when initializing sfl.zid file
+ * [#1976] Fix calltree switching from history
+ * [#2209] (Re)Fix cache for zid
+ * [#2209] Clean up debug messages
+ * [#2209] Clean debug messages
+ * [#2209] Fix trasnfering a call during a conference
+ * [#2209] Speex decode must return the number of bytes
+ * [#2209] Change frameSize speex 32kHz
+ * [#2209] Fix speex codec framesize
+ * [#2209] Reinit converterSamplingRate in RTP sessions
+ * [#2209] Change speex ultra wide band framesize
+ * [#1747] Add pixmap data
+ * [#2252] Fix Receiving a server error 488 crashes the callee
+ * [#2209] Fix iax low rate packate sending
+ * [#2209] Clean up debug messages
+ * [#2209] Add resampling changes for IAX
+ * [#2209] Clean up resampling code
+ * [#2209] Fix latency introduced by pulseaudio
+ * [#2209] Fix initialization of mainbuffer's internal sampling rate
+ * [#2176] Fix upsampling buffer size in audiolayer
+ * [#2209] Add dynamic converter sampling rate in audiortp sessions
+ * [#1747] Fixes runtime warnings
+ * [#1747] Remove from repo
+ * [#1747] register our icons to be used as stock icons
+ * [#2209] Fix number of byte in alsa's write to speaker
+ * [#2209] Fix putting non-resampled data in RTP's mainbuffer
+ * [#2209] Add alsa resampler
+ * [#2209] Add a samplerate converter in PulseLayer
+ * [#2209] Add mainbuffer's internal sampling rate and flushall method
+ * [#2176] Add mainbuffer stateInfo debug method
+ * [#2209] Resampling is optimal using SRC_LINEAR not SRC_FASTEST
+ * [#2176] Remove debug recordings
+ * [#2176] Fix Holding a conference participant on new calls
+ * [#2224] Add confID in callable object
+ * [#2176] Fix putting onhold a call participating to a conference when
+ pressing new call
+ * [#2176] Reset auidio buffers when adding streams (rtp, audiolayer)
+ * [#1976] Use xml to describe toolbars - Add a naviguation toolbar
+ * [#2176] Remove conference default_id in joinParticipant
+ * [#2176] Display error message in alsa's snd_pcm_avail_update call
+ * [#2176] Alsa mic avail data debug
+ * [#2176] Add some debug message for mic loss problem
+ * [#2176] Flush mic ring buffer when offholding a call
+ * [#2176] Reset ringbuffers' readpointer when adding main participant
+ * [#2176] Fix getAvailData algorithm
+ * [#2176] Reset ringbuffer's readpointer when adding a new participant
+ to a conference
+ * [#1744] Regex object renamed to Pattern. Previous attempt at
+ providing
+ * [#2176] Fix detach main participant problem when adding new one
+ * [#1976] Use right domain to translate
+ * [#1976] Add xml menu description
+ * [#2176] Store a list of confernece participant in client
+ * [#2176] Fix add participant, joinparticipant methods
+ * [#2181] Do not install dbus-c++ headers + add return value
+ * [#2176] Fix minor call handling instabilities
+ * [#2174] Fix incoming IP call contact address
+ * [#2211] Add test to protect NULL pointer
+ * [#1163] Add Advanced account configuration section
+ * [#2176] Add some usefull comments and debugging info
+ * [#2176] Add conditions to display security icons in conference
+ * [#2176] Fix detaching one participant while keeping communication to
+ others
+ * [#2176] Reenable userActive.svg in call tree
+ * [#2176] Make user active blue (not red)
+ * [#2176] Fix user active picture
+ * [#2176] Fix "hidden" merge conflict in sipvoiplink
+ * [#2176] Remove iax audio stream on peer hungup
+ * [#2174] Multiple UDP transports functional (TESTED with 2 accounts
+ and 3 calls)
+ * [#2176] Fix fix audio stream binding in iax
+ * [#2174] Create a default UDP transport + use tp selector for dialogs
+ also
+ * [#2176] Register iax audio stream in mainbuffer
+ * [#2176] Fix getAudioCodecName in IAXvoipLink
+ * [#2176] Fix iax account init
+ * [#2176] Handle multiple account using the same sip transport
+ * [#2165] Add .png files
+ * [#2176] Small fixes concerning dtmf
+ * [#2176] Fix make uninstall in codecs
+ * [#2174] remove stund makefile generation
+ * [#2176] Add conference lock
+ * [#2174] Add transport selector for multiple accounts
+ * [#2176] Change userActive picture from red to blue
+ * [#2176] Fix security pixbuff in calltree
+ * [#2176] Replace sfl.zid in .cache/sflphone instead of .sflphone
+ * [#2176] Fix add call description
+ * [#2176] Remove detach button from toolbar
+ * [#2176] Fix calltree call description state and state code in
+ conferences
+ * [#2176] Fix pulse audio double free
+ * [#2176] Fix conference selection
+ * [#2174] Clean up - remove stun settings in client network
+ configuration panel
+ * [#2174] Remove voviva stun code
+ * [#2174] Rsolve STUN with pjsip - DO NOT WORK
+ * [#2165] Add user svg
+ * [#2165] Debugging sip call failed
+ * [#929] Link against uuid if installed
+ * Oops
+ * Fixed bugs related to libsexy (with GTK < 2.16)
+ * [#929] Remove uuid-dev dependency in the core
+ * [#2165] Debugging no negociated codecs at communicatio start
+ * [#2165] Fix calltree bug (gtktreestore instead of gtkliststore)
+ * [#2165] Fix several merge problems
+ * Updated opensuse packaging script
+ * [#1163] Add missing figures
+ * [#1163] Update INSTALL file
+ * [#2165] Fix IAX
+ * [#2165] Add recordabe interface
+ * [#2165] Finish recording refactoring for call (not for conference)
+ * [#2165] Enable speaker recording for two different calls
+ simultanously
+ * [#2165] Implement call recording using the Recordable interface
+ * [#2165] Add get and set to AudioLayer's audio recorder
+ * [#2165] Add class recordable from which inherit call and conference
+ * [#2006] Fix G722 and Speex 8khz codec conferencing
+ * [#2006] add recording of audio buffers
+ * [#1163] Add general settings section
+ * [#1163] Fixes makefile error
+ * [#2006] Fix some minor issues
+ * [#2006] Drag a conference call on another conference call
+ (difference conferences)
+ * [#2006] Fix dragging a conference on itself
+ * [#1744] Integrating some of the needed regular expression patterns
+ in order
+ * COmplete call features
+ * [#1744] Added support for named subgroup in the Regex object. Also,
+ new
+ * [#1744] Adds thread safety features, compile() and setPattern()
+ methods to the Regex class.
+ * [#1744] Fix inconsistency in the finditer method from the last
+ commit.
+ * [#1744] Added regex pattern object built on top of libpcre. To be
+ used
+ * [#1744] Initial commit towards implementing RFC4568. Unimplemented
+ in the
+ * [#2157] Hide "security" and "advanced" tabs for IAX under account
+ * [#1163] Add call features section
+ * [#2006] Add joinConference capabilities
+ * [#2006] Add dbus joinConference signal
+ * [#2006] Drag a conference call onto a conference to add it
+ * [#1163] Add addressbook section
+ * [#2006] Drag a conference call onto a single call to create a
+ conference
+ * [#2006] Expand rows automatically
+ * [#2006] Add minimal multiple conference handling
+ * [#2006] Add atached/detached conference icons
+ * [#2006] Add function processRemainingParticipant
+ * [#2006] Deep refactoring, fix hangup bug
+ * [#1163] Update documentation - Accounts part
+ * [#1976] Integrate user doc to gnome client build system
+ * [#2122] Remove double inclusion in dbus-c++/src/Makefile.am
+ * Remove pjproject version number
+ * [#2006] Fix peerHungup
+ * [#1976] Make Yelp accessible from the GNOME client (need to install
+ the sflphone.xml first)
+ * [#2006] Fix multiconferencing hangup
+ * [#2006] Fix hangup calls in a conference
+ * [#2150] Make IAx2 reappear
+ * [#2006] Fix detach participant on multiple call
+ * [#2006] Can remove rining call from a conference
+ * [#2006] Reinit confID when removing a participant
+ * [#2006] Remove get isCurrentCAll in hangup/peerhungup (SipVoipLink)
+ * [#2006] Fix refuse call
+ * [#2006] Fix answerring incoming call
+ * [#2006] Refactor conference's participant list
+ * [#2101] Re-integrate test compilation in main build system
+ * [#2101] Make the test directory compile
+ * [#2136] Restore history functionality
+ * [#2006] Fix binding main participant to himself
+ * [#2006] Fix add current/incoming/onHold participant to an existing
+ conference
+ * [#2006] Fix add incoming calls to an already created conference
+ * [#2006] Fix remove stream
+ * [#2006] Fix detachParticipant/removeParticipant switchCall ids
+ * [#2006] Fix adding a call in conference having state "CURRENT"
+ * [#2006] Remove/add main participant from conferences
+ * [#2006] Hold/unHold conference
+ * [#2006] Detach a partcipant from drag n drop
+ * [#2006] Hangup a conference
+ * [#2006] Add hold/unhold conference dbus messages
+ * [#2034] gtk-ui fix under the "basic" tab.
+ * [#2006] Fix dragging calls on conference calls
+ * [#2006] Fix detach participant from a conference
+ * [#2034] Added default message is status bar under the account config
+ dialog
+ * [#2112] Fix a crashed caused when a non-md5 password was sent to
+ pjsip.
+ * [#2006] Detach participant by ID
+ * [#2006] Fix addParticipant method in managerImpl to handle
+ incoming/answered calls
+ * [#2006] Add addParticipant method in managerimpl and related dbus
+ messages
+ * [#2111] Added the ability to configure zrtp on sip.sflphone.org from
+ * [#2106] Fixed problem in the account assistant under gtk-ui. Also,
+ assistant.c
+ * [#2006] Fix dragging a conference call on another conference call
+ (same conference)
+ * [#1904] Small UI fix. Assistant was moved from "Call" to "Edit"
+ menu.
+ * [#1904] Fix a wrong label under gtk-ui.
+ * [#2034] Renaming and source code splitting.
+ * [#2034] Status bar added to account window to better reflect the
+ registration
+ * [#2006] Make calltree_remove_call recursive (for GtkTreeStore)
+ * [#1110] Small gtk-UI fix in the account window (alignment).
+ * [#2006] Fix remove conference, display children which are still
+ active
+ * [#2006] Recursive function call in calltree_update_call
+ * [#2006] Add multilayered capabilities to calltree (GtkTreeStore)
+ * [#2006] Implement remove conference in calltree
+ * [#2034] Now useless as Direct Ip calls settings moved under
+ Preferences.
+ * [#2034] Edit/add buttons were set insensitive all the time under
+ gtk-ui.
+ * [#1887] Information about the state of the current SIP call is
+ displayed
+ * [#2006] Add call tree remove callback
+ * [#2006] Fix create_conference function
+ * [#2006] Update conference_added_cb to add new conference to the list
+ * [#812] Added new tab under GTK-ui Preferences. Moving Direct Ip
+ Calls from
+ * [#2121] Disable temporarily test compilation
+ * [#2006] Fix conferencelist to handle conference_obj_t instead of
+ gchar
+ * [#2006] Add conference_obj structure
+ * [#2121] Update version
+ * [#2006] Fix conference selection
+ * [#2101] Use the new source tree to fetch the right object files
+ * [#2006] Add conference in calltree
+ * [#2006] Add Dbus signal conference added/removed/changed
+ * [#2006] Add getConferenceDetails call on dbus
+ * [#1904] Registration expire now appears as a spin box under gtk-ui.
+ * [#812] Fixing a segmentation fault caused by a non-existing account
+ ID
+ * [#2006] Add getConfList method over dbus
+ * [#2006] Add a conferencelist data structure in client-gnome
+ * [#812] Defaults value are now sent if a non-existing account is
+ requested
+ * [#2006] Add sflphone action sflphone_join_participant
+ * [#2006] Fix buffer read pointer problem deletion
+ * [pjsip] Attempt at fixing via header incompatibility with
+ Freeswitch.
+ * [#1797] forget something
+ * [#2006] Add call new state conferencing in deamon
+ * [#2006] Remove addParticipant method for conference, use
+ joinParticipant only
+ * [#1163] Update INSTALL documentation
+ * [#812] Msec/sec values were not taken into account.
+ * [#1797] Make pjproject-1.4 compile
+ * [#2006] Add Detach participant method
+ * [#2006] Dragndrop fully functional with INCOMING and HOLD call
+ * [#1797] Add pjproject-1.4
+ * [#1797] Remove pjproject-1.0.3
+ * [#2006] Get call state in conference related function
+ * [#2006] Add joinParticipant (conference) method in ManagerImpl
+ * [#2006] Add joinConference DBUS message
+ * [#2006] Store the previously selected call_id on dragndrop
+ * [#2006] Fix GValue pointer unref in selection callback
+ * [#2006] Store dragged call_id
+ * [#2006] Update drag_data_received_cb callback to manipulate CallIDs
+ * [#2006] Add dragndrop signals
+ * [#2006] Set calltree reordable
+ * [#812] Adds the ability to create a TLS listener in case the user
+ requests
+ * [#812] Adds the ability to configure local/published address from
+ * [#1883] Move switchCall in onHoldCall function
+ * [#812] Deals with the published address/port problem when
+ integrating TLS.
+ * [#1883] Switch call id in managerimpl when peerHungUp
+ * [#1883] Switch call id before hangup
+ * [#1883] Add usefull and permanent debug info for conference
+ cretion/deletion
+ * [#812] Fix various segmentation faults related to Direct IP kind of
+ calls.
+ * [#1883] Fix deletion of std::map elements using iterators
+ * [#2014] Add libzrtpcpp build dependency
+ * [#1883] Still some for loop test ambiguity (while loop instead)
+ * [#1883] Fix for loop initial test ambiguity (use while loop instead)
+ * [#1883] We must discard data in urgent ring buffer if data is get in
+ mainbuf
+ * [#1883] Fix availForGet same id for ringbuffer and readpointer
+ * [#812] Match "sips" as a Direct IP Call when the user enter a sip
+ uri
+ * [#812] Fix segmentation fault related to SIP URI creation.
+ * [#812] Towards integrating multiple tls listeners at the same time.
+ This
+ * [#1883] Add debug messages in conference and fix mainbufferTest
+ * [#812] gkt-ui fix. Private key must be fed as a filename and not as-
+ is.
+ * [#812] TLS integration within sipvoiplink and pjsip. Also,
+ configure.ac
+ * [#1883] Fix Alsa/Pulse mallocation
+ * [#1883] Fix data corruption in AudioRtp's micData buffer
+ * [#812] Full dbus integration for all the tls related options under
+ gtk-ui.
+ * [#1883] Fix memory leaks in audiortp session
+ * [#1883] Fix mem leaks in audio rtp
+ * [#812] Fix setAccountDetails where TLS_ENABLE was set to the value
+ * [#812] Small gtk-ui fix.
+ * [#811][#812] Small gtk-ui fix.
+ * [#812] Introduced a mechanism for configuration files that makes
+ possible
+ * [#812] New dbus bindings added. Also, configuration compliance was
+ enforced
+ * [#1881] Remove default buffer from MainBuffer (update unit-tests)
+ * [#1881] Add ring buffer read pointer tests
+ * [#1883] Fix issues in ringbuffer reader pointers
+ * [#2034] Implementing a new configuration dialogue for TLS transport
+ settings
+ * [#1883] Add some usefull debug and safety checks
+ * [#2028] Notify the client with libnotify when the zrtp negotiation
+ failed.
+ * [#811] Harmless no to throw an exception, an makes the application
+ less
+ * [#2028] A minidialog is showed to the user under sflphone-client-
+ gnome
+ * Removed useless file.
+ * Ignoring Makefile in src/widget
+ * [#2027] Fix segmentation fault when showMessage callback is called
+ after
+ * [#2026] keyExchange was set to ZRTP instead of "1"
+ * [#2024] Fix the wrong summary at the end of the assistant.
+ * [#1883] Fix mnagerimpl conference map insertion
+ * [#1883] Add Mutexes in MainBuffer
+ * [#811] Gtk ui was not presenting the right information about zrtp
+ for
+ * [#2023] security icons were not installed in sflphone-client-gnome.
+ * [#2021] Fix a mistake in the readme from sflphone-common that gives
+ wrong
+ * [#811] The current SRTP mode was not properly displayed for the
+ IP2IP
+ * [#1743] Re-implementation of the "automatically remove error dialogs
+ [...]"
+ * [#2017] [#2019] Fix the inability to dial a number and place a
+ registered
+ * [#811] Final re-integration of ZRTP support in the main branch from
+ 0.9.6
+ * [#1883] Fix map insertion methods
+ * [#811] Combo box now is now set to the active key exchange method
+ * [#811] ZRTP options now configurable back again from the Gtk UI.
+ IP2IP
+ * Updated hostname for git clone
+ * [#1883] Add minimal functionalities to create a conference
+ * [#811] re-integration of all the methods and signals on dbus.
+ ManagerImpl
+ * [#811] Got out of a precarious position were nothing would compile.
+ * [#1976] Build documentation squeleton with docbook
+ * [#1883] Add sflphone-client "addParticipant" button for conference
+ * [#1994] Better organize the source directory structure. New
+ subdirectories
+ * [#1883] Add a simple Conference class
+ * [#1882] Use static audio buffer in Pulse and ALSA layer (instead of
+ malloc)
+ * [#811] First commit toward re-integration and refactoring of ZRTP
+ * [#1882] Flush RTP ring buffer before entering mainloop
+ * [#1882] Fixed MainBuffer::UnBinCallID() in case there is no
+ ringbuffer
+ * [#1882] Test (and fixe) high level conference and mixing
+ functionalities
+ * [#1772] Apply patch to compile on fedora (sent by Marcin
+ Zajączkowski <mszpak@wp.pl>)
+ * [#1882] Update Bind, unBind call_id in MainBuffer
+ * [#1959] This adds the ability to store password as an MD5 Hash in
+ the
+ * [#1538] Fixes rules compilation
+ * [#1930][#1931] Fixed a mistake (again) related to index and
+ credential count
+ * [#1753] Remove ILBC from pjproject - Hacks in pjsip
+ * [#1930][#1931] Credential was not selected properly using realm
+ * [#1882] Finilize multiple reading pointer in RingBuffer
+ * [#1538] Remove configure from autogen.sh to respect debian upstream
+ authors policy
+ * [#1773] Remove generated files from repo
+ * [#1791] Use XDG_CACHE_HOME to save pid file
+ * [#1791] Fixes path to save history
+ * [#1791] Fix debian installation scripts
+ * [#1930][#1931] Settings are now taken into account in the server.
+ * [#1882] Add ringbuffer default ring buffer pointer in methods
+ involving mStart
+ * [#1882] Add default ringbuffer pointer
+ * [#1882] Add RingBuffer multiple read pointer basic functionnalities
+ * [#1882] Fix MainBuffer flushData unit test
+ * [#1930][#1931] Ability to save and retreive the configuration from
+ * [#1882] Added Multiple CallID mapping to MainBuffer
+ * [#1791] Not much
+ * [#1791] If XDG env variables are not null but empty, use default
+ ones
+ * [#1791] Make XDG_CONFIG_HOME writable
+ * [#1930][#1931] Partial commit. Not working yet. Cannot delete
+ account
+ * [#1881] Fixed alsa capture latency problem
+ * [#1881] Fixed Alsa capture temporarily
+ * [#1930] [#1931] Partial unbroken commit providing the ability to
+ * [#1881] MainBuffer implemented in AudioLayer/AudioRTP
+ * [#1881] Add discard and flush unit-tests
+ * [#1881] Add discard and flush functionnalites to MainRingBuffer
+ * [#1881] Add availForGet in MainBuffer
+ * [#1881] Add availForPut function to MainBuffer
+ * [#1880] Remove AudioRTP* pointer from SipVoIP (reapered while
+ merging master)
+ * [#1881] Add a map between call id and coresponding ring buffer
+ * [#1855] Refresh pot file and upload on Launchpad
+ * [#1881] MainBuffe now robust to false ids on getData and putData
+ * [#1881] Fix big big big memory leak
+ * [#1881] Add getData and putData to mainBuffer
+ * [#1881] Unit-test basic ring buffer functionnaities
+ * [#1881] Add class MainBuffer and basic buffer creation unit-tests
+ * [#1880] Fix call transfer (step2) issues
+ * [#1880] Moved AudioRtp* pointer from SIPVoIPLink to SIPCall class
+ * [#1791] Add postinst script to keep user data when migrating
+ config/history file
+ * [#1797] Make pjsip compile
+ * [#1777] Code indentation
+ * [#1791] Use XDG_DATA_HOME and XDG_CONFIG_HOME for sflphonedrc and
+ history + unit tests
+ * [#1746] Useless space does not appear anymore when volume sliders
+ and
+ * [#1643] GtkCheckMenuItem is used instead of icons for elements in
+ the
+ * [#1110] [#1668] STUN parameters are now located in the preferences,
+ under
+
+ -- Julien Bonjean <julien.bonjean@savoirfairelinux.com> Fri, 06 Nov 2009 11:23:15 -0500
+
+sflphone-common (0.9.6-SYSVER) karmic; urgency=low
** 0.9.6 **
@@ -65,7 +634,7 @@ sflphone-common (0.9.6-SYSVER) SYSTEM; urgency=low
-- SFLphone Automatic Build System <team@sflphone.org> Mon, 27 Jul 2009 09:53:00 -0400
-sflphone-common (0.9.6~rc2-SYSVER) SYSTEM; urgency=low
+sflphone-common (0.9.6~rc2-SYSVER) karmic; urgency=low
** 0.9.6~rc2 **
@@ -120,7 +689,7 @@ sflphone-common (0.9.6~rc2-SYSVER) SYSTEM; urgency=low
-- SFLphone Automatic Build System <team@sflphone.org> Fri, 17 Jul 2009 19:12:44 -0400
-sflphone-common (0.9.6~rc1-SYSVER) SYSTEM; urgency=low
+sflphone-common (0.9.6~rc1-SYSVER) karmic; urgency=low
** 0.9.6~rc1 **
@@ -228,7 +797,7 @@ sflphone-common (0.9.6~rc1-SYSVER) SYSTEM; urgency=low
-- SFLphone Automatic Build System <team@sflphone.org> Fri, 10 Jul 2009 15:49:56 -0400
-sflphone-common (0.9.6~beta-SYSVER) SYSTEM; urgency=low
+sflphone-common (0.9.6~beta-SYSVER) karmic; urgency=low
** 0.9.6~beta **
@@ -523,7 +1092,7 @@ sflphone-common (0.9.6~beta-SYSVER) SYSTEM; urgency=low
-- SFLphone Automatic Build System <team@sflphone.org> Tue, 23 Jun 2009 11:12:06 -0400
-sflphone-common (0.9.5-SYSVER) SYSTEM; urgency=low
+sflphone-common (0.9.5-SYSVER) karmic; urgency=low
** 0.9.5 release **
@@ -554,7 +1123,7 @@ sflphone-common (0.9.5-SYSVER) SYSTEM; urgency=low
-- SFLphone Automatic Build System <team@sflphone.org> Mon, 25 May 2009 11:34:40 -0400
-sflphone-common (0.9.5-SYSVER~rc2) SYSTEM; urgency=low
+sflphone-common (0.9.5-SYSVER~rc2) karmic; urgency=low
** 0.9.5 rc2 **
@@ -608,7 +1177,7 @@ sflphone-common (0.9.5-SYSVER~rc2) SYSTEM; urgency=low
-- SFLphone Automatic Build System <team@sflphone.org> Tue, 19 May 2009 12:08:03 -0400
-sflphone-common (0.9.5-0ubuntu1~rc1) SYSTEM; urgency=low
+sflphone-common (0.9.5-0ubuntu1~rc1) karmic; urgency=low
[ SFLphone Project ]
* [#1262] Updated changelogs for version 0.9.5-0ubuntu1 Snapshot 2009-
@@ -637,7 +1206,7 @@ sflphone-common (0.9.5-0ubuntu1~rc1) SYSTEM; urgency=low
-- Sflphone Project <sflphone@mtl.savoirfairelinux.net> Tue, 05 May 2009 19:16:09 -0400
-sflphone-common (0.9.5-0ubuntu1~beta) SYSTEM; urgency=low
+sflphone-common (0.9.5-0ubuntu1~beta) karmic; urgency=low
[ Julien Bonjean ]
* Updated Eclipse stuff
@@ -859,7 +1428,7 @@ sflphone-common (0.9.5-0ubuntu1~beta) SYSTEM; urgency=low
-- Sflphone Project <sflphone@mtl.savoirfairelinux.net> Mon, 27 Apr 2009 16:57:00 -0400
-sflphone-common (0.9.4-0ubuntu2) SYSTEM; urgency=low
+sflphone-common (0.9.4-0ubuntu2) karmic; urgency=low
[ Alexandre Savard ]
* Restore speex and GSM detection
@@ -869,7 +1438,7 @@ sflphone-common (0.9.4-0ubuntu2) SYSTEM; urgency=low
-- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Wed, 8 Apr 2009 11:29:15 -0500
-sflphone (0.9.4-0ubuntu1) SYSTEM; urgency=low
+sflphone (0.9.4-0ubuntu1) karmic; urgency=low
[ Emmanuel Milou ]
* Integrate DBus-c++ and libiax2 in the main build system
@@ -894,7 +1463,7 @@ sflphone (0.9.4-0ubuntu1) SYSTEM; urgency=low
-- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 3 Apr 2009 18:29:15 -0500
-sflphone (0.9.4-rc1) SYSTEM; urgency=low
+sflphone (0.9.4-rc1) karmic; urgency=low
[ Emmanuel Milou ]
* Fix bug while trying to hold/unhold several simultaneous call
@@ -908,7 +1477,7 @@ sflphone (0.9.4-rc1) SYSTEM; urgency=low
-- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 27 Mar 2009 18:29:15 -0500
-sflphone (0.9.4-0beta1) SYSTEM; urgency=low
+sflphone (0.9.4-0beta1) karmic; urgency=low
[ Alexandre Savard ]
* Display codec used during conversation on the GUI
@@ -924,7 +1493,7 @@ sflphone (0.9.4-0beta1) SYSTEM; urgency=low
-- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 20 Mar 2009 18:29:15 -0500
-sflphone (0.9.3-0ubuntu3) SYSTEM; urgency=low
+sflphone (0.9.3-0ubuntu3) karmic; urgency=low
[ Alexandre Savard ]
* Both playback and record streams in PA_STREAM_CORKED (pulseaudio)
@@ -949,7 +1518,7 @@ sflphone (0.9.3-0ubuntu3) SYSTEM; urgency=low
-- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Sat, 14 Feb 2009 13:29:15 -0500
-sflphone (0.9.3-0ubuntu2) SYSTEM; urgency=low
+sflphone (0.9.3-0ubuntu2) karmic; urgency=low
[ Emmanuel Milou ]
* Add compilation note in README
@@ -1018,7 +1587,7 @@ sflphone (0.9.3-0ubuntu2) SYSTEM; urgency=low
-- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Sat, 14 Feb 2009 13:29:15 -0500
-sflphone (0.9.3-0ubuntu1) SYSTEM; urgency=low
+sflphone (0.9.3-0ubuntu1) karmic; urgency=low
* Remove debug
* Join thread before leaving
@@ -1031,7 +1600,7 @@ sflphone (0.9.3-0ubuntu1) SYSTEM; urgency=low
-- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 06 Feb 2009 19:17:32 -0500
-sflphone (0.9.2-2ubuntu9) SYSTEM; urgency=low
+sflphone (0.9.2-2ubuntu9) karmic; urgency=low
[ Alexandre Savard ]
* Speex audio codec preprocessing initialization
@@ -1059,7 +1628,7 @@ sflphone (0.9.2-2ubuntu9) SYSTEM; urgency=low
-- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Thu, 05 Feb 2009 18:27:53 -0500
-sflphone (0.9.2-2ubuntu8) SYSTEM; urgency=low
+sflphone (0.9.2-2ubuntu8) karmic; urgency=low
[ Emmanuel Milou ]
* Update changelogs
@@ -1103,7 +1672,7 @@ sflphone (0.9.2-2ubuntu8) SYSTEM; urgency=low
-- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Wed, 04 Feb 2009 14:00:30 -0500
-sflphone (0.9.2-2ubuntu7) SYSTEM; urgency=low
+sflphone (0.9.2-2ubuntu7) karmic; urgency=low
[ Emmanuel Milou ]
* Update changelog to 0.9.2-6
@@ -1125,7 +1694,7 @@ sflphone (0.9.2-2ubuntu7) SYSTEM; urgency=low
-- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 23 Jan 2009 17:48:13 -0500
-sflphone (0.9.2-2ubuntu6) SYSTEM; urgency=low
+sflphone (0.9.2-2ubuntu6) karmic; urgency=low
[ Emmanuel Milou ]
* Migrate STUN configuration to the main config window
@@ -1159,7 +1728,7 @@ sflphone (0.9.2-2ubuntu6) SYSTEM; urgency=low
-- Emmanuel Milou <manu@sulfur.inside.savoirfairelinux.net> Fri, 16 Jan 2009 18:19:05 -0500
-sflphone (0.9.2-2ubuntu5) SYSTEM; urgency=low
+sflphone (0.9.2-2ubuntu5) karmic; urgency=low
* Fix memory leak in the pulseaudio callback
* Update debian package generation script
@@ -1175,7 +1744,7 @@ sflphone (0.9.2-2ubuntu5) SYSTEM; urgency=low
-- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Wed, 14 Jan 2009 21:17:20 -0500
-sflphone (0.9.2-2ubuntu4) SYSTEM; urgency=low
+sflphone (0.9.2-2ubuntu4) karmic; urgency=low
[ Emmanuel Milou ]
* add german translation
@@ -1185,7 +1754,7 @@ sflphone (0.9.2-2ubuntu4) SYSTEM; urgency=low
-- Yun Liu <yun.liu@savoirfairelinux.com> Thu, 08 Jan 2009 13:08:51 -0500
-sflphone (0.9.2-2ubuntu3) SYSTEM; urgency=low
+sflphone (0.9.2-2ubuntu3) karmic; urgency=low
[ Emmanuel Milou ]
* The main thread synchronizes the ringtone thread
@@ -1197,13 +1766,13 @@ sflphone (0.9.2-2ubuntu3) SYSTEM; urgency=low
-- Yun Liu <yun.liu@savoirfairelinux.com> Tue, 06 Jan 2009 16:18:38 -0500
-sflphone (0.9.2-2ubuntu2) SYSTEM; urgency=low
+sflphone (0.9.2-2ubuntu2) karmic; urgency=low
* Fix bug ticket #129
-- Yun Liu <yun.liu@savoirfairelinux.com> Wed, 5 Jan 2009 15:54:53 -0500
-sflphone (0.9.2-2ubuntu1) SYSTEM; urgency=low
+sflphone (0.9.2-2ubuntu1) karmic; urgency=low
* Migrate from eXosip library to pjsip
* Add multiple SIP accounts support