December 2014 Archives by author

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Starting: Mon Dec 1 05:37:16 CST 2014
Ending: Wed Dec 31 17:18:01 CST 2014
Messages: 552

• [asterisk-dev] AVPF support in asterisk   Nitesh Bansal
• [asterisk-dev] Adding the support for NACK in asterisk   Nitesh Bansal
• [asterisk-dev] Adding the support for NACK in asterisk   Nitesh Bansal
• [asterisk-dev] Adding the support for NACK in asterisk   Nitesh Bansal
• [asterisk-dev] Adding the support for NACK in asterisk   Nitesh Bansal
• [asterisk-dev] Slow Sip Processing   Ross Beer
• [asterisk-dev] ARI Extending Existing Feature: bridge control   Paul Belanger
• [asterisk-dev] ARI Extending Existing Feature: bridge control   Paul Belanger
• [asterisk-dev] ARI Extending Existing Feature: bridge control   Paul Belanger
• [asterisk-dev] ARI Extending Existing Feature: bridge control   Paul Belanger
• [asterisk-dev] git migration update   Paul Belanger
• [asterisk-dev] git migration update   Paul Belanger
• [asterisk-dev] git migration update   Russell Bryant
• [asterisk-dev] git migration update   Russell Bryant
• [asterisk-dev] [Code Review] 4093: Codec Format Is Not Included in the SDP Media Attributes When SLIN48 Codec Is Used   Frankie Chin
• [asterisk-dev] pjsip vs ca path   James Cloos
• [asterisk-dev] pjsip vs ca path   James Cloos
• [asterisk-dev] reviewboard gives 500 error   James Cloos
• [asterisk-dev] reviewboard gives 500 error   James Cloos
• [asterisk-dev] git migration update   Tzafrir Cohen
• [asterisk-dev] CDR: log? lib?   Tzafrir Cohen
• [asterisk-dev] [Code Review] 4212: res_pjsip_endpoint_identified_ip: Add 'show identify' and 'show identifies' cli commands   Joshua Colp
• [asterisk-dev] [Code Review] 4193: Stasis: allow for subscriptions to dictate message delivery on a threadpool for certain situations   Joshua Colp
• [asterisk-dev] [Code Review] 4215: sorcery: Add additional observer capabilities.   Joshua Colp
• [asterisk-dev] [Code Review] 4216: res_pjsip: Direct Media calls within private network sometimes get one way audio   Joshua Colp
• [asterisk-dev] [Code Review] 4211: Speed up loopback switches by avoiding unneeded lookups   Joshua Colp
• [asterisk-dev] [Code Review] 4218: res_pjsip_refer: Remove framehook when transfer is marked as completed as a result of joining a bridge   Joshua Colp
• [asterisk-dev] [Code Review] 4217: config: Create ast_variable_find_in_list().   Joshua Colp
• [asterisk-dev] [Code Review] 4218: res_pjsip_refer: Remove framehook when transfer is marked as completed as a result of joining a bridge   Joshua Colp
• [asterisk-dev] [Code Review] 4215: sorcery: Add additional observer capabilities.   Joshua Colp
• [asterisk-dev] [Code Review] 4216: res_pjsip: Direct Media calls within private network sometimes get one way audio   Joshua Colp
• [asterisk-dev] [Code Review] 4178: res_pjsip_outbound_publish: stack overflow when using non-default sorcery wizard   Joshua Colp
• [asterisk-dev] [Code Review] 4221: test framework: Fix race condition between AMI topic and Test Suite topic raising of AMI events   Joshua Colp
• [asterisk-dev] [Code Review] 4178: res_pjsip_outbound_publish: stack overflow when using non-default sorcery wizard   Joshua Colp
• [asterisk-dev] [Code Review] 4178: res_pjsip_outbound_publish: stack overflow when using non-default sorcery wizard   Joshua Colp
• [asterisk-dev] [Code Review] 4215: sorcery: Add additional observer capabilities.   Joshua Colp
• [asterisk-dev] [Code Review] 4243: ari: Add the ability to specify a linkedId when originating calls.   Joshua Colp
• [asterisk-dev] [Code Review] 4244: ari: Add originate with linkedid test.   Joshua Colp
• [asterisk-dev] [Code Review] 4244: ari: Add originate with linkedid test.   Joshua Colp
• [asterisk-dev] [Code Review] 4243: ari: Add the ability to specify a linkedId when originating calls.   Joshua Colp
• [asterisk-dev] [Code Review] 4243: ari: Add the ability to specify a linkedId when originating calls.   Joshua Colp
• [asterisk-dev] [Code Review] 4243: ari: Add the ability to specify a linkedId when originating calls.   Joshua Colp
• [asterisk-dev] [Code Review] 4243: ari: Add the ability to specify an originator when originating calls.   Joshua Colp
• [asterisk-dev] [Code Review] 4243: ari: Add the ability to specify an originator when originating calls.   Joshua Colp
• [asterisk-dev] [Code Review] 4244: ari: Add originate with originator test.   Joshua Colp
• [asterisk-dev] [Code Review] 4190: res_pjsip_config_wizard: Allow streamlined configuration of common pjsip scenarios   Joshua Colp
• [asterisk-dev] [Code Review] 4245: ARI/AMI: Include language in standard channel snapshot output   Joshua Colp
• [asterisk-dev] [Code Review] 4170: testsuite: Delete bridges on completion for a bunch of rest_api tests   Joshua Colp
• [asterisk-dev] [Code Review] 4178: res_pjsip_outbound_publish: stack overflow when using non-default sorcery wizard   Joshua Colp
• [asterisk-dev] [Code Review] 4190: res_pjsip_config_wizard: Allow streamlined configuration of common pjsip scenarios   Joshua Colp
• [asterisk-dev] [Code Review] 4190: res_pjsip_config_wizard: Allow streamlined configuration of common pjsip scenarios   Joshua Colp
• [asterisk-dev] [Code Review] 4243: ari: Add the ability to specify an originator when originating calls.   Joshua Colp
• [asterisk-dev] [Code Review] 4244: ari: Add originate with originator test.   Joshua Colp
• [asterisk-dev] Rules for binding with regards to RTP   Joshua Colp
• [asterisk-dev] Rules for binding with regards to RTP   Joshua Colp
• [asterisk-dev] [Code Review] 4248: res_pjsip_session: Queue BYE if an outstanding INVITE transaction exists   Joshua Colp
• [asterisk-dev] [Code Review] 4249: PJSIP Blind Transfer Direct Media Tests: Update to expect sane behavior   Joshua Colp
• [asterisk-dev] [Code Review] 4254: res_pjsip_session: Don't allow declined media streams to fail SDP negotiation on re-INVITE   Joshua Colp
• [asterisk-dev] [Code Review] 4255: Testsuite: Add test for SDP offer/answer during hold with a declined stream   Joshua Colp
• [asterisk-dev] [Code Review] 4254: res_pjsip_session: Don't allow declined media streams to fail SDP negotiation on re-INVITE   Joshua Colp
• [asterisk-dev] [Code Review] 4250: Sorcery: Log when a stale configuration remains in use   Joshua Colp
• [asterisk-dev] [Code Review] 4252: PJSIP: Allow use of 'inactive' stream types for hold   Joshua Colp
• [asterisk-dev] [Code Review] 4255: Testsuite: Add test for SDP offer/answer during hold with a declined stream   Joshua Colp
• [asterisk-dev] [Code Review] 4230: add capath support to res_pjsip   Joshua Colp
• [asterisk-dev] [Code Review] 4254: res_pjsip_session: Don't allow declined media streams to fail SDP negotiation on re-INVITE   Joshua Colp
• [asterisk-dev] [Code Review] 4255: Testsuite: Add test for SDP offer/answer during hold with a declined stream   Joshua Colp
• [asterisk-dev] [Code Review] 4248: res_pjsip_session: Queue BYE if an outstanding INVITE transaction exists   Joshua Colp
• [asterisk-dev] [Code Review] 4249: PJSIP Blind Transfer Direct Media Tests: Update to expect sane behavior   Joshua Colp
• [asterisk-dev] [Code Review] 4260: media: Fix crash when determining sample count of a frame during shutdown   Joshua Colp
• [asterisk-dev] [Code Review] 4257: chan_pjsip: Race condition between channel answer and bridge setup when using direct media   Joshua Colp
• [asterisk-dev] [Code Review] 4260: media: Fix crash when determining sample count of a frame during shutdown   Joshua Colp
• [asterisk-dev] Problem developing asterisk module with libcurl   Joshua Colp
• [asterisk-dev] [Code Review] 4190: res_pjsip_config_wizard: Allow streamlined configuration of common pjsip scenarios   Joshua Colp
• [asterisk-dev] AVPF support in asterisk   Joshua Colp
• [asterisk-dev] [Code Review] 4269: res_pjsip_config_wizard: fix test breakage   Joshua Colp
• [asterisk-dev] [Code Review] 4271: ARI: Allow interoperation of ASYNCGOTO and channels originated to Stasis()   Joshua Colp
• [asterisk-dev] [Code Review] 4277: Ensure that CHANNEL(pjsip, secure) gives expected value.   Joshua Colp
• [asterisk-dev] [Code Review] 4268: During a dual redirect, prevent a race condition that may cause one of the redirected channels to be hung up.   Joshua Colp
• [asterisk-dev] [Code Review] 4271: ARI: Allow interoperation of ASYNCGOTO and channels originated to Stasis()   Joshua Colp
• [asterisk-dev] [Code Review] 4288: res_pjsip_phoneprovi_provider: Fix reload   Joshua Colp
• [asterisk-dev] [Code Review] 4288: res_pjsip_phoneprovi_provider: Fix reload   Joshua Colp
• [asterisk-dev] PJSIP_AOR and PJSIP_CONTACT Dialplan Functions   Joshua Colp
• [asterisk-dev] PJSIP_AOR and PJSIP_CONTACT Dialplan Functions   Joshua Colp
• [asterisk-dev] [Code Review] 4286: rtp_engine: avoid payload types above 127   Joshua Colp
• [asterisk-dev] [Code Review] 4285: Bug fixes for ARI Originate/Continue with label support (Continuation of /r/4101)   Joshua Colp
• [asterisk-dev] [Code Review] 4293: testsuite: Add a test for PJSIP keep alive packets for connection oriented transports   Joshua Colp
• [asterisk-dev] [Code Review] 4294: testsuite: Add a test for user_eq_phone setting in PJSIP   Joshua Colp
• [asterisk-dev] PJSIP_AOR and PJSIP_CONTACT Dialplan Functions   Joshua Colp
• [asterisk-dev] [Code Review] 4300: bridge_native_rtp: Change local/remote message from debug/2 to verb/4   Joshua Colp
• [asterisk-dev] [Code Review] 4296: PJSIP: Fix bugs and improve documentation of remote attended transfers   Joshua Colp
• [asterisk-dev] [Code Review] 4283: Testsuite: Dual channel redirect tests   Joshua Colp
• [asterisk-dev] [Code Review] 4297: config: Add option to NOT preserve the effective context when changing a template.   Joshua Colp
• [asterisk-dev] [Code Review] 4298: Testsuite: Tests for Manager/Config/NoPreserveEffectiveContext   Joshua Colp
• [asterisk-dev] [Code Review] 4298: Testsuite: Tests for Manager/Config/NoPreserveEffectiveContext   Joshua Colp
• [asterisk-dev] [Code Review] 4301: res_pjsip_outbound_registration: Add 'pjsip send register' command and update behavior of 'send unregister'   Joshua Colp
• [asterisk-dev] [Code Review] 4286: rtp_engine: avoid payload types above 127   Joshua Colp
• [asterisk-dev] [Code Review] 4286: rtp_engine: avoid payload types above 127   Joshua Colp
• [asterisk-dev] Problem developing asterisk module with libcurl   David
• [asterisk-dev] Volume Control   Gaston Draque
• [asterisk-dev] ARI Stasis Application Examples and Tutorials   Samuel Galarneau
• [asterisk-dev] git migration update   Samuel Galarneau
• [asterisk-dev] Volume Control   Murthy Gandikota
• [asterisk-dev] Volume Control   Murthy Gandikota
• [asterisk-dev] Volume Control   Murthy Gandikota
• [asterisk-dev] Volume Control   Murthy Gandikota
• [asterisk-dev] cdr_adaptive_odbc does not work   Yuriy Gorlichenko
• [asterisk-dev] cdr_adaptive_odbc does not work   Yuriy Gorlichenko
• [asterisk-dev] [Code Review] 4243: ari: Add the ability to specify a linkedId when originating calls.   Scott Griepentrog
• [asterisk-dev] [Code Review] 4243: ari: Add the ability to specify an originator when originating calls.   Scott Griepentrog
• [asterisk-dev] [Code Review] 4244: ari: Add originate with originator test.   Scott Griepentrog
• [asterisk-dev] [Code Review] 4182: core: avoid rasterisk crash due to long identifier   Scott Griepentrog
• [asterisk-dev] [Code Review] 4262: bridge: channel ref leak on blond_nonfinal_enter   Scott Griepentrog
• [asterisk-dev] [Code Review] 4256: testsuite: check for channel leak on failed blonde transfer   Scott Griepentrog
• [asterisk-dev] [Code Review] 4262: bridge: channel ref leak on blond_nonfinal_enter   Scott Griepentrog
• [asterisk-dev] [Code Review] 4262: bridge: channel ref leak on blond_nonfinal_enter   Scott Griepentrog
• [asterisk-dev] [Code Review] 4280: sip_to_pjsip: improved ability to parse input without exception   Scott Griepentrog
• [asterisk-dev] [Code Review] 4101: Channel Originate/Continue via ARI support for labels in dialplan is incomplete   Scott Griepentrog
• [asterisk-dev] [Code Review] 4280: sip_to_pjsip: improved ability to parse input without exception   Scott Griepentrog
• [asterisk-dev] [Code Review] 4280: sip_to_pjsip: improved ability to parse input without exception   Scott Griepentrog
• [asterisk-dev] [Code Review] 4279: chan_sip: Send CANCEL via proxy if CANCEL is to be sent after an UPDATE   Scott Griepentrog
• [asterisk-dev] [Code Review] 4262: bridge: channel ref leak on blond_nonfinal_enter   Scott Griepentrog
• [asterisk-dev] [Code Review] 4286: rtp_engine: avoid payload types above 127   Scott Griepentrog
• [asterisk-dev] [Code Review] 4302: bridge: channel ref leak after failed blond transfer   Scott Griepentrog
• [asterisk-dev] [Code Review] 4286: rtp_engine: avoid payload types above 127   Scott Griepentrog
• [asterisk-dev] [Code Review] 4280: sip_to_pjsip: improved ability to parse input without exception   Scott Griepentrog
• [asterisk-dev] [Code Review] 4280: sip_to_pjsip: improved ability to parse input without exception   Scott Griepentrog
• [asterisk-dev] [Code Review] 4256: testsuite: check for channel leak on failed blonde transfer   Scott Griepentrog
• [asterisk-dev] [Code Review] 4256: testsuite: check for channel leak on failed blonde transfer   Scott Griepentrog
• [asterisk-dev] [Code Review] 4286: rtp_engine: avoid payload types above 127   Scott Griepentrog
• [asterisk-dev] [Code Review] 4286: rtp_engine: avoid payload types above 127   Scott Griepentrog
• [asterisk-dev] [Code Review] 4280: sip_to_pjsip: improved ability to parse input without exception   Scott Griepentrog
• [asterisk-dev] [Code Review] 4286: rtp_engine: avoid payload types above 127   Scott Griepentrog
• [asterisk-dev] [Code Review] 4216: res_pjsip: Direct Media calls within private network sometimes get one way audio   Kevin Harwell
• [asterisk-dev] [Code Review] 4216: res_pjsip: Direct Media calls within private network sometimes get one way audio   Kevin Harwell
• [asterisk-dev] [Code Review] 4215: sorcery: Add additional observer capabilities.   Kevin Harwell
• [asterisk-dev] [Code Review] 4216: res_pjsip: Direct Media calls within private network sometimes get one way audio   Kevin Harwell
• [asterisk-dev] [Code Review] 4178: res_pjsip_outbound_publish: stack overflow when using non-default sorcery wizard   Kevin Harwell
• [asterisk-dev] [Code Review] 4178: res_pjsip_outbound_publish: stack overflow when using non-default sorcery wizard   Kevin Harwell
• [asterisk-dev] [Code Review] 4178: res_pjsip_outbound_publish: stack overflow when using non-default sorcery wizard   Kevin Harwell
• [asterisk-dev] [Code Review] 4216: res_pjsip: Direct Media calls within private network sometimes get one way audio   Kevin Harwell
• [asterisk-dev] [Code Review] 4178: res_pjsip_outbound_publish: stack overflow when using non-default sorcery wizard   Kevin Harwell
• [asterisk-dev] [Code Review] 4178: res_pjsip_outbound_publish: stack overflow when using non-default sorcery wizard   Kevin Harwell
• [asterisk-dev] [Code Review] 4216: res_pjsip: Direct Media calls within private network sometimes get one way audio   Kevin Harwell
• [asterisk-dev] [Code Review] 4216: res_pjsip: Direct Media calls within private network sometimes get one way audio   Kevin Harwell
• [asterisk-dev] [Code Review] 4215: sorcery: Add additional observer capabilities.   Kevin Harwell
• [asterisk-dev] [Code Review] 4215: sorcery: Add additional observer capabilities.   Kevin Harwell
• [asterisk-dev] [Code Review] 4215: sorcery: Add additional observer capabilities.   Kevin Harwell
• [asterisk-dev] [Code Review] 4245: ARI/AMI: Include language in standard channel snapshot output   Kevin Harwell
• [asterisk-dev] [Code Review] 4178: res_pjsip_outbound_publish: stack overflow when using non-default sorcery wizard   Kevin Harwell
• [asterisk-dev] [Code Review] 4216: res_pjsip: Direct Media calls within private network sometimes get one way audio   Kevin Harwell
• [asterisk-dev] [Code Review] 4245: ARI/AMI: Include language in standard channel snapshot output   Kevin Harwell
• [asterisk-dev] [Code Review] 4257: chan_pjsip: Race condition between channel answer and bridge setup when using direct media   Kevin Harwell
• [asterisk-dev] [Code Review] 4257: chan_pjsip: Race condition between channel answer and bridge setup when using direct media   Kevin Harwell
• [asterisk-dev] [Code Review] 4257: chan_pjsip: Race condition between channel answer and bridge setup when using direct media   Kevin Harwell
• [asterisk-dev] [Code Review] 4260: media: Fix crash when determining sample count of a frame during shutdown   Kevin Harwell
• [asterisk-dev] [Code Review] 4261: res_pjsip_pubsub: Activate recreated persistent subscriptions   Kevin Harwell
• [asterisk-dev] [Code Review] 4265: res_pjsip_sdp_rtp: Native RTP bridging is chosen for RTP compatible channels when the DTMF mode is not compatible   Kevin Harwell
• [asterisk-dev] [Code Review] 4266: Testsuite - res_pjsip_sdp_rtp: Incompatible DTMF mode test   Kevin Harwell
• [asterisk-dev] [Code Review] 4265: res_pjsip_sdp_rtp: Native RTP bridging is chosen for RTP compatible channels when the DTMF mode is not compatible   Kevin Harwell
• [asterisk-dev] [Code Review] 4273: res_pjsip_outbound_registration: Prevent infinite authentication loops   Kevin Harwell
• [asterisk-dev] [Code Review] 4274: Testsuite: Ensure that outbound registrations with bad auth do not result in an infinite loop.   Kevin Harwell
• [asterisk-dev] [Code Review] 4270: testsuite: Allow tests to specify multiple minimum versions   Kevin Harwell
• [asterisk-dev] [Code Review] 4281: Testsuite: Single channel redirect tests   Kevin Harwell
• [asterisk-dev] [Code Review] 4265: res_pjsip_sdp_rtp: Native RTP bridging is chosen for RTP compatible channels when the DTMF mode is not compatible   Kevin Harwell
• [asterisk-dev] [Code Review] 4266: Testsuite - res_pjsip_sdp_rtp: Incompatible DTMF mode test   Kevin Harwell
• [asterisk-dev] [Code Review] 4281: Testsuite: Single channel redirect tests   Kevin Harwell
• [asterisk-dev] [Code Review] 4274: Testsuite: Ensure that outbound registrations with bad auth do not result in an infinite loop.   Kevin Harwell
• [asterisk-dev] [Code Review] 4280: sip_to_pjsip: improved ability to parse input without exception   Kevin Harwell
• [asterisk-dev] [Code Review] 4270: testsuite: Allow tests to specify multiple minimum versions   Kevin Harwell
• [asterisk-dev] [Code Review] 4280: sip_to_pjsip: improved ability to parse input without exception   Kevin Harwell
• [asterisk-dev] [Code Review] 4283: Testsuite: Dual channel redirect tests   Kevin Harwell
• [asterisk-dev] [Code Review] 4281: Testsuite: Single channel redirect tests   Kevin Harwell
• [asterisk-dev] [Code Review] 4281: Testsuite: Single channel redirect tests   Kevin Harwell
• [asterisk-dev] [Code Review] 4281: Testsuite: Single channel redirect tests   Kevin Harwell
• [asterisk-dev] [Code Review] 4283: Testsuite: Dual channel redirect tests   Kevin Harwell
• [asterisk-dev] [Code Review] 4296: PJSIP: Fix bugs and improve documentation of remote attended transfers   Kevin Harwell
• [asterisk-dev] [Code Review] 4211: Speed up loopback switches by avoiding unneeded lookups   Birger Harzenetter
• [asterisk-dev] [Code Review] 4211: Speed up loopback switches by avoiding unneeded lookups   Birger Harzenetter
• [asterisk-dev] [Code Review] 4273: res_pjsip_outbound_registration: Prevent infinite authentication loops   Alex Hermann
• [asterisk-dev] [Code Review] 4286: rtp_engine: avoid payload types above 127   Olle E Johansson
• [asterisk-dev] pjsip vs ca path   Olle E. Johansson
• [asterisk-dev] Adding the support for NACK in asterisk   Olle E. Johansson
• [asterisk-dev] git migration update   Olle E. Johansson
• [asterisk-dev] [Code Review] 4193: Stasis: allow for subscriptions to dictate message delivery on a threadpool for certain situations   Matt Jordan
• [asterisk-dev] [Code Review] 4218: res_pjsip_refer: Remove framehook when transfer is marked as completed as a result of joining a bridge   Matt Jordan
• [asterisk-dev] [Code Review] 4213: Stasis: Fix StasisStart and StasisEnd ordering and missing events   Matt Jordan
• [asterisk-dev] [Code Review] 4188: segfault when playing back voicemail under high concurrency with an IMAP backend   Matt Jordan
• [asterisk-dev] [Code Review] 4188: segfault when playing back voicemail under high concurrency with an IMAP backend   Matt Jordan
• [asterisk-dev] [Code Review] 4221: test framework: Fix race condition between AMI topic and Test Suite topic raising of AMI events   Matt Jordan
• [asterisk-dev] [Code Review] 4023: Allow passing options and command to MixMonitor when recording in ConfBridge   Matt Jordan
• [asterisk-dev] [Code Review] 4050: Add ability for Channel Drivers to provide Presence State information   Matt Jordan
• [asterisk-dev] [Code Review] 4226: Testsuite: Add tests for external interactions with ARI/Stasis   Matt Jordan
• [asterisk-dev] [Code Review] 4221: test framework: Fix race condition between AMI topic and Test Suite topic raising of AMI events   Matt Jordan
• [asterisk-dev] [Code Review] 4226: Testsuite: Add tests for external interactions with ARI/Stasis   Matt Jordan
• [asterisk-dev] [Code Review] 4234: Testsuite: Ensure that changes in /r/4233 result in Asterisk no longer crashing.   Matt Jordan
• [asterisk-dev] [Code Review] 4233: res_pjsip_session: Fix crash that would occur when rescheduling a reinvite due to a 491 response.   Matt Jordan
• [asterisk-dev] [Code Review] 4216: res_pjsip: Direct Media calls within private network sometimes get one way audio   Matt Jordan
• [asterisk-dev] [Code Review] 4216: res_pjsip: Direct Media calls within private network sometimes get one way audio   Matt Jordan
• [asterisk-dev] [Code Review] 4233: res_pjsip_session: Fix crash that would occur when rescheduling a reinvite due to a 491 response.   Matt Jordan
• [asterisk-dev] [Code Review] 3683: testsuite: Add a test for CELGenUserEvent   Matt Jordan
• [asterisk-dev] [Code Review] 4213: Stasis: Fix StasisStart and StasisEnd ordering and missing events   Matt Jordan
• [asterisk-dev] [Code Review] 4246: PJSIP: Stagger outbound qualifies   Matt Jordan
• [asterisk-dev] [Code Review] 4247: DEBUG_THREADS: Fix regression and lock tracking initialization problems.   Matt Jordan
• [asterisk-dev] [Code Review] 4178: res_pjsip_outbound_publish: stack overflow when using non-default sorcery wizard   Matt Jordan
• [asterisk-dev] [Code Review] 4259: DTMF atxfer: Setup recall channels as if the original transferrer originated the call.   Matt Jordan
• [asterisk-dev] [Code Review] 4247: DEBUG_THREADS: Fix regression and lock tracking initialization problems.   Matt Jordan
• [asterisk-dev] [Code Review] 4265: res_pjsip_sdp_rtp: Native RTP bridging is chosen for RTP compatible channels when the DTMF mode is not compatible   Matt Jordan
• [asterisk-dev] [Code Review] 4270: testsuite: Allow tests to specify multiple minimum versions   Matt Jordan
• [asterisk-dev] [Code Review] 4271: ARI: Allow interoperation of ASYNCGOTO and channels originated to Stasis()   Matt Jordan
• [asterisk-dev] [Code Review] 4270: testsuite: Allow tests to specify multiple minimum versions   Matt Jordan
• [asterisk-dev] [Code Review] 4270: testsuite: Allow tests to specify multiple minimum versions   Matt Jordan
• [asterisk-dev] [Code Review] 4279: chan_sip: Send CANCEL via proxy if CANCEL is to be sent after an UPDATE   Matt Jordan
• [asterisk-dev] [Code Review] 4264: PJSIP: Update transport method documentation   Matt Jordan
• [asterisk-dev] [Code Review] 4282: queue_log: Post QUEUESTART entry when Asterisk fully boots.   Matt Jordan
• [asterisk-dev] [Code Review] 4259: DTMF atxfer: Setup recall channels as if the original transferrer originated the call.   Matt Jordan
• [asterisk-dev] [Code Review] 4050: Add ability for Channel Drivers to provide Presence State information   Matt Jordan
• [asterisk-dev] [Code Review] 4270: testsuite: Allow tests to specify multiple minimum versions   Matt Jordan
• [asterisk-dev] [Code Review] 4230: add capath support to res_pjsip   Matt Jordan
• [asterisk-dev] [Code Review] 4293: testsuite: Add a test for PJSIP keep alive packets for connection oriented transports   Matt Jordan
• [asterisk-dev] [Code Review] 4293: testsuite: Add a test for PJSIP keep alive packets for connection oriented transports   Matt Jordan
• [asterisk-dev] [Code Review] 4292: app_macro: Don't restore the calling location on a channel redirect.   Matt Jordan
• [asterisk-dev] [Code Review] 4294: testsuite: Add a test for user_eq_phone setting in PJSIP   Matt Jordan
• [asterisk-dev] [Code Review] 4294: testsuite: Add a test for user_eq_phone setting in PJSIP   Matt Jordan
• [asterisk-dev] [Code Review] 4281: Testsuite: Single channel redirect tests   Matt Jordan
• [asterisk-dev] [Code Review] 4280: sip_to_pjsip: improved ability to parse input without exception   Matt Jordan
• [asterisk-dev] [Code Review] 4267: Testsuite: Add blind transfer tests for Stasis application interaction.   Matt Jordan
• [asterisk-dev] [Code Review] 4272: Testsuite: Verify that bridged channels originated into Stasis and AMI Redirect interoperate properly   Matt Jordan
• [asterisk-dev] [Code Review] 4256: testsuite: check for channel leak on failed blonde transfer   Matt Jordan
• [asterisk-dev] [Code Review] 4291: Testsuite: Add external bridging tests for Stasis application interaction - Stasis bridge to non-stasis application.   Matt Jordan
• [asterisk-dev] [Code Review] 4290: pjsip_options: Fix continued qualifies after endpoint/aor deletion.   Matt Jordan
• [asterisk-dev] [Code Review] 4289: pjsip_options: Fix duplicate qualify schedules on reload.   Matt Jordan
• [asterisk-dev] [Code Review] 4295: test_astobj2: Fix warning for missing trailing slash in category.   Matt Jordan
• [asterisk-dev] [Code Review] 4290: pjsip_options: Fix continued qualifies after endpoint/aor deletion.   Matt Jordan
• [asterisk-dev] [Code Review] 4293: testsuite: Add a test for PJSIP keep alive packets for connection oriented transports   Matt Jordan
• [asterisk-dev] [Code Review] 4294: testsuite: Add a test for user_eq_phone setting in PJSIP   Matt Jordan
• [asterisk-dev] [Code Review] 4302: bridge: channel ref leak after failed blond transfer   Matt Jordan
• [asterisk-dev] FW: Asterisk Developers Mailing List   Matthew Jordan
• [asterisk-dev] pjsip vs ca path   Matthew Jordan
• [asterisk-dev] [asterisk-commits] gtjoseph: branch 12 r428725 - in /branches/12/res: ./ res_pjsip/   Matthew Jordan
• [asterisk-dev] Slow Sip Processing   Matthew Jordan
• [asterisk-dev] reviewboard gives 500 error   Matthew Jordan
• [asterisk-dev] reviewboard gives 500 error   Matthew Jordan
• [asterisk-dev] reviewboard gives 500 error   Matthew Jordan
• [asterisk-dev] Asterisk 12 - Security Fix Only Notice   Matthew Jordan
• [asterisk-dev] Asterisk 12 - Security Fix Only! (aka: update repotools)   Matthew Jordan
• [asterisk-dev] confbridge feature request   Matthew Jordan
• [asterisk-dev] ARI Extending Existing Feature: bridge control   Matthew Jordan
• [asterisk-dev] Adding the support for NACK in asterisk   Matthew Jordan
• [asterisk-dev] Asterisk created mmlog   Matthew Jordan
• [asterisk-dev] ARI Extending Existing Feature: bridge control   Matthew Jordan
• [asterisk-dev] PJSIP_AOR and PJSIP_CONTACT Dialplan Functions   Matthew Jordan
• [asterisk-dev] [asterisk-commits] rmudgett: trunk r430028 - /trunk/channels/chan_vpb.cc   Matthew Jordan
• [asterisk-dev] Volume Control   Matthew Jordan
• [asterisk-dev] Volume Control   Matthew Jordan
• [asterisk-dev] Volume Control   Matthew Jordan
• [asterisk-dev] asterisknow-version   Matthew Jordan
• [asterisk-dev] git migration update   Matthew Jordan
• [asterisk-dev] git migration update   Matthew Jordan
• [asterisk-dev] Adding the support for NACK in asterisk   Matthew Jordan
• [asterisk-dev] PJSIP_AOR and PJSIP_CONTACT Dialplan Functions   Matthew Jordan
• [asterisk-dev] PJSIP_AOR and PJSIP_CONTACT Dialplan Functions   Matthew Jordan
• [asterisk-dev] PJSIP_AOR and PJSIP_CONTACT Dialplan Functions   Matthew Jordan
• [asterisk-dev] PJSIP_AOR and PJSIP_CONTACT Dialplan Functions   Matthew Jordan
• [asterisk-dev] [Code Review] 4217: config: Create ast_variable_find_in_list().   George Joseph
• [asterisk-dev] [Code Review] 4215: sorcery: Add additional observer capabilities.   George Joseph
• [asterisk-dev] [Code Review] 4212: res_pjsip_endpoint_identified_ip: Add 'show identify' and 'show identifies' cli commands   George Joseph
• [asterisk-dev] [Code Review] 4217: config: Create ast_variable_find_in_list().   George Joseph
• [asterisk-dev] [Code Review] 4190: res_pjsip_config_wizard: Allow streamlined configuration of common pjsip scenarios   George Joseph
• [asterisk-dev] [asterisk-commits] gtjoseph: branch 12 r428725 - in /branches/12/res: ./ res_pjsip/   George Joseph
• [asterisk-dev] [Code Review] 4190: res_pjsip_config_wizard: Allow streamlined configuration of common pjsip scenarios   George Joseph
• [asterisk-dev] [Code Review] 4215: sorcery: Add additional observer capabilities.   George Joseph
• [asterisk-dev] [Code Review] 4215: sorcery: Add additional observer capabilities.   George Joseph
• [asterisk-dev] [Code Review] 4215: sorcery: Add additional observer capabilities.   George Joseph
• [asterisk-dev] [Code Review] 4197: testsuite: Update cleanup-test-remnants to clean /var/tmp/asterisk-testsuite and ./logs/*   George Joseph
• [asterisk-dev] [Code Review] 4190: res_pjsip_config_wizard: Allow streamlined configuration of common pjsip scenarios   George Joseph
• [asterisk-dev] [Code Review] 4215: sorcery: Add additional observer capabilities.   George Joseph
• [asterisk-dev] [Code Review] 4215: sorcery: Add additional observer capabilities.   George Joseph
• [asterisk-dev] [Code Review] 4215: sorcery: Add additional observer capabilities.   George Joseph
• [asterisk-dev] [Code Review] 4190: res_pjsip_config_wizard: Allow streamlined configuration of common pjsip scenarios   George Joseph
• [asterisk-dev] [Code Review] 4190: res_pjsip_config_wizard: Allow streamlined configuration of common pjsip scenarios   George Joseph
• [asterisk-dev] [Code Review] 4190: res_pjsip_config_wizard: Allow streamlined configuration of common pjsip scenarios   George Joseph
• [asterisk-dev] [Code Review] 4190: res_pjsip_config_wizard: Allow streamlined configuration of common pjsip scenarios   George Joseph
• [asterisk-dev] [Code Review] 4251: loader: Move definition of ast_module_reload from _private.h to module.h   George Joseph
• [asterisk-dev] [Code Review] 4190: res_pjsip_config_wizard: Allow streamlined configuration of common pjsip scenarios   George Joseph
• [asterisk-dev] [Code Review] 4251: loader: Move definition of ast_module_reload from _private.h to module.h   George Joseph
• [asterisk-dev] [Code Review] 4190: res_pjsip_config_wizard: Allow streamlined configuration of common pjsip scenarios   George Joseph
• [asterisk-dev] [Code Review] 4172: Testsuite: tests for res_pjsip_config_wizard   George Joseph
• [asterisk-dev] [Code Review] 4269: res_pjsip_config_wizard: fix test breakage   George Joseph
• [asterisk-dev] [Code Review] 4269: res_pjsip_config_wizard: fix test breakage   George Joseph
• [asterisk-dev] [Code Review] 4275: config: make config_cache_remove public as ast_config_cache_remove   George Joseph
• [asterisk-dev] [Code Review] 4276: res_pjsip_config_wizard: Update to earlier patch for fix test breakage.   George Joseph
• [asterisk-dev] [Code Review] 4276: res_pjsip_config_wizard: Update to earlier patch for fix test breakage.   George Joseph
• [asterisk-dev] [Code Review] 4276: res_pjsip_config_wizard: Update to earlier patch for fix test breakage.   George Joseph
• [asterisk-dev] [Code Review] 4276: res_pjsip_config_wizard: Update to earlier patch for fix test breakage.   George Joseph
• [asterisk-dev] [Code Review] 4276: res_pjsip_config_wizard: Update to earlier patch for fix test breakage.   George Joseph
• [asterisk-dev] [Code Review] 4276: res_pjsip_config_wizard: Update to earlier patch for fix test breakage.   George Joseph
• [asterisk-dev] [Code Review] 4276: res_pjsip_config_wizard: Update to earlier patch for fix test breakage.   George Joseph
• [asterisk-dev] [Code Review] 4275: config: make config_cache_remove public as ast_config_cache_remove   George Joseph
• [asterisk-dev] [Code Review] 4276: res_pjsip_config_wizard: Update to earlier patch for fix test breakage.   George Joseph
• [asterisk-dev] [Code Review] 4288: res_pjsip_phoneprovi_provider: Fix reload   George Joseph
• [asterisk-dev] [Code Review] 4288: res_pjsip_phoneprovi_provider: Fix reload   George Joseph
• [asterisk-dev] [Code Review] 4288: res_pjsip_phoneprovi_provider: Fix reload   George Joseph
• [asterisk-dev] [Code Review] 4288: res_pjsip_phoneprovi_provider: Fix reload   George Joseph
• [asterisk-dev] [Code Review] 4289: pjsip_options: Fix duplicate qualify schedules on reload.   George Joseph
• [asterisk-dev] [Code Review] 4290: pjsip_options: Fix continued qualifies after endpoint/aor deletion.   George Joseph
• [asterisk-dev] PJSIP_AOR and PJSIP_CONTACT Dialplan Functions   George Joseph
• [asterisk-dev] git migration update   George Joseph
• [asterisk-dev] [Code Review] 4289: pjsip_options: Fix duplicate qualify schedules on reload.   George Joseph
• [asterisk-dev] [Code Review] 4295: test_astobj2: Fix warning for missing trailing slash in category.   George Joseph
• [asterisk-dev] [Code Review] 4289: pjsip_options: Fix duplicate qualify schedules on reload.   George Joseph
• [asterisk-dev] [Code Review] 4289: pjsip_options: Fix duplicate qualify schedules on reload.   George Joseph
• [asterisk-dev] [Code Review] 4290: pjsip_options: Fix continued qualifies after endpoint/aor deletion.   George Joseph
• [asterisk-dev] [Code Review] 4295: test_astobj2: Fix warning for missing trailing slash in category.   George Joseph
• [asterisk-dev] [Code Review] 4290: pjsip_options: Fix continued qualifies after endpoint/aor deletion.   George Joseph
• [asterisk-dev] [Code Review] 4297: config: Add option to NOT preserve the effective context when changing a template.   George Joseph
• [asterisk-dev] [Code Review] 4298: Testsuite: Tests for Manager/Config/NoPreserveEffectiveContext   George Joseph
• [asterisk-dev] PJSIP_AOR and PJSIP_CONTACT Dialplan Functions   George Joseph
• [asterisk-dev] PJSIP_AOR and PJSIP_CONTACT Dialplan Functions   George Joseph
• [asterisk-dev] [Code Review] 4300: bridge_native_rtp: Change local/remote message from debug/2 to verb/4   George Joseph
• [asterisk-dev] [Code Review] 4301: res_pjsip_outbound_registration: Add 'pjsip send register' command and update behavior of 'send unregister'   George Joseph
• [asterisk-dev] [Code Review] 4297: config: Add option to NOT preserve the effective context when changing a template.   George Joseph
• [asterisk-dev] [Code Review] 4297: config: Add option to NOT preserve the effective context when changing a template.   George Joseph
• [asterisk-dev] [Code Review] 4301: res_pjsip_outbound_registration: Add 'pjsip send register' command and update behavior of 'send unregister'   George Joseph
• [asterisk-dev] [Code Review] 4301: res_pjsip_outbound_registration: Add 'pjsip send register' command and update behavior of 'send unregister'   George Joseph
• [asterisk-dev] [Code Review] 4301: res_pjsip_outbound_registration: Add 'pjsip send register' command and update behavior of 'send unregister'   George Joseph
• [asterisk-dev] [Code Review] 4303: res_pjsip_outbound_registration: Fix reference leak.   George Joseph
• [asterisk-dev] [Code Review] 4304: res_pjsip_outbound_registration: Fix several reload issues.   George Joseph
• [asterisk-dev] [Code Review] 4305: pjsip cli: Fix sorting of contacts for 'pjsip list contacts'   George Joseph
• [asterisk-dev] ARI Extending Existing Feature: bridge control   Kaloyan Kovachev
• [asterisk-dev] ARI Extending Existing Feature: bridge control   Kaloyan Kovachev
• [asterisk-dev] [Code Review] 4204: Failure showing codecs via 'core show channeltype <tech>'   Brad Latus
• [asterisk-dev] [Code Review] 4258: Fix crash for sorcery misconfigs   David Lee
• [asterisk-dev] [Code Review] 4258: Fix crash for sorcery misconfigs   David Lee
• [asterisk-dev] [Code Review] 2826: Debug threads: avoid double-initialization of lock tracking   David Lee
• [asterisk-dev] ARI Extending Existing Feature: bridge control   Leif Madsen
• [asterisk-dev] git migration update   Leif Madsen
• [asterisk-dev] asterisknow-version   Leif Madsen
• [asterisk-dev] pjsip vs ca path   Mark Michelson
• [asterisk-dev] pjsip vs ca path   Mark Michelson
• [asterisk-dev] [Code Review] 4190: res_pjsip_config_wizard: Allow streamlined configuration of common pjsip scenarios   Mark Michelson
• [asterisk-dev] [Code Review] 4231: New AMI/ARI events for connected line updates on a channel   Mark Michelson
• [asterisk-dev] [Code Review] 4232: Testsuite: Tests for AMI NewConnectedLine and ARI ChannelConnectedLine events   Mark Michelson
• [asterisk-dev] [Code Review] 4233: res_pjsip_session: Fix crash that would occur when rescheduling a reinvite due to a 491 response.   Mark Michelson
• [asterisk-dev] [Code Review] 4234: Testsuite: Ensure that changes in /r/4233 result in Asterisk no longer crashing.   Mark Michelson
• [asterisk-dev] [Code Review] 4232: Testsuite: Tests for AMI NewConnectedLine and ARI ChannelConnectedLine events   Mark Michelson
• [asterisk-dev] [Code Review] 4233: res_pjsip_session: Fix crash that would occur when rescheduling a reinvite due to a 491 response.   Mark Michelson
• [asterisk-dev] [Code Review] 4232: Testsuite: Tests for AMI NewConnectedLine and ARI ChannelConnectedLine events   Mark Michelson
• [asterisk-dev] [Code Review] 4231: New AMI/ARI events for connected line updates on a channel   Mark Michelson
• [asterisk-dev] [Code Review] 4233: res_pjsip_session: Fix crash that would occur when rescheduling a reinvite due to a 491 response.   Mark Michelson
• [asterisk-dev] [Code Review] 4234: Testsuite: Ensure that changes in /r/4233 result in Asterisk no longer crashing.   Mark Michelson
• [asterisk-dev] Rules for binding with regards to RTP   Mark Michelson
• [asterisk-dev] Rules for binding with regards to RTP   Mark Michelson
• [asterisk-dev] [Code Review] 4232: Testsuite: Tests for AMI NewConnectedLine and ARI ChannelConnectedLine events   Mark Michelson
• [asterisk-dev] Rules for binding with regards to RTP   Mark Michelson
• [asterisk-dev] [Code Review] 4248: res_pjsip_session: Queue BYE if an outstanding INVITE transaction exists   Mark Michelson
• [asterisk-dev] [Code Review] 4249: PJSIP Blind Transfer Direct Media Tests: Update to expect sane behavior   Mark Michelson
• [asterisk-dev] [Code Review] 4247: DEBUG_THREADS: Fix regression and lock tracking initialization problems.   Mark Michelson
• [asterisk-dev] [Code Review] 4247: DEBUG_THREADS: Fix regression and lock tracking initialization problems.   Mark Michelson
• [asterisk-dev] [Code Review] 4247: DEBUG_THREADS: Fix regression and lock tracking initialization problems.   Mark Michelson
• [asterisk-dev] [Code Review] 4257: chan_pjsip: Race condition between channel answer and bridge setup when using direct media   Mark Michelson
• [asterisk-dev] [Code Review] 4254: res_pjsip_session: Don't allow declined media streams to fail SDP negotiation on re-INVITE   Mark Michelson
• [asterisk-dev] [Code Review] 4255: Testsuite: Add test for SDP offer/answer during hold with a declined stream   Mark Michelson
• [asterisk-dev] [Code Review] 4258: Fix crash for sorcery misconfigs   Mark Michelson
• [asterisk-dev] [Code Review] 4253: Testsuite: Add tests for use of 'inactive' stream direction for hold   Mark Michelson
• [asterisk-dev] [Code Review] 4253: Testsuite: Add tests for use of 'inactive' stream direction for hold   Mark Michelson
• [asterisk-dev] [Code Review] 4257: chan_pjsip: Race condition between channel answer and bridge setup when using direct media   Mark Michelson
• [asterisk-dev] [Code Review] 4251: loader: Move definition of ast_module_reload from _private.h to module.h   Mark Michelson
• [asterisk-dev] [Code Review] 4261: res_pjsip_pubsub: Activate recreated persistent subscriptions   Mark Michelson
• [asterisk-dev] [Code Review] 4261: res_pjsip_pubsub: Activate recreated persistent subscriptions   Mark Michelson
• [asterisk-dev] [Code Review] 4268: During a dual redirect, prevent a race condition that may cause one of the redirected channels to be hung up.   Mark Michelson
• [asterisk-dev] [Code Review] 4101: Channel Originate/Continue via ARI support for labels in dialplan is incomplete   Mark Michelson
• [asterisk-dev] [Code Review] 4273: res_pjsip_outbound_registration: Prevent infinite authentication loops   Mark Michelson
• [asterisk-dev] [Code Review] 4274: Testsuite: Ensure that outbound registrations with bad auth do not result in an infinite loop.   Mark Michelson
• [asterisk-dev] [Code Review] 4262: bridge: channel ref leak on blond_nonfinal_enter   Mark Michelson
• [asterisk-dev] [Code Review] 4271: ARI: Allow interoperation of ASYNCGOTO and channels originated to Stasis()   Mark Michelson
• [asterisk-dev] [Code Review] 4271: ARI: Allow interoperation of ASYNCGOTO and channels originated to Stasis()   Mark Michelson
• [asterisk-dev] [Code Review] 4277: Ensure that CHANNEL(pjsip, secure) gives expected value.   Mark Michelson
• [asterisk-dev] [Code Review] 4101: Channel Originate/Continue via ARI support for labels in dialplan is incomplete   Mark Michelson
• [asterisk-dev] [Code Review] 4274: Testsuite: Ensure that outbound registrations with bad auth do not result in an infinite loop.   Mark Michelson
• [asterisk-dev] [Code Review] 4274: Testsuite: Ensure that outbound registrations with bad auth do not result in an infinite loop.   Mark Michelson
• [asterisk-dev] [Code Review] 4101: Channel Originate/Continue via ARI support for labels in dialplan is incomplete   Mark Michelson
• [asterisk-dev] [Code Review] 4280: sip_to_pjsip: improved ability to parse input without exception   Mark Michelson
• [asterisk-dev] [Code Review] 4276: res_pjsip_config_wizard: Update to earlier patch for fix test breakage.   Mark Michelson
• [asterisk-dev] [Code Review] 4276: res_pjsip_config_wizard: Update to earlier patch for fix test breakage.   Mark Michelson
• [asterisk-dev] ARI Extending Existing Feature: bridge control   Mark Michelson
• [asterisk-dev] [Code Review] 4276: res_pjsip_config_wizard: Update to earlier patch for fix test breakage.   Mark Michelson
• [asterisk-dev] [Code Review] 4276: res_pjsip_config_wizard: Update to earlier patch for fix test breakage.   Mark Michelson
• [asterisk-dev] ARI Extending Existing Feature: bridge control   Mark Michelson
• [asterisk-dev] [Code Review] 4281: Testsuite: Single channel redirect tests   Mark Michelson
• [asterisk-dev] [Code Review] 4277: Ensure that CHANNEL(pjsip, secure) gives expected value.   Mark Michelson
• [asterisk-dev] [Code Review] 4268: During a dual redirect, prevent a race condition that may cause one of the redirected channels to be hung up.   Mark Michelson
• [asterisk-dev] [Code Review] 4273: res_pjsip_outbound_registration: Prevent infinite authentication loops   Mark Michelson
• [asterisk-dev] [Code Review] 4285: Bug fixes for ARI Originate/Continue with label support (Continuation of /r/4101)   Mark Michelson
• [asterisk-dev] [Code Review] 4284: Testsuite: Update ARI test for continuation and add new ARI test for origination to dialplan location   Mark Michelson
• [asterisk-dev] [Code Review] 4101: Channel Originate/Continue via ARI support for labels in dialplan is incomplete   Mark Michelson
• [asterisk-dev] [Code Review] 4284: Testsuite: Update ARI test for continuation and add new ARI test for origination to dialplan location   Mark Michelson
• [asterisk-dev] [Code Review] 4281: Testsuite: Single channel redirect tests   Mark Michelson
• [asterisk-dev] [Code Review] 4281: Testsuite: Single channel redirect tests   Mark Michelson
• [asterisk-dev] [Code Review] 4296: PJSIP: Fix bugs and improve documentation of remote attended transfers   Mark Michelson
• [asterisk-dev] ARI Extending Existing Feature: bridge control   Phil Mickelson
• [asterisk-dev] ARI Extending Existing Feature: bridge control   Phil Mickelson
• [asterisk-dev] ARI Extending Existing Feature: bridge control   Phil Mickelson
• [asterisk-dev] Asterisk 12 - Security Fix Only! (aka: update repotools)   Richard Mudgett
• [asterisk-dev] You have received the paymen   Max Mutrux
• [asterisk-dev] [Code Review] 4188: segfault when playing back voicemail under high concurrency with an IMAP backend   David Duncan Ross Palmer
• [asterisk-dev] [Code Review] 4050: Add ability for Channel Drivers to provide Presence State information   Jonathan Rose
• [asterisk-dev] ARI Extending Existing Feature: bridge control   Nir Simionovich
• [asterisk-dev] ARI Extending Existing Feature: bridge control   Nir Simionovich
• [asterisk-dev] ARI Extending Existing Feature: bridge control   Nir Simionovich
• [asterisk-dev] ARI Extending Existing Feature: bridge control   Nir Simionovich
• [asterisk-dev] ARI Extending Existing Feature: bridge control   Nir Simionovich
• [asterisk-dev] ARI Extending Existing Feature: bridge control   Nir Simionovich
• [asterisk-dev] ARI Extending Existing Feature: bridge control   Nir Simionovich
• [asterisk-dev] ARI Extending Existing Feature: bridge control   Nir Simionovich
• [asterisk-dev] ARI Extending Existing Feature: bridge control   Nir Simionovich
• [asterisk-dev] ARI Extending Existing Feature: bridge control   Nir Simionovich
• [asterisk-dev] ARI Extending Existing Feature: bridge control   Nir Simionovich
• [asterisk-dev] ARI Extending Existing Feature: bridge control   Nir Simionovich
• [asterisk-dev] ARI Stasis Application Examples and Tutorials   Nir Simionovich
• [asterisk-dev] git migration update   Nir Simionovich
• [asterisk-dev] [Code Review] 4201: app_record: prevent stripping 1/4 second audio when channel is hung up to end recording.   Ben Smithurst
• [asterisk-dev] Asterisk 11.15.0-rc1 Now Available   Asterisk Development Team
• [asterisk-dev] Asterisk 12.8.0-rc1 Now Available   Asterisk Development Team
• [asterisk-dev] Asterisk 13.1.0-rc1 Now Available   Asterisk Development Team
• [asterisk-dev] Asterisk 11.6-cert9, 11.14.2, 12.7.2, 13.0.2 Now Available (Security Release)   Asterisk Development Team
• [asterisk-dev] Asterisk 11.15.0-rc2 Now Available   Asterisk Development Team
• [asterisk-dev] Asterisk 12.8.0-rc2 Now Available   Asterisk Development Team
• [asterisk-dev] Asterisk 13.1.0-rc2 Now Available   Asterisk Development Team
• [asterisk-dev] Asterisk 11.15.0 Now Available   Asterisk Development Team
• [asterisk-dev] Asterisk 12.8.0 Now Available   Asterisk Development Team
• [asterisk-dev] Asterisk 13.1.0 Now Available   Asterisk Development Team
• [asterisk-dev] AST-2014-019: Remote Crash Vulnerability in WebSocket Server   Asterisk Security Team
• [asterisk-dev] Planned maintenance for community services on Tuesday, 9th of December 2014   Digium's Asterisk Development Team
• [asterisk-dev] Planned maintenance for community services on Sunday, 14th of December 2014   Digium's Asterisk Development Team
• [asterisk-dev] Planned maintenance for community services tonight, Tuesday the 16th of December 2014   Digium's Asterisk Development Team
• [asterisk-dev] Planned maintenance for community services tonight (19th) and this Sunday (21st)   Digium's Asterisk Development Team
• [asterisk-dev] ARI Extending Existing Feature: bridge control   Jonathan White
• [asterisk-dev] confbridge feature request   Jonathan White
• [asterisk-dev] confbridge feature request   Jonathan White
• [asterisk-dev] asterisknow-version   Jonathan White
• [asterisk-dev] [Code Review] 4230: add capath support to res_pjsip   cloos
• [asterisk-dev] [Code Review] 4230: add capath support to res_pjsip   cloos
• [asterisk-dev] [Code Review] 4050: Add ability for Channel Drivers to provide Presence State information   gareth
• [asterisk-dev] [Code Review] 4050: Add ability for Channel Drivers to provide Presence State information   gareth
• [asterisk-dev] [Code Review] 4023: Allow passing options and command to MixMonitor when recording in ConfBridge   gareth
• [asterisk-dev] [Code Review] 4050: Add ability for Channel Drivers to provide Presence State information   gareth
• [asterisk-dev] [Code Review] 4023: Allow passing options and command to MixMonitor when recording in ConfBridge   gareth
• [asterisk-dev] [Code Review] 4050: Add ability for Channel Drivers to provide Presence State information   gareth
• [asterisk-dev] [Code Review] 4101: Channel Originate/Continue via ARI support for labels in dialplan is incomplete   greenfieldtech
• [asterisk-dev] [Code Review] 4101: Channel Originate/Continue via ARI support for labels in dialplan is incomplete   greenfieldtech
• [asterisk-dev] [Code Review] 4101: Channel Originate/Continue via ARI support for labels in dialplan is incomplete   greenfieldtech
• [asterisk-dev] [Code Review] 4101: Channel Originate/Continue via ARI support for labels in dialplan is incomplete   greenfieldtech
• [asterisk-dev] [Code Review] 4101: Channel Originate/Continue via ARI support for labels in dialplan is incomplete   greenfieldtech
• [asterisk-dev] [Code Review] 4101: Channel Originate/Continue via ARI support for labels in dialplan is incomplete   greenfieldtech
• [asterisk-dev] [Code Review] 4101: Channel Originate/Continue via ARI support for labels in dialplan is incomplete   greenfieldtech
• [asterisk-dev] [Code Review] 4101: Channel Originate/Continue via ARI support for labels in dialplan is incomplete   greenfieldtech
• [asterisk-dev] [Code Review] 4101: Channel Originate/Continue via ARI support for labels in dialplan is incomplete   greenfieldtech
• [asterisk-dev] [Code Review] 4101: Channel Originate/Continue via ARI support for labels in dialplan is incomplete   greenfieldtech
• [asterisk-dev] [Code Review] 4101: Channel Originate/Continue via ARI support for labels in dialplan is incomplete   greenfieldtech
• [asterisk-dev] [Code Review] 4267: Testsuite: Add blind transfer tests for Stasis application interaction.   jbigelow
• [asterisk-dev] [Code Review] 4267: Testsuite: Add blind transfer tests for Stasis application interaction.   jbigelow
• [asterisk-dev] [Code Review] 4291: Testsuite: Add external bridging tests for Stasis application interaction - Stasis bridge to non-stasis application.   jbigelow
• [asterisk-dev] [Code Review] 4291: Testsuite: Add external bridging tests for Stasis application interaction - Stasis bridge to non-stasis application.   jbigelow
• [asterisk-dev] [Code Review] 4267: Testsuite: Add blind transfer tests for Stasis application interaction.   jbigelow
• [asterisk-dev] [Code Review] 4267: Testsuite: Add blind transfer tests for Stasis application interaction.   jbigelow
• [asterisk-dev] [Code Review] 4279: chan_sip: Send CANCEL via proxy if CANCEL is to be sent after an UPDATE   kwemheuer
• [asterisk-dev] [Code Review] 4279: chan_sip: Send CANCEL via proxy if CANCEL is to be sent after an UPDATE   kwemheuer
• [asterisk-dev] [Code Review] 4279: chan_sip: Send CANCEL via proxy if CANCEL is to be sent after an UPDATE   kwemheuer
• [asterisk-dev] [Code Review] 4279: chan_sip: Send CANCEL via proxy if CANCEL is to be sent after an UPDATE   kwemheuer
• [asterisk-dev] Asterisk created mmlog   bala murugan
• [asterisk-dev] [Code Review] 4242: app_meetme: Fix default values initialization when no configuration file is present   nerbos
• [asterisk-dev] [Code Review] 4242: app_meetme: Fix default values initialization when no configuration file is present   nerbos
• [asterisk-dev] [Code Review] 4242: app_meetme: Fix default values initialization when no configuration file is present   nerbos
• [asterisk-dev] [Code Review] 4242: app_meetme: Fix default values initialization when no configuration file is present   nerbos
• [asterisk-dev] [Code Review] 4141: Enable REF_DEBUG for ast_module_ref / ast_module_unref   opticron
• [asterisk-dev] [Code Review] 4213: Stasis: Fix StasisStart and StasisEnd ordering and missing events   opticron
• [asterisk-dev] [Code Review] 4197: testsuite: Update cleanup-test-remnants to clean /var/tmp/asterisk-testsuite and ./logs/*   opticron
• [asterisk-dev] [Code Review] 4213: Stasis: Fix StasisStart and StasisEnd ordering and missing events   opticron
• [asterisk-dev] [Code Review] 4226: Testsuite: Add tests for external interactions with ARI/Stasis   opticron
• [asterisk-dev] [Code Review] 4226: Testsuite: Add tests for external interactions with ARI/Stasis   opticron
• [asterisk-dev] [Code Review] 4230: add capath support to res_pjsip   opticron
• [asterisk-dev] [Code Review] 4226: Testsuite: Add tests for external interactions with ARI/Stasis   opticron
• [asterisk-dev] [Code Review] 4226: Testsuite: Add tests for external interactions with ARI/Stasis   opticron
• [asterisk-dev] [Code Review] 4232: Testsuite: Tests for AMI NewConnectedLine and ARI ChannelConnectedLine events   opticron
• [asterisk-dev] [Code Review] 4231: New AMI/ARI events for connected line updates on a channel   opticron
• [asterisk-dev] [Code Review] 4232: Testsuite: Tests for AMI NewConnectedLine and ARI ChannelConnectedLine events   opticron
• [asterisk-dev] [Code Review] 4213: Stasis: Fix StasisStart and StasisEnd ordering and missing events   opticron
• [asterisk-dev] [Code Review] 4226: Testsuite: Add tests for external interactions with ARI/Stasis   opticron
• [asterisk-dev] [Code Review] 4246: PJSIP: Stagger outbound qualifies   opticron
• [asterisk-dev] [Code Review] 4232: Testsuite: Tests for AMI NewConnectedLine and ARI ChannelConnectedLine events   opticron
• [asterisk-dev] [Code Review] 4246: PJSIP: Stagger outbound qualifies   opticron
• [asterisk-dev] [Code Review] 4246: PJSIP: Stagger outbound qualifies   opticron
• [asterisk-dev] [Code Review] 4182: core: avoid rasterisk crash due to long identifier   opticron
• [asterisk-dev] [Code Review] 4250: Sorcery: Log when a stale configuration remains in use   opticron
• [asterisk-dev] [Code Review] 4253: Testsuite: Add tests for use of 'inactive' stream direction for hold   opticron
• [asterisk-dev] [Code Review] 4252: PJSIP: Allow use of 'inactive' stream types for hold   opticron
• [asterisk-dev] [Code Review] 4254: res_pjsip_session: Don't allow declined media streams to fail SDP negotiation on re-INVITE   opticron
• [asterisk-dev] [Code Review] 4255: Testsuite: Add test for SDP offer/answer during hold with a declined stream   opticron
• [asterisk-dev] [Code Review] 4253: Testsuite: Add tests for use of 'inactive' stream direction for hold   opticron
• [asterisk-dev] [Code Review] 4250: Sorcery: Log when a stale configuration remains in use   opticron
• [asterisk-dev] [Code Review] 4252: PJSIP: Allow use of 'inactive' stream types for hold   opticron
• [asterisk-dev] [Code Review] 4253: Testsuite: Add tests for use of 'inactive' stream direction for hold   opticron
• [asterisk-dev] [Code Review] 4263: Fix characters unjustly cast to unsigned int before printf formatting   opticron
• [asterisk-dev] [Code Review] 4264: PJSIP: Update transport method documentation   opticron
• [asterisk-dev] [Code Review] 4155: PJSIP: Allow contact rewriting to fall back for reliable transports   opticron
• [asterisk-dev] [Code Review] 4266: Testsuite - res_pjsip_sdp_rtp: Incompatible DTMF mode test   opticron
• [asterisk-dev] [Code Review] 4265: res_pjsip_sdp_rtp: Native RTP bridging is chosen for RTP compatible channels when the DTMF mode is not compatible   opticron
• [asterisk-dev] [Code Review] 4271: ARI: Allow interoperation of ASYNCGOTO and channels originated to Stasis()   opticron
• [asterisk-dev] [Code Review] 4272: Testsuite: Verify that bridged channels originated into Stasis and AMI Redirect interoperate properly   opticron
• [asterisk-dev] [Code Review] 4271: ARI: Allow interoperation of ASYNCGOTO and channels originated to Stasis()   opticron
• [asterisk-dev] [Code Review] 4271: ARI: Allow interoperation of ASYNCGOTO and channels originated to Stasis()   opticron
• [asterisk-dev] [Code Review] 4271: ARI: Allow interoperation of ASYNCGOTO and channels originated to Stasis()   opticron
• [asterisk-dev] [Code Review] 4271: ARI: Allow interoperation of ASYNCGOTO and channels originated to Stasis()   opticron
• [asterisk-dev] [Code Review] 4267: Testsuite: Add blind transfer tests for Stasis application interaction.   opticron
• [asterisk-dev] [Code Review] 4264: PJSIP: Update transport method documentation   opticron
• [asterisk-dev] [Code Review] 4272: Testsuite: Verify that bridged channels originated into Stasis and AMI Redirect interoperate properly   opticron
• [asterisk-dev] [Code Review] 4231: New AMI/ARI events for connected line updates on a channel   rmudgett
• [asterisk-dev] [Code Review] 4242: app_meetme: Fix default values initialization when no configuration file is present   rmudgett
• [asterisk-dev] [Code Review] 4242: app_meetme: Fix default values initialization when no configuration file is present   rmudgett
• [asterisk-dev] [Code Review] 4242: app_meetme: Fix default values initialization when no configuration file is present   rmudgett
• [asterisk-dev] [Code Review] 4243: ari: Add the ability to specify a linkedId when originating calls.   rmudgett
• [asterisk-dev] [Code Review] 4243: ari: Add the ability to specify a linkedId when originating calls.   rmudgett
• [asterisk-dev] [Code Review] 4246: PJSIP: Stagger outbound qualifies   rmudgett
• [asterisk-dev] [Code Review] 4246: PJSIP: Stagger outbound qualifies   rmudgett
• [asterisk-dev] [Code Review] 4247: DEBUG_THREADS: Fix regression and lock tracking initialization problems.   rmudgett
• [asterisk-dev] [Code Review] 4247: DEBUG_THREADS: Fix regression and lock tracking initialization problems.   rmudgett
• [asterisk-dev] [Code Review] 4247: DEBUG_THREADS: Fix regression and lock tracking initialization problems.   rmudgett
• [asterisk-dev] [Code Review] 4247: DEBUG_THREADS: Fix regression and lock tracking initialization problems.   rmudgett
• [asterisk-dev] [Code Review] 4108: Weak References   rmudgett
• [asterisk-dev] [Code Review] 4247: DEBUG_THREADS: Fix regression and lock tracking initialization problems.   rmudgett
• [asterisk-dev] [Code Review] 4259: DTMF atxfer: Setup recall channels as if the original transferrer originated the call.   rmudgett
• [asterisk-dev] [Code Review] 4259: DTMF atxfer: Setup recall channels as if the original transferrer originated the call.   rmudgett
• [asterisk-dev] [Code Review] 4247: DEBUG_THREADS: Fix regression and lock tracking initialization problems.   rmudgett
• [asterisk-dev] [Code Review] 4247: DEBUG_THREADS: Fix regression and lock tracking initialization problems.   rmudgett
• [asterisk-dev] [Code Review] 4262: bridge: channel ref leak on blond_nonfinal_enter   rmudgett
• [asterisk-dev] [Code Review] 4259: DTMF atxfer: Setup recall channels as if the original transferrer originated the call.   rmudgett
• [asterisk-dev] [Code Review] 4271: ARI: Allow interoperation of ASYNCGOTO and channels originated to Stasis()   rmudgett
• [asterisk-dev] [Code Review] 4271: ARI: Allow interoperation of ASYNCGOTO and channels originated to Stasis()   rmudgett
• [asterisk-dev] [Code Review] 4271: ARI: Allow interoperation of ASYNCGOTO and channels originated to Stasis()   rmudgett
• [asterisk-dev] [Code Review] 4282: queue_log: Post QUEUESTART entry when Asterisk fully boots.   rmudgett
• [asterisk-dev] [Code Review] 4282: queue_log: Post QUEUESTART entry when Asterisk fully boots.   rmudgett
• [asterisk-dev] [Code Review] 4259: DTMF atxfer: Setup recall channels as if the original transferrer originated the call.   rmudgett
• [asterisk-dev] [Code Review] 4292: app_macro: Don't restore the calling location on a channel redirect.   rmudgett
• [asterisk-dev] [Code Review] 4292: app_macro: Don't restore the calling location on a channel redirect.   rmudgett
• [asterisk-dev] [Code Review] 4281: Testsuite: Single channel redirect tests   rmudgett
• [asterisk-dev] [Code Review] 4188: segfault when playing back voicemail under high concurrency with an IMAP backend   wdoekes
• [asterisk-dev] [Code Review] 4263: Fix characters unjustly cast to unsigned int before printf formatting   wdoekes
• [asterisk-dev] [Code Review] 4263: Fix characters unjustly cast to unsigned int before printf formatting   wdoekes
• [asterisk-dev] [Code Review] 4263: Fix characters unjustly cast to unsigned int before printf formatting   wdoekes
• [asterisk-dev] [Code Review] 4263: Fix characters unjustly cast to unsigned int before printf formatting   wdoekes
• [asterisk-dev] [Code Review] 4263: Fix characters unjustly cast to unsigned int before printf formatting   wdoekes
• [asterisk-dev] confbridge feature request   jonathan white

Last message date: Wed Dec 31 17:18:01 CST 2014
Archived on: Wed Dec 31 17:18:02 CST 2014

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