November 2014 Archives by author

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Starting: Sun Nov 2 01:13:24 CST 2014
Ending: Fri Nov 28 13:18:46 CST 2014
Messages: 396

• [asterisk-dev] SIP Deadlock/Pause   Ross Beer
• [asterisk-dev] SIP Deadlock/Pause   Ross Beer
• [asterisk-dev] [Code Review] 4093: Codec Format Is Not Included in the SDP Media Attributes When SLIN48 Codec Is Used   Frankie Chin
• [asterisk-dev] [Code Review] 4093: Codec Format Is Not Included in the SDP Media Attributes When SLIN48 Codec Is Used   Frankie Chin
• [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] [Code Review] 4103: chan_pjsip: Add 'moh_passthrough' option for passing through musiconhold requests.   Joshua Colp
• [asterisk-dev] [asterisk-commits] file: trunk r427112 - in /trunk: channels/ channels/pjsip/ configs/samples/ c...   Joshua Colp
• [asterisk-dev] Fwd: [asterisk-commits] mjordan: branch 12 r426995 - /branches/12/res/res_stasis.c   Joshua Colp
• [asterisk-dev] [Code Review] 4146: res_pjsip: If an endpoint is associated with the dialog place it on the messag early   Joshua Colp
• [asterisk-dev] [Code Review] 2964: res_pjsip_outbound_registration: Add "virtual line" support for automatic inbound matching   Joshua Colp
• [asterisk-dev] [Code Review] 4099: Optimistic SRTP Tests.   Joshua Colp
• [asterisk-dev] [Code Review] 4148: res_pjsip_multihomed: Provide logging during startup for an indication of what is going on   Joshua Colp
• [asterisk-dev] [Code Review] 4150: res_hep: fix major leak that occurs when config is missing or enabled=no   Joshua Colp
• [asterisk-dev] [Code Review] 4148: res_pjsip_multihomed: Provide logging during startup for an indication of what is going on   Joshua Colp
• [asterisk-dev] [Code Review] 4149: main/file.c: fix possible extra ast_module_unref to format modules   Joshua Colp
• [asterisk-dev] [Code Review] 4093: Codec Format Is Not Included in the SDP Media Attributes When SLIN48 Codec Is Used   Joshua Colp
• [asterisk-dev] [Code Review] 3603: func_jitterbuffer: fix audio failure caused by certain masquerade's   Joshua Colp
• [asterisk-dev] [Code Review] 4053: res_pjsip_history: A debugging module for busy systems   Joshua Colp
• [asterisk-dev] [Code Review] 4053: res_pjsip_history: A debugging module for busy systems   Joshua Colp
• [asterisk-dev] [Code Review] 4101: Channel Originate/Continue via ARI support for labels in dialplan is incomplete   Joshua Colp
• [asterisk-dev] [Code Review] 3867: [chan_sip] Default DTLS settings to use if peer misses own settings   Joshua Colp
• [asterisk-dev] [Code Review] 4101: Channel Originate/Continue via ARI support for labels in dialplan is incomplete   Joshua Colp
• [asterisk-dev] [Code Review] 4102: testsuite: add secure websocket test   Joshua Colp
• [asterisk-dev] [Code Review] 3603: func_jitterbuffer: fix audio failure caused by certain masquerade's   Joshua Colp
• [asterisk-dev] func_jitterbuffer handling of masquerades   Joshua Colp
• [asterisk-dev] [Code Review] 4139: stun: fix size of attribute string to match rfc   Joshua Colp
• [asterisk-dev] [Code Review] 4146: res_pjsip: If an endpoint is associated with the dialog place it on the messag early   Joshua Colp
• [asterisk-dev] Asterisk 14 - Remote URI Playback   Joshua Colp
• [asterisk-dev] func_jitterbuffer handling of masquerades   Joshua Colp
• [asterisk-dev] [Code Review] 4157: bridge_native_rtp: Fix T.38 directmedia fax test by always asking the remote peers to update themselves on native bridge stop   Joshua Colp
• [asterisk-dev] Notes from setting up SIP+TLS/RTP+DTLS   Joshua Colp
• [asterisk-dev] Notes from setting up SIP+TLS/RTP+DTLS   Joshua Colp
• [asterisk-dev] [Code Review] 4162: pbx: Fix crash in off-nominal when add_priority encounters a failure.   Joshua Colp
• [asterisk-dev] [Code Review] 4162: pbx: Fix crash in off-nominal when add_priority encounters a failure.   Joshua Colp
• [asterisk-dev] [Code Review] 4158: rtp_engine: Fix crash when endpoints send more RTCP report info blocks then we can handle   Joshua Colp
• [asterisk-dev] [Code Review] 4155: PJSIP: Allow contact rewriting to fall back for reliable transports   Joshua Colp
• [asterisk-dev] [Code Review] 4155: PJSIP: Allow contact rewriting to fall back for reliable transports   Joshua Colp
• [asterisk-dev] [Code Review] 4155: PJSIP: Allow contact rewriting to fall back for reliable transports   Joshua Colp
• [asterisk-dev] [Code Review] 4099: Optimistic SRTP Tests.   Joshua Colp
• [asterisk-dev] [Code Review] 4162: pbx: Fix crash in off-nominal when add_priority encounters a failure.   Joshua Colp
• [asterisk-dev] [Code Review] 4173: bridge: Protect bridge channel when changing state and make it smarter   Joshua Colp
• [asterisk-dev] [Code Review] 4093: Codec Format Is Not Included in the SDP Media Attributes When SLIN48 Codec Is Used   Joshua Colp
• [asterisk-dev] [Code Review] 4175: Fix race condition where identical SIP requests are processed by multiple threads (Asterisk 13)   Joshua Colp
• [asterisk-dev] [Code Review] 4174: Fix race condition where identical SIP requests are processed by multiple threads.   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] 4167: Allow mis-dialed DTMF-initiated transfers to be re-attempted.   Joshua Colp
• [asterisk-dev] [Code Review] 4155: PJSIP: Allow contact rewriting to fall back for reliable transports   Joshua Colp
• [asterisk-dev] [Code Review] 4177: app_confbridge: Play 'leader has left' sound file even when musiconhold is enabled   Joshua Colp
• [asterisk-dev] [Code Review] 4177: app_confbridge: Play 'leader has left' sound file even when musiconhold is enabled   Joshua Colp
• [asterisk-dev] [Code Review] 4177: app_confbridge: Play 'leader has left' sound file even when musiconhold is enabled   Joshua Colp
• [asterisk-dev] [Code Review] 4183: res_pjsip_phoneprov_provider: bug: call ast_sorcery_apply_config so config can come from realtime.   Joshua Colp
• [asterisk-dev] [Code Review] 4187: res_pjsip_refer: Ensure Refer-To is NULL terminated and parse it as a URI   Joshua Colp
• [asterisk-dev] [Code Review] 4187: res_pjsip_refer: Ensure Refer-To is NULL terminated and parse it as a URI   Joshua Colp
• [asterisk-dev] [Code Review] 4187: res_pjsip_refer: Ensure Refer-To is NULL terminated and parse it as a URI   Joshua Colp
• [asterisk-dev] [Code Review] 3992: res_pjsip_sdp_rtp: Add optimistic SRTP support.   Joshua Colp
• [asterisk-dev] [Code Review] 4099: Optimistic SRTP Tests.   Joshua Colp
• [asterisk-dev] [Code Review] 4196: bridge_basic: Fix features issues introduced by review 4167   Joshua Colp
• [asterisk-dev] [Code Review] 4185: sorcery: Make ast_sorcery_is_object_field_registered handle field names that are regexes.   Joshua Colp
• [asterisk-dev] [Code Review] 4185: sorcery: Make ast_sorcery_is_object_field_registered handle field names that are regexes.   Joshua Colp
• [asterisk-dev] [Code Review] 4101: Channel Originate/Continue via ARI support for labels in dialplan is incomplete   Joshua Colp
• [asterisk-dev] [Code Review] 4093: Codec Format Is Not Included in the SDP Media Attributes When SLIN48 Codec Is Used   Joshua Colp
• [asterisk-dev] [Code Review] 4185: sorcery: Make ast_sorcery_is_object_field_registered handle field names that are regexes.   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] 4204: Failure showing codecs via 'core show channeltype <tech>'   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] 4141: Enable REF_DEBUG for ast_module_ref / ast_module_unref   Corey Farrell
• [asterisk-dev] [Code Review] 4140: res_http_websockets: Module reference decrease below zero   Corey Farrell
• [asterisk-dev] Fwd: [asterisk-commits] mjordan: branch 12 r426995 - /branches/12/res/res_stasis.c   Corey Farrell
• [asterisk-dev] [Code Review] 4128: func_jitterbuffer: fix frame leaks   Corey Farrell
• [asterisk-dev] [Code Review] 4138: main/app.c, app_voicemail: Fix ast_writestream leaks   Corey Farrell
• [asterisk-dev] [Code Review] 4142: func_talkdetect: Fix stasis message leak in audiohook callback   Corey Farrell
• [asterisk-dev] [Code Review] 4143: testsuite: Close HTTP connections when test is complete to prevent reported leaks in tests/phoneprov/res_phoneprov and tests/manager/config   Corey Farrell
• [asterisk-dev] [Code Review] 4144: testsuite: Sort tests by name after loading   Corey Farrell
• [asterisk-dev] [Code Review] 4114: Prevent stringfields from accumulating unused memory   Corey Farrell
• [asterisk-dev] [Code Review] 4114: Prevent stringfields from accumulating unused memory   Corey Farrell
• [asterisk-dev] [Code Review] 4114: Prevent stringfields from accumulating unused memory   Corey Farrell
• [asterisk-dev] [Code Review] 4114: Prevent stringfields from accumulating unused memory   Corey Farrell
• [asterisk-dev] [Code Review] 4114: Prevent stringfields from accumulating unused memory   Corey Farrell
• [asterisk-dev] [Code Review] 4140: res_http_websockets: Module reference decrease below zero   Corey Farrell
• [asterisk-dev] [Code Review] 4142: func_talkdetect: Fix stasis message leak in audiohook callback   Corey Farrell
• [asterisk-dev] [Code Review] 4144: testsuite: Sort tests by name after loading   Corey Farrell
• [asterisk-dev] [Code Review] 4143: testsuite: Close HTTP connections when test is complete to prevent reported leaks in tests/phoneprov/res_phoneprov and tests/manager/config   Corey Farrell
• [asterisk-dev] [Code Review] 4149: main/file.c: fix possible extra ast_module_unref to format modules   Corey Farrell
• [asterisk-dev] [Code Review] 4150: res_hep: fix major leak that occurs when config is missing or enabled=no   Corey Farrell
• [asterisk-dev] [Code Review] 3603: func_jitterbuffer: fix audio failure caused by certain masquerade's   Corey Farrell
• [asterisk-dev] [Code Review] 4114: Prevent stringfields from accumulating unused memory   Corey Farrell
• [asterisk-dev] [Code Review] 4150: res_hep: fix major leak that occurs when config is missing or enabled=no   Corey Farrell
• [asterisk-dev] [Code Review] 4151: Fix compiler error when using ./configure --enable-dev-mode --enable-coverage   Corey Farrell
• [asterisk-dev] [Code Review] 4152: chan_console: Fix reference leaks to pvt   Corey Farrell
• [asterisk-dev] [Code Review] 4152: chan_console: Fix reference leaks to pvt   Corey Farrell
• [asterisk-dev] [Code Review] 4151: Fix compiler error when using ./configure --enable-dev-mode --enable-coverage   Corey Farrell
• [asterisk-dev] [Code Review] 4153: manager: HTTP connections leak references   Corey Farrell
• [asterisk-dev] [Code Review] 4149: main/file.c: fix possible extra ast_module_unref to format modules   Corey Farrell
• [asterisk-dev] [Code Review] 4151: Fix compiler error when using ./configure --enable-dev-mode --enable-coverage   Corey Farrell
• [asterisk-dev] [Code Review] 4152: chan_console: Fix reference leaks to pvt   Corey Farrell
• [asterisk-dev] [Code Review] 4151: Fix compiler error when using ./configure --enable-dev-mode --enable-coverage   Corey Farrell
• [asterisk-dev] [Code Review] 4153: manager: HTTP connections leak references   Corey Farrell
• [asterisk-dev] [Code Review] 4151: Fix compiler error when using ./configure --enable-dev-mode --enable-coverage   Corey Farrell
• [asterisk-dev] [Code Review] 4159: chan_sip: Add missing braces on if, else, while and for   Corey Farrell
• [asterisk-dev] [Code Review] 4160: chan_sip: Fix theoretical leak of p->refer   Corey Farrell
• [asterisk-dev] [Code Review] 4159: chan_sip: Add missing braces on if, else, while and for   Corey Farrell
• [asterisk-dev] [Code Review] 4160: chan_sip: Fix theoretical leak of p->refer   Corey Farrell
• [asterisk-dev] [Code Review] 4166: testsuite: tests/bridge/bridge_action leaves a channel open   Corey Farrell
• [asterisk-dev] [Code Review] 4170: testsuite: Delete bridges on completion for a bunch of rest_api tests   Corey Farrell
• [asterisk-dev] [Code Review] 4151: Fix compiler error when using ./configure --enable-dev-mode --enable-coverage   Corey Farrell
• [asterisk-dev] [Code Review] 4160: chan_sip: Fix theoretical leak of p->refer   Corey Farrell
• [asterisk-dev] [Code Review] 4170: testsuite: Delete bridges on completion for a bunch of rest_api tests   Corey Farrell
• [asterisk-dev] [Code Review] 4182: core: avoid rasterisk crash due to long identifier   Corey Farrell
• [asterisk-dev] [Code Review] 4166: testsuite: tests/bridge/bridge_action leaves a channel open   Corey Farrell
• [asterisk-dev] [Code Review] 4166: testsuite: tests/bridge/bridge_action leaves a channel open   Corey Farrell
• [asterisk-dev] [Code Review] 4182: core: avoid rasterisk crash due to long identifier   Corey Farrell
• [asterisk-dev] [Code Review] 4186: stringfields: Fix regression from fix for unintentional memory retention and another issue exposed by the fix   Corey Farrell
• [asterisk-dev] [Code Review] 4166: testsuite: tests/bridge/bridge_action leaves a channel open   Corey Farrell
• [asterisk-dev] [Code Review] 4166: testsuite: tests/bridge/bridge_action leaves a channel open   Corey Farrell
• [asterisk-dev] [Code Review] 4186: stringfields: Fix regression from fix for unintentional memory retention and another issue exposed by the fix   Corey Farrell
• [asterisk-dev] [Code Review] 4160: chan_sip: Fix theoretical leak of p->refer   Corey Farrell
• [asterisk-dev] [Code Review] 4182: core: avoid rasterisk crash due to long identifier   Corey Farrell
• [asterisk-dev] [Code Review] 4090: testsuite: add basic valgrind support   Corey Farrell
• [asterisk-dev] [Code Review] 4186: stringfields: Fix regression from fix for unintentional memory retention and another issue exposed by the fix   Corey Farrell
• [asterisk-dev] [Code Review] 4090: testsuite: add basic valgrind support   Corey Farrell
• [asterisk-dev] [Code Review] 4186: stringfields: Fix regression from fix for unintentional memory retention and another issue exposed by the fix   Corey Farrell
• [asterisk-dev] [Code Review] 4186: stringfields: Fix regression from fix for unintentional memory retention caused by ast_string_fields_copy   Corey Farrell
• [asterisk-dev] [Code Review] 4186: stringfields: Fix regression from fix for unintentional memory retention caused by ast_string_fields_copy   Corey Farrell
• [asterisk-dev] [Code Review] 4194: ast_str: Fix improper member access to struct ast_str members.   Corey Farrell
• [asterisk-dev] [Code Review] 4186: stringfields: Fix regression from fix for unintentional memory retention caused by ast_string_fields_copy   Corey Farrell
• [asterisk-dev] [Code Review] 4186: stringfields: Fix regression from fix for unintentional memory retention caused by ast_string_fields_copy   Corey Farrell
• [asterisk-dev] [Code Review] 4147: config: BUG: Restore ability for non-templates to be used as base objects in config.   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] 4182: core: avoid rasterisk crash due to long identifier   Scott Griepentrog
• [asterisk-dev] [Code Review] 4182: core: avoid rasterisk crash due to long identifier   Scott Griepentrog
• [asterisk-dev] [Code Review] 4139: stun: fix size of attribute string to match rfc   Scott Griepentrog
• [asterisk-dev] [Code Review] 4182: core: avoid rasterisk crash due to long identifier   Scott Griepentrog
• [asterisk-dev] [Code Review] 4182: core: avoid rasterisk crash due to long identifier   Scott Griepentrog
• [asterisk-dev] [Code Review] 4090: testsuite: add basic valgrind support   Scott Griepentrog
• [asterisk-dev] [Code Review] 4090: testsuite: add basic valgrind support   Scott Griepentrog
• [asterisk-dev] [Code Review] 4090: testsuite: add basic valgrind support   Scott Griepentrog
• [asterisk-dev] [Code Review] 3496: testsuite: add valgrind support   Scott Griepentrog
• [asterisk-dev] [Code Review] 4090: testsuite: add basic valgrind support   Scott Griepentrog
• [asterisk-dev] [Code Review] 4164: res_pjsip_outbound_registration: stack overflow when using non-default sorcery wizard   Kevin Harwell
• [asterisk-dev] [Code Review] 4164: res_pjsip_outbound_registration: stack overflow when using non-default sorcery wizard   Kevin Harwell
• [asterisk-dev] [Code Review] 4164: res_pjsip_outbound_registration: stack overflow when using non-default sorcery wizard   Kevin Harwell
• [asterisk-dev] [Code Review] 4164: res_pjsip_outbound_registration: 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] 4164: res_pjsip_outbound_registration: stack overflow when using non-default sorcery wizard   Kevin Harwell
• [asterisk-dev] [Code Review] 4168: Testsuite: transferdialattempts test   Kevin Harwell
• [asterisk-dev] [Code Review] 4156: Test Suite: Allow setting snaplen & buffer_size for packet capturing   Kevin Harwell
• [asterisk-dev] [Code Review] 4168: Testsuite: transferdialattempts test   Kevin Harwell
• [asterisk-dev] [Code Review] 4156: Test Suite: Allow setting snaplen & buffer_size for packet capturing   Kevin Harwell
• [asterisk-dev] [Code Review] 4156: Test Suite: Allow setting snaplen & buffer_size for packet capturing   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] 4211: Speed up loopback switches by avoiding unneeded lookups   Birger Harzenetter
• [asterisk-dev] unsubscribe   Peter Hyeroba
• [asterisk-dev] The state of Asterisk (new topic)   Olle E. Johansson
• [asterisk-dev] [Code Review] 4140: res_http_websockets: Module reference decrease below zero   Matt Jordan
• [asterisk-dev] [Code Review] 4142: func_talkdetect: Fix stasis message leak in audiohook callback   Matt Jordan
• [asterisk-dev] [Code Review] 4144: testsuite: Sort tests by name after loading   Matt Jordan
• [asterisk-dev] [Code Review] 4143: testsuite: Close HTTP connections when test is complete to prevent reported leaks in tests/phoneprov/res_phoneprov and tests/manager/config   Matt Jordan
• [asterisk-dev] [Code Review] 4150: res_hep: fix major leak that occurs when config is missing or enabled=no   Matt Jordan
• [asterisk-dev] [Code Review] 4157: bridge_native_rtp: Fix T.38 directmedia fax test by always asking the remote peers to update themselves on native bridge stop   Matt Jordan
• [asterisk-dev] [Code Review] 4158: rtp_engine: Fix crash when endpoints send more RTCP report info blocks then we can handle   Matt Jordan
• [asterisk-dev] [Code Review] 4153: manager: HTTP connections leak references   Matt Jordan
• [asterisk-dev] [Code Review] 4151: Fix compiler error when using ./configure --enable-dev-mode --enable-coverage   Matt Jordan
• [asterisk-dev] [Code Review] 4157: bridge_native_rtp: Fix T.38 directmedia fax test by always asking the remote peers to update themselves on native bridge stop   Matt Jordan
• [asterisk-dev] [Code Review] 4158: rtp_engine: Fix crash when endpoints send more RTCP report info blocks then we can handle   Matt Jordan
• [asterisk-dev] [Code Review] 4109: Documentation: CDR unanswered behavior   Matt Jordan
• [asterisk-dev] [Code Review] 4135: Resolve race condition that can result in ARI apps not being notified of transfers   Matt Jordan
• [asterisk-dev] [Code Review] 4101: Channel Originate/Continue via ARI support for labels in dialplan is incomplete   Matt Jordan
• [asterisk-dev] [Code Review] 4093: Codec Format Is Not Included in the SDP Media Attributes When SLIN48 Codec Is Used   Matt Jordan
• [asterisk-dev] [Code Review] 4167: Allow mis-dialed DTMF-initiated transfers to be re-attempted.   Matt Jordan
• [asterisk-dev] [Code Review] 4184: app_confbridge: Don't delay playing 'you have been kicked' prompts to end_marked users when there are only end_marked users in the conference   Matt Jordan
• [asterisk-dev] [Code Review] 4167: Allow mis-dialed DTMF-initiated transfers to be re-attempted.   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] 4172: Testsuite: tests for res_pjsip_config_wizard   Matt Jordan
• [asterisk-dev] [Code Review] 4145: testsuite: update pretty_print script for recent runtests.py changes.   Matt Jordan
• [asterisk-dev] [Code Review] 4184: app_confbridge: Don't delay playing 'you have been kicked' prompts to end_marked users when there are only end_marked users in the conference   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] 4168: Testsuite: transferdialattempts test   Matt Jordan
• [asterisk-dev] [Code Review] 4187: res_pjsip_refer: Ensure Refer-To is NULL terminated and parse it as a URI   Matt Jordan
• [asterisk-dev] [Code Review] 4187: res_pjsip_refer: Ensure Refer-To is NULL terminated and parse it as a URI   Matt Jordan
• [asterisk-dev] [Code Review] 4187: res_pjsip_refer: Ensure Refer-To is NULL terminated and parse it as a URI   Matt Jordan
• [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] 4145: testsuite: update pretty_print script for recent runtests.py changes.   Matt Jordan
• [asterisk-dev] [Code Review] 4172: Testsuite: tests for res_pjsip_config_wizard   Matt Jordan
• [asterisk-dev] [Code Review] 4186: stringfields: Fix regression from fix for unintentional memory retention caused by ast_string_fields_copy   Matt Jordan
• [asterisk-dev] [Code Review] 4196: bridge_basic: Fix features issues introduced by review 4167   Matt Jordan
• [asterisk-dev] [Code Review] 4196: bridge_basic: Fix features issues introduced by review 4167   Matt Jordan
• [asterisk-dev] [Code Review] 4196: bridge_basic: Fix features issues introduced by review 4167   Matt Jordan
• [asterisk-dev] [Code Review] 4196: bridge_basic: Fix features issues introduced by review 4167   Matt Jordan
• [asterisk-dev] [Code Review] 4196: bridge_basic: Fix features issues introduced by review 4167   Matt Jordan
• [asterisk-dev] [Code Review] 4199: test_channel_feature_hooks.c: Fix unit test for DTMF hooks.   Matt Jordan
• [asterisk-dev] [Code Review] 4201: app_record: prevent stripping 1/4 second audio when channel is hung up to end recording.   Matt Jordan
• [asterisk-dev] [Code Review] 4193: Stasis: allow for subscriptions to dictate message delivery on a threadpool for certain situations   Matt Jordan
• [asterisk-dev] Fwd: [asterisk-commits] mjordan: branch 12 r426995 - /branches/12/res/res_stasis.c   Matthew Jordan
• [asterisk-dev] [asterisk-commits] coreyfarrell: branch 11 r427024 - in /branches/11: ./ apps/ main/   Matthew Jordan
• [asterisk-dev] [asterisk-commits] file: trunk r427112 - in /trunk: channels/ channels/pjsip/ configs/samples/ c...   Matthew Jordan
• [asterisk-dev] Fwd: [asterisk-commits] mjordan: branch 12 r426995 - /branches/12/res/res_stasis.c   Matthew Jordan
• [asterisk-dev] Asterisk 14 - Remote URI Playback   Matthew Jordan
• [asterisk-dev] Numbered placeholders in dialplan's SPRINTF function   Matthew Jordan
• [asterisk-dev] Asterisk 14 - Remote URI Playback   Matthew Jordan
• [asterisk-dev] Asterisk 1.8.32.0-rc1 Created   Matthew Jordan
• [asterisk-dev] func_jitterbuffer handling of masquerades   Matthew Jordan
• [asterisk-dev] Asterisk 14 - Remote URI Playback   Matthew Jordan
• [asterisk-dev] Asterisk 14 - Remote URI Playback   Matthew Jordan
• [asterisk-dev] Asterisk 14 - Remote URI Playback   Matthew Jordan
• [asterisk-dev] Asterisk MoS and RTCP data [pardon the directness and language :-)]   Matthew Jordan
• [asterisk-dev] Process change proposal: allowing new features and improvements into release branches   Matthew Jordan
• [asterisk-dev] Process change proposal: allowing new features and improvements into release branches   Matthew Jordan
• [asterisk-dev] [Code Review] 4145: testsuite: update pretty_print script for recent runtests.py changes.   George Joseph
• [asterisk-dev] [Code Review] 4147: config: BUG: Restore ability for non-templates to be used as base objects in config.   George Joseph
• [asterisk-dev] [Code Review] 4147: config: BUG: Restore ability for non-templates to be used as base objects in config.   George Joseph
• [asterisk-dev] [Code Review] 4145: testsuite: update pretty_print script for recent runtests.py changes.   George Joseph
• [asterisk-dev] [Code Review] 4132: config: Make ast_config_text_file_save and 'dialplan save' escape semicolons in values.   George Joseph
• [asterisk-dev] [Code Review] 4134: testsute: Add 'add-relative-to-search-path' handler to TestModuleLoader   George Joseph
• [asterisk-dev] [Code Review] 4136: testsuite: Add escaped semicolons tests for UpdateConfig and dialplan save   George Joseph
• [asterisk-dev] [Code Review] 4165: 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] 4165: res_pjsip_config_wizard: Allow streamlined configuration of common pjsip scenarios.   George Joseph
• [asterisk-dev] [Code Review] 4183: res_pjsip_phoneprov_provider: bug: call ast_sorcery_apply_config so config can come from realtime.   George Joseph
• [asterisk-dev] [Code Review] 4183: res_pjsip_phoneprov_provider: bug: call ast_sorcery_apply_config so config can come from realtime.   George Joseph
• [asterisk-dev] [Code Review] 4165: res_pjsip_config_wizard: Allow streamlined configuration of common pjsip scenarios.   George Joseph
• [asterisk-dev] [Code Review] 4183: res_pjsip_phoneprov_provider: bug: call ast_sorcery_apply_config so config can come from realtime.   George Joseph
• [asterisk-dev] [Code Review] 4183: res_pjsip_phoneprov_provider: bug: call ast_sorcery_apply_config so config can come from realtime.   George Joseph
• [asterisk-dev] [Code Review] 4185: sorcery: Make ast_sorcery_is_object_field_registered handle field names that are regexes.   George Joseph
• [asterisk-dev] [Code Review] 4145: testsuite: update pretty_print script for recent runtests.py changes.   George Joseph
• [asterisk-dev] [Code Review] 4145: testsuite: update pretty_print script for recent runtests.py changes.   George Joseph
• [asterisk-dev] [Code Review] 4145: testsuite: update pretty_print script for recent runtests.py changes.   George Joseph
• [asterisk-dev] [Code Review] 4145: testsuite: update pretty_print script for recent runtests.py changes.   George Joseph
• [asterisk-dev] [Code Review] 4145: testsuite: update pretty_print script for recent runtests.py changes.   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] 4090: testsuite: add basic valgrind support   George Joseph
• [asterisk-dev] [Code Review] 4185: sorcery: Make ast_sorcery_is_object_field_registered handle field names that are regexes.   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] 4198: testsuite: Add --cleanup option to runtests.py   George Joseph
• [asterisk-dev] [Code Review] 4145: testsuite: update pretty_print script for recent runtests.py changes.   George Joseph
• [asterisk-dev] [Code Review] 4185: sorcery: Make ast_sorcery_is_object_field_registered handle field names that are regexes.   George Joseph
• [asterisk-dev] [Code Review] 4185: sorcery: Make ast_sorcery_is_object_field_registered handle field names that are regexes.   George Joseph
• [asterisk-dev] [Code Review] 4185: sorcery: Make ast_sorcery_is_object_field_registered handle field names that are regexes.   George Joseph
• [asterisk-dev] [Code Review] 4198: testsuite: Add --cleanup option to runtests.py   George Joseph
• [asterisk-dev] [Code Review] 4198: testsuite: Add --cleanup option to runtests.py   George Joseph
• [asterisk-dev] [Code Review] 4185: sorcery: Make ast_sorcery_is_object_field_registered handle field names that are regexes.   George Joseph
• [asterisk-dev] [Code Review] 4190: res_pjsip_config_wizard: Allow streamlined configuration of common pjsip scenarios   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] 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] 4215: sorcery: Add additional observer capabilities.   George Joseph
• [asterisk-dev] [Code Review] 4215: sorcery: Add additional observer capabilities.   George Joseph
• [asterisk-dev] [Code Review] 3867: [chan_sip] Default DTLS settings to use if peer misses own settings   Michael K.
• [asterisk-dev] [Code Review] 3867: [chan_sip] Default DTLS settings to use if peer misses own settings   Michael K.
• [asterisk-dev] Notes from setting up SIP+TLS/RTP+DTLS   Ben Klang
• [asterisk-dev] Notes from setting up SIP+TLS/RTP+DTLS   Ben Klang
• [asterisk-dev] Notes from setting up SIP+TLS/RTP+DTLS   Ben Klang
• [asterisk-dev] Asterisk consuming lots of memory in frame.c cache   James Lamanna
• [asterisk-dev] Asterisk consuming lots of memory in frame.c cache   James Lamanna
• [asterisk-dev] [Code Review] 4204: Failure showing codecs via 'core show channeltype <tech>'   Brad Latus
• [asterisk-dev] [Code Review] 3980: cel_odbc: Add microseconds precision in the eventtime column   Etienne Lessard
• [asterisk-dev] The state of Asterisk (new topic)   Leif Madsen
• [asterisk-dev] Process change proposal: allowing new features and improvements into release branches   Leif Madsen
• [asterisk-dev] Process change proposal: allowing new features and improvements into release branches   Leif Madsen
• [asterisk-dev] FW: Asterisk Developers Mailing List   Leif Madsen
• [asterisk-dev] [Code Review] 4135: Resolve race condition that can result in ARI apps not being notified of transfers   Mark Michelson
• [asterisk-dev] [Code Review] 4148: res_pjsip_multihomed: Provide logging during startup for an indication of what is going on   Mark Michelson
• [asterisk-dev] [Code Review] 4145: testsuite: update pretty_print script for recent runtests.py changes.   Mark Michelson
• [asterisk-dev] [Code Review] 4146: res_pjsip: If an endpoint is associated with the dialog place it on the messag early   Mark Michelson
• [asterisk-dev] [Code Review] 4123: app_agent_pool: Make agent alert interruptable by DTMF.   Mark Michelson
• [asterisk-dev] [Code Review] 4123: app_agent_pool: Make agent alert interruptable by DTMF.   Mark Michelson
• [asterisk-dev] [Code Review] 4135: Resolve race condition that can result in ARI apps not being notified of transfers   Mark Michelson
• [asterisk-dev] [Code Review] 4135: Resolve race condition that can result in ARI apps not being notified of transfers   Mark Michelson
• [asterisk-dev] [Code Review] 4164: res_pjsip_outbound_registration: stack overflow when using non-default sorcery wizard   Mark Michelson
• [asterisk-dev] [Code Review] 4162: pbx: Fix crash in off-nominal when add_priority encounters a failure.   Mark Michelson
• [asterisk-dev] [Code Review] 4160: chan_sip: Fix theoretical leak of p->refer   Mark Michelson
• [asterisk-dev] [Code Review] 4159: chan_sip: Add missing braces on if, else, while and for   Mark Michelson
• [asterisk-dev] Should internal timer implementing session timers be stopped and restarted on response to re-invite?   Mark Michelson
• [asterisk-dev] [Code Review] 4160: chan_sip: Fix theoretical leak of p->refer   Mark Michelson
• [asterisk-dev] [Code Review] 3992: res_pjsip_sdp_rtp: Add optimistic SRTP support.   Mark Michelson
• [asterisk-dev] [Code Review] 4167: Allow mis-dialed DTMF-initiated transfers to be re-attempted.   Mark Michelson
• [asterisk-dev] [Code Review] 4168: Testsuite: transferdialattempts test   Mark Michelson
• [asterisk-dev] [Code Review] 4099: Optimistic SRTP Tests.   Mark Michelson
• [asterisk-dev] [Code Review] 4164: res_pjsip_outbound_registration: stack overflow when using non-default sorcery wizard   Mark Michelson
• [asterisk-dev] [Code Review] 4163: Stasis: Fix StasisEnd message ordering   Mark Michelson
• [asterisk-dev] [Code Review] 4167: Allow mis-dialed DTMF-initiated transfers to be re-attempted.   Mark Michelson
• [asterisk-dev] [Code Review] 4174: Fix race condition where identical SIP requests are processed by multiple threads.   Mark Michelson
• [asterisk-dev] [Code Review] 4175: Fix race condition where identical SIP requests are processed by multiple threads (Asterisk 13)   Mark Michelson
• [asterisk-dev] [Code Review] 4163: Stasis: Fix StasisEnd message ordering   Mark Michelson
• [asterisk-dev] [Code Review] 4160: chan_sip: Fix theoretical leak of p->refer   Mark Michelson
• [asterisk-dev] [Code Review] 4166: testsuite: tests/bridge/bridge_action leaves a channel open   Mark Michelson
• [asterisk-dev] [Code Review] 4177: app_confbridge: Play 'leader has left' sound file even when musiconhold is enabled   Mark Michelson
• [asterisk-dev] Adding Required header to OK response will fail.   Mark Michelson
• [asterisk-dev] [Code Review] 4182: core: avoid rasterisk crash due to long identifier   Mark Michelson
• [asterisk-dev] [Code Review] 4166: testsuite: tests/bridge/bridge_action leaves a channel open   Mark Michelson
• [asterisk-dev] [Code Review] 4175: Fix race condition where identical SIP requests are processed by multiple threads (Asterisk 13)   Mark Michelson
• [asterisk-dev] [Code Review] 4174: Fix race condition where identical SIP requests are processed by multiple threads.   Mark Michelson
• [asterisk-dev] [Code Review] 4135: Resolve race condition that can result in ARI apps not being notified of transfers   Mark Michelson
• [asterisk-dev] [Code Review] 4167: Allow mis-dialed DTMF-initiated transfers to be re-attempted.   Mark Michelson
• [asterisk-dev] [Code Review] 4168: Testsuite: transferdialattempts test   Mark Michelson
• [asterisk-dev] [Code Review] 4168: Testsuite: transferdialattempts test   Mark Michelson
• [asterisk-dev] [Code Review] 4167: Allow mis-dialed DTMF-initiated transfers to be re-attempted.   Mark Michelson
• [asterisk-dev] [Code Review] 4168: Testsuite: transferdialattempts test   Mark Michelson
• [asterisk-dev] [Code Review] 4192: Update channel Stasis topics during a masquerade   Mark Michelson
• [asterisk-dev] [Code Review] 4192: Update channel Stasis topics during a masquerade   Mark Michelson
• [asterisk-dev] ringinuse seems to include busy; also performance of available queue member search   Mark Michelson
• [asterisk-dev] [Code Review] 4196: bridge_basic: Fix features issues introduced by review 4167   Mark Michelson
• [asterisk-dev] [Code Review] 4185: sorcery: Make ast_sorcery_is_object_field_registered handle field names that are regexes.   Mark Michelson
• [asterisk-dev] [Code Review] 4200: manager: Fix could not extend string messages.   Mark Michelson
• [asterisk-dev] [Code Review] 4193: Stasis: allow for subscriptions to dictate message delivery on a threadpool for certain situations   Mark Michelson
• [asterisk-dev] [Code Review] 4185: sorcery: Make ast_sorcery_is_object_field_registered handle field names that are regexes.   Mark Michelson
• [asterisk-dev] [Code Review] 4168: Testsuite: transferdialattempts test   Mark Michelson
• [asterisk-dev] [Code Review] 4198: testsuite: Add --cleanup option to runtests.py   Mark Michelson
• [asterisk-dev] Numbered placeholders in dialplan's SPRINTF function   Olivier
• [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] 4109: Documentation: CDR unanswered behavior   Jonathan Rose
• [asterisk-dev] [Code Review] 4120: res_pjsip_acl: contact ACL permits are being interpreted incorrectly   Jonathan Rose
• [asterisk-dev] Asterisk MoS and RTCP data [pardon the directness and language :-)]   Nir Simionovich
• [asterisk-dev] [Code Review] 4201: Prevent app_record stripping 1/4 second audio when channel is hung up to end recording.   Ben Smithurst
• [asterisk-dev] [Code Review] 4201: Prevent app_record stripping 1/4 second audio when channel is hung up to end recording.   Ben Smithurst
• [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] [Code Review] 4188: segfault when playing back voicemail under high concurrency with an IMAP backend   Ben Smithurst
• [asterisk-dev] Asterisk 1.8.32.0-rc1 Now Available   Asterisk Development Team
• [asterisk-dev] Asterisk 11.14.0-rc1 Now Available   Asterisk Development Team
• [asterisk-dev] Asterisk 12.7.0-rc1 Now Available   Asterisk Development Team
• [asterisk-dev] Asterisk 1.8.32.0-rc2 Now Available   Asterisk Development Team
• [asterisk-dev] Asterisk 11.14.0-rc2 Now Available   Asterisk Development Team
• [asterisk-dev] Asterisk 12.7.0-rc2 Now Available   Asterisk Development Team
• [asterisk-dev] Asterisk 1.8.32.0 Now Available   Asterisk Development Team
• [asterisk-dev] Asterisk 11.14.0 Now Available   Asterisk Development Team
• [asterisk-dev] Asterisk 12.7.0 Now Available   Asterisk Development Team
• [asterisk-dev] Asterisk 1.8.28-cert3, 1.8.32.1, 11.6-cert8, 11.14.1, 12.7.1, 13.0.1 Now Available (Security Release)   Asterisk Development Team
• [asterisk-dev] AST-2014-012: Mixed IP address families in access control lists may permit unwanted traffic.   Asterisk Security Team
• [asterisk-dev] AST-2014-013: PJSIP ACLs are not loaded on startup   Asterisk Security Team
• [asterisk-dev] AST-2014-014: High call load may result in hung channels in ConfBridge.   Asterisk Security Team
• [asterisk-dev] AST-2014-015: Remote Crash Vulnerability in PJSIP channel driver   Asterisk Security Team
• [asterisk-dev] AST-2014-016: Remote Crash Vulnerability in PJSIP channel driver   Asterisk Security Team
• [asterisk-dev] AST-2014-017: <font size="3" style="font-size: 12pt">Permission escalation through ConfBridge actions/dialplan functions</font>   Asterisk Security Team
• [asterisk-dev] AST-2014-018: AMI permission escalation through DB dialplan function   Asterisk Security Team
• [asterisk-dev] Planned maintenance for community services, Tuesday, 25th of November 2014   Digium's Asterisk Development Team
• [asterisk-dev] Should internal timer implementing session timers be stopped and restarted on response to re-invite?   Dave WOOLLEY
• [asterisk-dev] Should internal timer implementing session timers be stopped and restarted on response to re-invite?   Dave WOOLLEY
• [asterisk-dev] session-refresher ignored if UAC supports but does not request timers.   Dave WOOLLEY
• [asterisk-dev] Adding Required header to OK response will fail.   Dave WOOLLEY
• [asterisk-dev] ringinuse seems to include busy; also performance of available queue member search   Dave WOOLLEY
• [asterisk-dev] Asterisk 14 - Remote URI Playback   BJ Weschke
• [asterisk-dev] Asterisk 14 - Remote URI Playback   BJ Weschke
• [asterisk-dev] Asterisk 14 - Remote URI Playback   BJ Weschke
• [asterisk-dev] Process change proposal: allowing new features and improvements into release branches   BJ Weschke
• [asterisk-dev] Asterisk 14 - Remote URI Playback   Johan Wilfer
• [asterisk-dev] Asterisk 14 - Remote URI Playback   Johan Wilfer
• [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] 4156: Test Suite: Allow setting snaplen & buffer_size for packet capturing   jbigelow
• [asterisk-dev] [Code Review] 4156: Test Suite: Allow setting snaplen & buffer_size for packet capturing   jbigelow
• [asterisk-dev] [Code Review] 4156: Test Suite: Allow setting snaplen & buffer_size for packet capturing   jbigelow
• [asterisk-dev] FW: Asterisk Developers Mailing List   marcotasto
• [asterisk-dev] [Code Review] 4099: Optimistic SRTP Tests.   opticron
• [asterisk-dev] [Code Review] 4132: config: Make ast_config_text_file_save and 'dialplan save' escape semicolons in values.   opticron
• [asterisk-dev] [Code Review] 4155: PJSIP: Allow contact rewriting to fall back for reliable transports   opticron
• [asterisk-dev] [Code Review] 4135: Resolve race condition that can result in ARI apps not being notified of transfers   opticron
• [asterisk-dev] [Code Review] 4163: Stasis: Fix StasisEnd message ordering   opticron
• [asterisk-dev] [Code Review] 4155: PJSIP: Allow contact rewriting to fall back for reliable transports   opticron
• [asterisk-dev] [Code Review] 4163: Stasis: Fix StasisEnd message ordering   opticron
• [asterisk-dev] [Code Review] 4163: Stasis: Fix StasisEnd message ordering   opticron
• [asterisk-dev] [Code Review] 4155: PJSIP: Allow contact rewriting to fall back for reliable transports   opticron
• [asterisk-dev] [Code Review] 4155: PJSIP: Allow contact rewriting to fall back for reliable transports   opticron
• [asterisk-dev] [Code Review] 3603: func_jitterbuffer: fix audio failure caused by certain masquerade's   opticron
• [asterisk-dev] [Code Review] 4213: Stasis: Fix StasisStart and StasisEnd ordering and missing events   opticron
• [asterisk-dev] [Code Review] 4114: Prevent stringfields from accumulating unused memory   rmudgett
• [asterisk-dev] [Code Review] 4114: Prevent stringfields from accumulating unused memory   rmudgett
• [asterisk-dev] [Code Review] 4123: app_agent_pool: Make agent alert interruptable by DTMF.   rmudgett
• [asterisk-dev] [Code Review] 4137: res_pjsip: Add disable_tcp_switch option.   rmudgett
• [asterisk-dev] [Code Review] 4114: Prevent stringfields from accumulating unused memory   rmudgett
• [asterisk-dev] [Code Review] 4123: app_agent_pool: Make agent alert interruptable by DTMF.   rmudgett
• [asterisk-dev] [Code Review] 4123: app_agent_pool: Make agent alert interruptable by DTMF.   rmudgett
• [asterisk-dev] [Code Review] 4123: app_agent_pool: Make agent alert interruptable by DTMF.   rmudgett
• [asterisk-dev] [Code Review] 4184: app_confbridge: Don't delay playing 'you have been kicked' prompts to end_marked users when there are only end_marked users in the conference   rmudgett
• [asterisk-dev] [Code Review] 4186: stringfields: Fix regression from fix for unintentional memory retention and another issue exposed by the fix   rmudgett
• [asterisk-dev] [Code Review] 4186: stringfields: Fix regression from fix for unintentional memory retention caused by ast_string_fields_copy   rmudgett
• [asterisk-dev] [Code Review] 4194: ast_str: Fix improper member access to struct ast_str members.   rmudgett
• [asterisk-dev] [Code Review] 4194: ast_str: Fix improper member access to struct ast_str members.   rmudgett
• [asterisk-dev] [Code Review] 4196: bridge_basic: Fix features issues introduced by review 4167   rmudgett
• [asterisk-dev] [Code Review] 4196: bridge_basic: Fix features issues introduced by review 4167   rmudgett
• [asterisk-dev] [Code Review] 4199: test_channel_feature_hooks.c: Fix unit test for DTMF hooks.   rmudgett
• [asterisk-dev] [Code Review] 4200: manager: Fix could not extend string messages.   rmudgett
• [asterisk-dev] [Code Review] 4200: manager: Fix could not extend string messages.   rmudgett
• [asterisk-dev] [Code Review] 4200: manager: Fix could not extend string messages.   rmudgett
• [asterisk-dev] [Code Review] 4199: test_channel_feature_hooks.c: Fix unit test for DTMF hooks.   rmudgett
• [asterisk-dev] [Code Review] 4152: chan_console: Fix reference leaks to pvt   wdoekes
• [asterisk-dev] [Code Review] 4151: Fix compiler error when using ./configure --enable-dev-mode --enable-coverage   wdoekes
• [asterisk-dev] [Code Review] 4151: Fix compiler error when using ./configure --enable-dev-mode --enable-coverage   wdoekes
• [asterisk-dev] [Code Review] 4186: stringfields: Fix regression from fix for unintentional memory retention caused by ast_string_fields_copy   wdoekes

Last message date: Fri Nov 28 13:18:46 CST 2014
Archived on: Fri Nov 28 13:17:24 CST 2014

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