May 2014 Archives by thread
Starting: Thu May 1 07:04:20 CDT 2014
Ending: Sat May 31 04:25:47 CDT 2014
Messages: 468
- [asterisk-dev] [Code Review] 3514: res_pjsip_refer: Add Referred-By header on INVITE for blind transfers.
Matt Jordan
- [asterisk-dev] [Code Review] 3479: chan_pjsip: Call pickup test.
Matt Jordan
- [asterisk-dev] [Code Review] 3491: res_pjsip: Allow cipher to be specified by name
Joshua Colp
- [asterisk-dev] [Code Review] 3473: res_pjsip_sdp_rtp: Add tests for receiving same SDP when call is already held.
opticron
- [asterisk-dev] [Code Review] 3073: chan_sip+CEL: Add missing ANSWER and PICKUP events to INVITE/w/replaces pickup
wdoekes
- [asterisk-dev] [Code Review] 3477: Japanese language patch for app_voicemail.c and say.c
opticron
- [asterisk-dev] [Code Review] 3420: Testsuite: Call Files' Max Retries
opticron
- [asterisk-dev] [Code Review] 3490: Testsuite: Ensure that repeated device states and presence states behave as expected
Mark Michelson
- [asterisk-dev] [Code Review] 3482: func_presencestate: Make base64 encoded-ness consistent for consumers of presence state
Mark Michelson
- [asterisk-dev] [Code Review] 3505: app_chanspy: Fix a bug where barge mode only works on the first connection when multiple sessions are spied on for a channel
Joshua Colp
- [asterisk-dev] [Code Review] 3519: media_formats: Add legacy format API and move chan_iax2, chan_h323, and chan_misdn over.
Kevin Harwell
- [asterisk-dev] [Code Review] 3446: Parking: Add 'AnnounceChannel' to Park manager action. Change some announcement behavior for Park manager action.
rmudgett
- [asterisk-dev] [Code Review] 3494: ARI: Add the ability to raise an arbitrary User Event from the Applications resource
Matt Jordan
- [asterisk-dev] [Code Review] 3515: media_formats: Move chan_pjsip over.
Mark Michelson
- [asterisk-dev] [Code Review] 3518: media_formats: Move abstract jitterbuffer, audiohooks, smoother, and some core stuff over.
Mark Michelson
- [asterisk-dev] [Code Review] 3516: media_formats: Move chan_sip over.
Matt Jordan
- [asterisk-dev] [Code Review] 3512: media formats: Convert the translation core over
Mark Michelson
- [asterisk-dev] [Code Review] 3520: libpri: Add control of inband audio progress indication ie to the SETUP_ACKNOWLEDGE message.
rmudgett
- [asterisk-dev] [Code Review] 3521: chan_dahdi/sig_pri: Prevent unnecessary PROGRESS events when overlap dialing is enabled.
rmudgett
- [asterisk-dev] [Code Review] 3508: Prevent a queue member's state from getting stuck when using dynamic hints as 'state_interface'
Matt Jordan
- [asterisk-dev] [Code Review] 3501: testsuite: add tests for ari userevents
Matt Jordan
- [asterisk-dev] [Code Review] 3486: res_corosync: Fix module to work with Stasis
Matt Jordan
- [asterisk-dev] [Code Review] 3474: TLS and SRTP status not available in CLI
Jonathan Rose
- [asterisk-dev] [Code Review] 3513: Weak Reference Containers
rmudgett
- [asterisk-dev] [Code Review] 3409: app_queue: Fix for queue members receiving calls when in call and with ringinuse=no
Matt Jordan
- [asterisk-dev] [Code Review] 3388: media_formats: Move chan_mgcp, chan_unistim, and chan_skinny over.
Joshua Colp
- [asterisk-dev] [Code Review] 3410: media_formats: Move chan_multicast_rtp, chan_console, app_jack, and chan_ooh323 over.
Joshua Colp
- [asterisk-dev] [Code Review] 3522: Allow framehooks to be queried for what frame types they consume.
Joshua Colp
- [asterisk-dev] AMI Disconnect/Sudden "Asterisk Call Manager/1.3" received
Daniel McFarlane
- [asterisk-dev] [Code Review] 3428: Testsuite: ARI Playback Tones tests for channels and bridges
opticron
- [asterisk-dev] [Code Review] 3489: testsuite: Improve logging
opticron
- [asterisk-dev] [Code Review] 3506: format improvements: Port bridge_native_rtp over to new format capability API
Joshua Colp
- [asterisk-dev] Wrong entity field in NOTIFY??
Eugen Dedu
- [asterisk-dev] [Code Review] 3485: pjsip: Fix a bug where transferring to a parking extension causes calls to hang
Jonathan Rose
- [asterisk-dev] ISDN UDI Call with Dial command
Pawel Pastuszak
- [asterisk-dev] [Code Review] 3525: Testsuite: Fix ARI attended transfer test
opticron
- [asterisk-dev] [Code Review] 3529: fix case typo in H263+ mime
Guillaume Maudoux
- [asterisk-dev] [Code Review] 3530: Fix h264 SDP payload format
Guillaume Maudoux
- [asterisk-dev] [Code Review] 3531: app_chanspy: Fix broken barge test and address an unfree'd frame I noticed.
Jonathan Rose
- [asterisk-dev] [Code Review] 3534: Dahdi Dialtone detection regression
one47
- [asterisk-dev] Enquiry around a Segfault
Steve Davies
- [asterisk-dev] [Code Review] 3535: bridge_native_rtp: Reconfigure bridge on removal of framehook; don't send re-INVITE to hungup channel
Matt Jordan
- [asterisk-dev] [Code Review] 3536: res_musiconhold cleanup (part 1)
wdoekes
- [asterisk-dev] [Code Review] 3537: chan_sip+CEL: Add missing ANSWER and PICKUP events to INVITE/w/replaces pickup (for asterisk 12)
wdoekes
- [asterisk-dev] [Code Review] 3538: Partial fix to voicemail number maxmsg being overwritten.
Miguel Tavares
- [asterisk-dev] [Code Review] 3539: pbx.c: segfault on recursive replace
Scott Griepentrog
- [asterisk-dev] Digium TE820 on BSD: bunch of missed interrupts
Łukasz Wójcik
- [asterisk-dev] [Code Review] 3540: chan_local+app_dial: Propagagate call answered elsewhere over local channels.
wdoekes
- [asterisk-dev] Module pbx_lua not loading extensions.lua on startup
Dennis Guse
- [asterisk-dev] [Code Review] 3488: RAII_VAR: nested functions aren't portable. Adapting RAII_VAR to use clang/llvm blocks to get the same/similar functionality.
Matt Jordan
- [asterisk-dev] Asterisk CALLINGTON for SS7
Alberto Rinaudo
- [asterisk-dev] Segmentation fault error?
Mohammed Essaid Mezerreg
- [asterisk-dev] [Code Review] 3541: res_http_websocket: Create a websocket client
Kevin Harwell
- [asterisk-dev] [Code Review] 3542: Documentation: Wiki page for Maintenance and Upgrades, including sub pages.
rnewton
- [asterisk-dev] [Code Review] 3543: app_meetme: Don't interrupt MOH on waitmarked users.
rmudgett
- [asterisk-dev] [Code Review] 3438: Implement SIP TImer C in Asterisk
Olle E Johansson
- [asterisk-dev] [Code Review] 3439: chan_sip: Support a=rtcp attribute in SDP
Olle E Johansson
- [asterisk-dev] [Code Review] 3437: chan_sip: Add support for a few more 4xx error responses
Olle E Johansson
- [asterisk-dev] [Code Review] 2478: Support multiple Require: and Supported: headers in the same request
Olle E Johansson
- [asterisk-dev] [Code Review] 3546: DTMF emulation bad calculation that hurts RTP
Olle E Johansson
- [asterisk-dev] marcotasto at libero.it
marcotasto
- [asterisk-dev] [Code Review] 3547: Logger/CLI: Fix some aesthetic issues; clean up some chatty verbose messages
Matt Jordan
- [asterisk-dev] [Code Review] 3548: suspended destructions of pri spans following PRI_EVENT_REMOVED
Tzafrir Cohen
- [asterisk-dev] [Code Review] 3549: Replace __ast_answer with ast_raw_answer in app_control_answer
Paul Belanger
- [asterisk-dev] [Code Review] 3550: build: Allow autoconf/ast_ext_tool_check to handle cross-compiling better
George Joseph
- [asterisk-dev] [Code Review] 3551: app_agent_pool: Return to dialplan if the agent fails to ack the call.
rmudgett
- [asterisk-dev] [Code Review] 3331: Allows app_chanspy to whisper to a spyee's bridged peer (barge) even if the bridged party answers after initial spy invocation.
Matt Jordan
- [asterisk-dev] Unify (Siemens) OpenStage uaCSTA Functionality
Jonas Köritz
- [asterisk-dev] URI parsing
Kevin Harwell
- [asterisk-dev] Asterisk Leaks FileDescriptor in handle_recordfile - if Call Disconnect happens while playing beep
bala murugan
- [asterisk-dev] app_confbridge + USER_OPT_TALKER_DETECT
Jared Mauch
- [asterisk-dev] Asterisk 13 Feature Freeze Reminder
Matthew Jordan
- [asterisk-dev] [Code Review] 3554: repotools: Get rid of duplicate JIRA-1234 #comments
wdoekes
- [asterisk-dev] [Code Review] 3555: res_config_odbc: Fix old and new ast_string_field memory leaks.
wdoekes
- [asterisk-dev] [Code Review] 3557: odbc: Remove fixed size buffers.
Joshua Colp
- [asterisk-dev] [Code Review] 3559: sqlite3: Add ability to automatically retry query to busy database
Igor Goncharovsky
- [asterisk-dev] Asterisk 1.8.28.0-rc1 Now Available
Asterisk Development Team
- [asterisk-dev] Asterisk 11.10.0-rc1 Now Available
Asterisk Development Team
- [asterisk-dev] Asterisk 12.3.0-rc1 Now Available
Asterisk Development Team
- [asterisk-dev] [Code Review] 3560: Testsuite: Add PJSIP nominal attended transfer tests
opticron
- [asterisk-dev] [Code Review] 3561: testsuite: Tweak agent pool tests.
rmudgett
- [asterisk-dev] DTMF mute in RTP
Olle E. Johansson
- [asterisk-dev] [Code Review] 2150: Post trunk-resurrection fixes to libss7.
KNK
- [asterisk-dev] [Code Review] 3562: chan_sip: Start session timer at 200, not at INVITE.
wdoekes
- [asterisk-dev] [Code Review] 3397: testsuite: directory fixes to prevent untracked files from being created in the svn directory
wdoekes
- [asterisk-dev] [Code Review] 3563: TALK_DETECT: A channel function that raises AMI/ARI events when talking is detected
Matt Jordan
- [asterisk-dev] [Code Review] 3564: TALK_DETECT: Tests for Asterisk Test Suite
Matt Jordan
- [asterisk-dev] Sip call consciously without audio
Gunnar Hellstrom
- [asterisk-dev] [Code Review] 3567: [channels/chan_unistim.c]: Possible unlocked mutex
Peter Whisker
- [asterisk-dev] [Code Review] 3570: Testcase for r3562 (chan_sip: Start session timer at 200, not at INVITE)
wdoekes
- [asterisk-dev] [Code Review] 3571: res_pjsip_session: Fix leaking video RTP ports.
rmudgett
- [asterisk-dev] JIRA, Commit Messages, and Smart Commits (oh my)
Matthew Jordan
- [asterisk-dev] [Code Review] 3572: cli_aliases: Add 'pjsip reload' alias because it's nice. Plus, others for 11 syntax compatability.
rnewton
- [asterisk-dev] Asterisk 12.3.0-rc2 Now Available
Asterisk Development Team
- [asterisk-dev] [Code Review] 3573: [main/config.c] AMI action UpdateConfig EmptyCat clears all categories but the requested one
zvision
- [asterisk-dev] [Code Review] 3574: safe_asterisk: Cleanup and debian compatibility.
wdoekes
- [asterisk-dev] Asterisk 1.8.28.0 Now Available
Asterisk Development Team
- [asterisk-dev] Asterisk 11.10.0 Now Available
Asterisk Development Team
- [asterisk-dev] Asterisk 12.3.0 Now Available
Asterisk Development Team
- [asterisk-dev] [Code Review] 3575: config: Fix config files not reloading when only an included file changes.
rmudgett
- [asterisk-dev] [Code Review] 3576: astobj2: Split hash and rbtree impls into their own source files.
George Joseph
- [asterisk-dev] [Code Review] 3577: bridge_native_rtp: Use combined result of both channels to determine bridge type.
Joshua Colp
Last message date:
Sat May 31 04:25:47 CDT 2014
Archived on: Sat May 31 04:25:29 CDT 2014
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