April 2014 Archives by thread
      
      Starting: Tue Apr  1 06:24:30 CDT 2014
         Ending: Wed Apr 30 18:21:27 CDT 2014
         Messages: 654
     
- [asterisk-dev] [Code Review] 3320: media_formats: Move format	modules over.
 
Joshua Colp
- [asterisk-dev] [Code Review] 3362: func_periodic_hook: New	function for periodic hooks.
 
Russell Bryant
- [asterisk-dev] [Code Review] 3404: app_queue removing callers	from queue when reloading
 
Italo Rossi
- [asterisk-dev] [Code Review] 3314: Testsuite: BridgeWait's S	Option Test
 
Scott Emidy
- [asterisk-dev] [Code Review] 3392: Test for PJSIP_HEADER
 
Benjamin Keith Ford
- [asterisk-dev] [Code Review] 3329: AGI Exit Status Test
 
Benjamin Keith Ford
- [asterisk-dev] [Code Review] 3400: Nominal DISA Authentication	Test
 
Benjamin Keith Ford
- [asterisk-dev] [Code Review] 3348: Test Suite: MWI subscription	test for PJSIP
 
Joshua Colp
- [asterisk-dev] [Code Review] 3399: PJSIP DTMF INFO Test
 
opticron
- [asterisk-dev] [Code Review] 3313: Testsuite: Convert YAML Configuration Into Call File and Execute
 
opticron
- [asterisk-dev] [Code Review] 3403: Test for channel Pickup
 
opticron
- [asterisk-dev] [Code Review] 3408: app_voicemail: fix regression caused by dialplan function safety fixes - Asterisk 11 only
 
Corey Farrell
- [asterisk-dev] [Code Review] 3410: media_formats: Move chan_multicast_rtp, chan_console, app_jack, and chan_ooh323 over.
 
Kevin Harwell
- [asterisk-dev] [Code Review] 3411: Add some asserts that were handy when looking for a stasis cache problem.
 
rmudgett
- [asterisk-dev] [Code Review] 3412: testsuite: Add call setup tracking to masquerade supertest.
 
rmudgett
- [asterisk-dev] [Code Review] 3409: app_queue: Fix for queue members receiving calls when in call and with ringinuse=no
 
Shlomi Gutman
- [asterisk-dev] Asterisk Test Suite Git Migration
 
Kevin Harwell
- [asterisk-dev] [Code Review] 3383: TestSuite: Fix bouncing	show_subscriptions test
 
Kevin Harwell
- [asterisk-dev] [Code Review] 3413: media_formats: move app.c and	file.c over
 
Kevin Harwell
- [asterisk-dev] [Code Review] 3398: TestSuite: Fix tests	affected by 411086
 
Matt Jordan
- [asterisk-dev] [Code Review] 3414: internal_timing: Remove the option and always make it enabled if a timing module is loaded.
 
rmudgett
- [asterisk-dev] [Code Review] 3407: Test Suite: Nominal caller initiated blind transfer tests using PJSIP
 
opticron
- [asterisk-dev] [Code Review] 3415: bridging: Ensure proper locking	during snapshot creation
 
opticron
- [asterisk-dev] [Code Review] 3339: Testsuite: ARI test for playback of audio to a channel in a bridge.
 
Mark Michelson
- [asterisk-dev] [Code Review] 3363: Testsuite: Pluggable module	for testing realtime
 
Mark Michelson
- [asterisk-dev] UDP/TLS/RTP/SAVPF to RTP/SAVPF
 
jaflong jaflong
- [asterisk-dev] [Code Review] 3417: Add AMI events for all device state and presence state changes
 
Mark Michelson
- [asterisk-dev] [Code Review] 3418: Test the DeviceStateChange and PresenceStateChange AMI events
 
Mark Michelson
- [asterisk-dev] [Code Review] 3419: Test for PJSIP fast picture	update
 
Benjamin Keith Ford
- [asterisk-dev] [Code Review] 3420: Testsuite: Call Files' Max	Retries
 
Scott Emidy
- [asterisk-dev] [Code Review] 3421: Originated calls: Fix several	originate call problems.
 
rmudgett
- [asterisk-dev] [svn-commits] rmudgett: branch 1.8 r411715 - in	/branches/1.8: ./ channels/ configs/ includ...
 
Olle E. Johansson
- [asterisk-dev] [Code Review] 3422: testsuite: Fix Asterisk shutdown timeout in chan_sip session_timer tests by hanging up channels
 
Corey Farrell
- [asterisk-dev] [Code Review] 3377: ref count logs: Redo structure of log file, provide a python debugging tool
 
Matt Jordan
- [asterisk-dev] Asterisk 1.8 and SRV records
 
Mikael Fredin
- [asterisk-dev] [Code Review] 3406: AGI/Manager: Prevent multiple Newexten events from occuring from AGI application changes
 
Jonathan Rose
- [asterisk-dev] [Code Review] 3423: Internal timing: Add notice message about the -I and internal_timing option no longer needed.
 
rmudgett
- [asterisk-dev] [Code Review] 3424: mixmonitor: Add option to enable	a periodic beep
 
Russell Bryant
- [asterisk-dev] Audiohook dropping frames
 
Olle E. Johansson
- [asterisk-dev] [Code Review] 3426: Fix build failure on SmartOS /	Illumos / SunOS
 
Sebastian Wiedenroth
- [asterisk-dev] Dundi library
 
Klaus Darilion
- [asterisk-dev] [Code Review] 3427: ARI: Add tones playback resource
 
Jonathan Rose
- [asterisk-dev] [Code Review] 3428: Testsuite: ARI Playback Tones tests for channels and bridges
 
Jonathan Rose
- [asterisk-dev] [Code Review] 3300: Don't crash on lack of	bridged rtp instance
 
Russell Bryant
- [asterisk-dev] [Code Review] 2684: Fix exposure of	template-only config sections
 
Russell Bryant
- [asterisk-dev] [Code Review] 3429: monitor: use app options parsing	helper code
 
Russell Bryant
- [asterisk-dev] PJSIP in dialog OPTIONS method handling
 
Yaron Nachum
- [asterisk-dev] automated response
 
Joseph Shi
- [asterisk-dev] [Code Review] 3379: ARI: Make bridges/{bridgeId}/play queue sound files if sounds are already playing on the bridge instead of playing them simultaneously as they are called
 
Jonathan Rose
- [asterisk-dev] [Code Review] 3430: Improvements and bugfixes to	chan_unistim.c
 
Peter Whisker
- [asterisk-dev] [Code Review] 3431: Fix channel staging assertion	failure.
 
rmudgett
- [asterisk-dev] [Code Review] 3433: bridge_unreal: An alternative implementation for optimizing Unreal/Local channels.
 
Joshua Colp
- [asterisk-dev] TLS architecture chan_sip
 
Olle E. Johansson
- [asterisk-dev] [Code Review] 3349: Implement RFC-3966 TEL URI	incoming INVITE
 
wdoekes
- [asterisk-dev] [Code Review] 3357: testsuite: Add off-nominal subscription tests for PJSIP.
 
opticron
- [asterisk-dev] [Code Review] 3434: libpri: Make TE-PTP mode respond to MDL TEI check requests.
 
rmudgett
- [asterisk-dev] tcptls.c
 
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] 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] 2478: Support multiple Require: and Supported: headers in the same request
 
Olle E Johansson
- [asterisk-dev] [Code Review] 2872: Pre-review of work to handle SRTP lifetime and MKI in a less bad way, but not the best way
 
Olle E Johansson
- [asterisk-dev] [Code Review] 2227: Manage translation table between SIP and ISDN hangup causes
 
Olle E Johansson
- [asterisk-dev] [Code Review] 488: Add AMI actions for changing custom device state and generate manager events when device states changes
 
Olle E Johansson
- [asterisk-dev] [Code Review] 1164: [patch] Improve debug of	ast_hangup
 
Olle E Johansson
- [asterisk-dev] [Code Review] 1421: AMI :: Debug manager actions	in the CLI
 
Olle E Johansson
- [asterisk-dev] [Code Review] 3440: ARI: Get rid of \brief on ARI resource mustache struct documentation comments
 
Jonathan Rose
- [asterisk-dev] [Code Review] 3442: chan_sip: Add TELPHONECONTEXT channel variable for Request TEL URIs
 
Matt Jordan
- [asterisk-dev] [Code Review] 3441: testsuite: Add a test for TEL URI
 
Matt Jordan
- [asterisk-dev] [Code Review] 3443: Japanese language patch for app_voicemail.c and say.c, compatible with newly submitted Japanese sound files
 
Kevin McCoy
- [asterisk-dev] [Code Review] 1392: Fix app_sms regression
 
Matt Jordan
- [asterisk-dev] cross compile asterisk for openrisc 1200
 
Mohammed Essaid Mezerreg
- [asterisk-dev] [Code Review] 3405: Add ast_spinlock capability
 
George Joseph
- [asterisk-dev] Asterisk 11.9.0-rc2 Now Available
 
Asterisk Development Team
- [asterisk-dev] Asterisk 12.2.0-rc2 Now Available
 
Asterisk Development Team
- [asterisk-dev] [Code Review] 3444: Testsuite: PJSIP Callee	Initiated Nominal Blind Transfers
 
Scott Emidy
- [asterisk-dev] [Code Review] 3445: res_pjsip_refer: Channel variable SIPREFERTOHDR not being set during blind transfer
 
Kevin Harwell
- [asterisk-dev] [Code Review] 3446: Parking: Add 'AnnounceChannel' to Park manager action. Change some announcement behavior for Park manager action.
 
Jonathan Rose
- [asterisk-dev] [Code Review] 3447: Send real CallerID information with P-Asserted-Identity (RFC-3325)
 
Jonathan Rose
- [asterisk-dev] [Code Review] 3448: res_pjsip: Handle reloading when permanent contacts exist and qualify is configured
 
Joshua Colp
- [asterisk-dev] [Code Review] 3449: Testsuite: PJSIPQualify AMI	Action Test
 
Scott Emidy
- [asterisk-dev] [Code Review] 3450: Stasis: Ensure control's bridge	pointer has a ref
 
opticron
- [asterisk-dev] [Code Review] 3451: app_dial and app_queue: Make lock the forwarding channel while taking the channel snapshot.
 
rmudgett
- [asterisk-dev] [Code Review] 3337: Code for DTLS retransmission
 
Nitesh Bansal
- [asterisk-dev] [Code Review] 3454: When Asterisk initiates an ICE-based session, then it must send it's STUN check messages using role "ICE-CONTROLLING". Currently it uses "ICE-CONTROLLED". Though the role conflict get's resolved correctly, it does not conform to http://tools.ietf.org/html/rfc5245#section-5.2
 
Marko Seidenglanz
- [asterisk-dev] [Code Review] 3455: testsuite: Add chan_sip tests for sendrpid=pai/rpid tests (baseline and with trust_id_outbound)
 
Jonathan Rose
- [asterisk-dev] [Code Review] 3457: DISA Test - Invalid Extension
 
Benjamin Keith Ford
- [asterisk-dev] [Code Review] 3459: DISA Test - No Context
 
Benjamin Keith Ford
- [asterisk-dev] [Code Review] 3458: Testsuite: Off-Nominal Disa Bad	Authentication Test
 
Scott Emidy
- [asterisk-dev] [Code Review] 3460: DISA Test - No Authentication
 
Benjamin Keith Ford
- [asterisk-dev] [Code Review] 3462: PJSIP: Allow for multiple contacts to be set in a single contact= line in an aor section of pjsip.conf
 
Mark Michelson
- [asterisk-dev] [Code Review] 3463: RFC: astobj2 cached objects (alternative to weak containers)
 
Corey Farrell
- [asterisk-dev] [Code Review] 3464: Sounds: Various new sound sets were missing from the makefile and menuselect options
 
rnewton
- [asterisk-dev] [Code Review] 3466: HTTP: Add TCP_NODELAY to	accepted connections
 
opticron
- [asterisk-dev] Asterisk-11.5.1 Confbridge Dailout using pbx_exce new user not placed into current conference
 
hkc323
- [asterisk-dev] Menufile did not played when user press "*" using	Asterisk11.5.1 Confbridge
 
hkc323
- [asterisk-dev] [Code Review] 3471: Filesystem based dynamic MoH	classes
 
Vitezslav Novy
- [asterisk-dev] [Code Review] 1803: P-Asserted-Identity Privacy - fixed behaviour - trust peer
 
Matt Jordan
- [asterisk-dev] Asterisk 1.8.27.0-rc2 Now Available
 
Asterisk Development Team
- [asterisk-dev] Asterisk 11.9.0-rc3 Now Available
 
Asterisk Development Team
- [asterisk-dev] Asterisk 12.2.0-rc3 Now Available
 
Asterisk Development Team
- [asterisk-dev] [Code Review] 3472: res_pjsip_sdp_rtp: Fix issue with unholding when it shouldn't.
 
Joshua Colp
- [asterisk-dev] [Code Review] 3473: res_pjsip_sdp_rtp: Add tests for receiving same SDP when call is already held.
 
Joshua Colp
- [asterisk-dev] [Code Review] 1471: Implement Externaddr on a	sip device basis
 
wdoekes
- [asterisk-dev] Asterisk 1.8.27.0 Now Available
 
Asterisk Development Team
- [asterisk-dev] Asterisk 11.9.0 Now Available
 
Asterisk Development Team
- [asterisk-dev] Asterisk 12.2.0 Now Available
 
Asterisk Development Team
- [asterisk-dev] [Code Review] 3474: TLS and SRTP status not	available in CLI
 
Patrick Laimbock
- [asterisk-dev] [Code Review] 3475: pjsip realtime: increase the	size of some columns
 
Kevin Harwell
- [asterisk-dev] [Code Review] 3476: Memory Pre- and Post-Test	Condition
 
Benjamin Keith Ford
- [asterisk-dev] [Code Review] 3477: Japanese language patch for	app_voicemail.c and say.c
 
Kevin McCoy
- [asterisk-dev] [Code Review] 3479: chan_pjsip: Call pickup test.
 
Joshua Colp
- [asterisk-dev] [Code Review] 3478: chan_pjsip: Add call pickup	support.
 
Joshua Colp
- [asterisk-dev] Add new option to Queue function
 
Nguyen Hoang Son
- [asterisk-dev] [Code Review] 3480: chan_pjsip: Implement get_pvt_uniqueid channel technology callback.
 
Joshua Colp
- [asterisk-dev] [Code Review] 3481: Websocket: Add locking around session access and modification
 
opticron
- [asterisk-dev] [Code Review] 3482: func_presencestate: Make base64 encoded-ness consistent for consumers of presence state
 
Mark Michelson
- [asterisk-dev] [Code Review] 3485: pjsip: Fix a bug where transferring to a parking extension causes calls to hang
 
Jonathan Rose
- [asterisk-dev] asterisk-dev Digest, Vol 117, Issue 173
 
Nguyen Hoang Son
- [asterisk-dev] problem in cross compiling asterisk
 
Mohammed Essaid Mezerreg
- [asterisk-dev] [Code Review] 3486: res_corosync: Fix module to work	with Stasis
 
Matt Jordan
- [asterisk-dev] SIP Presence using SIP SIMPLE: How?
 
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.
 
Diederik de Groot
- [asterisk-dev] [Code Review] 3489: testsuite: Improve logging
 
Matt Jordan
- [asterisk-dev] [Code Review] 3490: Testsuite: Ensure that repeated device states and presence states behave as expected
 
Mark Michelson
- [asterisk-dev] A thread for format work
 
Matthew Jordan
- [asterisk-dev] [Code Review] 3491: res_pjsip: Allow cipher to be	specified by name
 
Joshua Colp
- [asterisk-dev] [Code Review] 3494: ARI: Add the ability to raise an arbitrary User Event from the Applications resource
 
Scott Griepentrog
- [asterisk-dev] [Code Review] 3495: res_stasis: Add missing PROGRESS	indications to functions
 
Joshua Colp
- [asterisk-dev] [Code Review] 3496: testsuite: add valgrind support
 
Scott Griepentrog
- [asterisk-dev] [Code Review] 3027: Valgrind support in TestSuite
 
Scott Griepentrog
- [asterisk-dev] [Code Review] 3501: testsuite: add tests for ari	userevents
 
Scott Griepentrog
- [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
 
Jonathan Rose
- [asterisk-dev] [Code Review] 3506: format improvements: Port bridge_native_rtp over to new format capability API
 
Matt Jordan
- [asterisk-dev] [Code Review] 3508: Prevent a queue member's state from getting stuck when using dynamic hints as 'state_interface'
 
sysreq
- [asterisk-dev] [Code Review] 3512: media formats: Convert the	translation core over
 
Kevin Harwell
- [asterisk-dev] [Code Review] 3513: Weak Reference Containers
 
George Joseph
- [asterisk-dev] [Code Review] 3515: media_formats: Move chan_pjsip	over.
 
Joshua Colp
- [asterisk-dev] [Code Review] 3516: media_formats: Move chan_sip	over.
 
Joshua Colp
- [asterisk-dev] [Code Review] 3517: media_formats: Move addons over.
 
Joshua Colp
- [asterisk-dev] [Code Review] 3518: media_formats: Move abstract jitterbuffer, audiohooks, smoother, and some core stuff over.
 
Joshua Colp
- [asterisk-dev] [Code Review] 3519: media_formats: Add legacy format API and move chan_iax2, chan_h323, and chan_misdn over.
 
Joshua Colp
- [asterisk-dev] [Code Review] 3514: res_pjsip_refer: Add Referred-By header on INVITE for blind transfers.
 
rmudgett
    
      Last message date: 
       Wed Apr 30 18:21:27 CDT 2014
    Archived on: Wed Apr 30 18:09:03 CDT 2014
    
   
     
     
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