December 2012 Archives by date

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Starting: Sat Dec 1 02:10:20 CST 2012
Ending: Mon Dec 31 16:15:32 CST 2012
Messages: 289

• [asterisk-dev] Asterisk 12 API improvements   Olle E. Johansson
• [asterisk-dev] Asterisk 12 API improvements   Antonio Goméz Soto
• [asterisk-dev] Asterisk 12 API improvements   Dan Jenkins
• [asterisk-dev] Asterisk 12 API improvements   Dan Jenkins
• [asterisk-dev] Asterisk 12 API improvements   Joshua Colp
• [asterisk-dev] Asterisk 11 and console verbosity   Pavel Troller
• [asterisk-dev] AST_BUILDOPT_SUM stability   Tzafrir Cohen
• [asterisk-dev] Asterisk 11 and console verbosity   Tilghman Lesher
• [asterisk-dev] [Code Review] Make inline documentation readable   Tilghman Lesher
• [asterisk-dev] AST_BUILDOPT_SUM stability   Tilghman Lesher
• [asterisk-dev] Asterisk 11 and console verbosity   Pavel Troller
• [asterisk-dev] [Code Review] Make inline documentation readable   wdoekes
• [asterisk-dev] Asterisk 12 API improvements   Olle E. Johansson
• [asterisk-dev] Asterisk 12 API improvements   Lenz Emilitri
• [asterisk-dev] Asterisk 12 API improvements   Lenz Emilitri
• [asterisk-dev] Asterisk 12 API improvements   Joshua Colp
• [asterisk-dev] New SIP Channel Driver Data Access Layer   Joshua Colp
• [asterisk-dev] New SIP Channel Driver Data Access Layer   Olle E. Johansson
• [asterisk-dev] [Code Review] res_srtp: Fix a crash caused by an attempt to dealloc a session pointer that was never alloced or has already been dealloced.   jrose
• [asterisk-dev] [Code Review] res_srtp: Fix a crash caused by an attempt to dealloc a session pointer that was never alloced or has already been dealloced.   jrose
• [asterisk-dev] [Code Review] app_voicemail: fix problem where we kill the msg_id of a message when changing its folder in odbc storage.   jrose
• [asterisk-dev] Asterisk 12 API improvements   Matthew Jordan
• [asterisk-dev] Asterisk 12 API improvements   David M. Lee
• [asterisk-dev] Asterisk 12 API improvements   Tilghman Lesher
• [asterisk-dev] Asterisk 12 API improvements   Olle E. Johansson
• [asterisk-dev] Asterisk 12 API improvements   David M. Lee
• [asterisk-dev] [Code Review]: app_voicemail: fix problem where we kill the msg_id of a message when changing its folder in odbc storage.   Alec Davis
• [asterisk-dev] [Code Review]: app_voicemail: fix problem where we kill the msg_id of a message when changing its folder in odbc storage.   Alec Davis
• [asterisk-dev] Asterisk 11 won't compile?   Tilghman Lesher
• [asterisk-dev] Asterisk 12 API improvements   Olle E. Johansson
• [asterisk-dev] AST_BUILDOPT_SUM stability   Tzafrir Cohen
• [asterisk-dev] [Code Review] res_srtp: Fix a crash caused by an attempt to dealloc a session pointer that was never alloced or has already been dealloced.   Matt Jordan
• [asterisk-dev] Asterisk 12 API improvements   Matthew Jordan
• [asterisk-dev] [Code Review]: app_voicemail: fix problem where we kill the msg_id of a message when changing its folder in odbc storage.   jrose
• [asterisk-dev] [Code Review] res_srtp: Fix a crash caused by an attempt to dealloc a session pointer that was never alloced or has already been dealloced.   opticron
• [asterisk-dev] [Code Review]: app_voicemail: fix problem where we kill the msg_id of a message when changing its folder in odbc storage.   Matt Jordan
• [asterisk-dev] [Code Review]: app_voicemail: fix problem where we kill the msg_id of a message when changing its folder in odbc storage.   jrose
• [asterisk-dev] AST_BUILDOPT_SUM stability   Jason Parker
• [asterisk-dev] AST_BUILDOPT_SUM stability   Tilghman Lesher
• [asterisk-dev] AST_BUILDOPT_SUM stability   Tzafrir Cohen
• [asterisk-dev] Updated API wiki page   David M. Lee
• [asterisk-dev] [Code Review]: res_srtp: Fix a crash caused by an attempt to dealloc a session pointer that was never alloced or has already been dealloced.   jrose
• [asterisk-dev] Updated API wiki page   Tim Ringenbach
• [asterisk-dev] Updated API wiki page   Tilghman Lesher
• [asterisk-dev] Updated API wiki page   David M. Lee
• [asterisk-dev] Updated API wiki page   Tim Ringenbach
• [asterisk-dev] [Code Review] Ensure that Min-SE is included if a 422 was ever returned   Mark Michelson
• [asterisk-dev] [Code Review]: Make inline documentation readable   Mark Michelson
• [asterisk-dev] [Code Review] Change dynamic features buffer to use ast_str instead of a fixed sized buffer   Mark Michelson
• [asterisk-dev] [Code Review]: Change dynamic features buffer to use ast_str instead of a fixed sized buffer   elguero
• [asterisk-dev] [Code Review] Change dynamic features buffer to use ast_str instead of a fixed sized buffer   elguero
• [asterisk-dev] [Code Review] Manage translation table between SIP and ISDN hangup causes   Olle E Johansson
• [asterisk-dev] Need help on "how to get channel outside asterisk?"   beiyan jin
• [asterisk-dev] [Code Review]: Ensure that Min-SE is included if a 422 was ever returned   opticron
• [asterisk-dev] [Code Review]: Ensure that Min-SE is included if a 422 was ever returned   opticron
• [asterisk-dev] A* 11.0.1: rtp stuttering (clock skew) with confbridge with 2 participants   matteo.fortini at sadel.it
• [asterisk-dev] DAHDI migration to Git   Russ Meyerriecks
• [asterisk-dev] DAHDI migration to Git   Dan Jenkins
• [asterisk-dev] DAHDI migration to Git   Olle E. Johansson
• [asterisk-dev] [Code Review] Manage translation table between SIP and ISDN hangup causes   Birger Harzenetter
• [asterisk-dev] [Code Review] Tab completion for 'acl show'   Russell Bryant
• [asterisk-dev] [Code Review]: Manage translation table between SIP and ISDN hangup causes   Olle E Johansson
• [asterisk-dev] [Code Review] Tab completion for 'acl show'   Matt Jordan
• [asterisk-dev] [Code Review]: Tab completion for 'acl show'   Russell Bryant
• [asterisk-dev] [Code Review]: Tab completion for 'acl show'   Matt Jordan
• [asterisk-dev] [Code Review] Tab completion for 'acl show'   Russell Bryant
• [asterisk-dev] [Code Review] Tab completion for 'acl show'   Matt Jordan
• [asterisk-dev] [Code Review] Manage translation table between SIP and ISDN hangup causes   Matt Jordan
• [asterisk-dev] [Code Review]: Manage translation table between SIP and ISDN hangup causes   Olle E Johansson
• [asterisk-dev] [Code Review] res_srtp: Fix a crash caused by an attempt to dealloc a session pointer that was never alloced or has already been dealloced.   jrose
• [asterisk-dev] [Code Review] res_srtp: Fix a crash caused by an attempt to dealloc a session pointer that was never alloced or has already been dealloced.   jrose
• [asterisk-dev] DAHDI migration to Git   Shaun Ruffell
• [asterisk-dev] DAHDI migration to Git   Shaun Ruffell
• [asterisk-dev] DAHDI migration to Git   Russell Bryant
• [asterisk-dev] [Code Review] res_srtp: Fix a crash caused by an attempt to dealloc a session pointer that was never alloced or has already been dealloced.   jrose
• [asterisk-dev] [Code Review]: Manage translation table between SIP and ISDN hangup causes   jcolp
• [asterisk-dev] DAHDI migration to Git   Shaun Ruffell
• [asterisk-dev] DAHDI migration to Git   Russell Bryant
• [asterisk-dev] [Code Review]: Manage translation table between SIP and ISDN hangup causes   Olle E Johansson
• [asterisk-dev] DAHDI migration to Git   Dan Jenkins
• [asterisk-dev] [Code Review]: Manage translation table between SIP and ISDN hangup causes   jcolp
• [asterisk-dev] DAHDI migration to Git   Olle E. Johansson
• [asterisk-dev] Updated API wiki page   David M. Lee
• [asterisk-dev] DAHDI migration to Git   Tzafrir Cohen
• [asterisk-dev] DAHDI migration to Git   David M. Lee
• [asterisk-dev] DAHDI migration to Git   Shaun Ruffell
• [asterisk-dev] DAHDI migration to Git   Dan Jenkins
• [asterisk-dev] DAHDI migration to Git   Dan Jenkins
• [asterisk-dev] DAHDI migration to Git   Dan Jenkins
• [asterisk-dev] DAHDI migration to Git   Faidon Liambotis
• [asterisk-dev] Question about starpy-1.0.2 ownership   Bob
• [asterisk-dev] asterisk 1.8.18.1 Now Available   Asterisk Development Team
• [asterisk-dev] asterisk 10.10.1 Now Available   Asterisk Development Team
• [asterisk-dev] asterisk 11.0.2 Now Available   Asterisk Development Team
• [asterisk-dev] asterisk 1.8.19.0-rc3 Now Available   Asterisk Development Team
• [asterisk-dev] asterisk 10.11.0-rc3 Now Available   Asterisk Development Team
• [asterisk-dev] asterisk 11.1.0-rc3 Now Available   Asterisk Development Team
• [asterisk-dev] Question about starpy-1.0.2 ownership   David M. Lee
• [asterisk-dev] Updated API wiki page   Tim Ringenbach
• [asterisk-dev] Question about starpy-1.0.2 ownership   Paul Belanger
• [asterisk-dev] DAHDI migration to Git   Paul Belanger
• [asterisk-dev] [Code Review] Fix MOH on simultaneous confbridge user entry to a new conference.   rmudgett
• [asterisk-dev] Realtime testers needed - Path header support (Oolong branch)   Olle E. Johansson
• [asterisk-dev] [Code Review] Add Path header support to chan_sip   Olle E Johansson
• [asterisk-dev] Tracking MOH on queues - ASTERISK-20742   Lenz Emilitri
• [asterisk-dev] Tracking MOH on queues - ASTERISK-20742   Matthew Jordan
• [asterisk-dev] [Code Review] Fix MOH on simultaneous confbridge user entry to a new conference.   rmudgett
• [asterisk-dev] [Code Review] Fix MOH on simultaneous confbridge user entry to a new conference.   rmudgett
• [asterisk-dev] [Code Review] Patch to add a SIP peer configuration feature to allow the peer's configured codecs to take precedence on an outgoing call.   Brent Eagles
• [asterisk-dev] [Code Review] Patch to add a SIP peer configuration feature to allow the peer's configured codecs to take precedence on an outgoing call.   Brent Eagles
• [asterisk-dev] [Code Review] Patch to add a SIP peer configuration feature to allow the peer's configured codecs to take precedence on an outgoing call.   jcolp
• [asterisk-dev] [Code Review] Who Hung Up Support for chan_motif   jcolp
• [asterisk-dev] Patch for codec g722 in asterisk 1.8   Burhan Khan
• [asterisk-dev] [Code Review] Who Hung Up Support for chan_motif   opticron
• [asterisk-dev] [Code Review] Handle Session-Expires less than local Min-SE in 200 OK   opticron
• [asterisk-dev] [Code Review] Handle Session-Expires less than local Min-SE in 200 OK   Matt Jordan
• [asterisk-dev] [Code Review]: Make inline documentation readable   Matt Jordan
• [asterisk-dev] [Code Review] Change dynamic features buffer to use ast_str instead of a fixed sized buffer   Matt Jordan
• [asterisk-dev] Asterisk 11 ccar problem   Shanavaz E A
• [asterisk-dev] Asterisk 11 ccar problem   Tilghman Lesher
• [asterisk-dev] [Code Review] Rewrite skinny dialing to remove threaded simpleswitch   wedhorn
• [asterisk-dev] [Code Review] Rewrite skinny dialing to remove threaded simpleswitch   wedhorn
• [asterisk-dev] [Code Review] Rewrite skinny dialing to remove threaded simpleswitch   wedhorn
• [asterisk-dev] Updated API wiki page   Paul Belanger
• [asterisk-dev] [Code Review] Revamp of terminal color codes   Tilghman Lesher
• [asterisk-dev] Asterisk 10 - Final Bug Fix Release   Matthew Jordan
• [asterisk-dev] [Code Review] Revamp of terminal color codes   wdoekes
• [asterisk-dev] New SIP Channel Driver Data Access Layer   Joshua Colp
• [asterisk-dev] Data Access Layer Design   Joshua Colp
• [asterisk-dev] [Code Review]: Handle Session-Expires less than local Min-SE in 200 OK   opticron
• [asterisk-dev] [Code Review]: Revamp of terminal color codes   Tilghman Lesher
• [asterisk-dev] Updated API wiki page   David M. Lee
• [asterisk-dev] StarPy   David M. Lee
• [asterisk-dev] StarPy   Russell Bryant
• [asterisk-dev] [Code Review]: Revamp of terminal color codes   wdoekes
• [asterisk-dev] StarPy   David M. Lee
• [asterisk-dev] Asterisk 10 - Final Bug Fix Release   David M. Lee
• [asterisk-dev] Asterisk 10 - Final Bug Fix Release   Jason Parker
• [asterisk-dev] asterisk 1.8.19.0 Now Available   Asterisk Development Team
• [asterisk-dev] Asterisk 10.11.0 Now Available   Asterisk Development Team
• [asterisk-dev] Asterisk 11.1.0 Now Available   Asterisk Development Team
• [asterisk-dev] Asterisk 1.8.20.0-rc1 Now Available   Asterisk Development Team
• [asterisk-dev] Asterisk 10.12.0-rc1 Now Available   Asterisk Development Team
• [asterisk-dev] Asterisk 11.2.0-rc1 Now Available   Asterisk Development Team
• [asterisk-dev] SIP Stack - Update   Matthew Jordan
• [asterisk-dev] SIP Stack - Update   James Cloos
• [asterisk-dev] [Code Review] Fix MOH on simultaneous confbridge user entry to a new conference.   jrose
• [asterisk-dev] [Code Review]: Fix MOH on simultaneous confbridge user entry to a new conference.   rmudgett
• [asterisk-dev] [Code Review]: Fix MOH on simultaneous confbridge user entry to a new conference.   jrose
• [asterisk-dev] [Code Review]: Fix MOH on simultaneous confbridge user entry to a new conference.   jrose
• [asterisk-dev] SIP Stack - Update   Faidon Liambotis
• [asterisk-dev] SIP Stack - Update   Joshua Colp
• [asterisk-dev] SIP Stack - Update   Shaun Ruffell
• [asterisk-dev] [Code Review] Threadpool Support   Mark Michelson
• [asterisk-dev] [Code Review] Fix MOH on simultaneous confbridge user entry to a new conference.   rmudgett
• [asterisk-dev] SIP Stack - Update   Joshua Colp
• [asterisk-dev] [Code Review] Rewrite skinny dialing to remove threaded simpleswitch   wedhorn
• [asterisk-dev] [Code Review]: Initial Bridging Test Object and sample tests   Mark Michelson
• [asterisk-dev] [Code Review] Rewrite skinny dialing to remove threaded simpleswitch   wedhorn
• [asterisk-dev] [Code Review]: Ensure that Min-SE is included if a 422 was ever returned   Mark Michelson
• [asterisk-dev] [Code Review] Ensure that Min-SE is included if a 422 was ever returned   Mark Michelson
• [asterisk-dev] [Code Review]: Ensure that Min-SE is included if a 422 was ever returned   Mark Michelson
• [asterisk-dev] [Code Review] Add test for proper handling of 422 responses   Mark Michelson
• [asterisk-dev] [Code Review] Patch to add a SIP peer configuration feature to allow the peer's configured codecs to take precedence on an outgoing call.   Mark Michelson
• [asterisk-dev] [Code Review] Patch to add a SIP peer configuration feature to allow the peer's configured codecs to take precedence on an outgoing call.   Mark Michelson
• [asterisk-dev] SIP Stack - Update   Shaun Ruffell
• [asterisk-dev] SIP Stack - Update   Shaun Ruffell
• [asterisk-dev] SIP Stack - Update   Joshua Colp
• [asterisk-dev] [Code Review] Fix MOH on simultaneous confbridge user entry to a new conference.   Mark Michelson
• [asterisk-dev] [Code Review]: Fix MOH on simultaneous confbridge user entry to a new conference.   jcolp
• [asterisk-dev] [Code Review]: Fix MOH on simultaneous confbridge user entry to a new conference.   Mark Michelson
• [asterisk-dev] [Code Review] a driver for single-port FXO cards based on Si3052 chip + Si3011/17/18 DAA (Motorola 52-6116-2A, PM560MS etc.)   Tzafrir Cohen
• [asterisk-dev] [Code Review] Fix AMI redirect action with two channels failing to redirect both channels.   rmudgett
• [asterisk-dev] [Code Review] Fix AMI redirect action with two channels failing to redirect both channels.   Mark Michelson
• [asterisk-dev] [Code Review] Revamp of terminal color codes   Tilghman Lesher
• [asterisk-dev] [Code Review] Revamp of terminal color codes   Tilghman Lesher
• [asterisk-dev] [Code Review] Add aliases for Directory   Tilghman Lesher
• [asterisk-dev] fail2ban with asterisk security log?   Russell Bryant
• [asterisk-dev] [Code Review] Add test for proper handling of 422 responses   opticron
• [asterisk-dev] [Code Review]: Add test for proper handling of 422 responses   opticron
• [asterisk-dev] [Code Review] Remove HAVE_DEV_URANDOM   Russell Bryant
• [asterisk-dev] [Code Review] Remove HAVE_DEV_URANDOM   David Lee
• [asterisk-dev] fail2ban with asterisk security log?   Andrew Latham
• [asterisk-dev] Proposed Review board Usage policy change   Mark Michelson
• [asterisk-dev] fail2ban with asterisk security log?   Mark Michelson
• [asterisk-dev] [Code Review]: Fix AMI redirect action with two channels failing to redirect both channels.   rmudgett
• [asterisk-dev] [Code Review] Fix AMI redirect action with two channels failing to redirect both channels.   opticron
• [asterisk-dev] [Code Review]: Fix AMI redirect action with two channels failing to redirect both channels.   rmudgett
• [asterisk-dev] [Code Review] Rewrite skinny dialing to remove threaded simpleswitch   opticron
• [asterisk-dev] Proposed Review board Usage policy change   Russell Bryant
• [asterisk-dev] [Code Review] Add aliases for Directory   Mark Michelson
• [asterisk-dev] Proposed Review board Usage policy change   Jonathan Rose
• [asterisk-dev] Proposed Review board Usage policy change   Paul Belanger
• [asterisk-dev] [Code Review] Rewrite skinny dialing to remove threaded simpleswitch   wedhorn
• [asterisk-dev] [Code Review]: Rewrite skinny dialing to remove threaded simpleswitch   wedhorn
• [asterisk-dev] [svn-commits] wedhorn: trunk r378006 - /trunk/channels/chan_skinny.c   Richard Mudgett
• [asterisk-dev] [svn-commits] wedhorn: trunk r378006 - /trunk/channels/chan_skinny.c   Damien Wedhorn
• [asterisk-dev] [Code Review] Add GroupPickup to Skinny   wedhorn
• [asterisk-dev] Please review Path header support   Olle E. Johansson
• [asterisk-dev] [Code Review]: Implement Externaddr on a sip device basis   Olle E Johansson
• [asterisk-dev] [Code Review] Implement Externaddr on a sip device basis   Olle E Johansson
• [asterisk-dev] [Code Review]: Expand MEETME_INFO() with new options   Olle E Johansson
• [asterisk-dev] [Code Review] Expand MEETME_INFO() with new options   Olle E Johansson
• [asterisk-dev] [svn-commits] wedhorn: trunk r378006 - /trunk/channels/chan_skinny.c   Richard Mudgett
• [asterisk-dev] [Code Review] Add tests for L() and S() bridging options   opticron
• [asterisk-dev] Proposed RTP engine API change   Mark Michelson
• [asterisk-dev] SIP Stack - Update   Faidon Liambotis
• [asterisk-dev] SIP Stack - Update   Tilghman Lesher
• [asterisk-dev] SIP Stack - Update   Shaun Ruffell
• [asterisk-dev] SIP Stack - Update   Tilghman Lesher
• [asterisk-dev] SIP Stack - Update   Faidon Liambotis
• [asterisk-dev] SIP Stack - Update   Joshua Colp
• [asterisk-dev] SIP Stack - Update   Frederic Van Espen
• [asterisk-dev] [Code Review] Add tests for L() and S() bridging options   Mark Michelson
• [asterisk-dev] SIP Stack - Update   Paul Belanger
• [asterisk-dev] [Code Review]: Add aliases for Directory   Tilghman Lesher
• [asterisk-dev] [Code Review] Add aliases for Directory   Tilghman Lesher
• [asterisk-dev] [Code Review] Add aliases for Directory   Tilghman Lesher
• [asterisk-dev] SIP Stack - Update   Jared Smith
• [asterisk-dev] [Code Review]: Add aliases for Directory   Mark Michelson
• [asterisk-dev] [Code Review] Rewrite skinny dialing to remove threaded simpleswitch   Brad Latus
• [asterisk-dev] [Code Review] Add aliases for Directory   Tilghman Lesher
• [asterisk-dev] [Code Review] Add aliases for Directory   Tilghman Lesher
• [asterisk-dev] [Code Review]: Add aliases for Directory   Tilghman Lesher
• [asterisk-dev] [Code Review] Add GroupPickup to Skinny   wedhorn
• [asterisk-dev] [Code Review] A name/value list mechanism for raising events.   Brent Eagles
• [asterisk-dev] [Code Review] A name/value list mechanism for raising events.   Russell Bryant
• [asterisk-dev] [Code Review] Add connected line test based on the new bridging test object   opticron
• [asterisk-dev] [Code Review] Add connected line test based on the new bridging test object   Mark Michelson
• [asterisk-dev] [Code Review]: app_voicemail: fix problem where we kill the msg_id of a message when changing its folder in odbc storage.   Alec Davis
• [asterisk-dev] [Code Review]: app_voicemail: fix problem where we kill the msg_id of a message when changing its folder in odbc storage.   jrose
• [asterisk-dev] [Code Review]: app_voicemail: fix problem where we kill the msg_id of a message when changing its folder in odbc storage.   Alec Davis
• [asterisk-dev] [Code Review]: app_voicemail: fix problem where we kill the msg_id of a message when changing its folder in odbc storage.   Alec Davis
• [asterisk-dev] Asterisk CLI verbosity from another remote machine   Julien Pawlak
• [asterisk-dev] Asterisk CLI verbosity from another remote machine   Karsten Wemheuer
• [asterisk-dev] Asterisk 11 crashing on Hangup()   Pavel Troller
• [asterisk-dev] SIP Stack - Update   Matthew Jordan
• [asterisk-dev] Asterisk 11 crashing on Hangup()   Pavel Troller
• [asterisk-dev] [Code Review]: app_voicemail: fix problem where we kill the msg_id of a message when changing its folder in odbc storage.   jrose
• [asterisk-dev] [Code Review]: A name/value list mechanism for raising events.   Mark Michelson
• [asterisk-dev] SIP Stack - Update   Paul Belanger
• [asterisk-dev] SIP Stack - Update   Matthew Jordan
• [asterisk-dev] [Code Review]: app_voicemail: fix problem where we kill the msg_id of a message when changing its folder in odbc storage.   Alec Davis
• [asterisk-dev] SIP Stack - Update   Russell Bryant
• [asterisk-dev] SIP Stack - Update   Matthew Jordan
• [asterisk-dev] [Code Review]: app_voicemail: fix problem where we kill the msg_id of a message when changing its folder in odbc storage.   jrose
• [asterisk-dev] [Code Review] testsuite bridge tests: Add some additional features to automon and automixmon tests   jrose
• [asterisk-dev] [Code Review] testsuite bridge tests: Add some additional features to automon and automixmon tests   Mark Michelson
• [asterisk-dev] new addon module   Meadows Hoa
• [asterisk-dev] new addon module   Sean Kelly
• [asterisk-dev] new addon module   Russell Bryant
• [asterisk-dev] Asterisk CLI verbosity from another remote machine   Julien Pawlak
• [asterisk-dev] [Code Review] A name/value list mechanism for raising events.   Brent Eagles
• [asterisk-dev] [Code Review] res_srtp: Fix a crash caused by an attempt to dealloc a session pointer that was never alloced or has already been dealloced.   Matt Jordan
• [asterisk-dev] [Code Review] res_srtp: Fix a crash caused by an attempt to dealloc a session pointer that was never alloced or has already been dealloced.   jrose
• [asterisk-dev] Loop start vs kewl start for T1 interface   Justin Killen
• [asterisk-dev] [Code Review] res_srtp: Fix a crash caused by an attempt to dealloc a session pointer that was never alloced or has already been dealloced.   Matt Jordan
• [asterisk-dev] Loop start vs kewl start for T1 interface   Tilghman Lesher
• [asterisk-dev] Loop start vs kewl start for T1 interface   Justin Killen
• [asterisk-dev] [Code Review]: res_srtp: Fix a crash caused by an attempt to dealloc a session pointer that was never alloced or has already been dealloced.   jrose
• [asterisk-dev] [Code Review] dahdi_transcode: A command line utility to convert audio to/from formats supported by the TC400M VPM   Sean Bright
• [asterisk-dev] [Code Review] res_sip and res_sip_session design review   Mark Michelson
• [asterisk-dev] [Code Review] dahdi_transcode: A command line utility to convert audio to/from formats supported by the Digium TC400M VPM   Sean Bright
• [asterisk-dev] SIP Stack - Update   Paul Belanger
• [asterisk-dev] [Code Review]: app_voicemail: fix problem where we kill the msg_id of a message when changing its folder in odbc storage.   Alec Davis
• [asterisk-dev] [Code Review]: app_voicemail: fix problem where we kill the msg_id of a message when changing its folder in odbc storage.   jrose
• [asterisk-dev] libpri 1.4.14 Now Available   Asterisk Development Team
• [asterisk-dev] [Code Review] Clean up AsyncAGI command loop.   Sean Bright
• [asterisk-dev] [Code Review]: Clean up AsyncAGI command loop.   rmudgett
• [asterisk-dev] [Code Review] A name/value list mechanism for raising events.   Mark Michelson
• [asterisk-dev] [Code Review] Fix AMI redirect action with two channels failing to redirect both channels.   Mark Michelson
• [asterisk-dev] [Code Review] Implement Externaddr on a sip device basis   Mark Michelson
• [asterisk-dev] [Code Review] Fix potential memory leaks in taskprocessor code   Mark Michelson
• [asterisk-dev] [Code Review] Revamp of terminal color codes   Tilghman Lesher
• [asterisk-dev] [Code Review] Revamp of terminal color codes   Tilghman Lesher
• [asterisk-dev] [Code Review] Threadpool support: Taskprocessor changes and tests   Matt Jordan
• [asterisk-dev] [Code Review] Threadpool support: Taskprocessor changes and tests   Matt Jordan
• [asterisk-dev] [Code Review] add queue pause device states for queue member/queues   p_lindheimer
• [asterisk-dev] [Code Review] Fix SIP Notify Messages To Have The Proper IP Address In The FROM Field   elguero
• [asterisk-dev] [Code Review] Fix The Queue Log Reporting That All Calls Ending As COMPLETECALLER When The h Extension Is Present   elguero
• [asterisk-dev] [Code Review] Ensure chan_sip rejects encrypted streams without crypto info   Matt Jordan
• [asterisk-dev] [Code Review]: Add Path header support to chan_sip   Matt Jordan
• [asterisk-dev] [Code Review] Add Path header support to chan_sip   Matt Jordan
• [asterisk-dev] [Code Review] Add test for proper handling of 422 responses   Matt Jordan
• [asterisk-dev] [Code Review] Expand MEETME_INFO() with new options   Matt Jordan
• [asterisk-dev] [Code Review]: Add connected line test based on the new bridging test object   opticron
• [asterisk-dev] [Code Review] Add connected line test based on the new bridging test object   opticron
• [asterisk-dev] [Code Review]: Add test for proper handling of 422 responses   opticron
• [asterisk-dev] [Code Review] Add test for proper handling of 422 responses   opticron

Last message date: Mon Dec 31 16:15:32 CST 2012
Archived on: Mon Dec 31 16:34:06 CST 2012

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