November 2011 Archives by author

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Starting: Tue Nov 1 01:18:10 CDT 2011
Ending: Wed Nov 30 19:33:34 CST 2011
Messages: 497

• [asterisk-dev] $ offer for 2 modifications in chan_dongle   S Adrian
• [asterisk-dev] asterisk busy tone   Yaprak Ayazoglu
• [asterisk-dev] Summary: SIP, NAT, security concerns, oh my!   Bruce B
• [asterisk-dev] Summary: SIP, NAT, security concerns, oh my!   Bruce B
• [asterisk-dev] Summary: SIP, NAT, security concerns, oh my!   Bruce B
• [asterisk-dev] [svn-commits] mjordan: testsuite/asterisk/trunk r2810 - /asterisk/trunk/tests/channels/SIP/...   Paul Belanger
• [asterisk-dev] [Code Review] Add a test for verifying when username information can be leaked via differing nat settings   Paul Belanger
• [asterisk-dev] [Code Review] Add #includeif statement   Paul Belanger
• [asterisk-dev] [Code Review] Add #tryinclude statement   Paul Belanger
• [asterisk-dev] Intended sip.conf matching behaviour?   Paul Belanger
• [asterisk-dev] [Code Review] update samples to use [public] context   Paul Belanger
• [asterisk-dev] JIRA Question   Russell Brown
• [asterisk-dev] Scheduler and event callbacks - what thread?   Russell Bryant
• [asterisk-dev] Developer   James Cloos
• [asterisk-dev] [Code Review] sparc64 compile error, undefined reference to `__sync_fetch_and_add_4'   Tzafrir Cohen
• [asterisk-dev] [Code Review]: sparc64 compile error, undefined reference to `__sync_fetch_and_add_4'   Tzafrir Cohen
• [asterisk-dev] Optional api and weak symbol problem   Tzafrir Cohen
• [asterisk-dev] [Code Review] chan_dahdi: create and destroy channels at run-time   Tzafrir Cohen
• [asterisk-dev] [Code Review] Separate verbose level for logging   Tzafrir Cohen
• [asterisk-dev] [Code Review]: Separate verbose level for logging   Tzafrir Cohen
• [asterisk-dev] [Code Review]: chan_dahdi: create and destroy channels at run-time   Tzafrir Cohen
• [asterisk-dev] [Code Review] chan_dahdi: create and destroy channels at run-time   Tzafrir Cohen
• [asterisk-dev] [Code Review] safe_asterisk: Allow overriding TTY and CONSOLE in startup.d   Tzafrir Cohen
• [asterisk-dev] asterisk-core-sounds-ru   Tzafrir Cohen
• [asterisk-dev] How to save CLI log to file   Saúl Ibarra Corretgé
• [asterisk-dev] $500 bounty for #18039: Realtime music restarts from beginning each time   Alistair Cunningham
• [asterisk-dev] $500 bounty for #18039: Realtime music restarts from beginning each time   Alistair Cunningham
• [asterisk-dev] $500 bounty for #18039: Realtime music restarts from beginning each time   Alistair Cunningham
• [asterisk-dev] Intended sip.conf matching behaviour?   Alistair Cunningham
• [asterisk-dev] Intended sip.conf matching behaviour?   Alistair Cunningham
• [asterisk-dev] [Code Review]: sparc64 compile error, undefined reference to `__sync_fetch_and_add_4'   Alec Davis
• [asterisk-dev] [asterisk-commits] wdoekes: trunk r343163 - in /trunk: ./ include/asterisk/ main/   Walter Doekes
• [asterisk-dev] changing sizeof to __alignof__   Walter Doekes
• [asterisk-dev] Summary: SIP, NAT, security concerns, oh my!   Walter Doekes
• [asterisk-dev] asterisk dial plan questions   Walter Doekes
• [asterisk-dev] Error loading module 'chan_sip.so': /usr/lib/asterisk/modules/chan_sip.so: un defined symbol   Walter Doekes
• [asterisk-dev] limiting SIP MESSAGE to 1300 bytes (was: fix odd logic in get_msg_text() and a couple of related quirks)   Walter Doekes
• [asterisk-dev] [Code Review] use autoconf to detect whether alignment should be enforced   Kevin Fleming
• [asterisk-dev] [Code Review] use __alignof__ instead of sizeof for alignment of ast_string_field_allocation   Kevin Fleming
• [asterisk-dev] externaddr in chan_sip.c - locking required?   Kevin P. Fleming
• [asterisk-dev] [Code Review] NetBorder Call Analyzer (Answering Machine Detection) support   Kevin P. Fleming
• [asterisk-dev] Memory failure - channels in meetme   Kevin P. Fleming
• [asterisk-dev] [asterisk-commits] wdoekes: trunk r343163 - in /trunk: ./ include/asterisk/ main/   Kevin P. Fleming
• [asterisk-dev] [asterisk-commits] wdoekes: trunk r343163 - in /trunk: ./ include/asterisk/ main/   Kevin P. Fleming
• [asterisk-dev] [Code Review]: Add #includeif statement   Kevin P. Fleming
• [asterisk-dev] Setvar= for manager accounts   Kevin P. Fleming
• [asterisk-dev] Optional api and weak symbol problem   Kevin P. Fleming
• [asterisk-dev] Optional api and weak symbol problem   Kevin P. Fleming
• [asterisk-dev] changing sizeof to __alignof__   Kevin P. Fleming
• [asterisk-dev] SIP, NAT, security concerns, oh my!   Kevin P. Fleming
• [asterisk-dev] Summary: SIP, NAT, security concerns, oh my!   Kevin P. Fleming
• [asterisk-dev] Memory failure - channels in meetme   Kevin P. Fleming
• [asterisk-dev] FOSDEM 2012 Telephony/Communications Devroom Call for Presenters   Kevin P. Fleming
• [asterisk-dev] FOSDEM 2012 Telephony/Communications Devroom Call for Presenters   Kevin P. Fleming
• [asterisk-dev] IAX Hardphone (again)   Kevin P. Fleming
• [asterisk-dev] bounty for ASTERISK-17474 streaming MusicOnHold bug   Luke Hamburg
• [asterisk-dev] How to save CLI log to file   SIP IMS
• [asterisk-dev] [Code Review]: Just remove the silly SIP registertrying option   Olle E Johansson
• [asterisk-dev] [Code Review] BLF Subscriptions Causes SIP Deadlock   Olle E Johansson
• [asterisk-dev] [Code Review]: chan_sip: removing some non-compliant code in v10 and minor fixes   Olle E Johansson
• [asterisk-dev] [Code Review]: Asterisk Support of SIP Connect 1.1   Olle E Johansson
• [asterisk-dev] [Code Review]: Asterisk Support of SIP Connect 1.1   Olle E Johansson
• [asterisk-dev] [Code Review]: Asterisk Support of SIP Connect 1.1   Olle E Johansson
• [asterisk-dev] [Code Review]: chan_sip: removing some non-compliant code in v10 and minor fixes   Olle E Johansson
• [asterisk-dev] [Code Review]: Asterisk Support of SIP Connect 1.1   Olle E Johansson
• [asterisk-dev] [Code Review] Separate verbose level for logging   Olle E Johansson
• [asterisk-dev] [Code Review]: Preserve DTMF length in main/features.c   Olle E. Johansson
• [asterisk-dev] New Asterisk community developers   Olle E. Johansson
• [asterisk-dev] Summary: SIP, NAT, security concerns, oh my!   Olle E. Johansson
• [asterisk-dev] Memory failure - channels in meetme   Olle E. Johansson
• [asterisk-dev] $500 bounty for #18039: Realtime music restarts from beginning each time   Olle E. Johansson
• [asterisk-dev] Asterisk World 2012 Call for Papers   Bryan Johns
• [asterisk-dev] tutor needed   Edward de Jong
• [asterisk-dev] how to link FreePBX IVR step to something inside extensions-custom.conf?   Edward de Jong
• [asterisk-dev] asterisk dial plan questions   Edward de Jong
• [asterisk-dev] ring groups   Edward de Jong
• [asterisk-dev] [Code Review] Add mute all participants; play participant count to ConfBridge   Matthew Jordan
• [asterisk-dev] Memory failure - channels in meetme   Matthew Jordan
• [asterisk-dev] [Bamboo] Matthew Jordan commented on Asterisk - Trunk - Ubuntu Lucid (10.04) 1227   Matthew Jordan
• [asterisk-dev] [Bamboo] Matthew Jordan commented on Asterisk - 10 - Ubuntu Lucid (10.04) - amd64 357   Matthew Jordan
• [asterisk-dev] [Bamboo] Matthew Jordan commented on Asterisk - Trunk - Ubuntu Lucid (10.04) - amd64 1250   Matthew Jordan
• [asterisk-dev] Asterisk / FreeSWITCH Developer required..   Robert Kirkby
• [asterisk-dev] asterisk-core-sounds-ru   Andrew Latham
• [asterisk-dev] [Code Review]: ensure that ast_string_field_pool base + used is always aligned   Tilghman Lesher
• [asterisk-dev] [Code Review]: Add #includeif statement   Tilghman Lesher
• [asterisk-dev] [Code Review] fix default udptl port range which differs in source / udptl.conf.sample   Tilghman Lesher
• [asterisk-dev] [Code Review]: Added VM_INFO dialplan function to return voicemail user information   Tilghman Lesher
• [asterisk-dev] [Code Review]: Added VM_INFO dialplan function to return voicemail user information   Tilghman Lesher
• [asterisk-dev] [Code Review] Eliminate redundant and possibly dangerous close/fclose pairs in chan_sip and tcptls.   Tilghman Lesher
• [asterisk-dev] [Code Review]: Restore SIP DTMF overlap dialing method.   Tilghman Lesher
• [asterisk-dev] tutor needed   Tilghman Lesher
• [asterisk-dev] [Code Review] Permit/deny negation, plus multiple-specification with commas   Tilghman Lesher
• [asterisk-dev] [Code Review] Update astdb documentation to how it actually works   Tilghman Lesher
• [asterisk-dev] [Code Review] Separate verbose level for logging   Tilghman Lesher
• [asterisk-dev] [Code Review] Separate verbose level for logging   Tilghman Lesher
• [asterisk-dev] [Code Review] Separate verbose level for logging   Tilghman Lesher
• [asterisk-dev] [Code Review] Separate verbose level for logging   Tilghman Lesher
• [asterisk-dev] [Code Review] Separate verbose level for logging   Tilghman Lesher
• [asterisk-dev] [Code Review] Separate verbose level for logging   Tilghman Lesher
• [asterisk-dev] [Code Review] Separate verbose level for logging   Tilghman Lesher
• [asterisk-dev] [Code Review] Separate verbose level for logging   Tilghman Lesher
• [asterisk-dev] [Code Review]: Separate verbose level for logging   Tilghman Lesher
• [asterisk-dev] [Code Review] Separate verbose level for logging   Tilghman Lesher
• [asterisk-dev] [Code Review] Separate verbose level for logging   Tilghman Lesher
• [asterisk-dev] [Code Review]: Separate verbose level for logging   Tilghman Lesher
• [asterisk-dev] [Code Review]: Separate verbose level for logging   Tilghman Lesher
• [asterisk-dev] [Code Review] Separate verbose level for logging   Tilghman Lesher
• [asterisk-dev] [Code Review] Separate verbose level for logging   Tilghman Lesher
• [asterisk-dev] [Code Review]: Separate verbose level for logging   Tilghman Lesher
• [asterisk-dev] [Code Review]: Separate verbose level for logging   Tilghman Lesher
• [asterisk-dev] [Code Review] Separate verbose level for logging   Tilghman Lesher
• [asterisk-dev] [Code Review] Separate verbose level for logging   Tilghman Lesher
• [asterisk-dev] [Code Review]: Separate verbose level for logging   Tilghman Lesher
• [asterisk-dev] [Code Review] Separate verbose level for logging   Tilghman Lesher
• [asterisk-dev] [Code Review] Separate verbose level for logging   Tilghman Lesher
• [asterisk-dev] [Code Review] Separate verbose level for logging   Tilghman Lesher
• [asterisk-dev] [Code Review] Separate verbose level for logging   Tilghman Lesher
• [asterisk-dev] [Code Review] safe_asterisk: Allow overriding TTY and CONSOLE in startup.d   Tilghman Lesher
• [asterisk-dev] [Code Review] update samples to use [public] context   Tilghman Lesher
• [asterisk-dev] [Code Review] NetBorder Call Analyzer (Answering Machine Detection) support   Konstantin M.
• [asterisk-dev] JIRA Question   Leif Madsen
• [asterisk-dev] [Code Review] Change the default nat setting to "yes"   Leif Madsen
• [asterisk-dev] How to get ast_debug() logged   Mark Michelson
• [asterisk-dev] [Code Review] Add mute all participants; play participant count to ConfBridge   Tony Mountifield
• [asterisk-dev] [Code Review] Add mute all participants; play participant count to ConfBridge   Tony Mountifield
• [asterisk-dev] [Code Review] Don't keep the STUN socket open between STUN monitor checks.   Richard Mudgett
• [asterisk-dev] [Code Review] app_queue change to have one devstate per device not per member / Janitorial cleanups   Matthew Nicholson
• [asterisk-dev] [Code Review] Fix several bugs with SDP parsing and well-formedness of responses   Matthew Nicholson
• [asterisk-dev] [Code Review] Make TestSuite Asterisk class create and install config files after instantiation   Matthew Nicholson
• [asterisk-dev] [Code Review]: Make TestSuite Asterisk class create and install config files after instantiation   Matthew Nicholson
• [asterisk-dev] [Code Review]: Asterisk Support of SIP Connect 1.1   Andrew Olmsted
• [asterisk-dev] [Code Review]: Asterisk Support of SIP Connect 1.1   Andrew Olmsted
• [asterisk-dev] [Code Review]: Asterisk Support of SIP Connect 1.1   Andrew Olmsted
• [asterisk-dev] [Code Review]: Asterisk Support of SIP Connect 1.1   Andrew Olmsted
• [asterisk-dev] [Code Review]: Asterisk Support of SIP Connect 1.1   Andrew Olmsted
• [asterisk-dev] Optional api and weak symbol problem   Yaroslav Panych
• [asterisk-dev] Optional api and weak symbol problem   Yaroslav Panych
• [asterisk-dev] Optional api and weak symbol problem   Yaroslav Panych
• [asterisk-dev] Optional api and weak symbol problem   Yaroslav Panych
• [asterisk-dev] Optional api and weak symbol problem   Yaroslav Panych
• [asterisk-dev] New Asterisk community developers   Jason Parker
• [asterisk-dev] SIP, NAT, security concerns, oh my!   Simon Perreault
• [asterisk-dev] [Code Review] Truncate host:port when checking SIP domains and make the URI parsing API more explicit about returning host:port and not a domain   Simon Perreault
• [asterisk-dev] [Code Review] Truncate host:port when checking SIP domains and make the URI parsing API more explicit about returning host:port and not a domain   Simon Perreault
• [asterisk-dev] [Code Review]: Truncate host:port when checking SIP domains and make the URI parsing API more explicit about returning host:port and not a domain   Simon Perreault
• [asterisk-dev] [Code Review] Truncate host:port when checking SIP domains and make the URI parsing API more explicit about returning host:port and not a domain   Simon Perreault
• [asterisk-dev] [Code Review] Add a unit test for ast_sockaddr_split_hostport   Simon Perreault
• [asterisk-dev] [Code Review] Don't keep the STUN socket open between STUN monitor checks.   Simon Perreault
• [asterisk-dev] [Code Review] Don't keep the STUN socket open between STUN monitor checks.   Simon Perreault
• [asterisk-dev] [Code Review] Don't keep the STUN socket open between STUN monitor checks.   Simon Perreault
• [asterisk-dev] [Code Review]: Don't keep the STUN socket open between STUN monitor checks.   Simon Perreault
• [asterisk-dev] [Code Review] Don't keep the STUN socket open between STUN monitor checks.   Simon Perreault
• [asterisk-dev] externaddr in chan_sip.c - locking required?   Ian Pilcher
• [asterisk-dev] externaddr in chan_sip.c - locking required?   Ian Pilcher
• [asterisk-dev] Scheduler and event callbacks - what thread?   Ian Pilcher
• [asterisk-dev] Scheduler and event callbacks - what thread?   Ian Pilcher
• [asterisk-dev] Summary: SIP, NAT, security concerns, oh my!   Matt Riddell
• [asterisk-dev] asterisk busy tone   Matt Riddell
• [asterisk-dev] IAX Hardphone (again)   Bill Shaw
• [asterisk-dev] IAX Hardphone (again)   Bill Shaw
• [asterisk-dev] IAX Hardphone (again)   Bill Shaw
• [asterisk-dev] Problem by enabling simultaneous two spans in MFC R2   Moises Silva
• [asterisk-dev] Adding languages to say.conf   Nir Simionovich
• [asterisk-dev] Optional api and weak symbol problem   Jared Smith
• [asterisk-dev] how to link FreePBX IVR step to something inside extensions-custom.conf?   Jamuel Starkey
• [asterisk-dev] asterisk dial plan questions   Jamuel Starkey
• [asterisk-dev] Developer   Jeff Strum
• [asterisk-dev] DAHDI-Linux 2.6.0-rc1 and DAHDI-Tools 2.6.0-rc1 Released   Asterisk Development Team
• [asterisk-dev] Asterisk 10.0.0-rc1 Now Available   Asterisk Development Team
• [asterisk-dev] Asterisk 1.8.8.0-rc3 Now Available   Asterisk Development Team
• [asterisk-dev] Asterisk 10.0.0-rc2 Now Available   Asterisk Development Team
• [asterisk-dev] Asterisk 1.8.8.0-rc4 Now Available   Asterisk Development Team
• [asterisk-dev] Asterisk / FreeSWITCH Developer required..   Steve Totaro
• [asterisk-dev] DAHDI analog: Monitoring call setup by backward audio connection   Pavel Troller
• [asterisk-dev] [Code Review] Restore SIP DTMF overlap dialing method.   Pavel Troller
• [asterisk-dev] [Code Review]: ensure that ast_string_field_pool base + used is always aligned   David Vossel
• [asterisk-dev] [Code Review]: ensure that ast_string_field_pool base + used is always aligned   David Vossel
• [asterisk-dev] [Code Review] Fix deadlock between subscription event RWLOCK and dialogs container lock in chan_sip.   David Vossel
• [asterisk-dev] [Code Review] Add mute all participants; play participant count to ConfBridge   David Vossel
• [asterisk-dev] [Code Review] BLF Subscriptions Causes SIP Deadlock   David Vossel
• [asterisk-dev] [Code Review] Add mute all participants; play participant count to ConfBridge   David Vossel
• [asterisk-dev] [Code Review] Add mute all participants; play participant count to ConfBridge   David Vossel
• [asterisk-dev] [Code Review] Fix deadlock between subscription event RWLOCK and dialogs container lock in chan_sip.   David Vossel
• [asterisk-dev] [Code Review] Fix deadlock between subscription event RWLOCK and dialogs container lock in chan_sip. (simplified)   David Vossel
• [asterisk-dev] [Code Review] BLF Subscriptions Causes SIP Deadlock   David Vossel
• [asterisk-dev] [Code Review]: Fix deadlock between subscription event RWLOCK and dialogs container lock in chan_sip. (simplified)   David Vossel
• [asterisk-dev] [Code Review]: BLF Subscriptions Causes SIP Deadlock   David Vossel
• [asterisk-dev] [Code Review]: BLF Subscriptions Causes SIP Deadlock   David Vossel
• [asterisk-dev] [Code Review] BLF Subscriptions Causes SIP Deadlock   David Vossel
• [asterisk-dev] [Code Review] Fix deadlock between subscription event RWLOCK and dialogs container lock in chan_sip. (simplified)   David Vossel
• [asterisk-dev] [Code Review] Fix deadlock between subscription event RWLOCK and dialogs container lock in chan_sip. (simplified)   David Vossel
• [asterisk-dev] Optional api and weak symbol problem   David Vossel
• [asterisk-dev] [Code Review] Fix deadlock between subscription event RWLOCK and dialogs container lock in chan_sip.   David Vossel
• [asterisk-dev] [Code Review]: Fix deadlock between subscription event RWLOCK and dialogs container lock in chan_sip.   David Vossel
• [asterisk-dev] [Code Review] Fix deadlock between subscription event RWLOCK and dialogs container lock in chan_sip.   David Vossel
• [asterisk-dev] [Code Review] Video format treated as audio when removed from the file playback scheduler   David Vossel
• [asterisk-dev] [Code Review] Add mute all participants; play participant count to ConfBridge   David Vossel
• [asterisk-dev] (no subject)   Charles Wang
• [asterisk-dev] [Code Review]: ensure that ast_string_field_pool base + used is always aligned   Terry Wilson
• [asterisk-dev] [Code Review]: ensure that ast_string_field_pool base + used is always aligned   Terry Wilson
• [asterisk-dev] [Code Review] NetBorder Call Analyzer (Answering Machine Detection) support   Terry Wilson
• [asterisk-dev] [Code Review] Mothership cmenuselect easteregg addition   Terry Wilson
• [asterisk-dev] [Code Review] ensure that ast_string_field_pool base + used is always aligned   Terry Wilson
• [asterisk-dev] [Code Review] Add hold status with CEL and setting a channel var HOLDING via ast_moh_stop/start   Terry Wilson
• [asterisk-dev] [Code Review] fix excess warnings after r1395 (and disallow empty names always)   Terry Wilson
• [asterisk-dev] [Code Review] Just remove the silly SIP registertrying option   Terry Wilson
• [asterisk-dev] [Code Review] fix sqlite crash on unset config_table / unbreak sqlite realtime_multi_handler   Terry Wilson
• [asterisk-dev] [Code Review] Just remove the silly SIP registertrying option   Terry Wilson
• [asterisk-dev] [Code Review] chan_sip registertrying lacks a global option   Terry Wilson
• [asterisk-dev] [Code Review] Add #includeif statement   Terry Wilson
• [asterisk-dev] [Code Review] Add mute all participants; play participant count to ConfBridge   Terry Wilson
• [asterisk-dev] [Code Review]: Just remove the silly SIP registertrying option   Terry Wilson
• [asterisk-dev] [Code Review]: Add mute all participants; play participant count to ConfBridge   Terry Wilson
• [asterisk-dev] [Code Review]: Add #includeif statement   Terry Wilson
• [asterisk-dev] Optional api and weak symbol problem   Terry Wilson
• [asterisk-dev] Optional api and weak symbol problem   Terry Wilson
• [asterisk-dev] [Code Review] Truncate host:port when checking SIP domains and make the URI parsing API more explicit about returning host:port and not a domain   Terry Wilson
• [asterisk-dev] [Code Review]: Truncate host:port when checking SIP domains and make the URI parsing API more explicit about returning host:port and not a domain   Terry Wilson
• [asterisk-dev] [Code Review]: Truncate host:port when checking SIP domains and make the URI parsing API more explicit about returning host:port and not a domain   Terry Wilson
• [asterisk-dev] [Code Review]: Truncate host:port when checking SIP domains and make the URI parsing API more explicit about returning host:port and not a domain   Terry Wilson
• [asterisk-dev] [Code Review] Add a unit test for ast_sockaddr_split_hostport   Terry Wilson
• [asterisk-dev] [Code Review] Add a unit test for ast_sockaddr_split_hostport   Terry Wilson
• [asterisk-dev] [Code Review] Truncate host:port when checking SIP domains and make the URI parsing API more explicit about returning host:port and not a domain   Terry Wilson
• [asterisk-dev] [Code Review]: Add a unit test for ast_sockaddr_split_hostport   Terry Wilson
• [asterisk-dev] [Code Review] Add a unit test for ast_sockaddr_split_hostport   Terry Wilson
• [asterisk-dev] Summary: SIP, NAT, security concerns, oh my!   Terry Wilson
• [asterisk-dev] Summary: SIP, NAT, security concerns, oh my!   Terry Wilson
• [asterisk-dev] [Code Review]: Truncate host:port when checking SIP domains and make the URI parsing API more explicit about returning host:port and not a domain   Terry Wilson
• [asterisk-dev] [Code Review]: chan_sip: removing some non-compliant code in v10 and minor fixes   Terry Wilson
• [asterisk-dev] [Code Review]: chan_sip: removing some non-compliant code in v10 and minor fixes   Terry Wilson
• [asterisk-dev] [Code Review]: chan_sip: removing some non-compliant code in v10 and minor fixes   Terry Wilson
• [asterisk-dev] Summary: SIP, NAT, security concerns, oh my!   Terry Wilson
• [asterisk-dev] [Code Review] chan_sip: removing some non-compliant code in v10 and minor fixes   Terry Wilson
• [asterisk-dev] [Code Review]: chan_sip: removing some non-compliant code in v10 and minor fixes   Terry Wilson
• [asterisk-dev] [Code Review]: chan_sip: removing some non-compliant code in v10 and minor fixes   Terry Wilson
• [asterisk-dev] [Code Review] Don't forget to rescan MOH files for cached realtime MOH classes   Terry Wilson
• [asterisk-dev] $500 bounty for #18039: Realtime music restarts from beginning each time   Terry Wilson
• [asterisk-dev] [Code Review]: Don't forget to rescan MOH files for cached realtime MOH classes   Terry Wilson
• [asterisk-dev] [Code Review] Don't read past then end of our int when calling write() in __ast_queue_frame   Terry Wilson
• [asterisk-dev] [Code Review]: Don't read past then end of our int when calling write() in __ast_queue_frame   Terry Wilson
• [asterisk-dev] [Code Review] Add a test for verifying when username information can be leaked via differing nat settings   Terry Wilson
• [asterisk-dev] [Code Review] Change the default nat setting to "yes"   Terry Wilson
• [asterisk-dev] [Code Review]: Change the default nat setting to "yes"   Terry Wilson
• [asterisk-dev] [Code Review] Update astdb documentation to how it actually works   Terry Wilson
• [asterisk-dev] [Code Review]: Add a test for verifying when username information can be leaked via differing nat settings   Terry Wilson
• [asterisk-dev] [Code Review]: Add a test for verifying when username information can be leaked via differing nat settings   Terry Wilson
• [asterisk-dev] [Code Review]: Add a test for verifying when username information can be leaked via differing nat settings   Terry Wilson
• [asterisk-dev] [Code Review] Add a test for verifying when username information can be leaked via differing nat settings   Terry Wilson
• [asterisk-dev] [Code Review]: Change the default nat setting to "yes"   Terry Wilson
• [asterisk-dev] [Code Review]: Change the default nat setting to "yes"   Terry Wilson
• [asterisk-dev] [Code Review] Change the default nat setting to "yes"   Terry Wilson
• [asterisk-dev] ring groups   Terry Wilson
• [asterisk-dev] [Code Review] Don't play a new song for every new hold within a call with caching realtime MOH   Terry Wilson
• [asterisk-dev] [Code Review] SendMessage strips extension from To: Header in SIP MESSAGE   Terry Wilson
• [asterisk-dev] [Code Review]: fix odd logic in get_msg_text() and a couple of related quirks   Terry Wilson
• [asterisk-dev] [Code Review]: fix odd logic in get_msg_text() and a couple of related quirks   Terry Wilson
• [asterisk-dev] [Code Review] unbreaking <sip:username> from-uri support for REGISTER   Terry Wilson
• [asterisk-dev] Google voice patch   sean darcy
• [asterisk-dev] Error loading module 'chan_sip.so': /usr/lib/asterisk/modules/chan_sip.so: un defined symbol   shalu dhamija
• [asterisk-dev] [Code Review] fix using external MP3 player and res_timing_dahdi   elguero
• [asterisk-dev] [Code Review] fix using external MP3 player with res_timing_dahdi being used as timer causing MOH stream to stop   elguero
• [asterisk-dev] [Code Review] Add PAUSED tag to queue log when adding queue member as paused   garlew
• [asterisk-dev] [Code Review] app_queue change to have one devstate per device not per member / Janitorial cleanups   irroot
• [asterisk-dev] [Code Review] Mothership cmenuselect easteregg addition   jrose
• [asterisk-dev] [Code Review] Mothership cmenuselect easteregg addition   jrose
• [asterisk-dev] [Code Review] Adds barriers to the menuselect easter egg   jrose
• [asterisk-dev] [Code Review] Adds barriers to the menuselect easter egg   jrose
• [asterisk-dev] [Code Review] Adds barriers to the menuselect easter egg   jrose
• [asterisk-dev] [Code Review] Adds barriers to the menuselect easter egg   jrose
• [asterisk-dev] [Code Review] Get CDR Congestion Logging Test (and improving whole concept behind CDRTestCase) working.   jrose
• [asterisk-dev] [Code Review] Get CDR Congestion Logging Test (and improving whole concept behind CDRTestCase) working.   jrose
• [asterisk-dev] [Code Review] Adds barriers to the menuselect easter egg   jrose
• [asterisk-dev] [Code Review] Adds barriers to the menuselect easter egg   jrose
• [asterisk-dev] [Code Review] Adds barriers to the menuselect easter egg   jrose
• [asterisk-dev] [Code Review] Adds barriers to the menuselect easter egg   jrose
• [asterisk-dev] [Code Review] Eliminate redundant and possibly dangerous close/fclose pairs in chan_sip and tcptls.   jrose
• [asterisk-dev] [Code Review] Eliminate redundant and possibly dangerous close/fclose pairs in chan_sip and tcptls.   jrose
• [asterisk-dev] [Code Review] Eliminate redundant and possibly dangerous close/fclose pairs in chan_sip and tcptls.   jrose
• [asterisk-dev] [Code Review]: Eliminate redundant and possibly dangerous close/fclose pairs in chan_sip and tcptls.   jrose
• [asterisk-dev] [Code Review]: Eliminate redundant and possibly dangerous close/fclose pairs in chan_sip and tcptls.   jrose
• [asterisk-dev] [Code Review] Move password entry for new Comedian Mail users to the end of setup.   jrose
• [asterisk-dev] [Code Review]: Eliminate redundant and possibly dangerous close/fclose pairs in chan_sip and tcptls.   jrose
• [asterisk-dev] [Code Review] Multiple recipients and urgency.   jrose
• [asterisk-dev] [Code Review] Multiple recipients and urgency.   jrose
• [asterisk-dev] [Code Review]: Eliminate redundant and possibly dangerous close/fclose pairs in chan_sip and tcptls.   jrose
• [asterisk-dev] [Code Review]: Eliminate redundant and possibly dangerous close/fclose pairs in chan_sip and tcptls.   jrose
• [asterisk-dev] [Code Review] Eliminate redundant and possibly dangerous close/fclose pairs in chan_sip and tcptls.   jrose
• [asterisk-dev] [Code Review]: Add mute all participants; play participant count to ConfBridge   jrose
• [asterisk-dev] [Code Review] Get CDR Congestion Logging Test (and improving whole concept behind CDRTestCase) working.   jrose
• [asterisk-dev] [Code Review]: Eliminate redundant and possibly dangerous close/fclose pairs in chan_sip and tcptls.   jrose
• [asterisk-dev] [Code Review] Eliminate redundant and possibly dangerous close/fclose pairs in chan_sip and tcptls.   jrose
• [asterisk-dev] [Code Review] PreDial - Ability to run dialplan on callee channel and caller channel right before actual Dial   kobaz
• [asterisk-dev] [Code Review]: PreDial - Ability to run dialplan on callee channel and caller channel right before actual Dial   kobaz
• [asterisk-dev] [Code Review]: PreDial - Ability to run dialplan on callee channel and caller channel right before actual Dial   kobaz
• [asterisk-dev] [Code Review] ast_indicate(chan, -1) don't stop playing tones   may213
• [asterisk-dev] [Code Review] ast_indicate(chan, -1) don't stop playing tones   may213
• [asterisk-dev] [Code Review] ast_indicate(chan, -1) don't stop playing tones   may213
• [asterisk-dev] [Code Review]: ast_indicate(chan, -1) don't stop playing tones   may213
• [asterisk-dev] [Code Review]: ast_indicate(chan, -1) don't stop playing tones   may213
• [asterisk-dev] [Code Review]: Add mute all participants; play participant count to ConfBridge   mjordan
• [asterisk-dev] [Code Review]: Add mute all participants; play participant count to ConfBridge   mjordan
• [asterisk-dev] [Code Review] BLF Subscriptions Causes SIP Deadlock   mjordan
• [asterisk-dev] [Code Review] Add mute all participants; play participant count to ConfBridge   mjordan
• [asterisk-dev] [Code Review]: Add mute all participants; play participant count to ConfBridge   mjordan
• [asterisk-dev] [Code Review] Fix deadlock between subscription event RWLOCK and dialogs container lock in chan_sip. (simplified)   mjordan
• [asterisk-dev] [Code Review]: Fix deadlock between subscription event RWLOCK and dialogs container lock in chan_sip. (simplified)   mjordan
• [asterisk-dev] [Code Review] BLF Subscriptions Causes SIP Deadlock   mjordan
• [asterisk-dev] [Code Review]: BLF Subscriptions Causes SIP Deadlock   mjordan
• [asterisk-dev] [Code Review] Added VM_INFO dialplan function to return voicemail user information   mjordan
• [asterisk-dev] [Code Review]: Added VM_INFO dialplan function to return voicemail user information   mjordan
• [asterisk-dev] [Code Review] app_voicemail [general] variables zonetag and locale are not set on mailbox until after reload   mjordan
• [asterisk-dev] [Code Review] Make TestSuite Asterisk class create and install config files after instantiation   mjordan
• [asterisk-dev] [Code Review]: Make TestSuite Asterisk class create and install config files after instantiation   mjordan
• [asterisk-dev] [Code Review]: Make TestSuite Asterisk class create and install config files after instantiation   mjordan
• [asterisk-dev] [Code Review] mxml puts quotes inside multiline opaque data   mjordan
• [asterisk-dev] [Code Review] Video format treated as audio when removed from the file playback scheduler   mjordan
• [asterisk-dev] [Code Review] Restore SIP DTMF overlap dialing method.   mjordan
• [asterisk-dev] [Code Review] Restore SIP DTMF overlap dialing method.   mjordan
• [asterisk-dev] [Code Review]: Restore SIP DTMF overlap dialing method.   mjordan
• [asterisk-dev] [Code Review] Restore SIP DTMF overlap dialing method.   mjordan
• [asterisk-dev] [Code Review] Update basic call parking test.   mjordan
• [asterisk-dev] [Code Review] Multiple recipients and urgency.   mjordan
• [asterisk-dev] [Code Review] Add mute all participants; play participant count to ConfBridge   mjordan
• [asterisk-dev] [Code Review] Change the default nat setting to "yes"   mjordan
• [asterisk-dev] [Code Review] Add a test for verifying when username information can be leaked via differing nat settings   mjordan
• [asterisk-dev] [Code Review] Change the default nat setting to "yes"   mjordan
• [asterisk-dev] [Code Review]: Add a test for verifying when username information can be leaked via differing nat settings   mjordan
• [asterisk-dev] [Code Review] Fix more res_jabber resource leaks   mjordan
• [asterisk-dev] [Code Review] Get CDR Congestion Logging Test (and improving whole concept behind CDRTestCase) working.   mjordan
• [asterisk-dev] [Code Review] SendMessage strips extension from To: Header in SIP MESSAGE   mjordan
• [asterisk-dev] [Code Review] Don't play a new song for every new hold within a call with caching realtime MOH   mjordan
• [asterisk-dev] [Code Review] Fix more res_jabber resource leaks   mjordan
• [asterisk-dev] [Code Review] Don't keep the STUN socket open between STUN monitor checks.   mjordan
• [asterisk-dev] [Code Review]: Don't keep the STUN socket open between STUN monitor checks.   mjordan
• [asterisk-dev] [Code Review] Doubly linked lists unit test and update to implementation.   mjordan
• [asterisk-dev] [Code Review] fix odd logic in get_msg_text() and a couple of related quirks   mjordan
• [asterisk-dev] [Code Review]: fix odd logic in get_msg_text() and a couple of related quirks   mjordan
• [asterisk-dev] [Code Review] fix odd logic in get_msg_text() and a couple of related quirks   mjordan
• [asterisk-dev] [Code Review] Get CDR Congestion Logging Test (and improving whole concept behind CDRTestCase) working.   mjordan
• [asterisk-dev] [Code Review] Fix more res_jabber resource leaks   opticron
• [asterisk-dev] [Code Review] BLF Subscriptions Causes SIP Deadlock   opticron
• [asterisk-dev] [Code Review]: BLF Subscriptions Causes SIP Deadlock   opticron
• [asterisk-dev] [Code Review] BLF Subscriptions Causes SIP Deadlock   opticron
• [asterisk-dev] [Code Review] BLF Subscriptions Causes SIP Deadlock   opticron
• [asterisk-dev] [Code Review]: BLF Subscriptions Causes SIP Deadlock   opticron
• [asterisk-dev] [Code Review] Don't forget to rescan MOH files for cached realtime MOH classes   opticron
• [asterisk-dev] [Code Review] Fix more res_jabber resource leaks   opticron
• [asterisk-dev] [Code Review]: Fix more res_jabber resource leaks   opticron
• [asterisk-dev] [Code Review]: Astobj2 locking enhancement   rmudgett
• [asterisk-dev] [Code Review] Fix deadlock between subscription event RWLOCK and dialogs container lock in chan_sip.   rmudgett
• [asterisk-dev] [Code Review] Fix deadlock between subscription event RWLOCK and dialogs container lock in chan_sip.   rmudgett
• [asterisk-dev] [Code Review]: Fix deadlock between subscription event RWLOCK and dialogs container lock in chan_sip.   rmudgett
• [asterisk-dev] [Code Review] Fix deadlock between subscription event RWLOCK and dialogs container lock in chan_sip.   rmudgett
• [asterisk-dev] [Code Review]: Fix deadlock between subscription event RWLOCK and dialogs container lock in chan_sip.   rmudgett
• [asterisk-dev] [Code Review] Fix deadlock between subscription event RWLOCK and dialogs container lock in chan_sip.   rmudgett
• [asterisk-dev] [Code Review]: Fix deadlock between subscription event RWLOCK and dialogs container lock in chan_sip.   rmudgett
• [asterisk-dev] [Code Review] Fix deadlock between subscription event RWLOCK and dialogs container lock in chan_sip. (simplified)   rmudgett
• [asterisk-dev] [Code Review]: Fix deadlock between subscription event RWLOCK and dialogs container lock in chan_sip.   rmudgett
• [asterisk-dev] [Code Review] Fix deadlock between subscription event RWLOCK and dialogs container lock in chan_sip.   rmudgett
• [asterisk-dev] [Code Review]: Fix deadlock between subscription event RWLOCK and dialogs container lock in chan_sip. (simplified)   rmudgett
• [asterisk-dev] [Code Review]: Fix deadlock between subscription event RWLOCK and dialogs container lock in chan_sip. (simplified)   rmudgett
• [asterisk-dev] [Code Review] Fix deadlock between subscription event RWLOCK and dialogs container lock in chan_sip. (simplified)   rmudgett
• [asterisk-dev] [Code Review]: Fix deadlock between subscription event RWLOCK and dialogs container lock in chan_sip. (simplified)   rmudgett
• [asterisk-dev] [Code Review] Fix deadlock between subscription event RWLOCK and dialogs container lock in chan_sip. (simplified)   rmudgett
• [asterisk-dev] [Code Review]: Fix deadlock between subscription event RWLOCK and dialogs container lock in chan_sip. (simplified)   rmudgett
• [asterisk-dev] [Code Review]: Fix deadlock between subscription event RWLOCK and dialogs container lock in chan_sip. (simplified)   rmudgett
• [asterisk-dev] [Code Review] Fix deadlock between subscription event RWLOCK and dialogs container lock in chan_sip. (simplified)   rmudgett
• [asterisk-dev] [Code Review] Fix deadlock between subscription event RWLOCK and dialogs container lock in chan_sip.   rmudgett
• [asterisk-dev] [Code Review] PreDial - Ability to run dialplan on callee channel and caller channel right before actual Dial   rmudgett
• [asterisk-dev] [Code Review] Fix deadlock between subscription event RWLOCK and dialogs container lock in chan_sip. (simplified)   rmudgett
• [asterisk-dev] [Code Review] Doubly linked lists unit test and update to implementation.   rmudgett
• [asterisk-dev] [Code Review]: ast_indicate(chan, -1) don't stop playing tones   rmudgett
• [asterisk-dev] [Code Review] ast_indicate(chan, -1) don't stop playing tones   rmudgett
• [asterisk-dev] [Code Review] Fix deadlock between subscription event RWLOCK and dialogs container lock in chan_sip.   rmudgett
• [asterisk-dev] [Code Review] Fix deadlock between subscription event RWLOCK and dialogs container lock in chan_sip.   rmudgett
• [asterisk-dev] [Code Review]: Fix deadlock between subscription event RWLOCK and dialogs container lock in chan_sip.   rmudgett
• [asterisk-dev] [Code Review]: Fix deadlock between subscription event RWLOCK and dialogs container lock in chan_sip.   rmudgett
• [asterisk-dev] [Code Review] Fix deadlock between subscription event RWLOCK and dialogs container lock in chan_sip.   rmudgett
• [asterisk-dev] [Code Review]: ast_indicate(chan, -1) don't stop playing tones   rmudgett
• [asterisk-dev] [Code Review] Add PAUSED tag to queue log when adding queue member as paused   rmudgett
• [asterisk-dev] [Code Review] Restore SIP DTMF overlap dialing method.   rmudgett
• [asterisk-dev] [Code Review] Restore SIP DTMF overlap dialing method.   rmudgett
• [asterisk-dev] [Code Review] Restore SIP DTMF overlap dialing method.   rmudgett
• [asterisk-dev] [Code Review] Restore SIP DTMF overlap dialing method.   rmudgett
• [asterisk-dev] [Code Review]: Restore SIP DTMF overlap dialing method.   rmudgett
• [asterisk-dev] [Code Review] Restore SIP DTMF overlap dialing method.   rmudgett
• [asterisk-dev] [Code Review] Add PAUSED tag to queue log when adding queue member as paused   rmudgett
• [asterisk-dev] [Code Review] Restore SIP DTMF overlap dialing method.   rmudgett
• [asterisk-dev] [Code Review] Update basic call parking test.   rmudgett
• [asterisk-dev] [Code Review] Don't keep the STUN socket open between STUN monitor checks.   rmudgett
• [asterisk-dev] [Code Review] fix ast_str_truncate signedness warning and documentation   rmudgett
• [asterisk-dev] [Code Review]: Don't keep the STUN socket open between STUN monitor checks.   rmudgett
• [asterisk-dev] [Code Review] Don't keep the STUN socket open between STUN monitor checks.   rmudgett
• [asterisk-dev] [Code Review]: Doubly linked lists unit test and update to implementation.   rmudgett
• [asterisk-dev] [Code Review] Doubly linked lists unit test and update to implementation.   rmudgett
• [asterisk-dev] [Code Review]: fix odd logic in get_msg_text() and a couple of related quirks   rmudgett
• [asterisk-dev] [Code Review] Don't keep the STUN socket open between STUN monitor checks.   rmudgett
• [asterisk-dev] [Code Review]: Don't keep the STUN socket open between STUN monitor checks.   rmudgett
• [asterisk-dev] [Code Review] chan_dahdi: create and destroy channels at run-time   rmudgett
• [asterisk-dev] [Code Review] Don't keep the STUN socket open between STUN monitor checks.   rmudgett
• [asterisk-dev] [Code Review] Don't keep the STUN socket open between STUN monitor checks.   rmudgett
• [asterisk-dev] [Code Review]: Don't keep the STUN socket open between STUN monitor checks.   rmudgett
• [asterisk-dev] [Code Review] Don't keep the STUN socket open between STUN monitor checks.   rmudgett
• [asterisk-dev] [Code Review]: Don't keep the STUN socket open between STUN monitor checks.   rmudgett
• [asterisk-dev] [Code Review] Don't keep the STUN socket open between STUN monitor checks.   rmudgett
• [asterisk-dev] [Code Review] chan_dahdi: create and destroy channels at run-time   rmudgett
• [asterisk-dev] [Code Review]: update samples to use [public] context   rmudgett
• [asterisk-dev] [Code Review]: Fix deadlock between subscription event RWLOCK and dialogs container lock in chan_sip.   schmidts
• [asterisk-dev] [Code Review] fix excess warnings after r1395 (and disallow empty names always)   schmidts
• [asterisk-dev] [Code Review] fix excess warnings after r1395 (and disallow empty names always)   schmidts
• [asterisk-dev] [Code Review] Fix deadlock between subscription event RWLOCK and dialogs container lock in chan_sip.   schmidts
• [asterisk-dev] [Code Review]: Fix deadlock between subscription event RWLOCK and dialogs container lock in chan_sip.   schmidts
• [asterisk-dev] [Code Review]: Fix deadlock between subscription event RWLOCK and dialogs container lock in chan_sip.   schmidts
• [asterisk-dev] [Code Review]: Fix deadlock between subscription event RWLOCK and dialogs container lock in chan_sip.   schmidts
• [asterisk-dev] [Code Review] Fix deadlock between subscription event RWLOCK and dialogs container lock in chan_sip. (simplified)   schmidts
• [asterisk-dev] [Code Review] ast_indicate(chan, -1) don't stop playing tones   schmidts
• [asterisk-dev] [Code Review] Fix deadlock between subscription event RWLOCK and dialogs container lock in chan_sip.   schmidts
• [asterisk-dev] [Code Review] rtp / rtcp set debug ip doesnt work correctly in 1.8   schmidts
• [asterisk-dev] [Code Review]: rtp / rtcp set debug ip doesnt work correctly in 1.8   schmidts
• [asterisk-dev] [Code Review] rtp / rtcp set debug ip doesnt work correctly in 1.8   schmidts
• [asterisk-dev] [Code Review] Added VM_INFO dialplan function to return voicemail user information   shawkris
• [asterisk-dev] [Code Review]: Added VM_INFO dialplan function to return voicemail user information   shawkris
• [asterisk-dev] [Code Review]: Added VM_INFO dialplan function to return voicemail user information   shawkris
• [asterisk-dev] [Code Review] Added VM_INFO dialplan function to return voicemail user information   shawkris
• [asterisk-dev] [Code Review]: Added VM_INFO dialplan function to return voicemail user information   shawkris
• [asterisk-dev] [Code Review]: Added VM_INFO dialplan function to return voicemail user information   shawkris
• [asterisk-dev] [Code Review] Added VM_INFO dialplan function to return voicemail user information   shawkris
• [asterisk-dev] [Code Review] Added VM_INFO dialplan function to return voicemail user information   shawkris
• [asterisk-dev] [Code Review] Added VM_INFO dialplan function to return voicemail user information   shawkris
• [asterisk-dev] New Asterisk community developers   anguri sudhakar
• [asterisk-dev] [Code Review]: Truncate host:port when checking SIP domains and make the URI parsing API more explicit about returning host:port and not a domain   ulogic
• [asterisk-dev] [Code Review]: ensure that ast_string_field_pool base + used is always aligned   wdoekes
• [asterisk-dev] [Code Review]: ensure that ast_string_field_pool base + used is always aligned   wdoekes
• [asterisk-dev] [Code Review] fix excess warnings after r1395 (and disallow empty names always)   wdoekes
• [asterisk-dev] [Code Review] fix sqlite crash on unset config_table / unbreak sqlite realtime_multi_handler   wdoekes
• [asterisk-dev] [Code Review]: fix excess warnings after r1395 (and disallow empty names always)   wdoekes
• [asterisk-dev] [Code Review] fix excess warnings after r1395 (and disallow empty names always)   wdoekes
• [asterisk-dev] [Code Review] Just remove the silly SIP registertrying option   wdoekes
• [asterisk-dev] [Code Review]: Just remove the silly SIP registertrying option   wdoekes
• [asterisk-dev] [Code Review] Add #includeif statement   wdoekes
• [asterisk-dev] [Code Review]: Add #includeif statement   wdoekes
• [asterisk-dev] [Code Review]: fix sqlite crash on unset config_table / unbreak sqlite realtime_multi_handler   wdoekes
• [asterisk-dev] [Code Review]: [patch] Improve debug of ast_hangup   wdoekes
• [asterisk-dev] [Code Review] SIP: peer matching by callbackextension   wdoekes
• [asterisk-dev] [Code Review] fix default udptl port range which differs in source / udptl.conf.sample   wdoekes
• [asterisk-dev] [Code Review]: sparc64 compile error, undefined reference to `__sync_fetch_and_add_4'   wdoekes
• [asterisk-dev] [Code Review]: Added VM_INFO dialplan function to return voicemail user information   wdoekes
• [asterisk-dev] [Code Review]: Added VM_INFO dialplan function to return voicemail user information   wdoekes
• [asterisk-dev] [Code Review] Added VM_INFO dialplan function to return voicemail user information   wdoekes
• [asterisk-dev] [Code Review]: Added VM_INFO dialplan function to return voicemail user information   wdoekes
• [asterisk-dev] [Code Review] app_voicemail [general] variables zonetag and locale are not set on mailbox until after reload   wdoekes
• [asterisk-dev] [Code Review]: sparc64 compile error, undefined reference to `__sync_fetch_and_add_4'   wdoekes
• [asterisk-dev] [Code Review] app_voicemail [general] variables zonetag and locale are not set on mailbox until after reload   wdoekes
• [asterisk-dev] [Code Review] regression test for r1570 (app_voicemail, bad load_config order)   wdoekes
• [asterisk-dev] [Code Review] use __alignof__ instead of sizeof for alignment of ast_string_field_allocation   wdoekes
• [asterisk-dev] [Code Review] Add a unit test for ast_sockaddr_split_hostport   wdoekes
• [asterisk-dev] [Code Review] Add a unit test for ast_sockaddr_split_hostport   wdoekes
• [asterisk-dev] [Code Review] Truncate host:port when checking SIP domains and make the URI parsing API more explicit about returning host:port and not a domain   wdoekes
• [asterisk-dev] [Code Review] Added VM_INFO dialplan function to return voicemail user information   wdoekes
• [asterisk-dev] [Code Review]: chan_sip: removing some non-compliant code in v10 and minor fixes   wdoekes
• [asterisk-dev] [Code Review] mxml puts quotes inside multiline opaque data   wdoekes
• [asterisk-dev] [Code Review]: Eliminate redundant and possibly dangerous close/fclose pairs in chan_sip and tcptls.   wdoekes
• [asterisk-dev] [Code Review] Eliminate redundant and possibly dangerous close/fclose pairs in chan_sip and tcptls.   wdoekes
• [asterisk-dev] [Code Review] Eliminate redundant and possibly dangerous close/fclose pairs in chan_sip and tcptls.   wdoekes
• [asterisk-dev] [Code Review]: Eliminate redundant and possibly dangerous close/fclose pairs in chan_sip and tcptls.   wdoekes
• [asterisk-dev] [Code Review] Eliminate redundant and possibly dangerous close/fclose pairs in chan_sip and tcptls.   wdoekes
• [asterisk-dev] [Code Review]: Remove need for registration strings in sip.conf   wdoekes
• [asterisk-dev] [Code Review] Don't read past then end of our int when calling write() in __ast_queue_frame   wdoekes
• [asterisk-dev] [Code Review] Don't read past then end of our int when calling write() in __ast_queue_frame   wdoekes
• [asterisk-dev] [Code Review] disallow tcp/tls transport in peers when tcpenable=no   wdoekes
• [asterisk-dev] [Code Review] fix odd logic in get_msg_text() and a couple of related quirks   wdoekes
• [asterisk-dev] [Code Review] unbreaking <sip:username> from-uri support for REGISTER   wdoekes
• [asterisk-dev] [Code Review] unbreaking <sip:username> from-uri support for REGISTER   wdoekes
• [asterisk-dev] [Code Review] Move password entry for new Comedian Mail users to the end of setup.   wdoekes
• [asterisk-dev] [Code Review] unbreaking <sip:username> from-uri support for REGISTER   wdoekes
• [asterisk-dev] [Code Review] unbreaking <sip:username> from-uri support for REGISTER   wdoekes
• [asterisk-dev] [Code Review] unbreaking <sip:username> from-uri support for REGISTER   wdoekes
• [asterisk-dev] [Code Review] Change the default nat setting to "yes"   wdoekes
• [asterisk-dev] [Code Review] Permit/deny negation, plus multiple-specification with commas   wdoekes
• [asterisk-dev] [Code Review] Add a test for verifying when username information can be leaked via differing nat settings   wdoekes
• [asterisk-dev] [Code Review] Eliminate redundant and possibly dangerous close/fclose pairs in chan_sip and tcptls.   wdoekes
• [asterisk-dev] [Code Review]: Eliminate redundant and possibly dangerous close/fclose pairs in chan_sip and tcptls.   wdoekes
• [asterisk-dev] [Code Review] Change the default nat setting to "yes"   wdoekes
• [asterisk-dev] [Code Review] Add a test for verifying when username information can be leaked via differing nat settings   wdoekes
• [asterisk-dev] [Code Review] fix ast_str_truncate signedness warning and documentation   wdoekes
• [asterisk-dev] [Code Review]: Get CDR Congestion Logging Test (and improving whole concept behind CDRTestCase) working.   wdoekes
• [asterisk-dev] [Code Review]: fix odd logic in get_msg_text() and a couple of related quirks   wdoekes
• [asterisk-dev] [Code Review]: fix odd logic in get_msg_text() and a couple of related quirks   wdoekes
• [asterisk-dev] [Code Review]: fix odd logic in get_msg_text() and a couple of related quirks   wdoekes
• [asterisk-dev] [Code Review]: fix odd logic in get_msg_text() and a couple of related quirks   wdoekes
• [asterisk-dev] [Code Review] Separate verbose level for logging   wdoekes
• [asterisk-dev] [Code Review] rtp / rtcp set debug ip doesnt work correctly in 1.8   wdoekes
• [asterisk-dev] [Code Review] rtp / rtcp set debug ip doesnt work correctly in 1.8   wdoekes
• [asterisk-dev] [Code Review] Separate verbose level for logging   wdoekes
• [asterisk-dev] [Code Review] Separate verbose level for logging   wdoekes
• [asterisk-dev] [Code Review] Eliminate redundant and possibly dangerous close/fclose pairs in chan_sip and tcptls.   wdoekes
• [asterisk-dev] [Code Review] make testsuite runtests a bit less warny   wdoekes

Last message date: Wed Nov 30 19:33:34 CST 2011
Archived on: Thu Dec 1 08:26:57 CST 2011

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