February 2014 Archives by author

• Messages sorted by: [ thread ] [ subject ] [ date ]
• More info on this list...

Starting: Sat Feb 1 10:24:12 CST 2014
Ending: Fri Feb 28 15:27:54 CST 2014
Messages: 641

• [asterisk-dev] [Code Review] 3225: Asterisk doesn't support the dynamic payload change in rtp mapping in the 200 OK response   Nitesh Bansal
• [asterisk-dev] Asterisk WebRTC one way audio issue   Nitesh Bansal
• [asterisk-dev] [Code Review] 3225: Asterisk doesn't support the dynamic payload change in rtp mapping in the 200 OK response   Nitesh Bansal
• [asterisk-dev] [Code Review] 3225: Asterisk doesn't support the dynamic payload change in rtp mapping in the 200 OK response   Nitesh Bansal
• [asterisk-dev] [Code Review] 3173: chan_sip refactor - sip_route   Paul Belanger
• [asterisk-dev] [Code Review] 3173: chan_sip refactor - sip_route   Paul Belanger
• [asterisk-dev] Asterisk 12.1.0-rc1 Now Available   Paul Belanger
• [asterisk-dev] [Code Review] 3250: chan_sip: Add incoming tel: uri support (rfc3966)   Paul Belanger
• [asterisk-dev] [Code Review] 2904: More consistent ARI error messages   Paul Belanger
• [asterisk-dev] [Code Review] 3279: Iterate through logger.conf [general] section   Paul Belanger
• [asterisk-dev] [Code Review] 3279: Iterate through logger.conf [general] section   Paul Belanger
• [asterisk-dev] [Code Review] 3279: Iterate through logger.conf [general] section   Paul Belanger
• [asterisk-dev] [Code Review] 3279: Iterate through logger.conf [general] section   Paul Belanger
• [asterisk-dev] CentOS packaging   Paul Belanger
• [asterisk-dev] [Code Review] 3278: ari: create bridges with specified unique id   Paul Belanger
• [asterisk-dev] [Code Review] 3278: ari: create bridges with specified unique id   Paul Belanger
• [asterisk-dev] [Code Review] 3279: Iterate through logger.conf [general] section   Russell Bryant
• [asterisk-dev] security breach via call-limit/groupcount   Marek Cervenka
• [asterisk-dev] [Code Review] 3192: chan_dahdi: handle DAHDI_EVENT_REMOVED on a pri D-Channel   Tzafrir Cohen
• [asterisk-dev] [Code Review] 3158: indications.conf: fix post-stutter dialtone for in, mx and ph, extra missing stutter   Tzafrir Cohen
• [asterisk-dev] [Code Review] 3110: live_ast: run wrapped programs with exec   Tzafrir Cohen
• [asterisk-dev] [Code Review] 3158: indications.conf: fix post-stutter dialtone for in, mx and ph, extra missing stutter   Tzafrir Cohen
• [asterisk-dev] [Code Review] 3192: chan_dahdi: handle DAHDI_EVENT_REMOVED on a pri D-Channel   Tzafrir Cohen
• [asterisk-dev] [Code Review] 3192: chan_dahdi: handle DAHDI_EVENT_REMOVED on a pri D-Channel   Tzafrir Cohen
• [asterisk-dev] [Code Review] 3192: chan_dahdi: handle DAHDI_EVENT_REMOVED on a pri D-Channel   Tzafrir Cohen
• [asterisk-dev] [Code Review] 3202: install_subst: helper script for installing with path substitution   Tzafrir Cohen
• [asterisk-dev] [Code Review] 3202: install_subst: helper script for installing with path substitution   Tzafrir Cohen
• [asterisk-dev] [Code Review] 3202: install_subst: helper script for installing with path substitution   Tzafrir Cohen
• [asterisk-dev] [Code Review] 3202: install_subst: helper script for installing with path substitution   Tzafrir Cohen
• [asterisk-dev] [Code Review] 3202: install_subst: helper script for installing with path substitution   Tzafrir Cohen
• [asterisk-dev] [Code Review] 3062: a systemd service   Tzafrir Cohen
• [asterisk-dev] [Code Review] 3062: a systemd service   Tzafrir Cohen
• [asterisk-dev] [Code Review] 3062: a systemd service   Tzafrir Cohen
• [asterisk-dev] [Code Review] 3202: install_subst: helper script for installing with path substitution   Tzafrir Cohen
• [asterisk-dev] [Code Review] 3062: a systemd service   Tzafrir Cohen
• [asterisk-dev] [Code Review] 3062: a systemd service   Tzafrir Cohen
• [asterisk-dev] [Code Review] 3202: install_subst: helper script for installing with path substitution   Tzafrir Cohen
• [asterisk-dev] CentOS packaging   Tzafrir Cohen
• [asterisk-dev] CentOS packaging   Tzafrir Cohen
• [asterisk-dev] CentOS packaging   Tzafrir Cohen
• [asterisk-dev] CentOS packaging   Tzafrir Cohen
• [asterisk-dev] CentOS packaging   Tzafrir Cohen
• [asterisk-dev] Installing Asterisk in Mac OS X Mavericks   Tzafrir Cohen
• [asterisk-dev] [Code Review] 3280: Makefile: replace -O6 with -O3   Tzafrir Cohen
• [asterisk-dev] [Code Review] 3120: res_ari: Provide transfer messages and enable transfers   Joshua Colp
• [asterisk-dev] [Code Review] 3145: res_ari: Add event tests for blind and attended transfers   Joshua Colp
• [asterisk-dev] [Code Review] 3093: res_pjsip_dialog_info: Add dialog-info+xml presence functionality.   Joshua Colp
• [asterisk-dev] [Code Review] 3175: timing: Improve performance for most timing implementations   Joshua Colp
• [asterisk-dev] [Code Review] 3178: media_formats: Moving of existing code around, implementation, and unit tests   Joshua Colp
• [asterisk-dev] [Code Review] 3175: timing: Improve performance for most timing implementations   Joshua Colp
• [asterisk-dev] [Code Review] 3184: Create sorcery instance registry   Joshua Colp
• [asterisk-dev] [Code Review] 3175: timing: Improve performance for most timing implementations   Joshua Colp
• [asterisk-dev] [Code Review] 3183: ARI: pass channel variables into originate as opposed to assigning after originate   Joshua Colp
• [asterisk-dev] [Code Review] 3184: Create sorcery instance registry   Joshua Colp
• [asterisk-dev] [Code Review] 3180: Documenation: Configuration section naming in pjsip.conf.sample needs a little clarification   Joshua Colp
• [asterisk-dev] [Code Review] 3183: ARI: pass channel variables into originate as opposed to assigning after originate   Joshua Colp
• [asterisk-dev] [Code Review] 3186: AMI Security Events: document the events; add optional IEs to the events   Joshua Colp
• [asterisk-dev] [Code Review] 3136: cli: pjsip show endpoint <endpoint> shows allow/disallow codecs the same   Joshua Colp
• [asterisk-dev] [Code Review] 3184: Create sorcery instance registry   Joshua Colp
• [asterisk-dev] [Code Review] 3136: cli: pjsip show endpoint <endpoint> shows allow/disallow codecs the same - OPTION 1   Joshua Colp
• [asterisk-dev] [Code Review] 3136: cli: pjsip show endpoint <endpoint> shows allow/disallow codecs the same - OPTION 1   Joshua Colp
• [asterisk-dev] [Code Review] 3193: cli: pjsip show endpoint <endpoint> shows allow/disallow codecs the same - OPTION 2   Joshua Colp
• [asterisk-dev] [Code Review] 3184: Create sorcery instance registry   Joshua Colp
• [asterisk-dev] [Code Review] 3184: Create sorcery instance registry   Joshua Colp
• [asterisk-dev] [Code Review] 3184: Create sorcery instance registry   Joshua Colp
• [asterisk-dev] [Code Review] 3178: media_formats: Moving of existing code around, implementation, and unit tests   Joshua Colp
• [asterisk-dev] [Code Review] 3178: media_formats: Moving of existing code around, implementation, and unit tests   Joshua Colp
• [asterisk-dev] [Code Review] 3178: media_formats: Moving of existing code around, implementation, and unit tests   Joshua Colp
• [asterisk-dev] [Code Review] 3178: media_formats: Moving of existing code around, implementation, and unit tests   Joshua Colp
• [asterisk-dev] [Code Review] 3175: timing: Improve performance for most timing implementations   Joshua Colp
• [asterisk-dev] [Code Review] 3199: scheduler: Remove hashtab usage.   Joshua Colp
• [asterisk-dev] [Code Review] 3199: scheduler: Remove hashtab usage.   Joshua Colp
• [asterisk-dev] [Code Review] 3199: scheduler: Remove hashtab usage.   Joshua Colp
• [asterisk-dev] Lightweight keepalive for websockets?   Joshua Colp
• [asterisk-dev] AMI and Sorcery   Joshua Colp
• [asterisk-dev] [Code Review] 3212: Makefile: fix so "make main" works without compiling deps (unless needed)   Joshua Colp
• [asterisk-dev] [Code Review] 3234: media_formats: Initial channel driver conversion and application conversion   Joshua Colp
• [asterisk-dev] [Code Review] 3244: res_pjsip_sdp_rtp: Apply packetization rules on inbound SDP handling   Joshua Colp
• [asterisk-dev] [Code Review] 3251: res_bucket_sounds: Add 'sounds' URI scheme implementation   Joshua Colp
• [asterisk-dev] [Code Review] 3207: HEP: Add a Homer Encapsulation Protocol (HEP) v3 capture agent module and a packet logger for PJSIP   Joshua Colp
• [asterisk-dev] [Code Review] 3102: res_pjsip_multihomed: Add multihomed support   Joshua Colp
• [asterisk-dev] [Code Review] 3256: res_rtp_asterisk: Undo regression from ASTERISK-23213 while working around PJNATH assertion abort from ASTERISK-22911   Joshua Colp
• [asterisk-dev] [Code Review] 3178: media_formats: Moving of existing code around, implementation, and unit tests   Joshua Colp
• [asterisk-dev] [Code Review] 3234: media_formats: Initial channel driver conversion and application conversion   Joshua Colp
• [asterisk-dev] [Code Review] 3265: media_formats: Move app_fax over   Joshua Colp
• [asterisk-dev] [Code Review] 3266: media_formats: Move codecs over.   Joshua Colp
• [asterisk-dev] [Code Review] 3256: res_rtp_asterisk: Undo regression from ASTERISK-23213 while working around PJNATH assertion abort from ASTERISK-22911   Joshua Colp
• [asterisk-dev] [Code Review] 3245: res_pjsip_send_to_voicemail: transferring to voicemail for digium phones   Joshua Colp
• [asterisk-dev] [Code Review] 3256: res_rtp_asterisk: Undo regression from ASTERISK-23213 while working around PJNATH assertion abort from ASTERISK-22911   Joshua Colp
• [asterisk-dev] [Code Review] 3245: res_pjsip_send_to_voicemail: transferring to voicemail for digium phones   Joshua Colp
• [asterisk-dev] [Code Review] 3267: pjsip: avoid edge case potential crash in answer()   Joshua Colp
• [asterisk-dev] [Code Review] 3256: res_rtp_asterisk: Undo regression from ASTERISK-23213 while working around PJNATH assertion abort from ASTERISK-22911   Joshua Colp
• [asterisk-dev] [Code Review] 3240: res_pjsip_exten_state: Presence for digium phones   Joshua Colp
• [asterisk-dev] [Code Review] 3267: pjsip: avoid edge case potential crash in answer()   Joshua Colp
• [asterisk-dev] [Code Review] 3267: pjsip: avoid edge case potential crash in answer()   Joshua Colp
• [asterisk-dev] [Code Review] 3272: func_audiohookinheritance: Check If Channel Was Specified   Joshua Colp
• [asterisk-dev] [Code Review] 3256: res_rtp_asterisk: Undo regression from ASTERISK-23213 while working around PJNATH assertion abort from ASTERISK-22911   Joshua Colp
• [asterisk-dev] [Code Review] 3256: res_rtp_asterisk: Undo regression from ASTERISK-23213 while working around PJNATH assertion abort from ASTERISK-22911   Joshua Colp
• [asterisk-dev] [Code Review] 3256: res_rtp_asterisk: Undo regression from ASTERISK-23213 while working around PJNATH assertion abort from ASTERISK-22911   Joshua Colp
• [asterisk-dev] [Code Review] 3275: res_rtp_asterisk: Fix the one way audio problems when resuming held calls with ICE   Joshua Colp
• [asterisk-dev] [Code Review] 3272: func_audiohookinheritance: Check If Channel Was Specified   Joshua Colp
• [asterisk-dev] PJSIP transport problem   Joshua Colp
• [asterisk-dev] PJSIP debug question   Joshua Colp
• [asterisk-dev] PJSIP transport problem   Joshua Colp
• [asterisk-dev] PJSIP transport problem   Joshua Colp
• [asterisk-dev] Asterisk 13: Media improvements update   Saúl Ibarra Corretgé
• [asterisk-dev] CentOS packaging   Sean Darcy
• [asterisk-dev] CentOS packaging   Sean Darcy
• [asterisk-dev] Proposal for PJSIP TOS/COS and DSCP   Malcolm Davenport
• [asterisk-dev] Asterisk for GSM core network (MSC)   Frederic Van Espen
• [asterisk-dev] Asterisk for GSM core network (MSC)   Frederic Van Espen
• [asterisk-dev] [Code Review] 3172: testsuite: chan_sip Record-Route test   Corey Farrell
• [asterisk-dev] [Code Review] 3173: chan_sip refactor - sip_route   Corey Farrell
• [asterisk-dev] [Code Review] 3173: chan_sip refactor - sip_route   Corey Farrell
• [asterisk-dev] [Code Review] 3173: chan_sip refactor - sip_route   Corey Farrell
• [asterisk-dev] [Code Review] 3173: chan_sip refactor - sip_route   Corey Farrell
• [asterisk-dev] [Code Review] 3173: chan_sip refactor - sip_route   Corey Farrell
• [asterisk-dev] [Code Review] 3175: timing: Improve performance for most timing implementations   Corey Farrell
• [asterisk-dev] [Code Review] 3175: timing: Improve performance for most timing implementations   Corey Farrell
• [asterisk-dev] [Code Review] 3183: ARI: pass channel variables into originate as opposed to assigning after originate   Corey Farrell
• [asterisk-dev] [Code Review] 3183: ARI: pass channel variables into originate as opposed to assigning after originate   Corey Farrell
• [asterisk-dev] [Code Review] 3183: ARI: pass channel variables into originate as opposed to assigning after originate   Corey Farrell
• [asterisk-dev] [Code Review] 3195: testsuite: fixes for run-local   Corey Farrell
• [asterisk-dev] [Code Review] 3202: install_subst: helper script for installing with path substitution   Corey Farrell
• [asterisk-dev] [Code Review] 3202: install_subst: helper script for installing with path substitution   Corey Farrell
• [asterisk-dev] [Code Review] 3172: testsuite: chan_sip Record-Route test   Corey Farrell
• [asterisk-dev] [Code Review] 3173: chan_sip refactor - sip_route   Corey Farrell
• [asterisk-dev] [Code Review] 3195: testsuite: fixes for run-local   Corey Farrell
• [asterisk-dev] [Code Review] 3209: Crash in ast_format_cmp on shutdown   Corey Farrell
• [asterisk-dev] [Code Review] 3202: install_subst: helper script for installing with path substitution   Corey Farrell
• [asterisk-dev] [Code Review] 3236: chan_sip: do not send empty Route header   Corey Farrell
• [asterisk-dev] [Code Review] 3241: Remove extra defines of AST_PBX_MAX_STACK   Corey Farrell
• [asterisk-dev] [Code Review] 3236: chan_sip: do not send empty Route header   Corey Farrell
• [asterisk-dev] [Code Review] 3241: Remove extra defines of AST_PBX_MAX_STACK   Corey Farrell
• [asterisk-dev] [Code Review] 3249: Testsuite: manager live_dangerously tests are missing a parameter for on_failure   Corey Farrell
• [asterisk-dev] [Code Review] 3250: chan_sip: Add incoming tel: uri support (rfc3966)   Corey Farrell
• [asterisk-dev] [Code Review] 3261: res_fax: Warn that minrate=2400 is not valid for V.27 instead of failing load   Corey Farrell
• [asterisk-dev] [Code Review] 3262: testsuite: Add dependency check for rawsocket access   Corey Farrell
• [asterisk-dev] [Code Review] 3263: Testsuite: Fix originate-cdr-disposition failure reporting   Corey Farrell
• [asterisk-dev] [Code Review] 3247: rtp_engine: lock channel during get_codecs calls to prevent NULL pvt dereference after channel masquerade   Corey Farrell
• [asterisk-dev] [Code Review] 3264: testsuite: manager/acl-login fails on some platforms   Corey Farrell
• [asterisk-dev] [Code Review] 3261: res_fax: Warn that minrate=2400 is not valid for V.27 instead of failing load   Corey Farrell
• [asterisk-dev] [Code Review] 3269: chan_sip: fix deadlock of monlock between unload_module and do_monitor   Corey Farrell
• [asterisk-dev] [Code Review] 3269: chan_sip: fix deadlock of monlock between unload_module and do_monitor   Corey Farrell
• [asterisk-dev] [Code Review] 3262: testsuite: Add dependency check for rawsocket access   Corey Farrell
• [asterisk-dev] [Code Review] 3263: Testsuite: Fix originate-cdr-disposition failure reporting   Corey Farrell
• [asterisk-dev] [Code Review] 3264: testsuite: manager/acl-login fails on some platforms   Corey Farrell
• [asterisk-dev] [Code Review] 3261: res_fax: Warn that minrate=2400 is not valid for V.27 instead of failing load   Corey Farrell
• [asterisk-dev] [Code Review] 3272: func_audiohookinheritance: Check If Channel Was Specified   Corey Farrell
• [asterisk-dev] [Code Review] 3249: Testsuite: manager live_dangerously tests are missing a parameter for on_failure   Corey Farrell
• [asterisk-dev] [Code Review] 3136: cli: pjsip show endpoint <endpoint> shows allow/disallow codecs the same   Scott Griepentrog
• [asterisk-dev] [Code Review] 3136: cli: pjsip show endpoint <endpoint> shows allow/disallow codecs the same   Scott Griepentrog
• [asterisk-dev] [Code Review] 3136: cli: pjsip show endpoint <endpoint> shows allow/disallow codecs the same   Scott Griepentrog
• [asterisk-dev] [Code Review] 3180: Documenation: Configuration section naming in pjsip.conf.sample needs a little clarification   Scott Griepentrog
• [asterisk-dev] [Code Review] 3182: testsuite: LinkedID Propagation test   Scott Griepentrog
• [asterisk-dev] [Code Review] 3182: testsuite: LinkedID Propagation test   Scott Griepentrog
• [asterisk-dev] [Code Review] 3180: Documenation: Configuration section naming in pjsip.conf.sample needs a little clarification   Scott Griepentrog
• [asterisk-dev] [Code Review] 3191: channel uniqueid phase 1: convert string uniqueid values to structure with time   Scott Griepentrog
• [asterisk-dev] [Code Review] 3136: cli: pjsip show endpoint <endpoint> shows allow/disallow codecs the same   Scott Griepentrog
• [asterisk-dev] [Code Review] 3136: cli: pjsip show endpoint <endpoint> shows allow/disallow codecs the same - OPTION 1   Scott Griepentrog
• [asterisk-dev] [Code Review] 3193: cli: pjsip show endpoint <endpoint> shows allow/disallow codecs the same - OPTION 2   Scott Griepentrog
• [asterisk-dev] [Code Review] 3182: testsuite: LinkedID Propagation test   Scott Griepentrog
• [asterisk-dev] [Code Review] 3193: cli: pjsip show endpoint <endpoint> shows allow/disallow codecs the same - OPTION 2   Scott Griepentrog
• [asterisk-dev] [Code Review] 3180: Documenation: Configuration section naming in pjsip.conf.sample needs a little clarification   Scott Griepentrog
• [asterisk-dev] [Code Review] 3198: testsuite: Don't continue if we cannot kill a (root?) running asterisk.   Scott Griepentrog
• [asterisk-dev] Proposal for PJSIP TOS/COS and DSCP   Scott Griepentrog
• [asterisk-dev] Proposal for PJSIP TOS/COS and DSCP   Scott Griepentrog
• [asterisk-dev] [Code Review] 3182: testsuite: LinkedID Propagation test   Scott Griepentrog
• [asterisk-dev] [Code Review] 3136: cli: pjsip show endpoint <endpoint> shows allow/disallow codecs the same - OPTION 1   Scott Griepentrog
• [asterisk-dev] [Code Review] 3208: possible null pointer dereference in format.c   Scott Griepentrog
• [asterisk-dev] [Code Review] 3209: Crash in ast_format_cmp on shutdown   Scott Griepentrog
• [asterisk-dev] [Code Review] 3169: Testsuite: Add basic NAT test   Scott Griepentrog
• [asterisk-dev] [Code Review] 3211: ARI: URI is case sensitive   Scott Griepentrog
• [asterisk-dev] [Code Review] 3213: ast_custom_escalating_function allocation leak   Scott Griepentrog
• [asterisk-dev] [Code Review] 3182: testsuite: LinkedID Propagation test   Scott Griepentrog
• [asterisk-dev] [Code Review] 3191: channel uniqueid phase 1: convert string uniqueid values to structure with time   Scott Griepentrog
• [asterisk-dev] [Code Review] 3191: channel uniqueid phase 1: convert string uniqueid values to structure with time   Scott Griepentrog
• [asterisk-dev] [Code Review] 3208: possible null pointer dereference in format.c   Scott Griepentrog
• [asterisk-dev] [Code Review] 3211: ARI: URI is case sensitive   Scott Griepentrog
• [asterisk-dev] [Code Review] 3182: testsuite: LinkedID Propagation test   Scott Griepentrog
• [asterisk-dev] [Code Review] 3182: testsuite: LinkedID Propagation test   Scott Griepentrog
• [asterisk-dev] [Code Review] 3182: testsuite: LinkedID Propagation test   Scott Griepentrog
• [asterisk-dev] [Code Review] 3191: channel uniqueid phase 1: convert string uniqueid values to structure with time   Scott Griepentrog
• [asterisk-dev] [Code Review] 3191: channel uniqueid phase 1: convert string uniqueid values to structure with time   Scott Griepentrog
• [asterisk-dev] [Code Review] 3191: channel uniqueid phase 1: convert string uniqueid values to structure with time   Scott Griepentrog
• [asterisk-dev] [Code Review] 3191: channel uniqueid phase 1: convert string uniqueid values to structure with time   Scott Griepentrog
• [asterisk-dev] [Code Review] 3191: channel uniqueid phases 1&2: convert string uniqueid values to structure with time, add ami origination & assignedid   Scott Griepentrog
• [asterisk-dev] [Code Review] 3243: testsuite: manager Originate test for assigning UniqueId on channel creation   Scott Griepentrog
• [asterisk-dev] [Code Review] 3242: starpy: add UniqueId parameter to ami.Originate()   Scott Griepentrog
• [asterisk-dev] [Code Review] 3191: channel uniqueid phases 1&2: convert string uniqueid values to structure with time, add ami origination & assignedid   Scott Griepentrog
• [asterisk-dev] [Code Review] 3213: ast_custom_escalating_function allocation leak   Scott Griepentrog
• [asterisk-dev] [Code Review] 3258: testsuite: eliminate sipp zombie   Scott Griepentrog
• [asterisk-dev] [Code Review] 3258: testsuite: eliminate sipp zombie   Scott Griepentrog
• [asterisk-dev] [Code Review] 3258: testsuite: eliminate sipp zombie   Scott Griepentrog
• [asterisk-dev] [Code Review] 3132: Test for allow=all sdp issue   Scott Griepentrog
• [asterisk-dev] [Code Review] 3267: pjsip: avoid edge case potential crash in answer()   Scott Griepentrog
• [asterisk-dev] [Code Review] 3242: starpy: add UniqueId parameter to ami.Originate()   Scott Griepentrog
• [asterisk-dev] [Code Review] 3243: testsuite: manager Originate test for assigning UniqueId on channel creation   Scott Griepentrog
• [asterisk-dev] [Code Review] 3191: channel uniqueid phases 1&2: convert string uniqueid values to structure with time, add ami origination & assignedid   Scott Griepentrog
• [asterisk-dev] [Code Review] 3243: testsuite: manager Originate test for assigning UniqueId on channel creation   Scott Griepentrog
• [asterisk-dev] [Code Review] 3193: cli: pjsip show endpoint <endpoint> shows allow/disallow codecs the same - OPTION 2   Scott Griepentrog
• [asterisk-dev] [Code Review] 3243: testsuite: manager Originate test for assigning UniqueId on channel creation   Scott Griepentrog
• [asterisk-dev] [Code Review] 3191: channel uniqueid phases 1&2: convert string uniqueid values to structure with time, add ami origination & assignedid   Scott Griepentrog
• [asterisk-dev] [Code Review] 3267: pjsip: avoid edge case potential crash in answer()   Scott Griepentrog
• [asterisk-dev] [Code Review] 3267: pjsip: avoid edge case potential crash in answer()   Scott Griepentrog
• [asterisk-dev] [Code Review] 3267: pjsip: avoid edge case potential crash in answer()   Scott Griepentrog
• [asterisk-dev] [Code Review] 3277: testsuite: ARI Originate test for assigning UniqueId on channel creation   Scott Griepentrog
• [asterisk-dev] [Code Review] 3278: ari: create bridges with specified unique id   Scott Griepentrog
• [asterisk-dev] [Code Review] 3277: testsuite: ARI Originate test for assigning UniqueId on channel creation   Scott Griepentrog
• [asterisk-dev] [Code Review] 3217: pjsip/subscribe/missing_aor: Fix test by using MessageWaiting AMI event   Kevin Harwell
• [asterisk-dev] Attended transfers and MOH   Kevin Harwell
• [asterisk-dev] Attended transfers and MOH   Kevin Harwell
• [asterisk-dev] [Code Review] 3226: channel.c: MOH is not working for transferee after attended transfer   Kevin Harwell
• [asterisk-dev] [Code Review] 3226: channel.c: MOH is not working for transferee after attended transfer   Kevin Harwell
• [asterisk-dev] [Code Review] 3239: res_pjsip_exten_state: Presence for digium phones   Kevin Harwell
• [asterisk-dev] [Code Review] 3240: res_pjsip_exten_state: Presence for digium phones   Kevin Harwell
• [asterisk-dev] [Code Review] 3245: res_pjsip_send_to_voicemail: transferring to voicemail for digium phones   Kevin Harwell
• [asterisk-dev] [Code Review] 3246: res_pjsip_send_to_voicemail: transferring to voicemail for digium phones   Kevin Harwell
• [asterisk-dev] [Code Review] 3170: app_confbridge: MOH is not working for transferee after attended sip transfer   Kevin Harwell
• [asterisk-dev] [Code Review] 3170: app_confbridge: MOH is not working for transferee after attended sip transfer   Kevin Harwell
• [asterisk-dev] [Code Review] 3226: channel.c: MOH is not working for transferee after attended transfer   Kevin Harwell
• [asterisk-dev] [Code Review] 3240: res_pjsip_exten_state: Presence for digium phones   Kevin Harwell
• [asterisk-dev] [Code Review] 3240: res_pjsip_exten_state: Presence for digium phones   Kevin Harwell
• [asterisk-dev] [Code Review] 3245: res_pjsip_send_to_voicemail: transferring to voicemail for digium phones   Kevin Harwell
• [asterisk-dev] [Code Review] 3245: res_pjsip_send_to_voicemail: transferring to voicemail for digium phones   Kevin Harwell
• [asterisk-dev] [Code Review] 3239: res_pjsip_exten_state: Presence for digium phones   Kevin Harwell
• [asterisk-dev] [Code Review] 3268: test_conditions: pre-/post- checks cleanup   Kevin Harwell
• [asterisk-dev] [Code Review] 3246: res_pjsip_send_to_voicemail: transferring to voicemail for digium phones   Kevin Harwell
• [asterisk-dev] [Code Review] 3246: res_pjsip_send_to_voicemail: transferring to voicemail for digium phones   Kevin Harwell
• [asterisk-dev] [Code Review] 3266: media_formats: Move codecs over.   Kevin Harwell
• [asterisk-dev] [Code Review] 3265: media_formats: Move app_fax over   Kevin Harwell
• [asterisk-dev] [Code Review] 3240: res_pjsip_exten_state: Presence for digium phones   Kevin Harwell
• [asterisk-dev] Asterisk 12.1.0-rc1 Now Available   Daniel Jenkins
• [asterisk-dev] Asterisk 12.1.0-rc1 Now Available   Daniel Jenkins
• [asterisk-dev] classifying SIP peers   Olle E. Johansson
• [asterisk-dev] classifying SIP peers   Olle E. Johansson
• [asterisk-dev] Attended transfers and MOH   Olle E. Johansson
• [asterisk-dev] Asterisk 13: Media improvements update   Olle E. Johansson
• [asterisk-dev] Asterisk 13: Media improvements update   Olle E. Johansson
• [asterisk-dev] Asterisk 13: Media improvements update   Olle E. Johansson
• [asterisk-dev] Asterisk 13: Media improvements update   Olle E. Johansson
• [asterisk-dev] [Code Review] 3183: ARI: pass channel variables into originate as opposed to assigning after originate   Matt Jordan
• [asterisk-dev] [Code Review] 3186: AMI Security Events: document the events; add optional IEs to the events   Matt Jordan
• [asterisk-dev] [Code Review] 3185: Logger: Allow creation and removal of dynamic logger channels   Matt Jordan
• [asterisk-dev] [Code Review] 3186: AMI Security Events: document the events; add optional IEs to the events   Matt Jordan
• [asterisk-dev] [Code Review] 3174: chan_iax2: Block unnecessary control frames to/from the wire.   Matt Jordan
• [asterisk-dev] [Code Review] 3183: ARI: pass channel variables into originate as opposed to assigning after originate   Matt Jordan
• [asterisk-dev] [Code Review] 3183: ARI: pass channel variables into originate as opposed to assigning after originate   Matt Jordan
• [asterisk-dev] [Code Review] 3174: chan_iax2: Block unnecessary control frames to/from the wire.   Matt Jordan
• [asterisk-dev] [Code Review] 3155: ConfBridge: Correct prompt playback target   Matt Jordan
• [asterisk-dev] [Code Review] 3172: testsuite: chan_sip Record-Route test   Matt Jordan
• [asterisk-dev] [Code Review] 3183: ARI: pass channel variables into originate as opposed to assigning after originate   Matt Jordan
• [asterisk-dev] [Code Review] 3182: testsuite: LinkedID Propagation test   Matt Jordan
• [asterisk-dev] [Code Review] 3183: ARI: pass channel variables into originate as opposed to assigning after originate   Matt Jordan
• [asterisk-dev] [Code Review] 3183: ARI: pass channel variables into originate as opposed to assigning after originate   Matt Jordan
• [asterisk-dev] [Code Review] 3186: AMI Security Events: document the events; add optional IEs to the events   Matt Jordan
• [asterisk-dev] [Code Review] 3193: cli: pjsip show endpoint <endpoint> shows allow/disallow codecs the same - OPTION 2   Matt Jordan
• [asterisk-dev] [Code Review] 3136: cli: pjsip show endpoint <endpoint> shows allow/disallow codecs the same - OPTION 1   Matt Jordan
• [asterisk-dev] [Code Review] 3183: ARI: pass channel variables into originate as opposed to assigning after originate   Matt Jordan
• [asterisk-dev] [Code Review] 3183: ARI: pass channel variables into originate as opposed to assigning after originate   Matt Jordan
• [asterisk-dev] [Code Review] 3174: chan_iax2: Block unnecessary control frames to/from the wire.   Matt Jordan
• [asterisk-dev] [Code Review] 3198: testsuite: Don't continue if we cannot kill a (root?) running asterisk.   Matt Jordan
• [asterisk-dev] [Code Review] 3185: Logger: Allow creation and removal of dynamic logger channels   Matt Jordan
• [asterisk-dev] [Code Review] 3185: Logger: Allow creation and removal of dynamic logger channels   Matt Jordan
• [asterisk-dev] [Code Review] 3183: ARI: pass channel variables into originate as opposed to assigning after originate   Matt Jordan
• [asterisk-dev] [Code Review] 3207: HEP: Add a Homer Encapsulation Protocol (HEP) v3 capture agent module and a packet logger for PJSIP   Matt Jordan
• [asterisk-dev] [Code Review] 3206: HEP: Add test for PJSIP HEP packet capture   Matt Jordan
• [asterisk-dev] [Code Review] 3206: HEP: Add test for PJSIP HEP packet capture   Matt Jordan
• [asterisk-dev] [Code Review] 3207: HEP: Add a Homer Encapsulation Protocol (HEP) v3 capture agent module and a packet logger for PJSIP   Matt Jordan
• [asterisk-dev] [Code Review] 3207: HEP: Add a Homer Encapsulation Protocol (HEP) v3 capture agent module and a packet logger for PJSIP   Matt Jordan
• [asterisk-dev] [Code Review] 3207: HEP: Add a Homer Encapsulation Protocol (HEP) v3 capture agent module and a packet logger for PJSIP   Matt Jordan
• [asterisk-dev] [Code Review] 3206: HEP: Add test for PJSIP HEP packet capture   Matt Jordan
• [asterisk-dev] [Code Review] 3205: Remove PJSIP's MWI-specific function calls   Matt Jordan
• [asterisk-dev] [Code Review] 3214: chan_sip: Set SIP_DEFER_BYE_ON_TRANSFER prior to calling ast_bridge_transfer_blind   Matt Jordan
• [asterisk-dev] [Code Review] 3214: chan_sip: Set SIP_DEFER_BYE_ON_TRANSFER prior to calling ast_bridge_transfer_blind   Matt Jordan
• [asterisk-dev] [Code Review] 3214: chan_sip: Set SIP_DEFER_BYE_ON_TRANSFER prior to calling ast_bridge_transfer_blind   Matt Jordan
• [asterisk-dev] [Code Review] 3214: chan_sip: Set SIP_DEFER_BYE_ON_TRANSFER prior to calling ast_bridge_transfer_blind   Matt Jordan
• [asterisk-dev] [Code Review] 3212: Makefile: fix so "make main" works without compiling deps (unless needed)   Matt Jordan
• [asterisk-dev] [Code Review] 3182: testsuite: LinkedID Propagation test   Matt Jordan
• [asterisk-dev] [Code Review] 3217: pjsip/subscribe/missing_aor: Fix test by using MessageWaiting AMI event   Matt Jordan
• [asterisk-dev] [Code Review] 3216: Add SIP User-Agent to contacts   Matt Jordan
• [asterisk-dev] [Code Review] 3218: PJSIP User Agent tests   Matt Jordan
• [asterisk-dev] [Code Review] 3219: testsuite: Add two basic tests for AgentLogin   Matt Jordan
• [asterisk-dev] [Code Review] 3221: testsuite: Test for Marked/Normal(Unmarked) user Conference Interaction   Matt Jordan
• [asterisk-dev] [Code Review] 3220: testsuite: Add basic tests for the DB function   Matt Jordan
• [asterisk-dev] [Code Review] 3222: pbx: If someone managed to get Asterisk to load with no dialplan at all, survive   Matt Jordan
• [asterisk-dev] [Code Review] 3206: HEP: Add test for PJSIP HEP packet capture   Matt Jordan
• [asterisk-dev] [Code Review] 3207: HEP: Add a Homer Encapsulation Protocol (HEP) v3 capture agent module and a packet logger for PJSIP   Matt Jordan
• [asterisk-dev] [Code Review] 3207: HEP: Add a Homer Encapsulation Protocol (HEP) v3 capture agent module and a packet logger for PJSIP   Matt Jordan
• [asterisk-dev] [Code Review] 3214: chan_sip: Set SIP_DEFER_BYE_ON_TRANSFER prior to calling ast_bridge_transfer_blind   Matt Jordan
• [asterisk-dev] [Code Review] 3217: pjsip/subscribe/missing_aor: Fix test by using MessageWaiting AMI event   Matt Jordan
• [asterisk-dev] [Code Review] 3228: buildsystem: Unbreak the build (infloop) on Asterisk 11+   Matt Jordan
• [asterisk-dev] [Code Review] 3222: pbx: If someone managed to get Asterisk to load with no dialplan at all, survive   Matt Jordan
• [asterisk-dev] [Code Review] 3219: testsuite: Add two basic tests for AgentLogin   Matt Jordan
• [asterisk-dev] [Code Review] 3220: testsuite: Add basic tests for the DB function   Matt Jordan
• [asterisk-dev] [Code Review] 3221: testsuite: Test for Marked/Normal(Unmarked) user Conference Interaction   Matt Jordan
• [asterisk-dev] [Code Review] 3221: testsuite: Test for Marked/Normal(Unmarked) user Conference Interaction   Matt Jordan
• [asterisk-dev] [Code Review] 3221: testsuite: Test for Marked/Normal(Unmarked) user Conference Interaction   Matt Jordan
• [asterisk-dev] [Code Review] 3182: testsuite: LinkedID Propagation test   Matt Jordan
• [asterisk-dev] [Code Review] 3191: channel uniqueid phase 1: convert string uniqueid values to structure with time   Matt Jordan
• [asterisk-dev] [Code Review] 3191: channel uniqueid phase 1: convert string uniqueid values to structure with time   Matt Jordan
• [asterisk-dev] [Code Review] 3244: res_pjsip_sdp_rtp: Apply packetization rules on inbound SDP handling   Matt Jordan
• [asterisk-dev] [Code Review] 3225: Asterisk doesn't support the dynamic payload change in rtp mapping in the 200 OK response   Matt Jordan
• [asterisk-dev] [Code Review] 3221: testsuite: Test for Marked/Normal(Unmarked) user Conference Interaction   Matt Jordan
• [asterisk-dev] [Code Review] 3234: media_formats: Initial channel driver conversion and application conversion   Matt Jordan
• [asterisk-dev] [Code Review] 3247: rtp_engine: lock channel during get_codecs calls to prevent NULL pvt dereference after channel masquerade   Matt Jordan
• [asterisk-dev] [Code Review] 3242: starpy: add UniqueId parameter to ami.Originate()   Matt Jordan
• [asterisk-dev] [Code Review] 3243: testsuite: manager Originate test for assigning UniqueId on channel creation   Matt Jordan
• [asterisk-dev] [Code Review] 3207: HEP: Add a Homer Encapsulation Protocol (HEP) v3 capture agent module and a packet logger for PJSIP   Matt Jordan
• [asterisk-dev] [Code Review] 3207: HEP: Add a Homer Encapsulation Protocol (HEP) v3 capture agent module and a packet logger for PJSIP   Matt Jordan
• [asterisk-dev] [Code Review] 3207: HEP: Add a Homer Encapsulation Protocol (HEP) v3 capture agent module and a packet logger for PJSIP   Matt Jordan
• [asterisk-dev] [Code Review] 3206: HEP: Add test for PJSIP HEP packet capture   Matt Jordan
• [asterisk-dev] [Code Review] 3206: HEP: Add test for PJSIP HEP packet capture   Matt Jordan
• [asterisk-dev] [Code Review] 3250: chan_sip: Add incoming tel: uri support (rfc3966)   Matt Jordan
• [asterisk-dev] [Code Review] 3256: res_rtp_asterisk: Undo regression from ASTERISK-23213 while working around PJNATH assertion abort from ASTERISK-22911   Matt Jordan
• [asterisk-dev] [Code Review] 3256: res_rtp_asterisk: Undo regression from ASTERISK-23213 while working around PJNATH assertion abort from ASTERISK-22911   Matt Jordan
• [asterisk-dev] [Code Review] 3263: Testsuite: Fix originate-cdr-disposition failure reporting   Matt Jordan
• [asterisk-dev] [Code Review] 3264: testsuite: manager/acl-login fails on some platforms   Matt Jordan
• [asterisk-dev] [Code Review] 3261: res_fax: Warn that minrate=2400 is not valid for V.27 instead of failing load   Matt Jordan
• [asterisk-dev] [Code Review] 3256: res_rtp_asterisk: Undo regression from ASTERISK-23213 while working around PJNATH assertion abort from ASTERISK-22911   Matt Jordan
• [asterisk-dev] [Code Review] 3256: res_rtp_asterisk: Undo regression from ASTERISK-23213 while working around PJNATH assertion abort from ASTERISK-22911   Matt Jordan
• [asterisk-dev] [Code Review] 3261: res_fax: Warn that minrate=2400 is not valid for V.27 instead of failing load   Matt Jordan
• [asterisk-dev] [Code Review] 3244: res_pjsip_sdp_rtp: Apply packetization rules on inbound SDP handling   Matt Jordan
• [asterisk-dev] [Code Review] 3247: rtp_engine: lock channel during get_codecs calls to prevent NULL pvt dereference after channel masquerade   Matt Jordan
• [asterisk-dev] [Code Review] 3275: res_rtp_asterisk: Fix the one way audio problems when resuming held calls with ICE   Matt Jordan
• [asterisk-dev] [Code Review] 3248: Fix for WebRTC over WSS not working   Matt Jordan
• [asterisk-dev] [Code Review] 3248: Fix for WebRTC over WSS not working   Matt Jordan
• [asterisk-dev] [Code Review] 3279: Iterate through logger.conf [general] section   Matt Jordan
• [asterisk-dev] [Code Review] 3280: Makefile: replace -O6 with -O3   Matt Jordan
• [asterisk-dev] [Code Review] 3280: Makefile: replace -O6 with -O3   Matt Jordan
• [asterisk-dev] [Code Review] 3279: Iterate through logger.conf [general] section   Matt Jordan
• [asterisk-dev] [Code Review] 3279: Iterate through logger.conf [general] section   Matt Jordan
• [asterisk-dev] [Code Review] 3275: res_rtp_asterisk: Fix the one way audio problems when resuming held calls with ICE   Matt Jordan
• [asterisk-dev] [Code Review] 3275: res_rtp_asterisk: Fix the one way audio problems when resuming held calls with ICE   Matt Jordan
• [asterisk-dev] [r400723-400741] ConfBridge now has the ability to set the language of announcements   Matthew Jordan
• [asterisk-dev] classifying SIP peers   Matthew Jordan
• [asterisk-dev] limiting local ICE candidates?   Matthew Jordan
• [asterisk-dev] classifying SIP peers   Matthew Jordan
• [asterisk-dev] Lightweight keepalive for websockets?   Matthew Jordan
• [asterisk-dev] BUG? Asterisk V10 SIP Message To: non numeric IP (mobile1.xyz.com) fails   Matthew Jordan
• [asterisk-dev] Fwd: patch   Matthew Jordan
• [asterisk-dev] Co-Op students at Digium!   Matthew Jordan
• [asterisk-dev] Asterisk 13: Media improvements update   Matthew Jordan
• [asterisk-dev] Second thoughts on proposed MWI behavior change in Asterisk 12   Matthew Jordan
• [asterisk-dev] New Asterisk Developer - George Joseph   Matthew Jordan
• [asterisk-dev] Asterisk 13: Media improvements update   Matthew Jordan
• [asterisk-dev] CentOS packaging   Matthew Jordan
• [asterisk-dev] CentOS packaging   Matthew Jordan
• [asterisk-dev] CentOS packaging   Matthew Jordan
• [asterisk-dev] Proposal for PJSIP TOS/COS and DSCP   George Joseph
• [asterisk-dev] [Code Review] 3136: cli: pjsip show endpoint <endpoint> shows allow/disallow codecs the same   George Joseph
• [asterisk-dev] [Code Review] 3184: Create sorcery instance registry   George Joseph
• [asterisk-dev] [Code Review] 3184: Create sorcery instance registry   George Joseph
• [asterisk-dev] [Code Review] 3143: pjsip_configuration: in ast_sip_auth_array_init, change assert(auths->names == NULL) to ast_sip_auth_array_destroy(auths)   George Joseph
• [asterisk-dev] [Code Review] 3184: Create sorcery instance registry   George Joseph
• [asterisk-dev] [Code Review] 3184: Create sorcery instance registry   George Joseph
• [asterisk-dev] [Code Review] 3184: Create sorcery instance registry   George Joseph
• [asterisk-dev] [Code Review] 3184: Create sorcery instance registry   George Joseph
• [asterisk-dev] [Code Review] 3184: Create sorcery instance registry   George Joseph
• [asterisk-dev] [Code Review] 3184: Create sorcery instance registry   George Joseph
• [asterisk-dev] [Code Review] 3104: PJSIP CLI Part 2   George Joseph
• [asterisk-dev] [Code Review] 3184: Create sorcery instance registry   George Joseph
• [asterisk-dev] [Code Review] 3184: Create sorcery instance registry   George Joseph
• [asterisk-dev] [Code Review] 3184: Create sorcery instance registry   George Joseph
• [asterisk-dev] [Code Review] 3184: Create sorcery instance registry   George Joseph
• [asterisk-dev] [Code Review] 3200: pjsip_cli: Memory leak in ast_sip_cli_print_sorcery_objectset   George Joseph
• [asterisk-dev] [Code Review] 3200: pjsip_cli: Memory leak in ast_sip_cli_print_sorcery_objectset   George Joseph
• [asterisk-dev] Proposal for PJSIP TOS/COS and DSCP   George Joseph
• [asterisk-dev] Proposal for PJSIP TOS/COS and DSCP   George Joseph
• [asterisk-dev] [Code Review] 3210: PJSIP_CLI: Add 'pjsip show registrations' and 'pjsip show contacts'   George Joseph
• [asterisk-dev] AMI and Sorcery   George Joseph
• [asterisk-dev] AMI and Sorcery   George Joseph
• [asterisk-dev] Second thoughts on proposed MWI behavior change in Asterisk 12   George Joseph
• [asterisk-dev] [Code Review] 3210: PJSIP_CLI: Add 'pjsip show registrations' and 'pjsip show contacts'   George Joseph
• [asterisk-dev] [Code Review] 3200: pjsip_cli: Memory leak in ast_sip_cli_print_sorcery_objectset   George Joseph
• [asterisk-dev] [Code Review] 3184: Create sorcery instance registry   George Joseph
• [asterisk-dev] Asterisk platform   George Joseph
• [asterisk-dev] Asterisk platform   George Joseph
• [asterisk-dev] [Code Review] 3254: sorcery: Create AST_SORCERY dialplan function (et. al.)   George Joseph
• [asterisk-dev] CentOS packaging   George Joseph
• [asterisk-dev] [Code Review] 3254: sorcery: Create AST_SORCERY dialplan function (et. al.)   George Joseph
• [asterisk-dev] [Code Review] 3254: sorcery: Create AST_SORCERY dialplan function (et. al.)   George Joseph
• [asterisk-dev] [Code Review] 3254: sorcery: Create AST_SORCERY dialplan function (et. al.)   George Joseph
• [asterisk-dev] [Code Review] 3254: sorcery: Create AST_SORCERY dialplan function (et. al.)   George Joseph
• [asterisk-dev] [Code Review] 3254: sorcery: Create AST_SORCERY dialplan function (et. al.)   George Joseph
• [asterisk-dev] Asterisk for GSM core network (MSC)   Treesa Fairy Joseph
• [asterisk-dev] Asterisk for GSM core network (MSC)   Treesa Fairy Joseph
• [asterisk-dev] AES-GCM mode SRTP   Kristian Kielhofner
• [asterisk-dev] Asterisk for GSM core network (MSC)   Kaloyan Kovachev
• [asterisk-dev] Asterisk for GSM core network (MSC)   Kaloyan Kovachev
• [asterisk-dev] Monitor() sync and CNG problems   Jaco Kroon
• [asterisk-dev] res_fax_spandsp segfaults during fax detection - FIXED?   Jaco Kroon
• [asterisk-dev] Lightweight keepalive for websockets?   Jeremy Lainé
• [asterisk-dev] Lightweight keepalive for websockets?   Jeremy Lainé
• [asterisk-dev] Lightweight keepalive for websockets?   Jeremy Lainé
• [asterisk-dev] CentOS packaging   Ben Langfeld
• [asterisk-dev] CentOS packaging   Ben Langfeld
• [asterisk-dev] CentOS packaging   Ben Langfeld
• [asterisk-dev] [Code Review] 3273: Correct cross-platform stat nanosecond code.   David Lee
• [asterisk-dev] [Code Review] 3273: Corrected cross-platform stat nanosecond code.   David Lee
• [asterisk-dev] [Code Review] 3273: Corrected cross-platform stat nanosecond code.   David Lee
• [asterisk-dev] [Code Review] 3273: Corrected cross-platform stat nanosecond code.   David Lee
• [asterisk-dev] Installing Asterisk in Mac OS X Mavericks   David M. Lee
• [asterisk-dev] CentOS packaging   David M. Lee
• [asterisk-dev] CentOS packaging   David M. Lee
• [asterisk-dev] [Code Review] 3194: res_config_pgsql: Fix ast_update2_realtime calls   Tilghman Lesher
• [asterisk-dev] Asterisk platform   Julian Lyndon-Smith
• [asterisk-dev] [Code Review] 3186: AMI Security Events: document the events; add optional IEs to the events   Leif Madsen
• [asterisk-dev] [Code Review] 3279: Iterate through logger.conf [general] section   Leif Madsen
• [asterisk-dev] Fwd: patch   Guillaume Maudoux
• [asterisk-dev] Installing Asterisk in Mac OS X Mavericks   Maruen Mehana
• [asterisk-dev] Installing Asterisk in Mac OS X Mavericks   Maruen Mehana
• [asterisk-dev] [Code Review] 3180: Documenation: Configuration section naming in pjsip.conf.sample needs a little clarification   Mark Michelson
• [asterisk-dev] [Code Review] 3180: Documenation: Configuration section naming in pjsip.conf.sample needs a little clarification   Mark Michelson
• [asterisk-dev] [Code Review] 3180: Documenation: Configuration section naming in pjsip.conf.sample needs a little clarification   Mark Michelson
• [asterisk-dev] [Code Review] 3180: Documenation: Configuration section naming in pjsip.conf.sample needs a little clarification   Mark Michelson
• [asterisk-dev] [Code Review] 3175: timing: Improve performance for most timing implementations   Mark Michelson
• [asterisk-dev] [Code Review] 3175: timing: Improve performance for most timing implementations   Mark Michelson
• [asterisk-dev] [Code Review] 3175: timing: Improve performance for most timing implementations   Mark Michelson
• [asterisk-dev] [Code Review] 3178: media_formats: Moving of existing code around, implementation, and unit tests   Mark Michelson
• [asterisk-dev] [Code Review] 3178: media_formats: Moving of existing code around, implementation, and unit tests   Mark Michelson
• [asterisk-dev] [Code Review] 3205: Remove PJSIP's MWI-specific function calls   Mark Michelson
• [asterisk-dev] [Code Review] 3207: HEP: Add a Homer Encapsulation Protocol (HEP) v3 capture agent module and a packet logger for PJSIP   Mark Michelson
• [asterisk-dev] [Code Review] 3206: HEP: Add test for PJSIP HEP packet capture   Mark Michelson
• [asterisk-dev] [Code Review] 3201: realtime: Fix ast_update2_realtime() on raspberry pi.   Mark Michelson
• [asterisk-dev] [Code Review] 3185: Logger: Allow creation and removal of dynamic logger channels   Mark Michelson
• [asterisk-dev] [Code Review] 3185: Logger: Allow creation and removal of dynamic logger channels   Mark Michelson
• [asterisk-dev] [Code Review] 3213: ast_custom_escalating_function allocation leak   Mark Michelson
• [asterisk-dev] [Code Review] 3209: Crash in ast_format_cmp on shutdown   Mark Michelson
• [asterisk-dev] [Code Review] 3214: chan_sip: Set SIP_DEFER_BYE_ON_TRANSFER prior to calling ast_bridge_transfer_blind   Mark Michelson
• [asterisk-dev] [Code Review] 3214: chan_sip: Set SIP_DEFER_BYE_ON_TRANSFER prior to calling ast_bridge_transfer_blind   Mark Michelson
• [asterisk-dev] [Code Review] 3216: Add SIP User-Agent to contacts   Mark Michelson
• [asterisk-dev] [Code Review] 3218: PJSIP User Agent tests   Mark Michelson
• [asterisk-dev] [Code Review] 3205: Remove PJSIP's MWI-specific function calls   Mark Michelson
• [asterisk-dev] [Code Review] 3216: Add SIP User-Agent to contacts   Mark Michelson
• [asterisk-dev] [Code Review] 3207: HEP: Add a Homer Encapsulation Protocol (HEP) v3 capture agent module and a packet logger for PJSIP   Mark Michelson
• [asterisk-dev] Attended transfers and MOH   Mark Michelson
• [asterisk-dev] [Code Review] 3216: Add SIP User-Agent to contacts   Mark Michelson
• [asterisk-dev] [Code Review] 3218: PJSIP User Agent tests   Mark Michelson
• [asterisk-dev] [Code Review] 3235: config: Add file size and nanosecond resolution fields to the cached modified config file information.   Mark Michelson
• [asterisk-dev] [Code Review] 3234: media_formats: Initial channel driver conversion and application conversion   Mark Michelson
• [asterisk-dev] [Code Review] 3234: media_formats: Initial channel driver conversion and application conversion   Mark Michelson
• [asterisk-dev] [Code Review] 3237: PJSIP: Allow for flexibility in configuration of unsolicited and solicited MWI notifications.   Mark Michelson
• [asterisk-dev] [Code Review] 3238: Testsuite: Test MWI aggregation   Mark Michelson
• [asterisk-dev] [Code Review] 3237: PJSIP: Allow for flexibility in configuration of unsolicited and solicited MWI notifications.   Mark Michelson
• [asterisk-dev] [Code Review] 3227: alembic: Add missing queue and cdr table creation scripts.   Mark Michelson
• [asterisk-dev] Second thoughts on proposed MWI behavior change in Asterisk 12   Mark Michelson
• [asterisk-dev] Second thoughts on proposed MWI behavior change in Asterisk 12   Mark Michelson
• [asterisk-dev] Second thoughts on proposed MWI behavior change in Asterisk 12   Mark Michelson
• [asterisk-dev] [Code Review] 3239: res_pjsip_exten_state: Presence for digium phones   Mark Michelson
• [asterisk-dev] [Code Review] 3240: res_pjsip_exten_state: Presence for digium phones   Mark Michelson
• [asterisk-dev] [Code Review] 3241: Remove extra defines of AST_PBX_MAX_STACK   Mark Michelson
• [asterisk-dev] [Code Review] 3234: media_formats: Initial channel driver conversion and application conversion   Mark Michelson
• [asterisk-dev] [Code Review] 3267: pjsip: avoid edge case potential crash in answer()   Mark Michelson
• [asterisk-dev] [Code Review] 3237: PJSIP: Allow for flexibility in configuration of unsolicited and solicited MWI notifications.   Mark Michelson
• [asterisk-dev] Asterisk platform   Frank W. Miller
• [asterisk-dev] Asterisk platform   Frank W. Miller
• [asterisk-dev] Asterisk platform   Frank W. Miller
• [asterisk-dev] 302 redirects ocassionally ignored; hypothesis: later queued busy preferred to earlier early media frame   Richard Mudgett
• [asterisk-dev] 302 redirects ocassionally ignored; hypothesis: later queued busy preferred to earlier early media frame   Richard Mudgett
• [asterisk-dev] [r400723-400741] ConfBridge now has the ability to set the language of announcements   Richard Mudgett
• [asterisk-dev] [Code Review] 3192: chan_dahdi: handle DAHDI_EVENT_REMOVED on a pri D-Channel   Richard Mudgett
• [asterisk-dev] PJSIP transport problem   Steve Murphy
• [asterisk-dev] PJSIP debug question   Steve Murphy
• [asterisk-dev] PJSIP transport problem   Steve Murphy
• [asterisk-dev] PJSIP transport problem   Steve Murphy
• [asterisk-dev] New Asterisk Developer - George Joseph   Rusty Newton
• [asterisk-dev] CentOS packaging   Jeffrey Ollie
• [asterisk-dev] [Code Review] 3250: chan_sip: Add incoming tel: uri support (rfc3966)   Geert Van Pamel
• [asterisk-dev] Asterisk Realtime - Static Peers w/ Qualify, long time bug   Trevor Peirce
• [asterisk-dev] CentOS packaging   Oron Peled
• [asterisk-dev] limiting local ICE candidates?   Daniel Pocock
• [asterisk-dev] classifying SIP peers   Daniel Pocock
• [asterisk-dev] classifying SIP peers   Daniel Pocock
• [asterisk-dev] Lightweight keepalive for websockets?   Daniel Pocock
• [asterisk-dev] [Code Review] 3188: format_wav: enhancing log message "Not a wav file" to be clear on what is supported   Jonathan Rose
• [asterisk-dev] Proposal for PJSIP TOS/COS and DSCP   Jonathan Rose
• [asterisk-dev] Proposal for PJSIP TOS/COS and DSCP   Jonathan Rose
• [asterisk-dev] [Code Review] 3255: testsuite: chan_sip ice crash test for ASTERISK-22911   Jonathan Rose
• [asterisk-dev] [Code Review] 3256: res_rtp_asterisk: Undo regression from ASTERISK-23213 while working around the abortion from ASTERISK-22911   Jonathan Rose
• [asterisk-dev] [Code Review] 3256: res_rtp_asterisk: Undo regression from ASTERISK-23213 while working around program abort from ASTERISK-22911   Jonathan Rose
• [asterisk-dev] [Code Review] 3256: res_rtp_asterisk: Undo regression from ASTERISK-23213 while working around PJNATH assertion abort from ASTERISK-22911   Jonathan Rose
• [asterisk-dev] [Code Review] 3256: res_rtp_asterisk: Undo regression from ASTERISK-23213 while working around PJNATH assertion abort from ASTERISK-22911   Jonathan Rose
• [asterisk-dev] [Code Review] 3256: res_rtp_asterisk: Undo regression from ASTERISK-23213 while working around PJNATH assertion abort from ASTERISK-22911   Jonathan Rose
• [asterisk-dev] [Code Review] 3256: res_rtp_asterisk: Undo regression from ASTERISK-23213 while working around PJNATH assertion abort from ASTERISK-22911   Jonathan Rose
• [asterisk-dev] [Code Review] 3256: res_rtp_asterisk: Undo regression from ASTERISK-23213 while working around PJNATH assertion abort from ASTERISK-22911   Jonathan Rose
• [asterisk-dev] [Code Review] 3256: res_rtp_asterisk: Undo regression from ASTERISK-23213 while working around PJNATH assertion abort from ASTERISK-22911   Jonathan Rose
• [asterisk-dev] [Code Review] 3256: res_rtp_asterisk: Undo regression from ASTERISK-23213 while working around PJNATH assertion abort from ASTERISK-22911   Jonathan Rose
• [asterisk-dev] [Code Review] 3256: res_rtp_asterisk: Undo regression from ASTERISK-23213 while working around PJNATH assertion abort from ASTERISK-22911   Jonathan Rose
• [asterisk-dev] [Code Review] 3275: res_rtp_asterisk: Fix the one way audio problems when resuming held calls with ICE   Jonathan Rose
• [asterisk-dev] [Code Review] 3256: res_rtp_asterisk: Undo regression from ASTERISK-23213 while working around PJNATH assertion abort from ASTERISK-22911   Jonathan Rose
• [asterisk-dev] [Code Review] 3275: res_rtp_asterisk: Fix the one way audio problems when resuming held calls with ICE   Jonathan Rose
• [asterisk-dev] [Code Review] 3275: res_rtp_asterisk: Fix the one way audio problems when resuming held calls with ICE   Jonathan Rose
• [asterisk-dev] [Code Review] 3275: res_rtp_asterisk: Fix the one way audio problems when resuming held calls with ICE   Jonathan Rose
• [asterisk-dev] [Code Review] 3255: testsuite: chan_sip ice crash test for ASTERISK-22911   Jonathan Rose
• [asterisk-dev] [Code Review] 3275: res_rtp_asterisk: Fix the one way audio problems when resuming held calls with ICE   Jonathan Rose
• [asterisk-dev] res_fax_spandsp segfaults during fax detection - FIXED?   Michal Rybarik
• [asterisk-dev] res_fax_spandsp segfaults during fax detection - FIXED?   Michal Rybárik
• [asterisk-dev] Asterisk for GSM core network (MSC)   Michal Rybárik
• [asterisk-dev] BUG? Asterisk V10 SIP Message To: non numeric IP (mobile1.xyz.com) fails   Johan Sandgren
• [asterisk-dev] Asterisk for GSM core network (MSC)   Stefan Schmidt
• [asterisk-dev] Asterisk for GSM core network (MSC)   Stefan Schmidt
• [asterisk-dev] [Code Review] 1194: Multiple parking lots parkedcall/transfers/reparking/hangup/recording no handled properly   Mitch Sharp
• [asterisk-dev] [Code Review] 3248: Fix for WebRTC over WSS not working   Moises Silva
• [asterisk-dev] CentOS packaging   Jared Smith
• [asterisk-dev] CentOS packaging   Jared Smith
• [asterisk-dev] CentOS packaging   Jared Smith
• [asterisk-dev] CentOS packaging   Jared Smith
• [asterisk-dev] CentOS packaging   Jared Smith
• [asterisk-dev] CentOS packaging   Jared Smith
• [asterisk-dev] CentOS packaging   Jared Smith
• [asterisk-dev] Lightweight keepalive for websockets?   Philippe Sultan
• [asterisk-dev] Asterisk 12.1.0-rc1 Now Available   Asterisk Development Team
• [asterisk-dev] Asterisk 11.8.0-rc2 Now Available   Asterisk Development Team
• [asterisk-dev] Asterisk 12.1.0-rc2 Now Available   Asterisk Development Team
• [asterisk-dev] Ast 11 / chan_sip: get_rpid/pai not setting CALLERID(num-pres)   Pavel Troller
• [asterisk-dev] Ast 11 / chan_sip: get_rpid/pai not setting CALLERID(num-pres)   Pavel Troller
• [asterisk-dev] Diagnosing signaling problems   Pavel Troller
• [asterisk-dev] 302 redirects ocassionally ignored; hypothesis: later queued busy preferred to earlier early media frame   Dave WOOLLEY
• [asterisk-dev] 302 redirects ocassionally ignored; hypothesis: later queued busy preferred to earlier early media frame   Dave WOOLLEY
• [asterisk-dev] 302 redirects ocassionally ignored; hypothesis: later queued busy preferred to earlier early media frame   Dave WOOLLEY
• [asterisk-dev] Second thoughts on proposed MWI behavior change in Asterisk 12   Brad Watkins
• [asterisk-dev] Second thoughts on proposed MWI behavior change in Asterisk 12   Brad Watkins
• [asterisk-dev] [r400723-400741] ConfBridge now has the ability to set the language of announcements   Jonathan White
• [asterisk-dev] CentOS packaging   Jonathan White
• [asterisk-dev] [Code Review] 2990: Documentation: Clarify "x" Option In chan_spy   Michael Young
• [asterisk-dev] [Code Review] 3272: func_audiohookinheritance: Check If Channel Was Specified   Michael Young
• [asterisk-dev] [Code Review] 1073: Adding the PICKUPSTATUS variable   junky
• [asterisk-dev] [Code Review] 3185: Logger: Allow creation and removal of dynamic logger channels   opticron
• [asterisk-dev] [Code Review] 3155: ConfBridge: Correct prompt playback target   opticron
• [asterisk-dev] [Code Review] 3185: Logger: Allow creation and removal of dynamic logger channels   opticron
• [asterisk-dev] [Code Review] 3185: Logger: Allow creation and removal of dynamic logger channels   opticron
• [asterisk-dev] [Code Review] 3185: Logger: Allow creation and removal of dynamic logger channels   opticron
• [asterisk-dev] [Code Review] 3182: testsuite: LinkedID Propagation test   opticron
• [asterisk-dev] [Code Review] 3155: ConfBridge: Correct prompt playback target   opticron
• [asterisk-dev] [Code Review] 3185: Logger: Allow creation and removal of dynamic logger channels   opticron
• [asterisk-dev] [Code Review] 3185: Logger: Allow creation and removal of dynamic logger channels   opticron
• [asterisk-dev] [Code Review] 3185: Logger: Allow creation and removal of dynamic logger channels   opticron
• [asterisk-dev] [Code Review] 3169: Testsuite: Add basic NAT test   opticron
• [asterisk-dev] [Code Review] 3132: Test for allow=all sdp issue   opticron
• [asterisk-dev] [Code Review] 3208: possible null pointer dereference in format.c   opticron
• [asterisk-dev] [Code Review] 3210: PJSIP_CLI: Add 'pjsip show registrations' and 'pjsip show contacts'   opticron
• [asterisk-dev] [Code Review] 3211: ARI: URI is case sensitive   opticron
• [asterisk-dev] [Code Review] 3185: Logger: Allow creation and removal of dynamic logger channels   opticron
• [asterisk-dev] [Code Review] 3182: testsuite: LinkedID Propagation test   opticron
• [asterisk-dev] [Code Review] 3216: Add SIP User-Agent to contacts   opticron
• [asterisk-dev] [Code Review] 3222: pbx: If someone managed to get Asterisk to load with no dialplan at all, survive   opticron
• [asterisk-dev] [Code Review] 3219: testsuite: Add two basic tests for AgentLogin   opticron
• [asterisk-dev] [Code Review] 3221: testsuite: Test for Marked/Normal(Unmarked) user Conference Interaction   opticron
• [asterisk-dev] [Code Review] 3221: testsuite: Test for Marked/Normal(Unmarked) user Conference Interaction   opticron
• [asterisk-dev] [Code Review] 3221: testsuite: Test for Marked/Normal(Unmarked) user Conference Interaction   opticron
• [asterisk-dev] [Code Review] 3161: res_sorcery_astdb.c: Fix regex handling and keep simple prefix matching performance.   opticron
• [asterisk-dev] [Code Review] 3161: res_sorcery_astdb.c: Fix regex handling and keep simple prefix matching performance.   opticron
• [asterisk-dev] [Code Review] 3161: res_sorcery_astdb.c: Fix regex handling and keep simple prefix matching performance.   opticron
• [asterisk-dev] [Code Review] 3234: media_formats: Initial channel driver conversion and application conversion   opticron
• [asterisk-dev] [Code Review] 3234: media_formats: Initial channel driver conversion and application conversion   opticron
• [asterisk-dev] [Code Review] 3251: res_bucket_sounds: Add 'sounds' URI scheme implementation   opticron
• [asterisk-dev] [Code Review] 3132: Test for allow=all sdp issue   opticron
• [asterisk-dev] [Code Review] 3238: Testsuite: Test MWI aggregation   opticron
• [asterisk-dev] [Code Review] 3246: res_pjsip_send_to_voicemail: transferring to voicemail for digium phones   opticron
• [asterisk-dev] [Code Review] 3248: Fix for WebRTC over WSS not working   opticron
• [asterisk-dev] [Code Review] 3249: Testsuite: manager live_dangerously tests are missing a parameter for on_failure   opticron
• [asterisk-dev] [Code Review] 3254: sorcery: Create AST_SORCERY dialplan function (et. al.)   opticron
• [asterisk-dev] [Code Review] 3246: res_pjsip_send_to_voicemail: transferring to voicemail for digium phones   opticron
• [asterisk-dev] [Code Review] 3174: chan_iax2: Block unnecessary control frames to/from the wire.   rmudgett
• [asterisk-dev] [Code Review] 3174: chan_iax2: Block unnecessary control frames to/from the wire.   rmudgett
• [asterisk-dev] [Code Review] 3174: chan_iax2: Block unnecessary control frames to/from the wire.   rmudgett
• [asterisk-dev] [Code Review] 3174: chan_iax2: Block unnecessary control frames to/from the wire.   rmudgett
• [asterisk-dev] [Code Review] 3155: ConfBridge: Correct prompt playback target   rmudgett
• [asterisk-dev] [Code Review] 3192: chan_dahdi: handle DAHDI_EVENT_REMOVED on a pri D-Channel   rmudgett
• [asterisk-dev] [Code Review] 3158: indications.conf: fix post-stutter dialtone for in, mx and ph, extra missing stutter   rmudgett
• [asterisk-dev] [Code Review] 3184: Create sorcery instance registry   rmudgett
• [asterisk-dev] [Code Review] 3184: Create sorcery instance registry   rmudgett
• [asterisk-dev] [Code Review] 3191: channel uniqueid phase 1: convert string uniqueid values to structure with time   rmudgett
• [asterisk-dev] [Code Review] 3184: Create sorcery instance registry   rmudgett
• [asterisk-dev] [Code Review] 3184: Create sorcery instance registry   rmudgett
• [asterisk-dev] [Code Review] 3184: Create sorcery instance registry   rmudgett
• [asterisk-dev] [Code Review] 3178: media_formats: Moving of existing code around, implementation, and unit tests   rmudgett
• [asterisk-dev] [Code Review] 3184: Create sorcery instance registry   rmudgett
• [asterisk-dev] [Code Review] 3184: Create sorcery instance registry   rmudgett
• [asterisk-dev] [Code Review] 3174: chan_iax2: Block unnecessary control frames to/from the wire.   rmudgett
• [asterisk-dev] [Code Review] 3200: pjsip_cli: Memory leak in ast_sip_cli_print_sorcery_objectset   rmudgett
• [asterisk-dev] [Code Review] 3200: pjsip_cli: Memory leak in ast_sip_cli_print_sorcery_objectset   rmudgett
• [asterisk-dev] [Code Review] 3155: ConfBridge: Correct prompt playback target   rmudgett
• [asterisk-dev] [Code Review] 3199: scheduler: Remove hashtab usage.   rmudgett
• [asterisk-dev] [Code Review] 3199: scheduler: Remove hashtab usage.   rmudgett
• [asterisk-dev] [Code Review] 3191: channel uniqueid phase 1: convert string uniqueid values to structure with time   rmudgett
• [asterisk-dev] [Code Review] 3191: channel uniqueid phase 1: convert string uniqueid values to structure with time   rmudgett
• [asterisk-dev] [Code Review] 3226: channel.c: MOH is not working for transferee after attended transfer   rmudgett
• [asterisk-dev] [Code Review] 3227: alembic: Add missing queue and cdr table creation scripts.   rmudgett
• [asterisk-dev] [Code Review] 3226: channel.c: MOH is not working for transferee after attended transfer   rmudgett
• [asterisk-dev] [Code Review] 3235: config: Add file size and nanosecond resolution fields to the cached modified config file information.   rmudgett
• [asterisk-dev] [Code Review] 3161: res_sorcery_astdb.c: Fix regex handling and keep simple prefix matching performance.   rmudgett
• [asterisk-dev] [Code Review] 3161: res_sorcery_astdb.c: Fix regex handling and keep simple prefix matching performance.   rmudgett
• [asterisk-dev] [Code Review] 3235: config: Add file size and nanosecond resolution fields to the cached modified config file information.   rmudgett
• [asterisk-dev] [Code Review] 3235: config: Add file size and nanosecond resolution fields to the cached modified config file information.   rmudgett
• [asterisk-dev] [Code Review] 3161: res_sorcery_astdb.c: Fix regex handling and keep simple prefix matching performance.   rmudgett
• [asterisk-dev] [Code Review] 3235: config: Add file size and nanosecond resolution fields to the cached modified config file information.   rmudgett
• [asterisk-dev] [Code Review] 3191: channel uniqueid phase 1: convert string uniqueid values to structure with time   rmudgett
• [asterisk-dev] [Code Review] 3273: Corrected cross-platform stat nanosecond code.   rmudgett
• [asterisk-dev] [Code Review] 3273: Corrected cross-platform stat nanosecond code.   rmudgett
• [asterisk-dev] [Code Review] 3180: Documenation: Configuration section naming in pjsip.conf.sample needs a little clarification   rnewton
• [asterisk-dev] [Code Review] 3180: Documenation: Configuration section naming in pjsip.conf.sample needs a little clarification   rnewton
• [asterisk-dev] [Code Review] 2990: Documentation: Clarify "x" Option In chan_spy   rnewton
• [asterisk-dev] [Code Review] 3136: cli: pjsip show endpoint <endpoint> shows allow/disallow codecs the same   rnewton
• [asterisk-dev] [Code Review] 3136: cli: pjsip show endpoint <endpoint> shows allow/disallow codecs the same   rnewton
• [asterisk-dev] [Code Review] 3180: Documenation: Configuration section naming in pjsip.conf.sample needs a little clarification   rnewton
• [asterisk-dev] [Code Review] 3188: format_wav: enhancing log message "Not a wav file" to be clear on what is supported   rnewton
• [asterisk-dev] [Code Review] 3188: format_wav: enhancing log message "Not a wav file" to be clear on what is supported   rnewton
• [asterisk-dev] [Code Review] 3188: format_wav: enhancing log message "Not a wav file" to be clear on what is supported   rnewton
• [asterisk-dev] [Code Review] 3180: Documenation: Configuration section naming in pjsip.conf.sample needs a little clarification   rnewton
• [asterisk-dev] [Code Review] 3173: chan_sip refactor - sip_route   wdoekes
• [asterisk-dev] [Code Review] 3173: chan_sip refactor - sip_route   wdoekes
• [asterisk-dev] [Code Review] 3173: chan_sip refactor - sip_route   wdoekes
• [asterisk-dev] [Code Review] 3194: res_config_pgsql: Fix ast_update2_realtime calls   wdoekes
• [asterisk-dev] [Code Review] 3195: testsuite: fixes for run-local   wdoekes
• [asterisk-dev] [Code Review] 3198: testsuite: Don't continue if we cannot kill a (root?) running asterisk.   wdoekes
• [asterisk-dev] [Code Review] 3201: realtime: Fix ast_update2_realtime() on raspberry pi.   wdoekes
• [asterisk-dev] [Code Review] 3201: realtime: Fix ast_update2_realtime() on raspberry pi.   wdoekes
• [asterisk-dev] [Code Review] 3202: install_subst: helper script for installing with path substitution   wdoekes
• [asterisk-dev] [Code Review] 3207: HEP: Add a Homer Encapsulation Protocol (HEP) v3 capture agent module and a packet logger for PJSIP   wdoekes
• [asterisk-dev] [Code Review] 3212: Makefile: fix so "make main" works without compiling deps (unless needed)   wdoekes
• [asterisk-dev] [Code Review] 3201: realtime: Fix ast_update2_realtime() on raspberry pi.   wdoekes
• [asterisk-dev] [Code Review] 3198: testsuite: Don't continue if we cannot kill a (root?) running asterisk.   wdoekes
• [asterisk-dev] [Code Review] 3194: res_config_pgsql: Fix ast_update2_realtime calls   wdoekes
• [asterisk-dev] [Code Review] 2757: unbreak safe_asterisk.conf parsing (r394939 / ASTERISK-21965)   wdoekes
• [asterisk-dev] [Code Review] 3212: Makefile: fix so "make main" works without compiling deps (unless needed)   wdoekes
• [asterisk-dev] [Code Review] 3212: Makefile: fix so "make main" works without compiling deps (unless needed)   wdoekes
• [asterisk-dev] [Code Review] 3228: buildsystem: Unbreak the build (infloop) on Asterisk 11+   wdoekes
• [asterisk-dev] [Code Review] 3228: buildsystem: Unbreak the build (infloop) on Asterisk 11+   wdoekes
• [asterisk-dev] [Code Review] 3235: config: Add file size and nanosecond resolution fields to the cached modified config file information.   wdoekes
• [asterisk-dev] [Code Review] 3161: res_sorcery_astdb.c: Fix regex handling and keep simple prefix matching performance.   wdoekes
• [asterisk-dev] [Code Review] 3236: chan_sip: do not send empty Route header   wdoekes
• [asterisk-dev] [Code Review] 3250: chan_sip: Add incoming tel: uri support (rfc3966)   wdoekes
• [asterisk-dev] [Code Review] 3250: chan_sip: Add incoming tel: uri support (rfc3966)   wdoekes
• [asterisk-dev] [Code Review] 3258: testsuite: eliminate sipp zombie   wdoekes
• [asterisk-dev] [Code Review] 3262: testsuite: Add dependency check for rawsocket access   wdoekes
• [asterisk-dev] [Code Review] 3258: testsuite: eliminate sipp zombie   wdoekes
• [asterisk-dev] [Code Review] 3273: Corrected cross-platform stat nanosecond code.   wdoekes
• [asterisk-dev] [Code Review] 3279: Iterate through logger.conf [general] section   wdoekes
• [asterisk-dev] [r400723-400741] ConfBridge now has the ability to set the language of announcements   jonathan white

Last message date: Fri Feb 28 15:27:54 CST 2014
Archived on: Fri Feb 28 15:18:52 CST 2014

• Messages sorted by: [ thread ] [ subject ] [ date ]
• More info on this list...