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Commit 84f8150b authored by Emmanuel Milou's avatar Emmanuel Milou
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Merge branch 'jbonjean'

parents 05741be6 4fbeeec5
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......@@ -23,7 +23,7 @@ LAUNCHPAD_PACKAGES=( "sflphone-client-gnome" "sflphone-common" )
REFERENCE_REPOSITORY="${ROOT_DIR}/sflphone-source-repository"
SOFTWARE_VERSION="0.9.7.beta"
SOFTWARE_VERSION="0.9.7.rc2"
VERSION_INDEX=1
......
......@@ -217,6 +217,7 @@ END
fi
done
cp ${DEBIAN_DIR}/changelog.generic ${DEBIAN_DIR}/changelog
done
# if push is activated
......
sflphone-client-gnome (0.9.7~rc1~ppa1~SYSTEM) SYSTEM; urgency=low
** 0.9.7~rc1~ppa1~SYSTEM **
* [#2462] Set explicitly the transport on incoming call too
* [#2462] fix typo
* [#2462] Use different address for SDP and call IP
* [#2462] Use published address in SIP-SDP
* [#2181] Fixed changelog files
* [#2181] Updated spec file
* [#2402] Fix pointer to int conversion warning (atoi)
* [#2402] Remove daemon warnings, make indent
* [#2459] Make sure the stream is opened when the call is answered
* [#2402] Add conference related picture in documentation
* [#2443] Not much ...
* [#2399] Fix dialing display problem
* [#2450] Fix incoming call already in conference crash
* [#2399] Display peer name on the first line and peer number on the
second
* [#2450] Handle 403 FORBIDDEN when refused
* [#2447] Bind offHold/onHold actions to button in gtk client
* [#2447] Bind hangup action to button for conference
* [#2447] Add conference action in gtk client's ToolBar
* [#2381] Disable the password hashing in config file
* [#2402] Cleanup
* [#2366] Set callback to null when deleting Pulseaudio streams
* [#1313] Fix main buffer unit test
* [#1313] Fix audio layer unit test
* [#2315] Hide pw in security tab, display when editing, sync with
basic tab
* [#1313] UnitTest change AudioRtpSession for AudioSymetricRtpSession
instance
* [#2402] Code cleanup
* [#2444] Add debug to catch occasional crash when loading client's
config
* [#2444] Add debug info to catch occasional crash when loading config
dialog
* [#2402] Restore Call menu translations
* [#2403] Use the published address if checked in GUI
* [#2442] Add protection test in sdp
* [#1841] Reapply pjsip patch concerning DNS SRV resolution
* [#2384] Tags incoming call as direct SIP call, if applicable
* [#2402] Change the monkey face
* [#2315] Enable user to display password in clear text
* [#2434] Force optimization level at 2
* [#2284] Fix dbus_get_all_ip_interface compilation warnings
* [#2431] Popup main window on incoming if applicable
* [$2402] Fix simple warnings
* [#2402] Fix implicit variable init order in LibraryManagerException
* [#2402] Fixing implicit variable initialization warnings in
AudioRtpSession
* [#2402] Revert atoi change, fixing codec list doubled entries
* [#2402] Fix gpointer to gint conversion
* [#2402] Fix pointer casting to integer different size warning in
codec list
* [#2402] Fix warning discarting qualifiers from pointer target
* [#2402] Fix gtk tree view assignement from incompatible type warning
* [#1669] Fix audio recording folder utf-8 non compatibility issue
* [#2414] Clean up debugs
* [#2414] Use transport set in iptoip Account and update it frm
preference
* [#2348] Use macro N_() to mark ui.xml strings as translatable
* [#2414] Rename getSipAddress/setSipAddress functions
* [#2407] Fix volume controls display
* [#2407] Fixes dialpad
* [#2383] Set ip to ip config when clicking apply button
* [#2404] Update call-to script - Maxime Chambreuil
* [#2405] Client handles unknown call in current state as well
* [#2383] Add DBUS signal to send IPtoIP local address and port as
string
* [#2383] Add Ip to IP config change apply call back
* Clonflict
* [#2402] Code cleanup
* [#2383] Do the same for IPtoIP (init localn ip with first in the
list)
* [#2383] Use first interface in the list if local addresss is not
defined
* [#2403] Clean up unuseful addresses/ports
* [#2403] Use the IP profile SIP port as global SIP port
* [#2383] Fix dbus_get_all_ip_interface warnings
* [#2383] Take into account sameAsLocal when loading published address
* [#2383] Tsake into account sameAsLocal option when saving published
address
* [#2383] Update local ip address in ip to ip config
* [#2383] Save ip 2 ip local port in config
* [#2406] Update toolbar at startup
* [#2284] Remove redefinition warnings + speex warnings
* [#2383] Fix security table in account config
* [#2383] Save ip 2 ip network interface parameters in config
* [#2403] Restore sip transport selector
* [#2383] Fix filling the Localt IP Address on account creation
* [#2383] Fix Gtk-Critical when checking STUN
* [#2383] Fix reopening account configuration display issue
* [#2383] Load IPtoIP local address and port in preference iptoiptab
* [#2383] Add LocalAddress and Localport in Preference IpToIp tab
* [#2403] Use the address and port associated to the account as often
as possible
* [#1753] Removed pjsip generated files
* [#1753] Removed remaining milenage lib references
* [#2383] Add _publishedSameasLocal variable in sipaccount
* [#2383] Add PUBLISHED_SAMEAS_LOCAL variable in config
* [#2383] Fix stun set active or not when opening config
* [#2181] Added RPM 64bits dbus patch
* [#2402] Code indentation
* [#2313] Force $(HOME).cache directory creation at startup
* [#2383] Separate network interface and published address in account
config
* [#2400] Change dbus service installation path to libdir
* [#2382] Move TLS related published address options in security tab
* [#2382] Indent accountconfigdialog.c
* [#2181] Install libdbus-c++ in $pkglib instead of $lib
* [#1753] Remove ILBC code and disable it by default in the configure
* [#1753] Remove milenage directory
* [#2382] Fix switching interaface instabilities
* [#2396] Save local ip in account creation wizard
* [#2284] Remove warning on hold
* [#2387] Fixes history searching and filtering
* [#1215] Add samplerate display in the GUI
* [#1663] Voicemail icon reflects voice messages
* [#2395] Fix account registration ( specifically with callcentric)
* [#2386] Strip "sip:" on incoming call, fixing history call back
* [#2181] Updated spec files
* [#1215] Display codec name in calltree instead of status bar
* [#2390] Move back nbCalls and stopStream higher in refuseCall
* [#2392] Fix ringtone during call in IAX
* [#2391] Stop audio streams when there is 0 calls only
* [#2391] Add debug when call state is not valid
* [#2390] Clear returns in IAXvoipLink::sendAudioFromMic() method
* [#2380] Fixing IncomingCallNotification not regular
* [#2339] Query conference at client startup
* [#2339] Working conference querying at startup
* [#2339] Add conference in call tree
* [#2339] Primitives to query conferences at client startup
* [#2320] Add account selection in history
* [#2355] Temporary solution: do not delete pointer when removing
account
* [#2380] Change algorithm in AudioRtp to trigger an
IncomingCallNotification
* [#2274] Comment sdebug in MainBuffer flush method
* [#2274] Add flushMain() in ManagerImpl::addStream
* [#2274] Add getBufferID() method in ring buffer
* [#2274] Fix warning, comment debug in ringbuffer's flush method
* [#2274] Use AudioLayer flushMain() and flushUrgent() in ALSA
* [#2274] Clean up unused variable warning
* [#2274] Protect minbudffer pointer on flushing
* [#2274] Fix playATone method which writing empty buffer in urgent
ringbuffer
* [#2274] Use audio layer flushUrgent and flushMain in createStreams
* [#2274] Use flush audio calls from audiolayer
* [#2274] Flush when peer answered call
* [#2375] Flush main buffer in iax when answering a call
* [#2274] Parse displayname using c++ string method
* [#2375] Flush main buffer when off holding calls
* [#2375] Flush main buffer mon RTP startup
* [#2376] Use now Pulseaudio module-cork-music-on-phone
* Updated OSC packaging
-- Julien Bonjean <julien.bonjean@savoirfairelinux.com> Fri, 20 Nov 2009 13:59:02 -0500
sflphone-client-gnome (0.9.7~beta~ppa1~SYSTEM) SYSTEM; urgency=low
** 0.9.7~beta~ppa1~SYSTEM **
......
sflphone-common (0.9.7~rc1~ppa1~SYSTEM) SYSTEM; urgency=low
** 0.9.7~rc1~ppa1~SYSTEM **
* [#2462] Set explicitly the transport on incoming call too
* [#2462] fix typo
* [#2462] Use different address for SDP and call IP
* [#2462] Use published address in SIP-SDP
* [#2181] Fixed changelog files
* [#2181] Updated spec file
* [#2402] Fix pointer to int conversion warning (atoi)
* [#2402] Remove daemon warnings, make indent
* [#2459] Make sure the stream is opened when the call is answered
* [#2402] Add conference related picture in documentation
* [#2443] Not much ...
* [#2399] Fix dialing display problem
* [#2450] Fix incoming call already in conference crash
* [#2399] Display peer name on the first line and peer number on the
second
* [#2450] Handle 403 FORBIDDEN when refused
* [#2447] Bind offHold/onHold actions to button in gtk client
* [#2447] Bind hangup action to button for conference
* [#2447] Add conference action in gtk client's ToolBar
* [#2381] Disable the password hashing in config file
* [#2402] Cleanup
* [#2366] Set callback to null when deleting Pulseaudio streams
* [#1313] Fix main buffer unit test
* [#1313] Fix audio layer unit test
* [#2315] Hide pw in security tab, display when editing, sync with
basic tab
* [#1313] UnitTest change AudioRtpSession for AudioSymetricRtpSession
instance
* [#2402] Code cleanup
* [#2444] Add debug to catch occasional crash when loading client's
config
* [#2444] Add debug info to catch occasional crash when loading config
dialog
* [#2402] Restore Call menu translations
* [#2403] Use the published address if checked in GUI
* [#2442] Add protection test in sdp
* [#1841] Reapply pjsip patch concerning DNS SRV resolution
* [#2384] Tags incoming call as direct SIP call, if applicable
* [#2402] Change the monkey face
* [#2315] Enable user to display password in clear text
* [#2434] Force optimization level at 2
* [#2284] Fix dbus_get_all_ip_interface compilation warnings
* [#2431] Popup main window on incoming if applicable
* [$2402] Fix simple warnings
* [#2402] Fix implicit variable init order in LibraryManagerException
* [#2402] Fixing implicit variable initialization warnings in
AudioRtpSession
* [#2402] Revert atoi change, fixing codec list doubled entries
* [#2402] Fix gpointer to gint conversion
* [#2402] Fix pointer casting to integer different size warning in
codec list
* [#2402] Fix warning discarting qualifiers from pointer target
* [#2402] Fix gtk tree view assignement from incompatible type warning
* [#1669] Fix audio recording folder utf-8 non compatibility issue
* [#2414] Clean up debugs
* [#2414] Use transport set in iptoip Account and update it frm
preference
* [#2348] Use macro N_() to mark ui.xml strings as translatable
* [#2414] Rename getSipAddress/setSipAddress functions
* [#2407] Fix volume controls display
* [#2407] Fixes dialpad
* [#2383] Set ip to ip config when clicking apply button
* [#2404] Update call-to script - Maxime Chambreuil
* [#2405] Client handles unknown call in current state as well
* [#2383] Add DBUS signal to send IPtoIP local address and port as
string
* [#2383] Add Ip to IP config change apply call back
* Clonflict
* [#2402] Code cleanup
* [#2383] Do the same for IPtoIP (init localn ip with first in the
list)
* [#2383] Use first interface in the list if local addresss is not
defined
* [#2403] Clean up unuseful addresses/ports
* [#2403] Use the IP profile SIP port as global SIP port
* [#2383] Fix dbus_get_all_ip_interface warnings
* [#2383] Take into account sameAsLocal when loading published address
* [#2383] Tsake into account sameAsLocal option when saving published
address
* [#2383] Update local ip address in ip to ip config
* [#2383] Save ip 2 ip local port in config
* [#2406] Update toolbar at startup
* [#2284] Remove redefinition warnings + speex warnings
* [#2383] Fix security table in account config
* [#2383] Save ip 2 ip network interface parameters in config
* [#2403] Restore sip transport selector
* [#2383] Fix filling the Localt IP Address on account creation
* [#2383] Fix Gtk-Critical when checking STUN
* [#2383] Fix reopening account configuration display issue
* [#2383] Load IPtoIP local address and port in preference iptoiptab
* [#2383] Add LocalAddress and Localport in Preference IpToIp tab
* [#2403] Use the address and port associated to the account as often
as possible
* [#1753] Removed pjsip generated files
* [#1753] Removed remaining milenage lib references
* [#2383] Add _publishedSameasLocal variable in sipaccount
* [#2383] Add PUBLISHED_SAMEAS_LOCAL variable in config
* [#2383] Fix stun set active or not when opening config
* [#2181] Added RPM 64bits dbus patch
* [#2402] Code indentation
* [#2313] Force $(HOME).cache directory creation at startup
* [#2383] Separate network interface and published address in account
config
* [#2400] Change dbus service installation path to libdir
* [#2382] Move TLS related published address options in security tab
* [#2382] Indent accountconfigdialog.c
* [#2181] Install libdbus-c++ in $pkglib instead of $lib
* [#1753] Remove ILBC code and disable it by default in the configure
* [#1753] Remove milenage directory
* [#2382] Fix switching interaface instabilities
* [#2396] Save local ip in account creation wizard
* [#2284] Remove warning on hold
* [#2387] Fixes history searching and filtering
* [#1215] Add samplerate display in the GUI
* [#1663] Voicemail icon reflects voice messages
* [#2395] Fix account registration ( specifically with callcentric)
* [#2386] Strip "sip:" on incoming call, fixing history call back
* [#2181] Updated spec files
* [#1215] Display codec name in calltree instead of status bar
* [#2390] Move back nbCalls and stopStream higher in refuseCall
* [#2392] Fix ringtone during call in IAX
* [#2391] Stop audio streams when there is 0 calls only
* [#2391] Add debug when call state is not valid
* [#2390] Clear returns in IAXvoipLink::sendAudioFromMic() method
* [#2380] Fixing IncomingCallNotification not regular
* [#2339] Query conference at client startup
* [#2339] Working conference querying at startup
* [#2339] Add conference in call tree
* [#2339] Primitives to query conferences at client startup
* [#2320] Add account selection in history
* [#2355] Temporary solution: do not delete pointer when removing
account
* [#2380] Change algorithm in AudioRtp to trigger an
IncomingCallNotification
* [#2274] Comment sdebug in MainBuffer flush method
* [#2274] Add flushMain() in ManagerImpl::addStream
* [#2274] Add getBufferID() method in ring buffer
* [#2274] Fix warning, comment debug in ringbuffer's flush method
* [#2274] Use AudioLayer flushMain() and flushUrgent() in ALSA
* [#2274] Clean up unused variable warning
* [#2274] Protect minbudffer pointer on flushing
* [#2274] Fix playATone method which writing empty buffer in urgent
ringbuffer
* [#2274] Use audio layer flushUrgent and flushMain in createStreams
* [#2274] Use flush audio calls from audiolayer
* [#2274] Flush when peer answered call
* [#2375] Flush main buffer in iax when answering a call
* [#2274] Parse displayname using c++ string method
* [#2375] Flush main buffer when off holding calls
* [#2375] Flush main buffer mon RTP startup
* [#2376] Use now Pulseaudio module-cork-music-on-phone
* Updated OSC packaging
-- Julien Bonjean <julien.bonjean@savoirfairelinux.com> Fri, 20 Nov 2009 14:00:02 -0500
sflphone-common (0.9.7~beta~ppa1~SYSTEM) SYSTEM; urgency=low
** 0.9.7~beta~ppa1~SYSTEM **
......
......@@ -125,6 +125,7 @@ make clean
%lang(zh_HK) %{_prefix}/share/locale/zh_HK/LC_MESSAGES/*.mo
%lang(it) %{_prefix}/share/locale/it/LC_MESSAGES/*.mo
%lang(pt_BR) %{_prefix}/share/locale/pt_BR/LC_MESSAGES/*.mo
%lang(da) %{_prefix}/share/locale/da/LC_MESSAGES/*.mo
%doc AUTHORS COPYING README
%doc %{_prefix}/share/man/man1/sflphone-client-gnome.1.gz
%doc %{_prefix}/share/man/man1/sflphone.1.gz
......
diff --git sflphone-common/src/dbus/Makefile.am sflphone-common/src/dbus/Makefile.am
index 5a2745e..5ca2e42 100644
--- sflphone-common/src/dbus/Makefile.am
+++ sflphone-common/src/dbus/Makefile.am
@@ -38,7 +38,7 @@ service_DATA = $(service_in_files:.service.in=.service)
# Rule to make the service file with bindir expanded
$(service_DATA): $(service_in_files) Makefile
- sed -e "s|libexec|$(prefix)/lib/sflphone|" $<> $@
+ sed -e "s|libexec|$(libdir)/sflphone|" $<> $@
EXTRA_DIST = *.xml README
......@@ -21,7 +21,6 @@ Packager: Julien Bonjean <julien.bonjean@savoirfairelinux.com>
BuildRoot: %{_tmppath}/%{name}
Source0: sflphone-common-%{version}.tar.gz
Patch0: sflphone-common-dbus-service-in-libdir.patch
BuildRequires: speex-devel
BuildRequires: gcc-c++
BuildRequires: expat
......@@ -95,7 +94,6 @@ Authors:
%prep
%setup -q
%patch0 -p1
%build
cd libs/pjproject
......@@ -127,7 +125,7 @@ make clean
%dir %{_libdir}/sflphone/plugins
%dir %{_prefix}/share/sflphone
%dir %{_prefix}/share/sflphone/ringtones
%{_libdir}/libdbus-*
%{_libdir}/sflphone/libdbus-*
%{_libdir}/sflphone/codecs/*
%{_libdir}/sflphone/plugins/*
%{_prefix}/share/dbus-1/services/org.sflphone.*
......
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