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sflphone-common (0.9.7~beta~ppa1~SYSTEM) SYSTEM; urgency=low
** 0.9.7~beta~ppa1~SYSTEM **
* [#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-SYSTEM) SYSTEM; urgency=low
** 0.9.6 **
* Documentation on echo test
* [redmine_down] codec names not displayed in total
* [redmine_down] crash when hanging up a dialing call because tries to
add it to history whereas no starttime
* [#1927] alternate every time screen changed to call history
* [#1886] clean code
* [#1886] debug messages when loading history removed
* [redmine_down] sflphone-kde icons
* [#1855] Update language files
* [#1502] Update version number
* [redmine_down] setHistory at close
* [#redmine_down] Handle PJ_DECLINE_SC as failure
* [#1923] Fix segmentation fault when adding a new account
* [#1923] Check on iterator before setting the config
* [#1904] Added mnemonic to tabs in sflphone-client-gnome.
* [#1905] The daemon was not sending the currentSelectedCodec signal
on dbus when answering a call.
* [#1922] Default values set to all account details
* [#1886] Spinbox reg expire enables apply, and address book is not
visible when disabled
* [#1905] Bug fix for segmentation fault caused by an empty string,
* [#1910] Warnings in test directory
* [#1919] Error fixed
* [#1855] Update russian translation - Hussein Abdallah
* [#1910] Remove files
* [#1919] fixed
* [#1777] Code indentation
* [#1918] fixed
* [#1917] fixed
* [#1910] Remove warnings compilation in src
* [#1886] removed AccountListModel in configskeleton
* [#1914]
* [#1911] check previous and new port
* [#1910] Remove compilation warnings in src/dbus and src/history
* [#1910] Remove compilation warnings in src/audio
* [1855] Update german translation - Sven Werlen
* [#1909] removed
* [#1906] Done
* [#1904] The registration expire value is now configurable from the
* Cleaned up debug messages.
* [#1886] separated initCallItem in two functions
* [#1886] reversed error in commit
* [#1886] clean debug
* [#1886] changed Name of classes and files
* [#1886] clean
* [#1870] In call_state_cb (dbus.c:126), _time_stop was overridden by
the actual time.
* [#1884] Added some new gpg flags to prevent tty warnings
* [#1886] Clean audio config dialog
* [#1886] No more compile warnings. + 1 comm
* [#1872] Check if the user input is smaller than PJ_MAX_HOSTNAME.
* [#1886]
* [#1785] Fixed build when no new commit
* [#1852] If chosen by the user, the hostname can now be solved and
used
* [#1871] * and # inverted back
* [#1869] Conditional compilation that checks if
* [#1309] removed test in main
* [#1425] Put actions in SFLPhone window class instead of ui view,
made a separate toolbar for screens.
-- SFLphone Automatic Build System <team@sflphone.org> Mon, 27 Jul 2009 09:53:00 -0400
sflphone-common (0.9.6~rc2-SYSTEM) SYSTEM; urgency=low
** 0.9.6~rc2 **
* [#1755] Remove generated file
* [#1753] restore ilbc ...
* [#1866] Methods getSipPort and setSipPort now have an effect on the
* [#1753] make pjsip compile without ilbc. Use ./autogen.sh --disable-
ilbc-codec
* [#1855] Fix error in russian translation
* [#1805] Remove the old flawed signal mechanism which was failing in
* [#1855] Refresh translation
* Spanish translation finished + po README files updated + echo's in
copy-in-clients
* [#1850] Yun made the chinese HK-CN translation
* [#1848] Fix transfer interface bug
* [#1862] At install, kde client installs only french translation file
* [#1841] A new fallback mechanism was added to the internal resolver
in PJSIP.
* Started AccountList model/view
* [#1855] Remove po subdir in Makefile.am
* [#1855] Fix typo error in sflphone-client-gnome
* [#1855] Do not generate Makefile in sflphone-common/po
* [#1855] Copy translation files into both clients dirs
* [#1855] Remove po dir from sflphone-common
* Comments added
* [#1860] mailbox->voicemail...
* make scripts executable
* [#1855] French translation
* [#1855] Chinese zh_HK partially filled...
* [#1859] An unnamed pipe monitored by poll() was added. When we want
to
* [#1855] Sven completed the first part of the german translation
* [#1855] Cantonese manually filled for already translated, almost
equal strings
* [#1855] Merge russian translation
* [#1855] Spanish manually filled for already translated, almost equal
strings
* [#1855] Update german translation in ./lang/de
* [#1858] This problem was fixed by removing a useless line in
* [#1855] merged existing translations in lang/ sflphone.po's
* [#1842] [#1843] An attempt at improving the expected behaviour that
can't
* [#1855] added po folder in gnome client and scripts for copying from
common lang folder to clients
* [#1853] Edit before call does nothing on call history
* Put most language entries possible in common. From 300 to 250
entries. Stays underscores problem. Scripts for copy in clients.
* commit to merge master
* [#1825] Changed "Bad authentification" to "Authentication Failed".
* common po files
* [#1753] Remove ILBC from pjproject
-- SFLphone Automatic Build System <team@sflphone.org> Fri, 17 Jul 2009 19:12:44 -0400
sflphone-common (0.9.6~rc1-SYSTEM) SYSTEM; urgency=low
** 0.9.6~rc1 **
* Update some version number
* [#1792] Creates .sflphone directory with permission 600. Also,
"chmod 600" after
* [#1810] GUI is now notified that the call failed. Also, a segfault
was
* [#1816] Address book search disabled when disabled address book and
enabled it back plus button stays triggered
* codeclistmodel + asynchronous loading of address book +
enable/disable address book
* [#1810] Now checking SDP answer after 200 OK. Still need to
implement full
* [#1794] Can't use the interface during a call
* Updated translation files
* Russian translation integrated
* Codec list model/view started.
* [#1807] Add configure.ac in pjproject-1.0.3
* [#1787] closeRtpSession added in some places where it should have
been
* Use Item class for contacts and accounts
* Comments + clean code
* [#1794] Improved debug messages
* [#1805] Replaced the old and unreliable mecanism that was was
waiting for
* [#1794] Can't use the interface during a call
* [#1787] For those cases where no registered SIP account is
configured
* [#1797] Make pjsip compile
* [#1787] Minor changes. Removed useless commented line. Changed order
of
* [#1777] Code indentation
* [#1797] Update package generation with new pjsip version
* [#1798] Does not hang up when the call is building up
* [#1797] Update .gitignore with new pjsip version
* [#1797] Remove generated files from repo
* [#1797] Main build system now uses pjproject-1.0.3
* [#1797] Add pjproject-1.0.3
* [#1797] Remove pjproject-1.0.2
* [#1796] Computing time optimization (samplerate conversion)
* [#1787] _audiortp->start() moved away from offhold(),
SIPCallAnswered()
* [#1312] Added new states for calls initialized by other clients
* [#1795] Crashes when adding a new account, checking it and applying
* [#1782] Missing icons
* [#1793] KDE client compilation problem
* Fake ringtone files can no longer be set.
* indentation
* [#1312] Able to fetch to differentiate incoming/ringing call state
* [#1784] Use DESTDIR variable in po Makefile - fix language file
installation
* [#1785] Fixed typo
* [#1785] Fixed changelog update
* [#1759] ./autogen.sh --prefix=/usr --with-debug to use optimization
level 0
* [#1773] Changed snapshot naming convention
* [#1773] Removed gpg agent use, added repository cache cleaning
* [#1759] Use optimization level 0 for repository, 2 for packages
* [#1777] Code indentation/formatting
* Translated new features in french
* [#1785] Added missing changelog entry
* [#1781] Window title is SFLPhone
* [#1777] Add code indentation/formatting in the buil system
* [#1774] Can't set voicemail number in KDE account creation wizard
* [#1775] Can't modify account information for account created with
the wizard
* [#1771] Add a "Default" button in context menu to disable chosen
prior account
* [#1705]
* [#1224] Remove generated file from the repo
* [#1224] Remove generated file from the repo
* [#1762] distclean target should remove kconfig generated files
(settings.h, settings.cpp). Rename them?
* [#1761] clear history button should really clear history
* Dialpad works.
* Implemented Dialpad widget instead of building it in main view.
* Removed last occurence of the old config dialog, that made the build
crash.
* [#1755] Do not consider G722 as a dynamic payload elsewhere than in
RTP layer
* [#1753] Remove ilbc Makefile generation
* [#1756] Implement a kde configuration dialog with kconfig xt and
kconfigdialog class
* [#1755] fix audiocodec folder parsing problem
* [#1450] Reinit timestamp comparison in RTP, create session in
newOutgoingCall
* [#1753] Remove milenage third party code from pjsip
* New Config Dialog integrated in GUI.(without codecs)
* [#1753] Remove ILBC codec
* kconfig started, tr2i18n -> i18n, icons folder, accountList changed
* [#1705] Fixed Audio RTP thread creation/start
* [#1714] Fix codec negociation result handling
* [#1678] Fix audiortp payload setting
* [#1678] Put bac putData method in rtp
* [#1669] gtk_file_chooser_get_filename() support UTF-8 by default
* [#1735] Add conditions to sdp update call if call declined
* [#1737] substr of recordings destination folder to remove "file://"
should be done in client rather than in daemon
* [#1731] Enlarge audio stream buffer size
* [#1714] Missing true
* [#1317] Fixed Mandriva timeout
* [#1317] Changed tag convention
* [#1317] Cleaned git-dch
-- SFLphone Automatic Build System <team@sflphone.org> Fri, 10 Jul 2009 15:49:56 -0400
sflphone-common (0.9.6~beta-SYSTEM) SYSTEM; urgency=low
** 0.9.6~beta **
* spec files for mandriva and opensuse updated with buildrequires
libqt4-dev >=4.3
* [#1700] Cannot build on ubuntu 8.10 and a few other distribs
* [#1502] Update version number where applicable
* [#1642] Update client icons
* [#1450] Clean up useless debug and comments in sipvoiplink and
audiortp
* [#1450] Remove Semaphore object in AudioRtp thread deletion
* [#1450] Audio RTP init now synchronized with Sip/SDP
* [#1693] kde client crashes when changing codecs order/activation
* [#1450] Deep refactoring of audiortp
* [#1450] setRtpSessionRemoteIp
* [#1689] getCallList at start
* [#1224] Change path in package files
* [#1450] Audio RTP initialized only once, payload and remote ip set
at runtime
* [#1450] Add setRtpSessionMedia and setRtpSessionRemoteIp address
* [#1642] Make GNOME GUI fresher and younger ;)
* [#1686] Status bar displaying used account
* added sflphone-kde icon so that it compiles
* [#1659] Ending a call causes the daemon to crash
* corrected introspection XMLs, po files...
* [#1211] g722 media descriptor in codecDescriptor
* [#1310] Install sflphoned in $(prefix)/lib/sflphone
* [#1502] Do not install test binaries and dbus utilitaries
* [#1224] hack for pjsip build system!
* [#1224] Remove pjsip binaries from repo
* [#1224] Upgrade to pjsip 1.0.2
* [#1658] About SFLphone (bugs)
* [#1658] About SFLphone
* [#1660] Displaying all dialed numbers in a call
* Tested status bar.
* [#790] Optimize pulse audio streams parameters
* [#1678] Some usefull debug messages for mutex/semaphore deadlock
problem
* [#1669] Add/remove some usefull/unusefull debug
* [#1665] Fix latency related to pulse audio stream openning/closing
* [#1457] Make the menus and panels accessible in french
* [#1457] Improve broken keyboard accessibility in menus and conf
panels
* [#961] Instanciate only once the searchbar icons
* [#961] Restore transfer fonction
* [#961] Filter on the history type OK
* [#961] Fix compilation problems on hardy/intrepid
* [#1157] Commit missing files
* [#790] Reduce number of start/stop streams call on pulse audio
* [#1639] kde client crashes when no account registered
* [#1620] Fix the searchbar
* [#1620] Get back caltree as it was during gtkcritical area
* [#1620] Add history filter reinit function
* [#1335] Add a missing label in address book preferences
* [#1561] Update russian translation - Hussein Abdallah
* [#1605] Fix edit menu french translation
* [#961] Enable to search in the history according to the call type
* [#1449] Searchbar does not work anymore
* [#961] Add popup menu on the entry primary icon for history
* [#1317] Fixed KDE client package dependency
* [#936] speex 32 khz integration completed
* [#936] Use 320 frame size
* [#936] Test using a frame size at 320 smpls
* [#1214] Enable / Disable history
* [#1607] Fix compilation problem for ubuntu 8.10 (libsexy)
* [#1313] Implement processDataEncode processDataDecode in audiortp
* [#1613] codec list order can't be set
* Better handling of localisation + added languages + corrected
warnings + begginning of new config dialog with kconfig + 14px
account leds
* [#1214] Save and load history according to the limit timestamp +
unit tests
* [1609] Fix call number copy/paste feature
* [1607] Restore clear action icon in searchbar
* [#936] Try to decode using 1280 samples
* [#936] Add some debug
* [#936] Add .cpp file
* [#936] Oops Forgot speex 32 khz
* [#1214] Add configuration panel for history + D-Bus calls
* [#1313] Test rtp thread function, frame size, nbbytes, resampling
* [#790] Flush audio data before closing audio streams
* [#1214] History displays local time
* [#1214] Skip empty field on display
* [#1214] Associate an account to an history entry
* [#1342] Get addressbook options sensitive/non-sensitive
* [#1211] Clean up and comments
* [#1211] Get back to 20 ms framesize
* [#1211] Use sendImmediate instead of putData in RTP
* [#1211] Fix nb byte available in RTP
* [#1211] Clear condition on maxNbSamples in RTP
* [#1211] Fix max byte available in RTP session
* [#1211] G722: Use 160 samples per frame instead of 320
* [#1211] Test using a dynamic payload
* [#1211] Test using a dynamic payload type
* [#1211] Rename size variable (nb_samples, nb_bytes)
* [#1211] Test g722 ip-to-ip sending twice the data lenth
* [#1211] Test g722 ip-to-ip
* [#1214] Do not select an history item by default at startup
* [#1214] Remove some compilation warnings
* [#1214] Handle empty field - remove g_print
* [#1214] Add each history item only once
* [#1214] Handle call timestamps properlier
* [#1214] Do not need timestamp files anymore
* [#1214] Use the saved date for history entry
* Clean up
* [#1214] Client doesn't crash if the D-Bus call fails
* [#1214] Client is able to save its history - still some glitches
* [#1211] Forgot 16000 for g722
* [#1211] G722 initialization
* [#1214] Save name/number, successfully load the history if no fields
are empty
* [#1499] Fixed destination directory bug
* [#1214] Restore all the functionalities; peer name/number way more
easy to handle !!
* [#1214] Add callable_object instead of call_t, refactoring
* [#1211] Test with polycom soundstation 16000
* [#1211] Remove C like inline function in g722 codec
* [#1342] Finalize gnome client preference window formating
* [#1214] Retrieve the history when the gnome client startsup
* [#1306] Implement localization for KDE client
* [#1593] enable accounts apply button when account checked/unchecked
* [#1214] Implement the dbus calls on server side
* [#1214] Add serialized/unserialized functions to pass data on DBUS
* [#1342] Formating gnome client configuration windows
* [#1214] Save sucessfully a map of history items
* [#1499] Removed multiple jobs compilation for KDE client (2)
* [#1214] Load history from file into memory, add unit tests
* [#1534] Throws a length_error exception in case URL exceeds
std::string max_size
* [#1499] Removed multiple jobs compilation for KDE client
* [#1565] make account leds smaller
* [1430] Fix dbus debug
* [#1562] crashes when trying to change item of a call of state "OVER"
* [#1116] Fix compilation bug
* [#1317] Added mandriva and opensuse-11 64 bits
* [#1108] Add messges in main window concerning transfer success
failure
* [#1116] Fix compilation problems
* [#1211] g722 Makefile
* [#1108] Client side transferFailed/trasferSucceded signals handling
* [#1211] G722 mostly completed,
* [#1555] make bigger toolbar (24x24)
* [#1551] remove default mailbox number in wizard and disable mailbox
button when first account doesn't have mailbox number
* [#1342] Re-add sflphone manpages
* [#1116] Fix compilation on non-jaunty distros
* [#1317] Fixed opensuse startup sleep
* [#1108] Add a signal in the client to notify successful or failed
transfer
* [#1108] Dbus signals concerning call transfer success/failure
* [#1317] Added opensuse to automatic build system
* [#1223] Fix manpages bug
* [#1060] german translation glitch
* Clean up some gnome client warnings
* [#1547] replace ugly account leds by beautiful icons
* [#1548] add close button that hides windowand just hide on clicking
the cross
* [#1549] put introspec XMLs in the client's source
* [#1312] Implement getCallList D-BUS method
* [#1116] Clear text in history and contacts
* [#1499] KDE integration
* [#1469] Modify header linkers in dbus-c++'s Makefile.am's
* [#1469] Remove examples folder from dbus-c++
* [#1214] History integration in build system; unit test squeleton
* [#1317] Cleaning
* [#1469] Remove configure stuff in dbus-c++
* [#1469] Add unofficial mainline dbus-c++
* [#1469] Remove dbus-c++ from freedesktop
* [#1430] Bring account changed signal/callback back to normal
* [#1060] Update german translation - Sven Werlen
* [#1430] Add marshaller one string define
* [#1430] Send account change signal broadcast using account id
* [#1430] Remove condition on setRegistrationState, cause stun to
crash
* [#1317] Centralized version handling
* [#1317] Fixed version number on sfl-git-dch
* [#1317] Refactoring for new distributions
* [#1215] Fix account order at startup if latency
* [#1088] Restore sip dns srv
* [#1214] Add squeleton for history manager
* [#1430] Add accout id to accout changed method
* [#1430] No connectionStatusNotification (account changed) if no
changes
* [#1538] Add COPYING file
* [#1430] Add audio rtp thread tests
* [#1317] Changed version detection
* [#1538] Document license in libs/stund
* [#1317] Added version files
* [#1538] Apply François patches - debian packages
* [#1317] Updated spec files
* add files
* [#1538] Apply François patches - debian packages
* [#1535] Change program file structure (directory src...)
* [#1317] Updated build system scripts
* [#1317] Cleaning
* [#1317] Copied introspect files to gnome client
* [#1317] Added opensuse to build-system : first-shot
* [#1317] Remove spec files from configure
* [#1317] Added missing prefix
* removed debug for daemon account fix
* [#1430] Add a connection reference which most likely belong to
libdbus
* [#1430] Use shared connection instead of private
* make daemon find the account, added userMatch
* Clean code, add comments...
* [#1317] Fixed packaging rules
* [#1317] Updated autogen
* Updated autogen.sh for pjsip
* [#1526] Set accounts order
* [#1317] Fixed pjsip lib dirs
* [#1317] Updated debian packaging for new pjsip configuration script
* [#1317] Switch to autogenerated guess and sub files
* [#1317] Updated pjsip inclusion in build system
* [#1317] Replaced pjsip guess and sub files
* [#1317] Fixed compilation issues on opensuse 11
* [#1505] account list seem to crash the application when clicking
Apply very fast...
* [#1456] Add a flag to be replaced in the control files
* [#1456] Added version dependancy handling
* put account alias in AccountWidgetItem rather than in the item with
" " before.
* [#1034] The KDE client should start sflphoned if it is not started
* [#1500] Handle options for notifications and display on incoming
call.
* [#1443] Client should not crash when receive an unexpected
stateChanged signal
* [#1403] Do not stop the notification anymore
* [#1456] Added version dependancy handling
* [#1426] Daemon crashes when get alsa plugin
* [#1422] Improved error messages
* commit for merge
* [#1424] Change logo in tray icon and put a different one when
incoming call
* [#1425] first part done, window title...
* [#1413] add manpages creating and installing in build system
* [#1417] The client should start the account creation wizard if
started for the first time (if config file doesn't exist)
* [#1421] Make volume bars horizontal when dialpad is hidden.
* Changed main window title and fixed a mistake in sflphone_const.h
* [#1412] make debian package building work
* changelog changed.
* Changed addAccount method in gnome client.
* Debian and man folders added.
* [#1388] Change project name from sflphone_kde to sflphone-client-kde
* Better handle of kabc check.
* [#1351] Automatic generation of dbus interfaces in makefile
generated by cmake
* [#1307] Implement "edit before call" in history and address book.
* [#1344] change action_call label in call history from "call" to
"call back".
* [#1308] Implement Hook feature in kde client
* Improved build system.
* #1219 : Add address book configuration page
* Better handling of registration to the daemon.
* #1039 : Add tray icon in kde.
* Issue no 1216 : Double click on item in history or address book
causes call.
* display peer name in call list and call history when called from
address book.
* Address book functionnal with photo displayed.
* Help menu kde available but actions disappeared. All fonctions in
view.
* Address book functionnal but ugly and making its own sort in the
complete address book.
* Account choice on right click, clean out includes, page address
book, fixed bugs...
* Wizard, double click, context menu...
* Removed sflphone_kde.kdevelop.filelist
* Added account creation wizard and translated interface in english.
* Transfer functionnal but ugly.
* transfer not functionnal
* Bug fixed : unholding (UNHOLD_CURRENT, UNHOLD_RECORD)
* Commit functional for push. With install.sh
* Before merge.
* Problem with enable accounts. Account display increased.
* Functional with codec order working , playDTMF.
* Commit functional.
* sflphone_kde/build added in .gitignore.
* complete commit for checkout previous.
* Commit before checkout previous version to check the display
bug(little font everywhere...)
* Functionnal client. Rest : history icons, config icons and
functionalities
* commit before merge asavard for isRecording.
* Call and Automate fusion done and seems to work.
* Commiting before putting Automate class in Call class.
* Functionnal main window without recording, history, voicemail, kio
widgets.
* client kde avec kdevelop.
* Config Dialog almost finished.
* Base of QT client
-- SFLphone Automatic Build System <team@sflphone.org> Tue, 23 Jun 2009 11:12:06 -0400
sflphone-common (0.9.5-SYSTEM) SYSTEM; urgency=low
** 0.9.5 release **
* [#1060] FIx bug in chinese translation
* [#1313] git add rtpTest.cpp rtpTest.h
* [#1313] Add init/close rtp tests
* [#1313] Basic instanciation of the rtp layer
* [#1449] Gtk-Critical concerning history filters and new calls
* [#1400] Make the match with the hostname instead of username
* [#1324] Change status bar label for "Using %s (%s)"
* [#1403] Icon size: 60x60 px
* [#1403] Do not remove notification, improve icon quality
* [#1403] Add smaller icon for gnome notifications
* [#1403] Prevent crash when hangup && no notification
* [#1403] Remove all actions on notifications; code refactoring
* [#1451] Use stun.sflphone.org as default STUN server
* [#1060] New po files - need to be translated
* [#1060] Update french translation - Rebuild template file
* [#1456] Add a flag to be replaced in the control files
* [#1454] Make cppunit optional; remove from build deps in control
files
* [#1401] Add libexpat1-dev dependency in control files
* [#1448] Take off these ugly debug messages
* [#1448] fixed getTelephoneTone and getTelephoneFile() called
repeatedly
* [#1406] add liblog4c-dev in build-depends
* [#1409] Restore .desktop icon
-- SFLphone Automatic Build System <team@sflphone.org> Mon, 25 May 2009 11:34:40 -0400
sflphone-common (0.9.5-SYSTEM~rc2) SYSTEM; urgency=low
** 0.9.5 rc2 **
* [#1422] Improved error message
* [#1402] Fix pjsip build
* [#1404] Clear GTK-Critical Bug at client startup
* [#1422] Added automatic VM shutdown when building on more than one
VM
* [#1422] Fixed some issues with new changelog generation script
* [#1422] Moved distribution update to specific file
* [#1422] Dropped git-dch, replace by home made implementation
* [#1402] Fix pjsip build
* [#1404] Clear GTK-Critical Bug at client startup
* Changes for name based dbus connection
* Clean changelogs
* [#1343] Gnome: Implement a callback system to handle focus on
different widgets
* Debus Session
* Refactoring Python code, PEP8
* [#1430] Get back dbus_g_proxy_new_for_name
* [#1430] Get back DBUS_BUS_SESSION type
* [#1430] Dbus fixed owner message binding
* Second test with DBUS owner
* [#1404] Gnome -> Preferences -> Hooks
* [#1404] Gnome -> Preferences -> Recordings
* [#1404] Call History
* [#1404] Gnome -> Preferences -> Address Book
* [#1404] IF the first notification option disable the second
notification
* Dbus with fixed owner does not automatically start the deamon
* Add codec debug tests in pysflphone
* [#1407] Some print info
* [#1407] Add a scenario to pick_up action
* Test client dbus connection to a fixed owner
* Add python dbus test suite
* [#1161] Modified version handling in build system
* [#1314] Test pulse audio and audio streams connect and disconnect
* [#1402] Add info message after configure
* [#1402] Build the daemon with the local pjsip library (vs the
installed one)
* [#1009] Fix Codec Sampling Rate set to zeros
* [#1314] Add mutex to pulse layer audio streams
* [#1314] Refactoring pulseaudio stream to test connect disconnect
* [#1314] Refactoring of pulselayer to test conect/disconnect
* Add debug messages in debus calls concerning account
* [#1314] Add some return values to audio init functions
* [#1406] add liblog4c-dev in build-depends
* [#1409] Restore .desktop icon
* Bug #1405: Fix strings as requested.
* Bug #1404: Fix strings in preferences panel.
-- 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 Project ]
* [#1262] Updated changelogs for version 0.9.5-0ubuntu1 Snapshot 2009-
05-05
[ Emmanuel Milou ]
* Add some python CLI client code; not really functional
* [#1108] Fix peerHungup method for IP to IP call
[ Alexandre Savard ]
* [#1108] Correct setting of SIP contact for direct IP call
* [#1108] SIP user agent handles incoming REFER
[ Emmanuel Milou ]
* Remove website from repository
* Update translation
[ Alexandre Savard ]
* Sflphone icon's tooltip changed for "configured" instead of
"registered"
[ Emmanuel Milou ]
* Update translation
[ Sflphone Project ]
-- Sflphone Project <sflphone@mtl.savoirfairelinux.net> Tue, 05 May 2009 19:16:09 -0400
sflphone-common (0.9.5-0ubuntu1~beta) SYSTEM; urgency=low
[ Julien Bonjean ]
* Updated Eclipse stuff
* Improved addressbook config window
* Added sflphone Eclipse stuff
* Implemented addressbook list server side
* Moved dbus stuff in dbus directory
* Updated addressbook configuration
[ Emmanuel Milou ]
* Remove unuseful installation scripts. Use apt-get build-dep sflphone
instead
* fix bug #1090
[ Alexandre Savard ]
* defining speex 16khz
[ Emmanuel Milou ]
* Remove unuseful file from build system
* Start dns srv resolver
[ Alexandre Savard ]
* Basic ogg/vorbis initialization
[ Emmanuel Milou ]
* Handle incoming IP-to-IP invite correctly
[ Alexandre Savard ]
* speex wideband 16000
[ Emmanuel Milou ]
* Better handling of incoming IP to IP call
* DNS SRV resolution functional
* Implement IAX2 incoming URL
* Allow user to make IP call without any accounts configured
* Add a contextual menu to edit a number from the contacts tab
* Add comments, tooltip and new button to the contextual menu
* add delete event, migrate to GTK 2.16 for sexy icons
* Resolve ticket #1118
* Update suse spec file
* Add phone number cleanup functions, unit tests and panel
configuration
* Add pertinent test that fails
* fix dependencies for suse package
* Add contextual edit menu in history - #1120
[ Alexandre Savard ]
* Temporary comit: make speex wideband (16 khz)
* Temporary: shared object for speex narrow band
* Temporary: speex narrowband and wideband coexist
[ Julien Bonjean ]
* Fixed bug when no book selected
* Fixed addressbook related compilation warnings
* Fixed GTK client remaining compilation warnings
* Fixed segfault when book removed since last sflphone run
* Fixed bug when book is unreachable (ldap error)
[ Alexandre Savard ]
* Fix codec list in audio config window
* Active/inactive speex codec by payload
[ Julien Bonjean ]
* Updated gitignore
* Added some comments
[ Emmanuel Milou ]
* Add callto: handler script for browsers and al.
* Integrate test compilation in the daemon build-system
[ Julien Bonjean ]
* Fixed g_object_unref warning for pixbuf
* Cleaned too verbose output
* Fixed toolbar update warning
* Added support for asynchornous books open (first shot)
[ Emmanuel Milou ]
* Add a DBus call to fetch the call details from a call ID - Ticket
#928
[ Julien Bonjean ]
* Improved async open books
* Fixed bug #1139
[ Emmanuel Milou ]
* Add a way to save account order
* commit missing files
[ Julien Bonjean ]
* Introduced log4c (ticket #1162)
[ Emmanuel Milou ]
* Load/save account order functionnal - ticket #813
[ Alexandre Savard ]
* Add CELT codec (#1143)
* Make celt frame size 256 (*1143)
[ Julien Bonjean ]
* Switched everything to log4c (ticket #1162)
* Updated eclipse settings
[ Emmanuel Milou ]
* Restore adding account - ticket #1172
* Add liblog4c dependecy - ticket #1179
[ Alexandre Savard ]
* Double maxAvailByte for frame size in rtp (#1143)
[ Emmanuel Milou ]
* Add User-Agent SIP header - Ticket #1173
[ Julien Bonjean ]
* Fixed autoresize issue (#708)
[ Emmanuel Milou ]
* Remove libcppuint dependency for the debian packages
* Look for libsexy only if gtk version < 2.16 - Ticket #1116
* Remove libsexy dependency for jaunty. ticket #1116
[ Julien Bonjean ]
* Introduced unit tests (#1146)
* Updated gitignore
* Fixed Makefile (#1146)
[ Emmanuel Milou ]
* [TICKET #1112] Add a test on the voice buffer to send through iax
packets
* Remove doublon in dependencies
* Remove warnings from the client test framework
* Update version number to 0.9.5~beta
* Update build-package script
* Add check dependency in build-deps control file field
* Create debian files for the new sflphone-client-gnome
* [TICKET #1212] Add Replaces field in control files
* [TICKET #1212] Fix manpages installation path
* [TICKET #1212] Add maintainer scripts to create alternatives
* [#1212] Update the manpages generation - edit preinst maintainer
script
* [#1212] Fix reference error in manpage
* [#1212] Add missing files on the client side
* [#1212] Fix debian docs files - no TODO file
* [1212] Fix manpage creation problem
* [#1220] Generate client-side glue files and marshaller at
compilation time
* [#1220] Generate server-side glue files at compilation time
* [#1212] Change binary name to sflphone-client-gnome
* [#1212] Update .gitignore to fit the new working tree
* [#1220] Explicitly generate glue files before building the library
* [#1220] Compile dbus directory before audio
* [#1212] Create sflphone-common at the root of the repository
* [#1212] Re-add pjproject
* [#1212] Remove Makefile from repo
* [#1220] Fix Makefile.am
* [#1212] New working directory functional
* [#1212] Update .gitignore
* [#1212] Hack to make pjsip compile..
* [#1220] Use non-installed binary for dbusxx-xml2cpp
* [#1212] Add descriptive files, remove unuseful scripts from tools/
[ Alexandre Savard ]
* Restore speex codecs
* add frame size for celt (#1143)
* add framesize to codec, independant from audiolayer (#1143)
* use codec frame size in rtp (#1143)
* compute fixed_codec_framesize (#1143)
* do not resample if not required (#1143)
* add condition on resampling for decoder (#1143)
* add a condition on bytesAvail == 0 from mic data
* no maximum in rtp decode (#1143)
* compute maximum for decoding (#1143)
[ Emmanuel Milou ]
* [#1146] Implement unitary tests on the client-side
[ Alexandre Savard ]
* use float instead of int to compute max nb of sample (#1143)
* add nbSampleMax for unresampled data (#1143)
* make thread sleep during 5 ms insead of 20 (#1143)
* use unix usleep (#1143)
* 50 usecond thread!!!!! (#1143)
* try with the smallest compression (#1143)
* use timer set at framesize (#1143)
[ Emmanuel Milou ]
* [#1161] Restore changelog version
[ Alexandre Savard ]
* Remove celt stuff
[ Emmanuel Milou ]
* [#1161] Update changelog
* [#1220] Add Conflicts: sflphone in debian control files
* [#1179] Add liblog4c3 runtime dependency
* [#1212] FIx typo error in dependency list for itnrepid
* [#1212] FIx .desktop file to point on the right exec
* [#1212] Modify changelog replacing tag
[ Sflphone Project ]
* "[#1262] Updated changelogs for version 0.9.5-0ubuntu1~beta"
[ Emmanuel Milou ]
* [#1212] restore changelogs
[ Sflphone Project ]
* [#1262] Updated changelogs for version 0.9.5-0ubuntu1 Snapshot 2009-
04-27
[ Emmanuel Milou ]
* [#1212] restore changelogs
[ Sflphone Project ]
* [#1262] Updated changelogs for version 0.9.5-0ubuntu1~beta
[ Emmanuel Milou ]
* [#1212] restore changelogs
[ Sflphone Project ]
-- Sflphone Project <sflphone@mtl.savoirfairelinux.net> Mon, 27 Apr 2009 16:57:00 -0400
sflphone-common (0.9.4-0ubuntu2) SYSTEM; urgency=low
[ Alexandre Savard ]
* Restore speex and GSM detection
[ Emmanuel Milou ]
* Fix bug #1090
-- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Wed, 8 Apr 2009 11:29:15 -0500
sflphone (0.9.4-0ubuntu1) SYSTEM; urgency=low
[ Emmanuel Milou ]
* Integrate DBus-c++ and libiax2 in the main build system
* Clean up in the working repository
* Reorder hooks configuration panel
* Protect case when no codecs are active
* Fix some return values
* Add unitary tests for the hook manager (premisces)
[Yun Liu]
* Update chinese translation
[Sven Werlen]
* Update german translation
[Hussein Abdallah]
* Update russian translation
[Maxime Chambreuil]
* Update spanish translation
-- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 3 Apr 2009 18:29:15 -0500
sflphone (0.9.4-rc1) SYSTEM; urgency=low
[ Emmanuel Milou ]
* Fix bug while trying to hold/unhold several simultaneous call
* Improve address book build system
* Implement SIP url popup on incoming call
* Improve GTK+ panel configuration
[ Julien Bonjean ]
* GTK+ client refactoring
* GTK+ clean up
* Address book improvment
-- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 27 Mar 2009 18:29:15 -0500
sflphone (0.9.4-0beta1) SYSTEM; urgency=low
[ Alexandre Savard ]
* Display codec used during conversation on the GUI
* Enable/disable STUN parameters at runtime
* Refactor search bar use
[ Emmanuel Milou ]
* Build system fixes
* Implement SIP re-invite
* Implement IP to IP call
[ Julien Bonjean ]
* Integrate GNOME address book based on evolution data server
-- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 20 Mar 2009 18:29:15 -0500
sflphone (0.9.3-0ubuntu3) SYSTEM; urgency=low
[ Alexandre Savard ]
* Both playback and record streams in PA_STREAM_CORKED (pulseaudio)
* Use PLUGHW device for ALSA capture
* Functional IAX and SIP recording for voicemail
* Use the less CPU-consuming interpolator algorithm for resampling
* Display in GTK GUI the codec used in conversation
* GTK GUI use ASCII instread of utf-8
* Add record menus in GTK GUI
* Put on hold when dialing a new number
* AccountID's are saved in the history
[ Emmanuel Milou ]
* Integrate DBUS C++, libiax2 in the git repository
* Update website
* Use libspeexdsp only if available on the system
* Updated .gitignore file
[Cyrille Béraud]
* Account assistant manager improvment
* Add an email request when creating a new account to receive voicemails
-- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Sat, 14 Feb 2009 13:29:15 -0500
sflphone (0.9.3-0ubuntu2) SYSTEM; urgency=low
[ Emmanuel Milou ]
* Add compilation note in README
* Use default ALSA plugin for capture
* Fix the ALSA capture problem one more time
* Clean up debug messages in dbus.c
* Add libspeexdsp dependency
* Remove implicit declaration compilation warnings
* Fix links in the website, add release note
* Change capture for the website front page
* Add alsa devel dependency in build-depends control file field
* Clean up, indentation, try to handle latency problems in iax/pulseaudio
* Remove pjsip generated files from the repo
* Use the previous declared curAlias function in accountwindow
* Fix bug in history call duration when the call fails
* Remove runtime warning in the GTK+ client
* Add librsvg2-common dependency to load SVG under KDE
* Refresh .gitignore
* Update locales files + french translation
* Add configuration panel for future noise reduction
* Add configuration panel for audio record module
* Daemon less verbose; accounts don't try to access STUn options anymore
* Fix typo in configwindow
* Add content in the official website
* use a GTK_STOCK icon for the record button
* Complete description text in the assistant manager
* Add libtool flags in client configure.ac
* Remove unuseful dependency (snd)
* Fix SIP transfer problems
* Remove previous version of PJSIP from the repo
* Upgrade PJSIP to version 1.0.1
* Add the new website source in the repository
* Use libspeexdsp for silence detection only if available
[ Loïc Faure-Lacroix ]
* Ajout du logo gpl3
* Ajout des images
* Ajout de la section screenshot pour le site
* Ajout du favicon dans le header
* Modification des cartes
[ Alexandre Savard ]
* Clean up <speex/libspeexdsp>
* Small cleanup
* Save Wave fixed
* Fix new call button when recording
* libspeexdsp added
* Recording: default home folder at startup
* Minor changes to config window
* IAX recording fixed
* Set / get recording path, still need some GTK for client
* AudioRecord file name format
* Now recording in HOME folder
[ Cyrille Béraud ]
* Fix bug in reqaccount.c
[ Maxime Chambreuil ]
* Update spanish translation
[Yun Liu ]
* Update chinese translation
[ Hussein Abdallah ]
* Update russian translation
-- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Sat, 14 Feb 2009 13:29:15 -0500
sflphone (0.9.3-0ubuntu1) SYSTEM; urgency=low
* Remove debug
* Join thread before leaving
* Fix implicit declaration in reqaccount
* Add REST code to build the request to server
* Fix GValue initialization warnings
* Update version number, fix implicit declaration, fix GTK markup
warnings
* Apply patch to create custom SIP account from our own server
-- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 06 Feb 2009 19:17:32 -0500
sflphone (0.9.2-2ubuntu9) SYSTEM; urgency=low
[ Alexandre Savard ]
* Speex audio codec preprocessing initialization
* peer hung up segmentation fault solved
* Stop recording when transfering
* Terminate only one call
* Add isRecording() function
* Fix call_icon GTK client
* Fix SIPCallClose() function, recorded file now close properly
* Function terminateSIPCall added in sipvoiplink and managerimpl
* Fix thread destructor
* setRecordingOption function implement in audiorecord
* Record now implemented in Call class
* Record interface complete (on hold erase previous recording)
* Added recButton in client
* Added: record button related icons
* Record button added
* Overload AudioRecord::recData to get mic and speaker data mixed
* Recording now in audiortp::run() method
* Audio recording working in AudioRTP: receiveSessionForSpeaker
* Open/close a wave file when pulse audio stream start/stop
[ Emmanuel Milou ]
* Fix path for GTK+ icons; clean up
-- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Thu, 05 Feb 2009 18:27:53 -0500
sflphone (0.9.2-2ubuntu8) SYSTEM; urgency=low
[ Emmanuel Milou ]
* Update changelogs
* Fix bug in merge and in Makefile.am
* Terminate only one call
* Disable PJsip shutdown when changing STUN parameters
* Function terminateSIPCall added in sipvoiplink and managerimpl
* Add a timer to the alsa thread to not jam the CPU load
* Fix bug in sipvoiplink.cpp
* Clean shutdown of pulseaudio on quiting
* Fix DTMF at first start with Pulseaudio
* Remove zeroconf from the build system
* Add a library manager + exception handling
* Clean up in the working directory
* Better handling of capture XRUNs
* Restore mic adjust volume on ALSA layer
* Protect device ALSA operation if not opened
* Fix the switching layer bug
* Use dynamic_cast<> to use audiolayer-specific methods
* Open the audio devices only once at startup
* Refactoring of the ALSA part
* Functional plug-in manager
* Use a C++ thread to handle tones and DTMF in ALSA
* Restore IAXVoIPLink, restore Mutex
* Make the plugins registering against the plugin manager
* Migrate to 1->N relationship between voiplink and accounts
* API plugin for registration
* Use C++ thread in SIP, move everything in sipvoiplink
* Complete singleton pattern for the plugin manager
* Add -Wno-return-type compilation flag to remove warnings; Update
version number in configure.ac
* Add the dynamic loading for the plugin framework; integate unittest
[ Yun Liu ]
* Update rpm spec file
* modify build package script and spec file for suse
[ Alexandre Savard ]
* Add audiorecorder plugin and testaudiorecorder
* Add audio Recording class, edit global.h
-- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Wed, 04 Feb 2009 14:00:30 -0500
sflphone (0.9.2-2ubuntu7) SYSTEM; urgency=low
[ Emmanuel Milou ]
* Update changelog to 0.9.2-6
* Fix some dbus-glib implementation details on the client side
* Init history after dbus initialization
* Add error checking in useragent; Clean sipvoiplink
* Prevent crash when trying to call an empty number
* Set the volume of the playback stream to PA_VOLUME_NORM at startup
* Fix GTK+ generic value double initialization
* Fix jaunty control file dependency problems
* Fix jaunty control file dependency problems
[ Yun Liu ]
* Fix bug ticket # 137
* Tolerant to gsm library of OpenSuse 11
[ Sven Werlen ]
* Update german translation
-- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 23 Jan 2009 17:48:13 -0500
sflphone (0.9.2-2ubuntu6) SYSTEM; urgency=low
[ Emmanuel Milou ]
* Migrate STUN configuration to the main config window
* Update french translation
* Other tiny memory leaks
* Fix memory leak in sampleconverter.cpp
* Generate packages from the release branch
* update the build package script
* modify the control files with architecture=any
* Remove valgring uninitialized value
* IAX and SIP use the same global variables to set account
configuration ; fix broken code
[ Maxime Chambreuil ]
* Update spanish translation
[ Hussein Abdallah ]
* Update russian translation
[ Yun Liu ]
* Update translation files
* Fix the bug when user uncheck the account which fails in the
previous registration
* Add stun error status
* Fix bug ticket #143
* Script for auto-install dependencies
* Fix bug ticket #140
* Fix bug ticket 141
* Fix the reregister process when user change the details of an
account
-- Emmanuel Milou <manu@sulfur.inside.savoirfairelinux.net> Fri, 16 Jan 2009 18:19:05 -0500
sflphone (0.9.2-2ubuntu5) SYSTEM; urgency=low
* Fix memory leak in the pulseaudio callback
* Update debian package generation script
* Warnings removal in GTK+ client
* Clean adjust volume method in alsalayer
* Plug the sflphone playback volume control to the pulseaudio volume
manager
* Display the date in history according to the current locale
* Generate the changelog according to the git commit messages
* Complete header in chinese translation file
* Use the right gpg key to sign the packages
* add debian jaunty jackalope support
-- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Wed, 14 Jan 2009 21:17:20 -0500
sflphone (0.9.2-2ubuntu4) SYSTEM; urgency=low
[ Emmanuel Milou ]
* add german translation
[ Yun Liu ]
* Fix GUI crash in Ubuntu8.10 64bit system
-- Yun Liu <yun.liu@savoirfairelinux.com> Thu, 08 Jan 2009 13:08:51 -0500
sflphone (0.9.2-2ubuntu3) SYSTEM; urgency=low
[ Emmanuel Milou ]
* The main thread synchronizes the ringtone thread
* disable custom ringtone for the ALSA layer
* Fix the Makefile.am in man directory, add a SEE ALSO section
[ Yun Liu ]
* Fix daemon crash caused by the previous patch ( for bug ticket #129)
-- Yun Liu <yun.liu@savoirfairelinux.com> Tue, 06 Jan 2009 16:18:38 -0500
sflphone (0.9.2-2ubuntu2) SYSTEM; 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
* Migrate from eXosip library to pjsip
* Add multiple SIP accounts support
* Fix ringtones problems
* Add a pulseaudio support
* Improve audio quality with ALSA
* Add chinese translation
* Improve spanish translation
* Migrate to a maintained C++ DBus bindings
* Clean and improve the build system
* Add build-dependency on Perl because we need pod2man to generate manpages
-- Yun Liu <yun.liu@savoirfairelinux.com> Wed, 26 Nov 2008 09:47:53 -0500
sflphone (0.9.1) unstable; urgency=low
* Add a search tool in the history
* Migrate some gtk_entry_new to sexy_icon_entry_new
* Bug fix (Ticket #78): The voicemail password isn't displayed anymore in
the history tab
* Add the SIP registration expire value in the user file.
-- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Thu, 22 May 2008 11:14:25 -0500
sflphone (0.9.0) unstable; urgency=low
* Add history features
* Call date
* Call duration
* Mouse events in the history tab
* Smooth switch from the history tab to the calls tab
* Remove most of GTK-Critical warnings
-- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Wed, 13 May 2008 16:58:25 -0500
sflphone (0.9-2008-06-06) unstable; urgency=low
* Audio bug correction: capture stopped after a few minutes of conversation
with USB Plantronics sound card
-- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Tue, 06 May 2008 16:58:25 -0500
sflphone (0.9-2008-05-06) unstable; urgency=low
* Bug correction: account creation with the assistant
* GTK+ warnings removal
* libnotify warnings removal
* Remove aliasing on the SFLphone logo
-- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Mon, 05 May 2008 16:58:25 -0500
sflphone (0.9) unstable; urgency=low
* Clean dependencies ( removal of libboost )
* Several GTK improvement and updates
-account window
-configuration window
* Migrate from GtkCheckMenuItem to GtkImageMenuItem
* ALSA standard I/O transfers: MMAP instead of R/W
* Fix speex audio quality
* IAX2 protocol
-Fix hold/unhold situation
-Add on hold music
* SIP protocol
-Ringtone on incoming call
-Fix transfer situation
* Add desktop notification ( libnotify )
* Improve the system tray icon behaviour
* Improve registration error handling
* Register/unregister from the account window takes effect without starting back SFLphone
* Compilation warnings removal
* Call history
* Add an account configuration wizard
-- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Wed, 30 Apr 2008 16:58:25 -0500
sflphone (0.8.2) unstable; urgency=low
* Internationalization of the GTK GUI
* English / French
* STUN support
* Slight modifications of the graphical interface ( tooltips, dialpad, ...)
-- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 21 Mar 2008 11:37:53 -0500
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