baresip: update to baresip-1.1.0

[Yorick Hardy <yorickhardy%gmail.com@localhost>]

Module Name:	pkgsrc-wip
Committed By:	Yorick Hardy <yorickhardy%gmail.com@localhost>
Pushed By:	yhardy
Date:		Sat May 1 11:54:28 2021 +0200
Changeset:	432c55e4d92227cdcd61001b79365a377538c2a7

Modified Files:
	baresip/Makefile
	baresip/PLIST
	baresip/distinfo

Log Message:
baresip: update to baresip-1.1.0

The CHANGELOG.md file does not show this release; this is the change log
provided in the released files:

==Baresip Changelog

== [Unreleased]

- cons: emulate key-release
- Correct reverse domain name notation
- gtk with account_uri_complete
- bump version to 1.1.0
- ui: fix leaking of cmd_ctx
- DTMF tones for A B C D
- account: use a fixed username for the template
- contact: update contacts template
- config: disable ctrl_dbus in config template
- Module event
- add event UA_EVENT_MODULE to tell to app when snapshot has been written
- ringtone: generated busy and ringback tone
- audio: prevent restart of rx_thread on call termination
- modules: update auplay/ausrc modules
- Auplay remove inheritance
- h264: add doxygen comment
- vidloop: add VIDEO_SRATE
- vidloop: check error
- vidloop: add vidframe_clear
- vidloop: split enable_codec into encoder/decoder
- Ausrc remove inheritance
- ua: remove prev call
- sndfile: get number of bytes from auframe
- plc: check format of struct auframe
- speex_pp: check format of struct auframe
- webrtc_aec: use format from struct auframe
- README: update codecs and RFCs
- menu: use uri complete for command dialdir
- video: check for video display before calling handler
- Changed name and made public
- menu: return call-id for dial and dialdir
- Fixes for account uri complete
- Avoid compiler warnings:
- Avoid compiler warnings (I haven't found anything wrong with the code)
- vidfilt: fix warning
- vidfilt: split parameters into encode/decode
- snapshot: fix warnings
- video: group functions from vidutil.c
- avfilter: fix warnings
- vumeter: use format from audio frame
- replaced ua_uri_complete with account_uri_complete
- aulevel: move to librem
- omx: fix warning
- vidisp: remove inheritance
- docs: change video settings to match the default values
- menu: select call in cmd_find_call()
- menu: use menu_stop_play()
- main: unload app modules in signal handler
- avformat: replace const double with double
- avformat: clean up ifdefs
- ci: drop ubuntu 16.04 support - end of life
- avformat: proper code formatting
- avcodec: add avcodec prefix to log messages
- avcodec: check length of H265 packet
- x11grab: remove vidsrc inheritance
- v4l2: remove vs inheritance
- vidsrc: remove concept of baseclass/inheritance
- ua,menu: remove uag_find_call_state
- Updated homepage
- sdl: correct aspect-ratio in fullscreen mode
- vidloop: add vidisp parameters
- auloop: use auframe_size
- audio: use auframe_size
- Auplay use auframe
- Docs examples config
- Serreg fixes
- Update config.c
- contact: use uag_find_requri()
- ua: use new tls function to set cafile and path
- config: add sip_capath config line
- Call event answered fixes alsa issue
- ctrl_dbus: send DBUS signal when dbus interface is ready
- Multicast call priority
- Menu fixes for play tones2
- gst: add missing include unistd.h
- multicast: cleanup function description and fix doxygen warning
- menu: remove call resume for command hangup
- ua: add a generic filter API for calls
- Merge pull request #1288 from cspiel1/remove_call_resume_on_termination
- menu: remove call resume on termination
- multicast: fix build error when using HAVE_PTHREAD=
- alsa_play.c add suggestion to use dmix
- readme.md: added multicast module
- audiounit: fix typo
- update copyright year
- config cleanup
- update copyright year
- conf: add call_hold_other_calls config option
- config.c: added rtmp to config template
- main.c: update year
- The avformat_decoder should be optional
- src/audio: set started false with audio_stop
- readme: update baresip fork links
- ausine: mono support and stereo_left/right option
- menu: fix incoming calls are not selected on call termination
- test: remove mock_aucodec, using g711 instead
- opengl: remove deprecated module
- Added account_dtmfmode and account_set_dtmfmode API functions
- avcodec: remove support for MPEG4 codec
- call: start streams asynchronously (issue #1261)
- audio: remove special handling of Comfort Noise
- multicast: fix one doxygen warning
- menu: update doxygen comment
- menu: correct hangupall command for parallel call feature
- menu: on call termination select another active call
- ua: correct doxygen of uag_hold_resume()
- menu: simplify cmd_hangupall()
- support for sending of DTMF INFO
- Menu optional call parameter
- cleanup tabs and spaces
- ua: correct doxygen for uag_hold_others()
- ua: add doxygen for call find functions
- menu: add doxygen to cmd_hangup(), cmd_hold(), cmd_resume()
- menu: command accept searches all User-Agents for an incoming call
- ua: add function uag_find_call_state()
- menu: print correct warning for hangup, accept, hold, resume
- menu: add optional parameter call-id to cmd_call_resume()
- menu: add optional parameter call-id to cmd_call_hold()
- menu: add optional parameter call-id to cmd_hangup()
- menu: add optional parameter call-id to cmd_answerdir()
- menu: add utility function that decodes complex command parameters
- menu: use SDP_SENDRECV for cmd_answerdir() as fallback
- menu: add optional parameter call-id to cmd_answer()
- ua: add call find per call-id function
- call: call_info() prints also the call-id
- ua: in ua_print_calls() print User-Agent info in header
- menu: ua NULL check for answer command
- replace spaces with tab
- removed newline
- undid httpreq spacing
- fixed line too long
- moved multicast template to end of config template
- ua: fix uag_hold_others use of wrong list element
- added multicast enabled message
- updated date and added multicast to signaling
- Merge pull request #1248 from webstean/patch-2
- Added newline to multicast comment
- Menu ensure only one established call
- Call resume on hangup
- menu: for call answer search all UAs for calls to put on hold
- ua: ua_answer() should answer same call like ua_hold_answer()
- ua: make ua_find_call_state() global usable
- Add multicast_listener to config template
- Update config template to include multicast module
- menu: if a call becomes established then put others on hold
- ua: add uag_hold_others()
- Fix multiple resumed calls
- Merge pull request #1241 from cHuberCoffee/cmd_hangupall
- RFC: Make avformat decode mjpeg v4l2 with vaapi
- ua: add doxygen for new uag_hold_resume()
- menu: fix missing callid of menu at call closed
- menu: use uag_hold_resume to ensure only one active call
- ua: on call resume check for other active calls
- menu: new hangupall command with direction parameter
- readme: update supported compilers and ssl libs
- menu: fix redial
- Fix spaces
- Multicast module
- menu: use print backend pointer pf correctly
- menu: start ringback only once for parallel calls
- jack: support port pattern in config file
- config: disables server verification if sip_verify_server is missing
- ua: for UA selection allow arbitrary aor for regint=0 accounts
- Ctrl dbus synchronize
- event: encode also remote audio direction
- Merge pull request #1235 from cspiel1/event_add_string_for_UA_EVENT_CUSTOM
- event: add string for UA_EVENT_CUSTOM
- Mimic ifdef on avutil version for hwcontext
- Fix to tabs and improve checks
- src/config: show sip_cafile warning only if sip_verify_server is enabled
- Avoid compiler warnings using casts
- test: disable SIP TLS server verification
- config,ua: add config flag disable SIP TLS server verification
- alsa/play: snd_pcm_writei error codes are negative
- alsa: fix clang warnings "conversion loses integer precision"
- Intelligent call answer
- Remove uag next
- Merge pull request #1219 from cspiel1/message_reply_once
- menu: update switch_audio_player
- Make vaapi/mjpeg options of avformat
- src/config: no sip_cafile wording
- message: reply only once
- src/ua: only warn if tls_add_ca fails, same as undefined cafile
- src/config: add sip_cafile warning and enable by default
- ua: change log message from warning to info
- video: fix video payload text
- Make avformat decode mjpeg v4l2 with vaapi
- ua: improve UA selection for incoming calls
- ua: limit account matches for incoming calls to non-registrar accounts
- ua: check for NULL parameter in uag_find_msg()
- ua: early exit for AF_UNSPEC in uri_match_af()
- ua: use sip_transp_decode() in uri_match_transport()
- ua: use arrays in uri_host_local()
- test: add test for deny UDP peer-to-peer call
- ua: improve UA selection for incoming calls
- Sip message to application
- opus: Ensure (re)init of fmtp strings
- ctrl_dbus: generate dbus interface during build
- mod_gtk: switch to gtk 3
- menu: set_answer_mode: apply all uas
- menu: find_call: search all user-agents
- menu: fix usage of ua
- isac: remove deprecated module
- menu: cmd_print_calls: print all uas
- Fix interaction between CLI menu and GTK menu
- menu: rename menu_current() to menu_uacur()
- webrtc_aec: fix compilation with gcc 4.9 (fix #1193)
- win32: add cons module, fixes #1197
- ua: remove ua_aor() -- use account_aor() instead
- gtk: use account_aor()
- menu: use account_aor()
- presence: use account_aor()
- modules: use account_aor()
- account: fix video codes decode
- core: use account_aor()
- Merge pull request #1198 from baresip/av1
- Avoid unused parameter warning
- debug_cmd: add UA_EVENT_CUSTOM
- fix decoder changed debug text
- cairo: minor debug tuning
- menu: add uadelall to delete all user agents
- use account_aor()
- mctrl: remove support for media-control (deprecated)
- update doxygen comments
- ua: minor cleanup
- ua: split struct uag from instance
- README: add RFC 5373
- menu: fix segfault on last account deletion
- call: extend SIP auto answer support for incoming calls
- Sip auto answer caller
- win32: remove timer.c
- ua: give a nice name to 'global' struct
- ua: remove ua_cur
- move uag_current to menu module
- menu: pass ua from mqtt to menu via opaque data
- Sip autoanswer callee
- ua: for answer-mode early also send INCOMING event
- gst: The error handler call for end of stream is now
- mk: also detect mqtt.so in SYSROOT_ALT
- contact: add ua_lookup_domain
- video: minor tuning of pipeline text
- gst: playback of read only audio files failed
- gtk: make a local pointer to current ua
- menu: clean up usage of uag_current()
- call: correction of remote video direction info at SDP-offer
- debug_cmd: print all user-agents
- presence: one command with status as argument
- ua: rename presence status to pstat
- ua: remove LIBRE_HAVE_SIPTRACE check, always enabled
- update doxygen comments
- mk: update doxygen config file
- menu: initialize menu with zeros
- Re mk cross build2
- net: make fallback DNS ignored message debug only
- mixausrc: improve logging
- mixausrc: fix shorten-64-to-32 warnings
- config: template for osx/ios
- Supressed clang zero length array warning
- Added ctx param to video_stop/video_stop_source and set ctx to null
- avformat: add empty line after base class
- Make macos warnings into errors
- disable mixausrc until warnings are fixed
- clang shorten-64-to-32 warnings
- Mixausrc
- aufile: fix warning on OSX
- alsa: print warning if running, fixed #1162
- Don't default stunuser/pass to account authuser/pass
- Audio file info
- gitignore: clangd cache, compile_commands.json and cleanup
- Merge pull request #1167 from baresip/video_display
- Reordered video_stop_display
- Expose video_stop_display() to API
- Video dir rename
- ci: use baresip/rem repo
- stream: add function to send a RTP dummy packet
- Play aufile extended support
- video: move video related start/stop/update into video file
- aufile: add audio player to write speaker data to wav file
- Fix compiler warnings
- play: fix warning
- play ausrc
- README: add more status badges
- README: replace travis status badge
- menu: fix uint16_t scode
- config: revert dirent.h changes
- audio: fix HAVE_PTHREAD audio_destructor
- gst ready for file play
- debug_cmd: mem_deref of player fixes segfault
- net: remove deprecated net_domain()
- update contact examples
- fix freeze on hangup
- menu: make audio files configurable
- aptx: declare variable outside for-loop
- fix warnings on openbsd
- jack: declare variable outside for loop
- account: declare variable outside for loop
- coreaudio: declare variable outside for loop
- menu: initialize menu.play fixes segfault
- ausine: declare variable outside for loop
- timer: remove tmr_jiffies_usec (replaced by libre)
- Adaptive jbuf
- Update build.yml
- mqtt: allow to separate pub from sub topic base
- video: fix warning
- mqtt: fix printing port and add tls support
- httpreq: in cmd_setauth check if parameter was given
- Merge pull request #1132 from baresip/pr-dependency-action
- ci: add pull request dependency checkouts
- audio: remove redundant union
- menu: use menu_ as prefix for global symbols
- menu: use menu_ as prefix for global symbols
- ci: add apt-get update
- menu: module refactoring
- audio, video, stream: check payload type before put to jbuf
- Cmd dialdir
- Cmd acceptdir
- event: add register fallback to event string and class name
- avformat: use %u for unsigned
- modify event type and check if peeruri null
- event: move code from ua.c
- Valgrind ci
- h264 cleanup, second part
- h264 cleanup
- Merge pull request #1113 from baresip/github-actions-v2
- ci: remove travis
- ci: add github actions - replaces travisci
- qtcapture: remove deprecated module
- test: prepare for dualstack
- test: add mock dns_server_add_aaaa
- make EXTRA_MODULES last, not first
- httpreq: fix cmd_settimeout
- test: bind network to localhost, a fix for #1090
- modules/webrtc_aec: link flags fixes
- menu: commands in alphabetical order
- httpreq: fix warning about unused args
- serreg: fix warnings about unused argument
- menu: fix warnings about unused argument
- Add a HTTP request module with authorization
- Menu: corrections for ring tones and call status by means of a global call counter
- mk: remove dirent.h
- Updating .vcxproj file for windows builds
- ccheck: change license to BSD license
- Merge pull request #1095 from baresip/websocket
- Serial registration
- Ctrl dbus
- README: remove references to creytiv.com
- Branch of baresip that includes Alfred's sip websocket patch
- Merge pull request #1091 from baresip/debian
- ua, menu: new command to print certificate issuer and subject
- .gitignore: add ctags and Vim swp files

=== Contributors (many thanks)

- [alfredh](https://github.com/alfredh)
- [robert-scheck](https://github.com/robert-scheck)
- [mbattista](https://github.com/mbattista)
- [cspiel1](https://github.com/cspiel1)
- [juha-h](https://github.com/juha-h)
- [ahinrichs](https://github.com/ahinrichs)
- [jurjen-van-dijk](https://github.com/jurjen-van-dijk)
- [sreimers](https://github.com/sreimers)
- [cHuberCoffee](https://github.com/cHuberCoffee)
- [webstean](https://github.com/webstean)
- [viric](https://github.com/viric)
- [agramner](https://github.com/agramner)
- [weili-jiang](https://github.com/weili-jiang)
- [thillux](https://github.com/thillux)
- [wkiswk](https://github.com/wkiswk)
- [philippbachmann08](https://github.com/philippbachmann08)
- [ursfassler](https://github.com/ursfassler)
- [RobertMi21](https://github.com/RobertMi21)
- [alberanid](https://github.com/alberanid)
- [agranig](https://github.com/agranig)
- [nanguantong](https://github.com/nanguantong)
- [johnjuuljensen](https://github.com/johnjuuljensen)

To see a diff of this commit:
https://wip.pkgsrc.org/cgi-bin/gitweb.cgi?p=pkgsrc-wip.git;a=commitdiff;h=432c55e4d92227cdcd61001b79365a377538c2a7

Please note that diffs are not public domain; they are subject to the
copyright notices on the relevant files.

diffstat:
 baresip/Makefile | 2 +-
 baresip/PLIST    | 9 +++++++++
 baresip/distinfo | 8 ++++----
 3 files changed, 14 insertions(+), 5 deletions(-)

diffs:
diff --git a/baresip/Makefile b/baresip/Makefile
index ba3bf814e8..8f390b2f7f 100644
--- a/baresip/Makefile
+++ b/baresip/Makefile
@@ -1,6 +1,6 @@
 # $NetBSD: Makefile,v 1.2 2014/09/05 08:06:00 thomasklausner Exp $
 
-DISTNAME=	baresip-1.0.0
+DISTNAME=	baresip-1.1.0
 CATEGORIES=	net audio
 MASTER_SITES=	${MASTER_SITE_GITHUB:=baresip/}
 GITHUB_TAG=	v${PKGVERSION_NOREV}
diff --git a/baresip/PLIST b/baresip/PLIST
index 443afa4f32..c77956afb0 100644
--- a/baresip/PLIST
+++ b/baresip/PLIST
@@ -31,6 +31,8 @@ ${PLIST.ilbc}lib/baresip/modules/ilbc.so
 ${PLIST.jack}lib/baresip/modules/jack.so
 lib/baresip/modules/l16.so
 lib/baresip/modules/menu.so
+lib/baresip/modules/mixausrc.so
+lib/baresip/modules/multicast.so
 lib/baresip/modules/mwi.so
 lib/baresip/modules/natpmp.so
 ${PLIST.opus}lib/baresip/modules/opus.so
@@ -39,8 +41,10 @@ lib/baresip/modules/plc.so
 ${PLIST.portaudio}lib/baresip/modules/portaudio.so
 lib/baresip/modules/presence.so
 ${PLIST.pulseaudio}lib/baresip/modules/pulse.so
+lib/baresip/modules/rtcpsummary.so
 ${PLIST.sdl2}lib/baresip/modules/sdl.so
 lib/baresip/modules/selfview.so
+lib/baresip/modules/serreg.so
 ${PLIST.sndfile}lib/baresip/modules/sndfile.so
 ${PLIST.speex}lib/baresip/modules/speex_pp.so
 lib/baresip/modules/srtp.so
@@ -58,6 +62,7 @@ ${PLIST.libvpx}lib/baresip/modules/vp9.so
 lib/baresip/modules/vumeter.so
 ${PLIST.x11}lib/baresip/modules/x11.so
 ${PLIST.x11}lib/baresip/modules/x11grab.so
+share/baresip/autoanswer.wav
 share/baresip/busy.wav
 share/baresip/callwaiting.wav
 share/baresip/error.wav
@@ -76,6 +81,10 @@ share/baresip/sound6.wav
 share/baresip/sound7.wav
 share/baresip/sound8.wav
 share/baresip/sound9.wav
+share/baresip/sounda.wav
+share/baresip/soundb.wav
+share/baresip/soundc.wav
+share/baresip/soundd.wav
 share/baresip/soundroute.wav
 share/baresip/soundstar.wav
 share/examples/baresip/accounts
diff --git a/baresip/distinfo b/baresip/distinfo
index 36ca9c0ae9..0d81fbfc71 100644
--- a/baresip/distinfo
+++ b/baresip/distinfo
@@ -1,8 +1,8 @@
 $NetBSD: distinfo,v 1.1 2014/08/10 00:26:01 thomasklausner Exp $
 
-SHA1 (baresip-1.0.0.tar.gz) = 575024228abf9527da8e21dbf36142043935379d
-RMD160 (baresip-1.0.0.tar.gz) = fe40d85cf978e9c17e04dde0dfd23e78f0e63018
-SHA512 (baresip-1.0.0.tar.gz) = 798ea9cd9892fe72e20d50c9a3b45b6b4c195da6d2f4052977276a7ae5ef90ac4f0c2bd6c5e48f53a4370a23f0e005dd73d1f46571965fc1bde5014b97bdc258
-Size (baresip-1.0.0.tar.gz) = 989461 bytes
+SHA1 (baresip-1.1.0.tar.gz) = 0151b56368699e7ca6605e46e71e6821e1275366
+RMD160 (baresip-1.1.0.tar.gz) = b53256dfa00c874e2789da7f5c5b489d722a5192
+SHA512 (baresip-1.1.0.tar.gz) = 82616ddfb344c4a48f742a92e9fcdc1fdd3b281950fadee0f3c3c6401d6f31e2232e9a64e5aa0bd8fc54dec02ad4c4573ff6c5a71c0929d89f83e136d35f2a3a
+Size (baresip-1.1.0.tar.gz) = 1105338 bytes
 SHA1 (patch-modules_ilbc_ilbc.c) = a2a7d685c4989bf910a9d5b8582d1261fce32e1c
 SHA1 (patch-modules_v4l2_v4l2.c) = 71ba2d1e5c8ba61eb011bd2b6b9e0d9cdaec5797