May 2012 Archives by author

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Starting: Tue May 1 00:45:15 CDT 2012
Ending: Thu May 31 16:37:04 CDT 2012
Messages: 482

• [asterisk-dev] EWS - Calendar Integration Probelm   Paul Belanger
• [asterisk-dev] [Code Review] Log 'stun set debug on' messages as ast_debug   Paul Belanger
• [asterisk-dev] Missed ability to filter on sip debug   Paul Belanger
• [asterisk-dev] [Code Review] chan_jingle2: New Jingle + Google Talk channel driver   Paul Belanger
• [asterisk-dev] [Code Review] Add tests for the IAX2 implementation of the HANGUPCAUSE hash   Paul Belanger
• [asterisk-dev] [Code Review] Add IAX2 support for the new HANGUPCAUSE hash   Paul Belanger
• [asterisk-dev] [svn-commits] mmichelson: branch 1.8 r367002 - in /branches/1.8: channels/ include/asterisk...   Paul Belanger
• [asterisk-dev] asterisk.org documentation   Paul Belanger
• [asterisk-dev] asterisk.org documentation   Paul Belanger
• [asterisk-dev] cdr documentation - new fields   Paul Belanger
• [asterisk-dev] RFC: chan_sip SDP parsing changes in behavior   Paul Belanger
• [asterisk-dev] RFC: chan_sip SDP parsing changes in behavior   Paul Belanger
• [asterisk-dev] Astricon 2012 presentations   Paul Belanger
• [asterisk-dev] [Code Review] Add test for correct HANGUPCAUSE after SIP 404   Paul Belanger
• [asterisk-dev] [Code Review] ExternalIVR: Add IPv6 support   Sean Bright
• [asterisk-dev] [Code Review]: ExternalIVR: Add IPv6 support   Sean Bright
• [asterisk-dev] [Code Review] ExternalIVR: Add IPv6 support   Sean Bright
• [asterisk-dev] Asterisk disconnect active connection when SMS is arriving   Russell Bryant
• [asterisk-dev] [Code Review]: Pinequeue: Play queue prompts in the background - making call available to agent faster   Russell Bryant
• [asterisk-dev] [Code Review] chan_jingle2: New Jingle + Google Talk channel driver   Russell Bryant
• [asterisk-dev] Opensips/Asterisk developer   Ronnie Bulanek
• [asterisk-dev] cdr documentation - new fields   Marek Cervenka
• [asterisk-dev] [Code Review] Macro AST_PKG_CONFIG_CHECK   Tzafrir Cohen
• [asterisk-dev] [Code Review] pass BUILD_CFLGAS and BUILD_LDFLAGS to menuselect (asterisk part)   Tzafrir Cohen
• [asterisk-dev] [Code Review] pass BUILD_CFLGAS and BUILD_LDFLAGS to menuselect (menuselect part)   Tzafrir Cohen
• [asterisk-dev] [Code Review] pass BUILD_CFLGAS and BUILD_LDFLAGS to menuselect (asterisk part)   Tzafrir Cohen
• [asterisk-dev] [Code Review]: Macro AST_PKG_CONFIG_CHECK   Tzafrir Cohen
• [asterisk-dev] [Code Review] Macro AST_PKG_CONFIG_CHECK   Tzafrir Cohen
• [asterisk-dev] [Code Review] Macro AST_PKG_CONFIG_CHECK   Tzafrir Cohen
• [asterisk-dev] httpd bind port changed to 8080 in 1.8 [was: Re: [svn-commits] mmichelson: branch 1.8 r353770 - in /branches/1.8: ./ configs/ include/asteri...]   Tzafrir Cohen
• [asterisk-dev] [Code Review]: ICE, STUN, and TURN Support   Joshua Colp
• [asterisk-dev] [Code Review]: ICE, STUN, and TURN Support   Joshua Colp
• [asterisk-dev] [Code Review] ICE, STUN, and TURN Support   Joshua Colp
• [asterisk-dev] [Code Review] ConfbridgeActionExec AMI Command   Joshua Colp
• [asterisk-dev] [Code Review] chan_jingle2: New Jingle + Google Talk channel driver   Joshua Colp
• [asterisk-dev] [Code Review] chan_jingle2: New Jingle + Google Talk channel driver   Joshua Colp
• [asterisk-dev] [Code Review]: chan_jingle2: New Jingle + Google Talk channel driver   Joshua Colp
• [asterisk-dev] [Code Review]: chan_jingle2: New Jingle + Google Talk channel driver   Joshua Colp
• [asterisk-dev] [Code Review] WebSocket HTTP Module   Joshua Colp
• [asterisk-dev] [Code Review] ICE, STUN, and TURN Support   Joshua Colp
• [asterisk-dev] [Code Review]: WebSocket HTTP Module   Joshua Colp
• [asterisk-dev] [Code Review]: WebSocket HTTP Module   Joshua Colp
• [asterisk-dev] [Code Review] WebSocket HTTP Module   Joshua Colp
• [asterisk-dev] [Code Review]: WebSocket HTTP Module   Joshua Colp
• [asterisk-dev] [Code Review] WebSocket HTTP Module   Joshua Colp
• [asterisk-dev] RFC: chan_sip SDP parsing changes in behavior   Saúl Ibarra Corretgé
• [asterisk-dev] Urgent development consultancy wanted   Alistair Cunningham
• [asterisk-dev] Urgent development consultancy wanted   Alistair Cunningham
• [asterisk-dev] Urgent development consultancy wanted   Alistair Cunningham
• [asterisk-dev] $300 bounty for #19837: Asterisk crashing regularly in 1.8.11.1   Alistair Cunningham
• [asterisk-dev] $300 bounty for #19837: Asterisk crashing regularly in 1.8.11.1   Alistair Cunningham
• [asterisk-dev] Scenario where generated ringtone continues after call answered.   Steve Davies
• [asterisk-dev] [Code Review] Fix local channel chains optimizing themselves out of a call.   Alec Davis
• [asterisk-dev] [Code Review]: Fix local channel chains optimizing themselves out of a call.   Alec Davis
• [asterisk-dev] [Code Review]: Fix local channel chains optimizing themselves out of a call.   Alec Davis
• [asterisk-dev] [Code Review]: Fix local channel chains optimizing themselves out of a call.   Alec Davis
• [asterisk-dev] Fwd: [JIRA] Commented: (ASTERISK-19846) sip users/peers not matched on incoming invite when there are multiple A records in DNS   John Fawcett
• [asterisk-dev] Fwd: [JIRA] Commented: (ASTERISK-19846) sip users/peers not matched on incoming invite when there are multiple A records in DNS   John Fawcett
• [asterisk-dev] [Code Review] pass BUILD_CFLGAS and BUILD_LDFLAGS to menuselect (asterisk part)   Kevin Fleming
• [asterisk-dev] [Code Review] pass BUILD_CFLGAS and BUILD_LDFLAGS to menuselect (menuselect part)   Kevin Fleming
• [asterisk-dev] [Code Review] Macro AST_PKG_CONFIG_CHECK   Kevin Fleming
• [asterisk-dev] [Code Review] Macro AST_PKG_CONFIG_CHECK   Kevin Fleming
• [asterisk-dev] [Code Review]: Fix a variety of memory leaks   Kevin Fleming
• [asterisk-dev] [Code Review]: Pinequeue: Play queue prompts in the background - making call available to agent faster   Kevin Fleming
• [asterisk-dev] [Code Review] Help mitigate reinvite glares in the SIP channel driver   Kevin Fleming
• [asterisk-dev] [Code Review] Improve SDP parsing warning messages and RFC compliance   Kevin Fleming
• [asterisk-dev] [Code Review]: Improve SDP parsing warning messages and RFC compliance   Kevin Fleming
• [asterisk-dev] [Code Review]: Help mitigate reinvite glares in the SIP channel driver   Kevin Fleming
• [asterisk-dev] [Code Review] Help mitigate reinvite glares in the SIP channel driver   Kevin Fleming
• [asterisk-dev] [Code Review] Help mitigate reinvite glares: Attempt 2   Kevin Fleming
• [asterisk-dev] [Code Review] Help mitigate reinvite glares: Attempt 2   Kevin Fleming
• [asterisk-dev] directmediapermit/deny doesn't work in any way approaching correct for any Asterisk version and is poorly named, so we should rework it in trunk   Kevin P. Fleming
• [asterisk-dev] Pinequeue - to hold or not to hold   Kevin P. Fleming
• [asterisk-dev] [Code Review] Adjust to allow for Digium phones' send to voicemail feature   Kevin P. Fleming
• [asterisk-dev] Urgent development consultancy wanted   Kevin P. Fleming
• [asterisk-dev] Digium's new Community Support Manager - Rusty Newton   Kevin P. Fleming
• [asterisk-dev] RFC: chan_sip SDP parsing changes in behavior   Kevin P. Fleming
• [asterisk-dev] $300 bounty for #19837: Asterisk crashing regularly in 1.8.11.1   Kevin P. Fleming
• [asterisk-dev] RFC: chan_sip SDP parsing changes in behavior   Kevin P. Fleming
• [asterisk-dev] RFC: chan_sip SDP parsing changes in behavior   Kevin P. Fleming
• [asterisk-dev] RFC: chan_sip SDP parsing changes in behavior   Kevin P. Fleming
• [asterisk-dev] RFC: chan_sip SDP parsing changes in behavior   Kevin P. Fleming
• [asterisk-dev] [Code Review] Patch to detect/parse ANI-II / ANI2 / OLI from SIP INVITE messages   Rob Gagnon
• [asterisk-dev] [Code Review] Patch to detect/parse ANI-II / ANI2 / OLI from SIP INVITE messages   Rob Gagnon
• [asterisk-dev] [Code Review]: Patch to detect/parse ANI-II / ANI2 / OLI from SIP INVITE messages   Rob Gagnon
• [asterisk-dev] [Code Review] Patch to detect/parse ANI-II / ANI2 / OLI from SIP INVITE messages   Rob Gagnon
• [asterisk-dev] [Code Review] Patch to detect/parse ANI-II / ANI2 / OLI from SIP INVITE messages   Rob Gagnon
• [asterisk-dev] [Code Review] Patch to detect/parse ANI-II / ANI2 / OLI from SIP INVITE messages   Rob Gagnon
• [asterisk-dev] [Code Review] Patch to detect/parse ANI-II / ANI2 / OLI from SIP INVITE messages   Rob Gagnon
• [asterisk-dev] [Code Review]: Patch to detect/parse ANI-II / ANI2 / OLI from SIP INVITE messages   Rob Gagnon
• [asterisk-dev] [Code Review] Patch to detect/parse ANI-II / ANI2 / OLI from SIP INVITE messages   Rob Gagnon
• [asterisk-dev] [Code Review] Patch to detect/parse ANI-II / ANI2 / OLI from SIP INVITE messages   Rob Gagnon
• [asterisk-dev] [Code Review]: Patch to detect/parse ANI-II / ANI2 / OLI from SIP INVITE messages   Rob Gagnon
• [asterisk-dev] [Code Review] Patch to detect/parse ANI-II / ANI2 / OLI from SIP INVITE messages   Rob Gagnon
• [asterisk-dev] [Code Review]: Patch to detect/parse ANI-II / ANI2 / OLI from SIP INVITE messages   Rob Gagnon
• [asterisk-dev] [Code Review] Patch to detect/parse ANI-II / ANI2 / OLI from SIP INVITE messages   Rob Gagnon
• [asterisk-dev] [Code Review] Remove AST_FLAG_ANSWERED_ELSEWHERE, duplicating the functionality of AST_CAUSE_ANSWERED_ELSEWHERE   Birger Harzenetter
• [asterisk-dev] [Code Review] Remove AST_FLAG_ANSWERED_ELSEWHERE, duplicating the functionality of AST_CAUSE_ANSWERED_ELSEWHERE   Birger Harzenetter
• [asterisk-dev] [Code Review]: Remove AST_FLAG_ANSWERED_ELSEWHERE, duplicating the functionality of AST_CAUSE_ANSWERED_ELSEWHERE   Birger Harzenetter
• [asterisk-dev] [Code Review] Remove AST_FLAG_ANSWERED_ELSEWHERE, duplicating the functionality of AST_CAUSE_ANSWERED_ELSEWHERE   Birger Harzenetter
• [asterisk-dev] [Code Review] Extend SIP REFER message originated by Transfer with SIPAddHeader-added headers   Olle E Johansson
• [asterisk-dev] [Code Review] ICE, STUN, and TURN Support   Olle E Johansson
• [asterisk-dev] [Code Review]: Pinequeue: Play queue prompts in the background - making call available to agent faster   Olle E Johansson
• [asterisk-dev] [Code Review]: Pinequeue: Play queue prompts in the background - making call available to agent faster   Olle E Johansson
• [asterisk-dev] [Code Review] ConfbridgeActionExec AMI Command   Olle E Johansson
• [asterisk-dev] [Code Review]: Pinequeue: Play queue prompts in the background - making call available to agent faster   Olle E Johansson
• [asterisk-dev] Fwd: [JIRA] Commented: (ASTERISK-19846) sip users/peers not matched on incoming invite when there are multiple A records in DNS   Olle E. Johansson
• [asterisk-dev] Fwd: [JIRA] Commented: (ASTERISK-19846) sip users/peers not matched on incoming invite when there are multiple A records in DNS   Olle E. Johansson
• [asterisk-dev] Pinequeue - to hold or not to hold   Olle E. Johansson
• [asterisk-dev] Pinequeue - to hold or not to hold   Olle E. Johansson
• [asterisk-dev] Pinequeue - to hold or not to hold   Olle E. Johansson
• [asterisk-dev] Pinequeue - to hold or not to hold   Olle E. Johansson
• [asterisk-dev] Pinequeue - to hold or not to hold   Olle E. Johansson
• [asterisk-dev] Wrong SIP to SIP SIP Cause mapping   Olle E. Johansson
• [asterisk-dev] RFC: chan_sip SDP parsing changes in behavior   Olle E. Johansson
• [asterisk-dev] RFC: chan_sip SDP parsing changes in behavior   Olle E. Johansson
• [asterisk-dev] [Code Review] Rework the leave_voicemail_contexts test   Matt Jordan
• [asterisk-dev] [Code Review] Properly support the d() option in VoiceMail   Matt Jordan
• [asterisk-dev] [Code Review] Rework the leave_voicemail_contexts test   Matt Jordan
• [asterisk-dev] [Code Review] Properly support the d() option in VoiceMail   Matt Jordan
• [asterisk-dev] [Code Review] Properly support the d() option in VoiceMail   Matt Jordan
• [asterisk-dev] [Code Review]: Rework the leave_voicemail_contexts test   Matt Jordan
• [asterisk-dev] [Code Review]: Rework the leave_voicemail_contexts test   Matt Jordan
• [asterisk-dev] [Code Review] ConfbridgeActionExec AMI Command   Matt Jordan
• [asterisk-dev] [Code Review] Add CEL testing to the testsuite   Matt Jordan
• [asterisk-dev] [Code Review]: ConfbridgeActionExec AMI Command   Matt Jordan
• [asterisk-dev] [Code Review]: ConfbridgeActionExec AMI Command   Matt Jordan
• [asterisk-dev] [Code Review]: ConfbridgeActionExec AMI Command   Matt Jordan
• [asterisk-dev] [Code Review] ConfbridgeActionExec AMI Command   Matt Jordan
• [asterisk-dev] [Code Review] ConfbridgeExecAction AMI Command   Matt Jordan
• [asterisk-dev] [Code Review] chan_jingle2: New Jingle + Google Talk channel driver   Matt Jordan
• [asterisk-dev] [Code Review] Add voicemail message IDs where needed on startup; Add message IDs to ODBC voicemail   Matt Jordan
• [asterisk-dev] [Code Review]: ConfbridgeExecAction AMI Command   Matt Jordan
• [asterisk-dev] [Code Review] ConfbridgeExecAction AMI Command   Matt Jordan
• [asterisk-dev] [Code Review] A new higher-level API for working with Asterisk configs, with example code in app_skel.c and udptl.c   Matt Jordan
• [asterisk-dev] [Code Review] Fix a variety of memory leaks   Matt Jordan
• [asterisk-dev] [Code Review]: Fix a variety of memory leaks   Matt Jordan
• [asterisk-dev] [Code Review] Fix a variety of memory leaks   Matt Jordan
• [asterisk-dev] [Code Review] Enable support for early media in AMI originate action   Matt Jordan
• [asterisk-dev] [Code Review] [confbridge] Behavioural correction for hold-music state when users join/part conferences in varying combinations   Matt Jordan
• [asterisk-dev] [Code Review] Add CEL testing to the testsuite   Matt Jordan
• [asterisk-dev] [Code Review] Digium Phones "send to voicemail" tests   Matt Jordan
• [asterisk-dev] [Code Review] Adjust to allow for Digium phones' send to voicemail feature   Matt Jordan
• [asterisk-dev] [Code Review]: ConfbridgeExecAction AMI Command   Matt Jordan
• [asterisk-dev] [Code Review]: ConfbridgeExecAction AMI Command   Matt Jordan
• [asterisk-dev] [Code Review]: Digium Phones "send to voicemail" tests   Matt Jordan
• [asterisk-dev] [Code Review]: Fix a variety of memory leaks   Matt Jordan
• [asterisk-dev] [Code Review] Fix a variety of memory leaks   Matt Jordan
• [asterisk-dev] [Code Review]: Add voicemail message IDs where needed on startup; Add message IDs to ODBC voicemail   Matt Jordan
• [asterisk-dev] [Code Review]: Fix a variety of memory leaks   Matt Jordan
• [asterisk-dev] [Code Review] Adjust to allow for Digium phones' send to voicemail feature   Matt Jordan
• [asterisk-dev] [Code Review] Update a peer's lastmsgssent value appropriately   Matt Jordan
• [asterisk-dev] [Code Review]: Update a peer's lastmsgssent value appropriately   Matt Jordan
• [asterisk-dev] [Code Review] Add voicemail message IDs where needed on startup; Add message IDs to ODBC voicemail   Matt Jordan
• [asterisk-dev] [Code Review] Add voicemail message IDs where needed on startup; Add message IDs to ODBC voicemail   Matt Jordan
• [asterisk-dev] [Code Review] Add unique message IDs to IMAP voicemail   Matt Jordan
• [asterisk-dev] [Code Review] A new higher-level API for working with Asterisk configs, with example code in app_skel.c and udptl.c   Matt Jordan
• [asterisk-dev] [Code Review] Add documentation to function CHANNEL for options echocan_mode and buffers   Matt Jordan
• [asterisk-dev] [Code Review] Add unique message IDs to IMAP voicemail   Matt Jordan
• [asterisk-dev] EWS - Calendar Integration Probelm   Matthew Jordan
• [asterisk-dev] directmediapermit/deny doesn't work in any way approaching correct for any Asterisk version and is poorly named, so we should rework it in trunk   Matthew Jordan
• [asterisk-dev] directmediapermit/deny doesn't work in any way approaching correct for any Asterisk version and is poorly named, so we should rework it in trunk   Matthew Jordan
• [asterisk-dev] [Code Review]: ConfbridgeActionExec AMI Command   Matthew Jordan
• [asterisk-dev] Reload module   Matthew Jordan
• [asterisk-dev] Reload module   Matthew Jordan
• [asterisk-dev] asterisk.org documentation   Matthew Jordan
• [asterisk-dev] Wrong SIP to SIP SIP Cause mapping   Matthew Jordan
• [asterisk-dev] Wrong SIP to SIP SIP Cause mapping   Matthew Jordan
• [asterisk-dev] asterisk.org documentation   Matthew Jordan
• [asterisk-dev] cdr documentation - new fields   Matthew Jordan
• [asterisk-dev] RFC: chan_sip SDP parsing changes in behavior   Matthew Jordan
• [asterisk-dev] RFC: chan_sip SDP parsing changes in behavior   Matthew Jordan
• [asterisk-dev] Reload failed because retrieve_conf encountered an error: 1   Matthew Jordan
• [asterisk-dev] Remote MWI notification issue   Michael Keuter
• [asterisk-dev] Remote MWI notification issue   Michael Keuter
• [asterisk-dev] app Swift() hangs   Justin Killen
• [asterisk-dev] Asterisk disconnect active connection when SMS is arriving   Eugen Konkov
• [asterisk-dev] EWS - Calendar Integration Probelm   Bharat Lalcheta
• [asterisk-dev] [Code Review] unchecked_return audit and fixes   Tilghman Lesher
• [asterisk-dev] [Code Review]: unchecked_return audit and fixes   Tilghman Lesher
• [asterisk-dev] [Code Review]: unchecked_return audit and fixes   Tilghman Lesher
• [asterisk-dev] [Code Review] unchecked_return audit and fixes   Tilghman Lesher
• [asterisk-dev] [Code Review]: Pinequeue: Play queue prompts in the background - making call available to agent faster   Tilghman Lesher
• [asterisk-dev] [Code Review] Fix a variety of memory leaks   Tilghman Lesher
• [asterisk-dev] [Code Review]: Fix a variety of memory leaks   Tilghman Lesher
• [asterisk-dev] Missed ability to filter on sip debug   Konstantin M.
• [asterisk-dev] Add ability to set up a header (via SIPAddHeader) to Progress()   Konstantin M.
• [asterisk-dev] [Code Review]: Pinequeue: Play queue prompts in the background - making call available to agent faster   Leif Madsen
• [asterisk-dev] Pinequeue - to hold or not to hold   Leif Madsen
• [asterisk-dev] [Code Review] chan_jingle2: New Jingle + Google Talk channel driver   Leif Madsen
• [asterisk-dev] [Code Review]: chan_jingle2: New Jingle + Google Talk channel driver   Leif Madsen
• [asterisk-dev] [Code Review] Extend SIP REFER message originated by Transfer with SIPAddHeader-added headers   Marquis
• [asterisk-dev] [Code Review]: Extend SIP REFER message originated by Transfer with SIPAddHeader-added headers   Marquis
• [asterisk-dev] [Code Review]: Generate VMWI neon pulses from FXS module to light NEON lamp on older 'non intellegent phones'   Mark Michelson
• [asterisk-dev] [Code Review] Properly handle the linkedid for local channels and fix race condition for LINKEDID_END CEL event.   Mark Michelson
• [asterisk-dev] [Code Review] Properly handle the linkedid for local channels and fix race condition for LINKEDID_END CEL event.   Mark Michelson
• [asterisk-dev] [Code Review] ExternalIVR: Add IPv6 support   Mark Michelson
• [asterisk-dev] directmediapermit/deny doesn't work in any way approaching correct for any Asterisk version and is poorly named, so we should rework it in trunk   Mark Michelson
• [asterisk-dev] [Code Review] Add voicemail message IDs where needed on startup; Add message IDs to ODBC voicemail   Mark Michelson
• [asterisk-dev] [Code Review] Add voicemail message IDs where needed on startup; Add message IDs to ODBC voicemail   Mark Michelson
• [asterisk-dev] [Code Review] Add voicemail message IDs where needed on startup; Add message IDs to ODBC voicemail   Mark Michelson
• [asterisk-dev] [Code Review] Call ID logging phase III part 1 - channel driver specific logging (SIP, IAX2, and DAHDI)   Mark Michelson
• [asterisk-dev] [Code Review] Rework the leave_voicemail_contexts test   Mark Michelson
• [asterisk-dev] [Code Review] Update Unit Tests For Security Events Framework API - Trunk   Mark Michelson
• [asterisk-dev] [Code Review] Rework the leave_voicemail_contexts test   Mark Michelson
• [asterisk-dev] [Code Review]: Extend SIP REFER message originated by Transfer with SIPAddHeader-added headers   Mark Michelson
• [asterisk-dev] [Code Review] Add replacement for SIP_CAUSE   Mark Michelson
• [asterisk-dev] [Code Review] Fix local channel chains optimizing themselves out of a call.   Mark Michelson
• [asterisk-dev] [Code Review] ICE, STUN, and TURN Support   Mark Michelson
• [asterisk-dev] [Code Review] [confbridge] Behavioural correction for hold-music state when users join/part conferences in varying combinations   Mark Michelson
• [asterisk-dev] Remote MWI notification issue   Mark Michelson
• [asterisk-dev] [Code Review] Log 'stun set debug on' messages as ast_debug   Mark Michelson
• [asterisk-dev] [Code Review] Fix broken reinvite glare scenario   Mark Michelson
• [asterisk-dev] [Code Review] Add predial support to FollowMe.   Mark Michelson
• [asterisk-dev] [Code Review] Fix SIP directmedia's use of ACL so that the directmedia reachability of peers is checked in a sensible fashion   Mark Michelson
• [asterisk-dev] [Code Review] Fix potential deadlock between masquerade and chan_local.   Mark Michelson
• [asterisk-dev] [Code Review] Fix broken reinvite glare scenario   Mark Michelson
• [asterisk-dev] [Code Review] Add replacement for SIP_CAUSE   Mark Michelson
• [asterisk-dev] httpd bind port changed to 8080 in 1.8 [was: Re: [svn-commits] mmichelson: branch 1.8 r353770 - in /branches/1.8: ./ configs/ include/asteri...]   Mark Michelson
• [asterisk-dev] [Code Review] Adjust to allow for Digium phones' send to voicemail feature   Mark Michelson
• [asterisk-dev] [Code Review] Digium Phones "send to voicemail" tests   Mark Michelson
• [asterisk-dev] [Code Review] Digium Phones "send to voicemail" tests   Mark Michelson
• [asterisk-dev] [Code Review] Adjust to allow for Digium phones' send to voicemail feature   Mark Michelson
• [asterisk-dev] [Code Review]: Enable support for early media in AMI originate action   Mark Michelson
• [asterisk-dev] [Code Review] add test to assert that asterisk replies 481 to an invite with a to-tag   Mark Michelson
• [asterisk-dev] [Code Review] Fix for SIP peer's allowtransfer setting not being used   Mark Michelson
• [asterisk-dev] [Code Review] app_queue: Improvements to per member 'ringinuse' (formerly 'ignorebusy') option   Mark Michelson
• [asterisk-dev] [Code Review]: Digium Phones "send to voicemail" tests   Mark Michelson
• [asterisk-dev] [Code Review]: app_queue: Improvements to per member 'ringinuse' (formerly 'ignorebusy') option   Mark Michelson
• [asterisk-dev] [Code Review] Digium Phones "send to voicemail" tests   Mark Michelson
• [asterisk-dev] [Code Review]: Adjust to allow for Digium phones' send to voicemail feature   Mark Michelson
• [asterisk-dev] [Code Review] Fix a variety of memory leaks   Mark Michelson
• [asterisk-dev] [Code Review] Fix Dial I option ignored if dial forked and one fork redirects.   Mark Michelson
• [asterisk-dev] [Code Review]: Add voicemail message IDs where needed on startup; Add message IDs to ODBC voicemail   Mark Michelson
• [asterisk-dev] [Code Review] Fix a variety of memory leaks   Mark Michelson
• [asterisk-dev] [Code Review]: Add voicemail message IDs where needed on startup; Add message IDs to ODBC voicemail   Mark Michelson
• [asterisk-dev] [Code Review]: Add voicemail message IDs where needed on startup; Add message IDs to ODBC voicemail   Mark Michelson
• [asterisk-dev] [Code Review] chan_sip: Fix directmedia's use of ACL to limit remotebridging to certain host addresses for trunk   Mark Michelson
• [asterisk-dev] [Code Review] app_queue: Improvements to per member 'ringinuse' (formerly 'ignorebusy') option   Mark Michelson
• [asterisk-dev] [Code Review] chan_sip: Fix directmedia's use of ACL to limit remotebridging to certain host addresses for trunk   Mark Michelson
• [asterisk-dev] [Code Review] Start a library of common SIPp scenarios for the testsuite   Mark Michelson
• [asterisk-dev] [Code Review]: add test to assert that asterisk replies 481 to an invite with a to-tag   Mark Michelson
• [asterisk-dev] [Code Review]: add test to assert that asterisk replies 481 to an invite with a to-tag   Mark Michelson
• [asterisk-dev] Urgent development consultancy wanted   Mark Michelson
• [asterisk-dev] [Code Review] Add voicemail message IDs where needed on startup; Add message IDs to ODBC voicemail   Mark Michelson
• [asterisk-dev] [Code Review] Add voicemail message IDs where needed on startup; Add message IDs to ODBC voicemail   Mark Michelson
• [asterisk-dev] [Code Review] Update a peer's lastmsgssent value appropriately   Mark Michelson
• [asterisk-dev] [svn-commits] mmichelson: branch 1.8 r367002 - in /branches/1.8: channels/ include/asterisk...   Mark Michelson
• [asterisk-dev] [Code Review] Add unique message IDs to IMAP voicemail   Mark Michelson
• [asterisk-dev] [Code Review] Help mitigate reinvite glares in the SIP channel driver   Mark Michelson
• [asterisk-dev] [Code Review] add test to assert that asterisk replies 481 to an invite with a to-tag   Mark Michelson
• [asterisk-dev] [Code Review] Masquerade super test   Mark Michelson
• [asterisk-dev] [Code Review] chan_sip: Fix directmedia's use of ACL to limit remotebridging to certain host addresses for trunk   Mark Michelson
• [asterisk-dev] [Code Review]: chan_sip: Fix directmedia's use of ACL to limit remotebridging to certain host addresses for trunk   Mark Michelson
• [asterisk-dev] [Code Review] Masquerade super test   Mark Michelson
• [asterisk-dev] [Code Review] Fix for SIP peer's allowtransfer setting not being used   Mark Michelson
• [asterisk-dev] [Code Review] Fix Dial I option ignored if dial forked and one fork redirects.   Mark Michelson
• [asterisk-dev] [Code Review]: Help mitigate reinvite glares in the SIP channel driver   Mark Michelson
• [asterisk-dev] [Code Review]: Help mitigate reinvite glares in the SIP channel driver   Mark Michelson
• [asterisk-dev] [Code Review] Add IAX2 support for the new HANGUPCAUSE hash   Mark Michelson
• [asterisk-dev] [Code Review] Patch to detect/parse ANI-II / ANI2 / OLI from SIP INVITE messages   Mark Michelson
• [asterisk-dev] [Code Review] Patch to detect/parse ANI-II / ANI2 / OLI from SIP INVITE messages   Mark Michelson
• [asterisk-dev] [Code Review] Add tests for the IAX2 implementation of the HANGUPCAUSE hash   Mark Michelson
• [asterisk-dev] [Code Review]: Help mitigate reinvite glares in the SIP channel driver   Mark Michelson
• [asterisk-dev] [Code Review]: Help mitigate reinvite glares in the SIP channel driver   Mark Michelson
• [asterisk-dev] [Code Review] Help mitigate reinvite glares in the SIP channel driver   Mark Michelson
• [asterisk-dev] [Code Review] Help mitigate reinvite glares in the SIP channel driver   Mark Michelson
• [asterisk-dev] [Code Review]: Help mitigate reinvite glares in the SIP channel driver   Mark Michelson
• [asterisk-dev] [Code Review] logger: Callid logging phase IV - chan_local, chan_agent, bridging and autoservice.   Mark Michelson
• [asterisk-dev] [Code Review] WebSocket HTTP Module   Mark Michelson
• [asterisk-dev] [Code Review] Help mitigate reinvite glares: Attempt 2   Mark Michelson
• [asterisk-dev] [Code Review] logger: Callid logging phase IV - chan_local, chan_agent, bridging and autoservice.   Mark Michelson
• [asterisk-dev] [Code Review]: Add unique message IDs to IMAP voicemail   Mark Michelson
• [asterisk-dev] [Code Review]: Add unique message IDs to IMAP voicemail   Mark Michelson
• [asterisk-dev] [Code Review] Add unique message IDs to IMAP voicemail   Mark Michelson
• [asterisk-dev] Wrong SIP to SIP SIP Cause mapping   Ryan Mitchell
• [asterisk-dev] [Code Review]: ConfbridgeActionExec AMI Command   Tony Mountifield
• [asterisk-dev] Wrong SIP to SIP SIP Cause mapping   alexandre Moutot
• [asterisk-dev] Wrong SIP to SIP SIP Cause mapping   alexandre Moutot
• [asterisk-dev] Wrong SIP to SIP SIP Cause mapping   alexandre Moutot
• [asterisk-dev] Wrong SIP to SIP SIP Cause mapping   alexandre Moutot
• [asterisk-dev] Maximum allowed/recommended size of AMI action?   Richard Mudgett
• [asterisk-dev] "Sending Complete" IE status availability ?   Richard Mudgett
• [asterisk-dev] Maximum allowed/recommended size of AMI action?   Yaroslav Panych
• [asterisk-dev] Maximum allowed/recommended size of AMI action?   Yaroslav Panych
• [asterisk-dev] Reload module   Yaroslav Panych
• [asterisk-dev] Reload module   Yaroslav Panych
• [asterisk-dev] Remote MWI notification issue   Josh Patten
• [asterisk-dev] Remote MWI notification issue   Josh Patten
• [asterisk-dev] Remote MWI notification issue   Josh Patten
• [asterisk-dev] Remote MWI notification issue   Josh Patten
• [asterisk-dev] [Code Review] Adjust to allow for Digium phones' send to voicemail feature   Simon Perreault
• [asterisk-dev] directmediapermit/deny doesn't work in any way approaching correct for any Asterisk version and is poorly named, so we should rework it in trunk   Jonathan Rose
• [asterisk-dev] directmediapermit/deny doesn't work in any way approaching correct for any Asterisk version and is poorly named, so we should rework it in trunk   Jonathan Rose
• [asterisk-dev] directmediapermit/deny doesn't work in any way approaching correct for any Asterisk version and is poorly named, so we should rework it in trunk   Jonathan Rose
• [asterisk-dev] directmediapermit/deny doesn't work in any way approaching correct for any Asterisk version and is poorly named, so we should rework it in trunk   Jonathan Rose
• [asterisk-dev] Pinequeue - to hold or not to hold   Jonathan Rose
• [asterisk-dev] Urgent development consultancy wanted   Shaun Ruffell
• [asterisk-dev] Urgent development consultancy wanted   Shaun Ruffell
• [asterisk-dev] Remote MWI notification issue   Stefan Schmidt
• [asterisk-dev] sqlite3 DB memory leaking?   Stefan Schmidt
• [asterisk-dev] sqlite3 DB memory leaking?   Stefan Schmidt
• [asterisk-dev] Disable global atxfernoanswertimeout   Thomas Sevestre
• [asterisk-dev] sqlite3 DB memory leaking?   Jamuel Starkey
• [asterisk-dev] [Code Review] [confbridge] Behavioural correction for hold-music state when users join/part conferences in varying combinations   Neil Tallim
• [asterisk-dev] Asterisk 1.8.12.0-rc3 Now Available   Asterisk Development Team
• [asterisk-dev] Asterisk 10.4.0-rc3 Now Available   Asterisk Development Team
• [asterisk-dev] Asterisk 1.8.12.0 Now Available   Asterisk Development Team
• [asterisk-dev] Asterisk 10.4.0 Now Available   Asterisk Development Team
• [asterisk-dev] Asterisk 1.8.13.0-rc1 Now Available   Asterisk Development Team
• [asterisk-dev] Asterisk 10.5.0-rc1 Now Available   Asterisk Development Team
• [asterisk-dev] Planned service outage for community services   Asterisk Development Team
• [asterisk-dev] Certified Asterisk 1.8.11-cert2; Asterisk 1.8.12.1, 10.4.1 Now Available (Security Release)   Asterisk Development Team
• [asterisk-dev] Asterisk 1.8.12.2 Now Available   Asterisk Development Team
• [asterisk-dev] Asterisk 10.4.2 Now Available   Asterisk Development Team
• [asterisk-dev] Asterisk 1.8.13.0-rc2 Now Available   Asterisk Development Team
• [asterisk-dev] Asterisk 10.5.0-rc2 Now Available   Asterisk Development Team
• [asterisk-dev] AST-2012-007: Remote crash vulnerability in IAX2 channel driver.   Asterisk Security Team
• [asterisk-dev] AST-2012-008: Skinny Channel Driver Remote Crash Vulnerability   Asterisk Security Team
• [asterisk-dev] "Sending Complete" IE status availability ?   Timo Teras
• [asterisk-dev] Merci   Steve Totaro
• [asterisk-dev] An idea: Interception macro for registering overlap dialling   Pavel Troller
• [asterisk-dev] "Sending Complete" IE status availability ?   Pavel Troller
• [asterisk-dev] "Sending Complete" IE status availability ?   Pavel Troller
• [asterisk-dev] Wrong SIP to SIP SIP Cause mapping   Pavel Troller
• [asterisk-dev] Wrong SIP to SIP SIP Cause mapping   Pavel Troller
• [asterisk-dev] Wrong SIP to SIP SIP Cause mapping   Pavel Troller
• [asterisk-dev] Wrong SIP to SIP SIP Cause mapping   Pavel Troller
• [asterisk-dev] [Code Review]: ICE, STUN, and TURN Support   Terry Wilson
• [asterisk-dev] EWS - Calendar Integration Probelm   Terry Wilson
• [asterisk-dev] [Code Review] Properly handle the linkedid for local channels and fix race condition for LINKEDID_END CEL event.   Terry Wilson
• [asterisk-dev] [Code Review] Properly handle the linkedid for local channels and fix race condition for LINKEDID_END CEL event.   Terry Wilson
• [asterisk-dev] [Code Review] Extend SIP REFER message originated by Transfer with SIPAddHeader-added headers   Terry Wilson
• [asterisk-dev] [Code Review] ICE, STUN, and TURN Support   Terry Wilson
• [asterisk-dev] [Code Review]: ICE, STUN, and TURN Support   Terry Wilson
• [asterisk-dev] [Code Review] Two new CEL-related fixes   Terry Wilson
• [asterisk-dev] [Code Review] Add CEL testing to the testsuite   Terry Wilson
• [asterisk-dev] [Code Review] Two new CEL-related fixes   Terry Wilson
• [asterisk-dev] [Code Review]: Two new CEL-related fixes   Terry Wilson
• [asterisk-dev] [Code Review]: Two new CEL-related fixes   Terry Wilson
• [asterisk-dev] [Code Review]: Two new CEL-related fixes   Terry Wilson
• [asterisk-dev] [Code Review]: Add CEL testing to the testsuite   Terry Wilson
• [asterisk-dev] [Code Review] Add CEL testing to the testsuite   Terry Wilson
• [asterisk-dev] [Code Review] A new higher-level API for working with Asterisk configs, with example code in app_skel.c and udptl.c   Terry Wilson
• [asterisk-dev] [Code Review] A new higher-level API for working with Asterisk configs, with example code in app_skel.c and udptl.c   Terry Wilson
• [asterisk-dev] [Code Review]: A new higher-level API for working with Asterisk configs, with example code in app_skel.c and udptl.c   Terry Wilson
• [asterisk-dev] [Code Review] Change ao2 global array to ao2 global object holder.   Terry Wilson
• [asterisk-dev] [Code Review]: A new higher-level API for working with Asterisk configs, with example code in app_skel.c and udptl.c   Terry Wilson
• [asterisk-dev] [Code Review] Don't use a variable after calling ASTOBJ_UNREF on it.   Terry Wilson
• [asterisk-dev] [Code Review] A new higher-level API for working with Asterisk configs, with example code in app_skel.c and udptl.c   Terry Wilson
• [asterisk-dev] [Code Review] Two new CEL-related fixes   Terry Wilson
• [asterisk-dev] [Code Review]: Add CEL testing to the testsuite   Terry Wilson
• [asterisk-dev] [Code Review]: Don't use a variable after calling ASTOBJ_UNREF on it.   Terry Wilson
• [asterisk-dev] [Code Review]: Two new CEL-related fixes   Terry Wilson
• [asterisk-dev] [Code Review]: Two new CEL-related fixes   Terry Wilson
• [asterisk-dev] [Code Review] Two new CEL-related fixes   Terry Wilson
• [asterisk-dev] [Code Review] A new higher-level API for working with Asterisk configs, with example code in app_skel.c and udptl.c   Terry Wilson
• [asterisk-dev] [Code Review]: A new higher-level API for working with Asterisk configs, with example code in app_skel.c and udptl.c   Terry Wilson
• [asterisk-dev] Pinequeue - to hold or not to hold   Bryant Zimmerman
• [asterisk-dev] [Code Review] chan_jingle2: New Jingle + Google Talk channel driver   sean darcy
• [asterisk-dev] [Code Review] chan_jingle2: New Jingle + Google Talk channel driver   sean darcy
• [asterisk-dev] [Code Review] Update Unit Tests For Security Events Framework API - Trunk   elguero
• [asterisk-dev] [Code Review] Fix broken reinvite glare scenario   elguero
• [asterisk-dev] [Code Review] Fix for SIP peer's allowtransfer setting not being used   elguero
• [asterisk-dev] [Code Review]: Fix for SIP peer's allowtransfer setting not being used   elguero
• [asterisk-dev] [Code Review] Fix for SIP peer's allowtransfer setting not being used   elguero
• [asterisk-dev] [Code Review] Fix for SIP peer's allowtransfer setting not being used   elguero
• [asterisk-dev] [Code Review] Add documentation to function CHANNEL for options echocan_mode and buffers   elguero
• [asterisk-dev] [Code Review] Add documentation to function CHANNEL for options echocan_mode and buffers   elguero
• [asterisk-dev] Merci   yaherve at gmail.com
• [asterisk-dev] [Code Review]: Call ID logging phase III part 1 - channel driver specific logging (SIP, IAX2, and DAHDI)   jrose
• [asterisk-dev] [Code Review] Fix SIP directmedia's use of ACL so that the directmedia reachability of peers is checked in a sensible fashion   jrose
• [asterisk-dev] [Code Review] unchecked_return audit and fixes   jrose
• [asterisk-dev] [Code Review]: unchecked_return audit and fixes   jrose
• [asterisk-dev] [Code Review]: unchecked_return audit and fixes   jrose
• [asterisk-dev] [Code Review] unchecked_return audit and fixes   jrose
• [asterisk-dev] [Code Review] unchecked_return audit and fixes   jrose
• [asterisk-dev] [Code Review]: unchecked_return audit and fixes   jrose
• [asterisk-dev] [Code Review]: unchecked_return audit and fixes   jrose
• [asterisk-dev] [Code Review] unchecked_return audit and fixes   jrose
• [asterisk-dev] [Code Review] UNINIT audit and fixes   jrose
• [asterisk-dev] [Code Review] Call ID logging phase III part 1 - channel driver specific logging (SIP, IAX2, and DAHDI)   jrose
• [asterisk-dev] [Code Review] app_queue: Improvements to per member 'ringinuse' (formerly 'ignorebusy') option   jrose
• [asterisk-dev] [Code Review] app_queue: Improvements to per member 'ringinuse' (formerly 'ignorebusy') option   jrose
• [asterisk-dev] [Code Review] chan_sip: Fix directmedia's use of ACL to limit remotebridging to certain host addresses for trunk   jrose
• [asterisk-dev] [Code Review] chan_sip: Fix directmedia's use of ACL to limit remotebridging to certain host addresses for trunk   jrose
• [asterisk-dev] [Code Review] logger: Call ID logging phase III part 1 split - chan_iax2 and chan_dahdi   jrose
• [asterisk-dev] [Code Review]: app_queue: Improvements to per member 'ringinuse' (formerly 'ignorebusy') option   jrose
• [asterisk-dev] [Code Review] app_queue: Improvements to per member 'ringinuse' (formerly 'ignorebusy') option   jrose
• [asterisk-dev] [Code Review] Fix SIP directmedia's use of ACL so that the directmedia reachability of peers is checked in a sensible fashion   jrose
• [asterisk-dev] [Code Review] chan_sip: Fix directmedia's use of ACL to limit remotebridging to certain host addresses for trunk   jrose
• [asterisk-dev] [Code Review]: chan_sip: Fix directmedia's use of ACL to limit remotebridging to certain host addresses for trunk   jrose
• [asterisk-dev] [Code Review] chan_sip: Fix directmedia's use of ACL to limit remotebridging to certain host addresses for trunk   jrose
• [asterisk-dev] [Code Review] app_queue: Improvements to per member 'ringinuse' (formerly 'ignorebusy') option   jrose
• [asterisk-dev] [Code Review]: chan_sip: Fix directmedia's use of ACL to limit remotebridging to certain host addresses for trunk   jrose
• [asterisk-dev] [Code Review] chan_sip: Fix directmedia's use of ACL to limit remotebridging to certain host addresses for trunk   jrose
• [asterisk-dev] [Code Review]: logger: Call ID logging phase III part 1 split - chan_iax2 and chan_dahdi   jrose
• [asterisk-dev] [Code Review]: logger: Call ID logging phase III part 1 split - chan_iax2 and chan_dahdi   jrose
• [asterisk-dev] [Code Review]: chan_sip: Fix directmedia's use of ACL to limit remotebridging to certain host addresses for trunk   jrose
• [asterisk-dev] [Code Review] logger: Call ID logging phase III part 1 split - chan_iax2 and chan_dahdi   jrose
• [asterisk-dev] [Code Review] logger: Call ID logging phase III part 1 split - chan_iax2 and chan_dahdi   jrose
• [asterisk-dev] [Code Review]: logger: Call ID logging phase III part 1 split - chan_iax2 and chan_dahdi   jrose
• [asterisk-dev] [Code Review] logger: Callid logging phase 4: specific changes for chan_local, chan_agent, bridging and autoservice.   jrose
• [asterisk-dev] [Code Review] logger: Callid logging phase IV: specific changes for chan_local, chan_agent, bridging and autoservice.   jrose
• [asterisk-dev] [Code Review] logger: Callid logging phase IV - chan_local, chan_agent, bridging and autoservice.   jrose
• [asterisk-dev] [Code Review]: logger: Callid logging phase IV - chan_local, chan_agent, bridging and autoservice.   jrose
• [asterisk-dev] [Code Review]: logger: Call ID logging phase III part 1 split - chan_iax2 and chan_dahdi   jrose
• [asterisk-dev] [Code Review] Switch public voicemail APIs over to use unique message IDs instead of message indexes.   opticron
• [asterisk-dev] [Code Review]: Add replacement for SIP_CAUSE   opticron
• [asterisk-dev] [Code Review] Add replacement for SIP_CAUSE   opticron
• [asterisk-dev] [Code Review] ConfbridgeActionExec AMI Command   opticron
• [asterisk-dev] [Code Review] Properly support the d() option in VoiceMail   opticron
• [asterisk-dev] [Code Review] UNINIT audit and fixes   opticron
• [asterisk-dev] [Code Review] Run predial routine on local; 2 channel where you would expect.   opticron
• [asterisk-dev] [Code Review] Fix broken reinvite glare scenario   opticron
• [asterisk-dev] [Code Review] ConfbridgeExecAction AMI Command   opticron
• [asterisk-dev] [Code Review] ConfbridgeExecAction AMI Command   opticron
• [asterisk-dev] [Code Review] Fix a variety of memory leaks   opticron
• [asterisk-dev] [Code Review] Add IAX2 support for the new HANGUPCAUSE hash   opticron
• [asterisk-dev] [Code Review] Add tests for the IAX2 implementation of the HANGUPCAUSE hash   opticron
• [asterisk-dev] [Code Review] Add tests for the IAX2 implementation of the HANGUPCAUSE hash   opticron
• [asterisk-dev] [Code Review] Add IAX2 support for the new HANGUPCAUSE hash   opticron
• [asterisk-dev] [Code Review] Add tests for the IAX2 implementation of the HANGUPCAUSE hash   opticron
• [asterisk-dev] [Code Review] Don't use a variable after calling ASTOBJ_UNREF on it.   opticron
• [asterisk-dev] [Code Review] Start a library of common SIPp scenarios for the testsuite   opticron
• [asterisk-dev] [Code Review] Digium Phones "send to voicemail" tests   opticron
• [asterisk-dev] [Code Review] ICE, STUN, and TURN Support   opticron
• [asterisk-dev] [Code Review] chan_sip: Fix directmedia's use of ACL to limit remotebridging to certain host addresses for trunk   opticron
• [asterisk-dev] [Code Review]: chan_sip: Fix directmedia's use of ACL to limit remotebridging to certain host addresses for trunk   opticron
• [asterisk-dev] [Code Review] Fix for SIP peer's allowtransfer setting not being used   opticron
• [asterisk-dev] [Code Review]: Add IAX2 support for the new HANGUPCAUSE hash   opticron
• [asterisk-dev] [Code Review] Add IAX2 support for the new HANGUPCAUSE hash   opticron
• [asterisk-dev] [Code Review] Remove AST_FLAG_ANSWERED_ELSEWHERE, duplicating the functionality of AST_CAUSE_ANSWERED_ELSEWHERE   opticron
• [asterisk-dev] [Code Review] Improve SDP parsing warning messages and RFC compliance   opticron
• [asterisk-dev] [Code Review] Add chan_dahdi Who Hung Up? implementation for analog and PRI/ISDN   opticron
• [asterisk-dev] [Code Review] Add unique message IDs to IMAP voicemail   opticron
• [asterisk-dev] [Code Review] Add test for correct HANGUPCAUSE after SIP 404   opticron
• [asterisk-dev] [Code Review] Add cause reporting to sig_ss7 for chan_dahdi   opticron
• [asterisk-dev] [Code Review] logger: Call ID logging phase III part 1 split - chan_iax2 and chan_dahdi   opticron
• [asterisk-dev] [Code Review] Fix local channel chains optimizing themselves out of a call.   rmudgett
• [asterisk-dev] [Code Review] Properly handle the linkedid for local channels and fix race condition for LINKEDID_END CEL event.   rmudgett
• [asterisk-dev] [Code Review] Fix local channel chains optimizing themselves out of a call.   rmudgett
• [asterisk-dev] [Code Review]: Fix local channel chains optimizing themselves out of a call.   rmudgett
• [asterisk-dev] [Code Review] Fix local channel chains optimizing themselves out of a call.   rmudgett
• [asterisk-dev] [Code Review]: Fix local channel chains optimizing themselves out of a call.   rmudgett
• [asterisk-dev] [Code Review]: Fix local channel chains optimizing themselves out of a call.   rmudgett
• [asterisk-dev] [Code Review] Two new CEL-related fixes   rmudgett
• [asterisk-dev] [Code Review]: Two new CEL-related fixes   rmudgett
• [asterisk-dev] [Code Review] Two new CEL-related fixes   rmudgett
• [asterisk-dev] [Code Review]: Two new CEL-related fixes   rmudgett
• [asterisk-dev] [Code Review] Run predial routine on local; 2 channel where you would expect.   rmudgett
• [asterisk-dev] [Code Review] Add predial support to FollowMe.   rmudgett
• [asterisk-dev] [Code Review] Fix potential deadlock between masquerade and chan_local.   rmudgett
• [asterisk-dev] [Code Review]: Fix potential deadlock between masquerade and chan_local.   rmudgett
• [asterisk-dev] [Code Review] Fix Dial I option ignored if dial forked and one fork redirects.   rmudgett
• [asterisk-dev] [Code Review] Change ao2 global array to ao2 global object holder.   rmudgett
• [asterisk-dev] [Code Review] Masquerade super test   rmudgett
• [asterisk-dev] [Code Review] logger: Call ID logging phase III part 1 split - chan_iax2 and chan_dahdi   rmudgett
• [asterisk-dev] [Code Review]: Fix Dial I option ignored if dial forked and one fork redirects.   rmudgett
• [asterisk-dev] [Code Review] Fix Dial I option ignored if dial forked and one fork redirects.   rmudgett
• [asterisk-dev] [Code Review] Don't use a variable after calling ASTOBJ_UNREF on it.   rmudgett
• [asterisk-dev] [Code Review] Two new CEL-related fixes   rmudgett
• [asterisk-dev] [Code Review]: Don't use a variable after calling ASTOBJ_UNREF on it.   rmudgett
• [asterisk-dev] [Code Review]: Two new CEL-related fixes   rmudgett
• [asterisk-dev] [Code Review] Two new CEL-related fixes   rmudgett
• [asterisk-dev] [Code Review]: Masquerade super test   rmudgett
• [asterisk-dev] [Code Review] Masquerade super test   rmudgett
• [asterisk-dev] [Code Review]: Add IAX2 support for the new HANGUPCAUSE hash   rmudgett
• [asterisk-dev] [Code Review] Patch to detect/parse ANI-II / ANI2 / OLI from SIP INVITE messages   rmudgett
• [asterisk-dev] [Code Review] add test to assert that asterisk replies 481 to an invite with a to-tag   wdoekes
• [asterisk-dev] [Code Review]: add test to assert that asterisk replies 481 to an invite with a to-tag   wdoekes
• [asterisk-dev] [Code Review] Add test for correct HANGUPCAUSE after SIP 404   wdoekes
• [asterisk-dev] [Code Review] Add test for correct HANGUPCAUSE after SIP 404   wdoekes
• [asterisk-dev] [Code Review] Add test for correct HANGUPCAUSE after SIP 404   wdoekes
• [asterisk-dev] [Code Review] WebSocket HTTP Module   wdoekes
• [asterisk-dev] [Code Review] WebSocket HTTP Module   wdoekes
• [asterisk-dev] [Code Review] WebSocket HTTP Module   wdoekes
• [asterisk-dev] [Code Review]: Add test for correct HANGUPCAUSE after SIP 404   wdoekes
• [asterisk-dev] [Code Review] Add test for correct HANGUPCAUSE after SIP 404   wdoekes
• [asterisk-dev] Reload failed because retrieve_conf encountered an error: 1   white.heron white
• [asterisk-dev] asterisk-dev Digest, Vol 94, Issue 36   bell_n_tell786 at yahoo.com
• [asterisk-dev] asterisk-dev Digest, Vol 94, Issue 37   bell_n_tell786 at yahoo.com

Last message date: Thu May 31 16:37:04 CDT 2012
Archived on: Thu May 31 16:47:36 CDT 2012

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