September 2011 Archives by author

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Starting: Thu Sep 1 03:49:09 CDT 2011
Ending: Fri Sep 30 22:51:34 CDT 2011
Messages: 677

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• [asterisk-dev] CID Problem in Saudi Arabia   Mohamed Abdullah
• [asterisk-dev] CID Problem in Saudi Arabia   Mohamed Abdullah
• [asterisk-dev] CID Problem in Saudi Arabia (Russ Meyerriecks)   Mohamed Abdullah
• [asterisk-dev] [Code Review] automagically set a pointer to null after it is freed with ast_free   Benny Amorsen
• [asterisk-dev] [Code Review]: automagically set a pointer to null after it is freed with ast_free   Benny Amorsen
• [asterisk-dev] [Code Review] automagically set a pointer to null after it is freed with ast_free   Benny Amorsen
• [asterisk-dev] Asterisk Documentation Update   Jay Ashworth
• [asterisk-dev] Any plans to package the new FreePBX 2.9 into the repository?   Bruce B
• [asterisk-dev] [Code Review] Enable manager by default for the testsuite   Paul Belanger
• [asterisk-dev] [Code Review] Additional tests for app_voicemail   Paul Belanger
• [asterisk-dev] unexpected behavior between chan_sip and chan_iax2 configuration files   Paul Belanger
• [asterisk-dev] [Code Review] chan_iax2: authdebug is now disabled by default   Paul Belanger
• [asterisk-dev] [Bamboo] Paul Belanger commented on Asterisk - 10 - Ubuntu Lucid (10.04) 110   Paul Belanger
• [asterisk-dev] [Bamboo] Paul Belanger commented on Asterisk - Trunk - Ubuntu Lucid (10.04) 926   Paul Belanger
• [asterisk-dev] [Code Review] Initial tests for app_voicemail   Paul Belanger
• [asterisk-dev] [svn-commits] schmidts: trunk r334683 - /trunk/main/features.c   Paul Belanger
• [asterisk-dev] [Code Review] Additional tests for app_voicemail   Paul Belanger
• [asterisk-dev] unexpected behavior between chan_sip and chan_iax2 configuration files   Paul Belanger
• [asterisk-dev] [Code Review] TestCase.ami_connect clean up   Paul Belanger
• [asterisk-dev] [Code Review] Iterate though cdr.conf setting   Paul Belanger
• [asterisk-dev] unexpected behavior between chan_sip and chan_iax2 configuration files   Paul Belanger
• [asterisk-dev] [Code Review]: Iterate though cdr.conf setting   Paul Belanger
• [asterisk-dev] [Code Review] Iterate though cdr.conf setting   Paul Belanger
• [asterisk-dev] [Code Review] Adding the PICKUPSTATUS variable   Paul Belanger
• [asterisk-dev] [Code Review]: Move unsupported modules to their own directory   Paul Belanger
• [asterisk-dev] [Code Review] do parse defaultlanguage from asterisk.conf   Paul Belanger
• [asterisk-dev] [Code Review] Iterate over the [csv] section of cdr.conf   Paul Belanger
• [asterisk-dev] [Code Review]: Iterate over the [csv] section of cdr.conf   Paul Belanger
• [asterisk-dev] [Code Review] Iterate over the [csv] section of cdr.conf   Paul Belanger
• [asterisk-dev] [Code Review] TestSuite pre- and post-test condition check framework   Paul Belanger
• [asterisk-dev] [Code Review] TestSuite archiving of Asterisk logs on test run failure   Paul Belanger
• [asterisk-dev] [Code Review]: TestSuite archiving of Asterisk logs on test run failure   Paul Belanger
• [asterisk-dev] [Code Review]: TestSuite archiving of Asterisk logs on test run failure   Paul Belanger
• [asterisk-dev] [Code Review] Clean up dnsmgr.conf parsing   Paul Belanger
• [asterisk-dev] [Code Review] Clean up dnsmgr.conf parsing   Paul Belanger
• [asterisk-dev] [Code Review]: Clean up dnsmgr.conf parsing   Paul Belanger
• [asterisk-dev] [Code Review] Clean up dsp.conf parsing   Paul Belanger
• [asterisk-dev] [Code Review]: Clean up dnsmgr.conf parsing   Paul Belanger
• [asterisk-dev] [Code Review] Clean up dnsmgr.conf parsing   Paul Belanger
• [asterisk-dev] [Code Review] Clean up cel.conf parsing   Paul Belanger
• [asterisk-dev] [Code Review] Clean up cel.conf parsing   Paul Belanger
• [asterisk-dev] [svn-commits] mnicholson: branch 1.8 r335618 - in /branches/1.8/main: manager.c pbx.c   Paul Belanger
• [asterisk-dev] [Code Review]: Clean up cel.conf parsing   Paul Belanger
• [asterisk-dev] [Code Review] TestSuite pre- and post-test condition to check for locks   Paul Belanger
• [asterisk-dev] [svn-commits] mnicholson: branch 1.8 r335618 - in /branches/1.8/main: manager.c pbx.c   Paul Belanger
• [asterisk-dev] [Code Review] TestSuite pre- and post-test condition check framework   Paul Belanger
• [asterisk-dev] [Code Review] TestSuite pre- and post-test condition check framework   Paul Belanger
• [asterisk-dev] [Code Review] TestSuite archiving of Asterisk logs on test run failure   Paul Belanger
• [asterisk-dev] [Code Review] New DUNDi cli commands to list cache entries   Paul Belanger
• [asterisk-dev] [Code Review] SIP Dialog pre- / post-test condition checking   Paul Belanger
• [asterisk-dev] [Bamboo] Paul Belanger commented on Asterisk - 1.8 - FreeBSD 8.1 275   Paul Belanger
• [asterisk-dev] [svn-commits] tzafrir: branch 10 r335717 - /branches/10/main/asterisk.c   Paul Belanger
• [asterisk-dev] [Bamboo] Paul Belanger commented on Asterisk - Trunk - Ubuntu Lucid (10.04) 992   Paul Belanger
• [asterisk-dev] [Code Review] Tests for SIP handling of 484 Address Incomplete; Incomplete application   Paul Belanger
• [asterisk-dev] [Code Review]: Add test execution modes to the Asterisk Testsuite   Paul Belanger
• [asterisk-dev] [Code Review]: Add test execution modes to the Asterisk Testsuite   Paul Belanger
• [asterisk-dev] [Code Review]: Generate security events in chan_sip using new Security Events Framework   Paul Belanger
• [asterisk-dev] [svn-commits] irroot: branch 10 r337261 - in /branches/10: CHANGES apps/app_originate.c   Paul Belanger
• [asterisk-dev] [svn-commits] oej: branch 10 r336042 - in /branches/10: CHANGES CREDITS apps/app_meetme.c   Paul Belanger
• [asterisk-dev] [Code Review] New DUNDi cli commands to list cache entries   Paul Belanger
• [asterisk-dev] [svn-commits] oej: branch 10 r336042 - in /branches/10: CHANGES CREDITS apps/app_meetme.c   Paul Belanger
• [asterisk-dev] [Code Review] Clean up udptl.conf parsing   Paul Belanger
• [asterisk-dev] [Code Review] Clean up udptl.conf parsing   Paul Belanger
• [asterisk-dev] [Code Review] rework cdr_csv parsing (2nd attempt)   Paul Belanger
• [asterisk-dev] [Code Review] Clean up logger.conf parsing   Paul Belanger
• [asterisk-dev] [Code Review] Include test output in results XML file   Paul Belanger
• [asterisk-dev] [Code Review]: Include test output in results XML file   Paul Belanger
• [asterisk-dev] [Code Review] Clean up http.conf parsing   Paul Belanger
• [asterisk-dev] [Code Review] Clean up manager.conf parsing   Paul Belanger
• [asterisk-dev] [Code Review] Clean up asterisk.conf parsing   Paul Belanger
• [asterisk-dev] [Code Review]: Clean up manager.conf parsing   Paul Belanger
• [asterisk-dev] [Code Review]: Clean up udptl.conf parsing   Paul Belanger
• [asterisk-dev] [Code Review]: Clean up asterisk.conf parsing   Paul Belanger
• [asterisk-dev] [Code Review]: Clean up asterisk.conf parsing   Paul Belanger
• [asterisk-dev] Outbound Call Implementation   Paul Belanger
• [asterisk-dev] AstriCon 2011 Developer Days - Topics Requested   Paul Belanger
• [asterisk-dev] [svn-commits] transnexus: branch 1.8 r338137 - /branches/1.8/apps/app_osplookup.c   Paul Belanger
• [asterisk-dev] [svn-commits] transnexus: branch 1.8 r338137 - /branches/1.8/apps/app_osplookup.c   Paul Belanger
• [asterisk-dev] Bounty   Paul Belanger
• [asterisk-dev] [svn-commits] transnexus: branch 1.8 r338137 - /branches/1.8/apps/app_osplookup.c   Paul Belanger
• [asterisk-dev] [svn-commits] transnexus: branch 1.8 r338137 - /branches/1.8/apps/app_osplookup.c   Paul Belanger
• [asterisk-dev] [Code Review] Add #includeif statement   Paul Belanger
• [asterisk-dev] [Code Review] Add #includeif statement   Paul Belanger
• [asterisk-dev] [Code Review] Add #includeif statement   Paul Belanger
• [asterisk-dev] [svn-commits] rmudgett: trunk r338665 - in /trunk: ./ channels/chan_sip.c   Paul Belanger
• [asterisk-dev] asterisk 1.4 developer needed   Theo Belder
• [asterisk-dev] asterisk 1.4 developer needed   Theo Belder
• [asterisk-dev] Asterisk segfault bug   Stephan Bosch
• [asterisk-dev] Enhancement patch for app_voicemail   Russell Brown
• [asterisk-dev] A new kind of SIP attack ?   Tom Browning
• [asterisk-dev] A new kind of SIP attack ?   Tom Browning
• [asterisk-dev] A new kind of SIP attack ?   Tom Browning
• [asterisk-dev] [Code Review] Fix a crash in res_ais   Russell Bryant
• [asterisk-dev] [Code Review] Set default tonezone for SIP devices   Russell Bryant
• [asterisk-dev] [Code Review] Set default tonezone for SIP devices   Russell Bryant
• [asterisk-dev] [Code Review]: Fix a crash in res_ais   Russell Bryant
• [asterisk-dev] asterisk 1.4 developer needed   Russell Bryant
• [asterisk-dev] [Code Review] Fix crashes in ast_rtcp_write()   Russell Bryant
• [asterisk-dev] [Code Review]: Fix crashes in ast_rtcp_write()   Russell Bryant
• [asterisk-dev] [Code Review] automagically set a pointer to null after it is freed with ast_free   Russell Bryant
• [asterisk-dev] Outbound Call Implementation   Russell Bryant
• [asterisk-dev] Outbound Call Implementation   Russell Bryant
• [asterisk-dev] AstriCon 2011 Developer Days - Topics Requested   Russell Bryant
• [asterisk-dev] AstriCon 2011 Developer Days - Topics Requested   Russell Bryant
• [asterisk-dev] Bounty   Russell Bryant
• [asterisk-dev] Bounty   CDR
• [asterisk-dev] Bounty   CDR
• [asterisk-dev] Bounty   CDR
• [asterisk-dev] Bounty   CDR
• [asterisk-dev] Bounty   CDR
• [asterisk-dev] asterisk-dev Digest, Vol 86, Issue 126   Careem
• [asterisk-dev] documentation issue: snmpd agentx permissions   Stephane Chazelas
• [asterisk-dev] AST.pdf and AST.txt   Stephane Chazelas
• [asterisk-dev] DAHDI frame type   Kelvin Chua
• [asterisk-dev] DAHDI frame type   Kelvin Chua
• [asterisk-dev] Asterisk Documentation Update   Tzafrir Cohen
• [asterisk-dev] dahdi-linux 'make test' fails   Tzafrir Cohen
• [asterisk-dev] dahdi_speed   Tzafrir Cohen
• [asterisk-dev] [Code Review] Add #includeif statement   Saúl Ibarra Corretgé
• [asterisk-dev] Outbound Call Implementation   Andre Courchesne
• [asterisk-dev] The MulticastRTP channel needs more control of output   John R. Covert
• [asterisk-dev] NAT mapping while binding to IPv6   Klaus Darilion
• [asterisk-dev] Asterisk Documentation Update   Malcolm C. Davenport
• [asterisk-dev] Asterisk Documentation Update   Malcolm C. Davenport
• [asterisk-dev] Asterisk Documentation Update   Malcolm C. Davenport
• [asterisk-dev] [Code Review] Prevent segfault when asterisk restarts. Happens if call arrives before fully booted.   Alec Davis
• [asterisk-dev] [Code Review] Prevent segfault when asterisk restarts. Happens if call arrives before fully booted.   Alec Davis
• [asterisk-dev] [Code Review] log Asterisk Version number, Build etc into log file   Alec Davis
• [asterisk-dev] [Code Review]: log Asterisk Version number, Build etc into log file   Alec Davis
• [asterisk-dev] [Code Review] log Asterisk Version number, Build etc into log file   Alec Davis
• [asterisk-dev] [Code Review] log Asterisk Version number, Build etc into log file   Alec Davis
• [asterisk-dev] [Code Review] Prevent segfault when asterisk restarts. Happens if call arrives before fully booted.   Alec Davis
• [asterisk-dev] [Code Review] log Asterisk Version number, Build etc into log file   Alec Davis
• [asterisk-dev] [Code Review] Adding the PICKUPSTATUS variable   Alec Davis
• [asterisk-dev] [Code Review]: Adding the PICKUPSTATUS variable   Alec Davis
• [asterisk-dev] How to pass a parameter to the DAHDI kernel driver ?   Alec Davis
• [asterisk-dev] uncached realtime sip friends cannot set transport-protocol in sipregs   Walter Doekes
• [asterisk-dev] uncached realtime sip friends cannot set transport-protocol in sipregs   Walter Doekes
• [asterisk-dev] uncached realtime sip friends cannot set transport-protocol in sipregs   Walter Doekes
• [asterisk-dev] (correction to) uncached realtime sip friends cannot set transport-protocol in sipregs   Walter Doekes
• [asterisk-dev] uncached realtime sip friends cannot set transport-protocol in sipregs   Walter Doekes
• [asterisk-dev] uncached realtime sip friends cannot set transport-protocol in sipregs   Walter Doekes
• [asterisk-dev] addons/format_mp3.c - why is OUTSCALE 4096?   Walter Doekes
• [asterisk-dev] passive recording of FXO interface (tap to old analog extension)   Krzysztof Drewicz
• [asterisk-dev] New Opportunities   Steve Edwards
• [asterisk-dev] New Opportunities   Kory Ferbet
• [asterisk-dev] jira queue   Kevin P. Fleming
• [asterisk-dev] NAT mapping while binding to IPv6   Kevin P. Fleming
• [asterisk-dev] NAT mapping while binding to IPv6   Kevin P. Fleming
• [asterisk-dev] Reviewboard and commit policy change   Kevin P. Fleming
• [asterisk-dev] NAT mapping while binding to IPv6   Kevin P. Fleming
• [asterisk-dev] unexpected behavior between chan_sip and chan_iax2 configuration files   Kevin P. Fleming
• [asterisk-dev] Development Hero of the week: Paul Belanger   Kevin P. Fleming
• [asterisk-dev] Development Hero of the week: Paul Belanger   Kevin P. Fleming
• [asterisk-dev] How to pass a parameter to the DAHDI kernel driver ?   Kevin P. Fleming
• [asterisk-dev] How to pass a parameter to the DAHDI kernel driver ?   Kevin P. Fleming
• [asterisk-dev] How to pass a parameter to the DAHDI kernel driver ?   Kevin P. Fleming
• [asterisk-dev] Channel lock in "core show channel" CLI command ?   Kevin P. Fleming
• [asterisk-dev] Channel lock in "core show channel" CLI command ?   Kevin P. Fleming
• [asterisk-dev] [Code Review] automagically set a pointer to null after it is freed with ast_free   Kevin P. Fleming
• [asterisk-dev] After a BYE, chan_sip need 32s to destroy a SIP dialog. bug or feature?   Kevin P. Fleming
• [asterisk-dev] [Code Review]: automagically set a pointer to null after it is freed with ast_free   Kevin P. Fleming
• [asterisk-dev] [Code Review] automagically set a pointer to null after it is freed with ast_free   Kevin P. Fleming
• [asterisk-dev] Enabling strictrtp by default in 10   Kevin P. Fleming
• [asterisk-dev] [Code Review]: automagically set a pointer to null after it is freed with ast_free   Kevin P. Fleming
• [asterisk-dev] New Asterisk community developer   Kevin P. Fleming
• [asterisk-dev] [svn-commits] transnexus: branch 1.8 r338137 - /branches/1.8/apps/app_osplookup.c   Kevin P. Fleming
• [asterisk-dev] Asterisk 1.8 SIP_CAUSE performance regression   Gunnar
• [asterisk-dev] Learning To Code for Asterisk   Luke Hamburg
• [asterisk-dev] Learning To Code for Asterisk   Luke Hamburg
• [asterisk-dev] Bounty   Robert Huddleston
• [asterisk-dev] Bounty   Robert Huddleston
• [asterisk-dev] [Code Review] Update to chan_unistinm functionality   IgorG
• [asterisk-dev] [Code Review] Allow codecs to be negated in allow and disallow lines.   Olle E Johansson
• [asterisk-dev] [Code Review]: log Asterisk Version number, Build etc into log file   Olle E Johansson
• [asterisk-dev] [Code Review] No review yet - Set channel variables with setvar= when manager user originates calls   Olle E Johansson
• [asterisk-dev] [Code Review]: No review yet - Set channel variables with setvar= when manager user originates calls   Olle E Johansson
• [asterisk-dev] [Code Review] No review yet - Set channel variables with setvar= when manager user originates calls   Olle E Johansson
• [asterisk-dev] [Code Review] SIP 484 handling fix and new Incomplete app control frame   Olle E Johansson
• [asterisk-dev] [Code Review]: SIP 484 handling fix and new Incomplete app control frame   Olle E Johansson
• [asterisk-dev] [Code Review] Add the Cancel Reason Header if a pickup occurs   Olle E Johansson
• [asterisk-dev] [Code Review] AMI :: Debug manager actions in the CLI   Olle E Johansson
• [asterisk-dev] [Code Review]: AMI :: Debug manager actions in the CLI   Olle E Johansson
• [asterisk-dev] [Code Review]: No review yet - Set channel variables with setvar= when manager user originates calls   Olle E Johansson
• [asterisk-dev] [Code Review]: AMI :: Debug manager actions in the CLI   Olle E Johansson
• [asterisk-dev] [Code Review] Plenty of review done - Set channel variables with setvar= when manager user originates calls   Olle E Johansson
• [asterisk-dev] [Code Review]: Add manager event for early media   Olle E Johansson
• [asterisk-dev] [Code Review] Set default tonezone for SIP devices   Olle E Johansson
• [asterisk-dev] [Code Review] Set default tonezone for SIP devices   Olle E Johansson
• [asterisk-dev] [Code Review] Add a SIPpeerstatus command to chan_sip's manager interface.   Olle E Johansson
• [asterisk-dev] [Code Review] Plenty of review done - Set channel variables with setvar= when manager user originates calls   Olle E Johansson
• [asterisk-dev] [Code Review] Plenty of review done - Set channel variables with setvar= when manager user originates calls   Olle E Johansson
• [asterisk-dev] [Code Review]: Plenty of review done - Set channel variables with setvar= when manager user originates calls   Olle E Johansson
• [asterisk-dev] [Code Review] Add manager event for early media   Olle E Johansson
• [asterisk-dev] [Code Review]: Add manager event for early media   Olle E Johansson
• [asterisk-dev] [Code Review] AMI :: Debug manager actions in the CLI   Olle E Johansson
• [asterisk-dev] [Code Review] Add CLI-command "cdr pgsql status" based on "cdr mysql status"   Olle E Johansson
• [asterisk-dev] [Code Review]: [patch] Improve debug of ast_hangup   Olle E Johansson
• [asterisk-dev] [Code Review] Add CLI-command "cdr pgsql status" based on "cdr mysql status"   Olle E Johansson
• [asterisk-dev] [Code Review]: Dialplan function for manager account checks - AMI_CLIENT()   Olle E Johansson
• [asterisk-dev] [Code Review] Manager originate: Don't fall back to "s" when given extension does not exist   Olle E Johansson
• [asterisk-dev] [Code Review] Make pbx_start NOT default to s at default if extension not found   Olle E Johansson
• [asterisk-dev] [Code Review]: Make pbx_start NOT default to s at default if extension not found   Olle E Johansson
• [asterisk-dev] [Code Review] Manager originate: Don't fall back to "s" when given extension does not exist   Olle E Johansson
• [asterisk-dev] [Code Review]: Manager originate: Don't fall back to "s" when given extension does not exist   Olle E Johansson
• [asterisk-dev] [Code Review]: Make pbx_start NOT default to s at default if extension not found   Olle E Johansson
• [asterisk-dev] [Code Review]: Make pbx_start NOT default to s at default if extension not found   Olle E Johansson
• [asterisk-dev] [Code Review] Make pbx_start NOT default to s at default if extension not found   Olle E Johansson
• [asterisk-dev] [Code Review] Make pbx_start NOT default to s at default if extension not found   Olle E Johansson
• [asterisk-dev] [Code Review] Change default strictrtp to "yes" - enabled - in rtp.conf.sample and res_rtp_asterisk.c   Olle E Johansson
• [asterisk-dev] [Code Review]: Change default strictrtp to "yes" - enabled - in rtp.conf.sample and res_rtp_asterisk.c   Olle E Johansson
• [asterisk-dev] [Code Review]: Plenty of review done - Set channel variables with setvar= when manager user originates calls   Olle E Johansson
• [asterisk-dev] [Code Review]: Plenty of review done - Set channel variables with setvar= when manager user originates calls   Olle E Johansson
• [asterisk-dev] [Code Review] AMI - Set channel variables with setvar= when manager user originates calls   Olle E Johansson
• [asterisk-dev] [Code Review] Preserve DTMF length in main/features.c   Olle E Johansson
• [asterisk-dev] [Code Review] Add PAUSED tag to queue log when adding queue member as paused   Olle E Johansson
• [asterisk-dev] [Code Review] Fix possible memory leak with SIP realtime regs   Olle E Johansson
• [asterisk-dev] [Code Review] rtptimeout not working per peer   Olle E Johansson
• [asterisk-dev] [Code Review] AGI Streamfile does not restart MOH.   Olle E Johansson
• [asterisk-dev] [Code Review] Implement Externaddr on a sip device basis   Olle E Johansson
• [asterisk-dev] [Code Review] Enable support for early media in AMI originate action   Olle E Johansson
• [asterisk-dev] [Code Review]: Enable support for early media in AMI originate action   Olle E Johansson
• [asterisk-dev] chan_sip 1.8 & 1.6: Can't CANCEL a call attempt if MESSAGE has been sent back   Olle E. Johansson
• [asterisk-dev] uncached realtime sip friends cannot set transport-protocol in sipregs   Olle E. Johansson
• [asterisk-dev] Reviewboard and commit policy change   Olle E. Johansson
• [asterisk-dev] uncached realtime sip friends cannot set transport-protocol in sipregs   Olle E. Johansson
• [asterisk-dev] Reviewboard and commit policy change   Olle E. Johansson
• [asterisk-dev] uncached realtime sip friends cannot set transport-protocol in sipregs   Olle E. Johansson
• [asterisk-dev] uncached realtime sip friends cannot set transport-protocol in sipregs   Olle E. Johansson
• [asterisk-dev] unexpected behavior between chan_sip and chan_iax2 configuration files   Olle E. Johansson
• [asterisk-dev] uncached realtime sip friends cannot set transport-protocol in sipregs   Olle E. Johansson
• [asterisk-dev] unexpected behavior between chan_sip and chan_iax2 configuration files   Olle E. Johansson
• [asterisk-dev] A new kind of SIP attack ?   Olle E. Johansson
• [asterisk-dev] A new kind of SIP attack ?   Olle E. Johansson
• [asterisk-dev] Development Hero of the week: Paul Belanger   Olle E. Johansson
• [asterisk-dev] Development Hero of the week: Paul Belanger   Olle E. Johansson
• [asterisk-dev] AST.pdf and AST.txt   Olle E. Johansson
• [asterisk-dev] Enabling strictrtp by default in 10   Olle E. Johansson
• [asterisk-dev] SIP pedantic and multiline headers   Olle E. Johansson
• [asterisk-dev] [svn-commits] oej: branch 10 r336042 - in /branches/10: CHANGES CREDITS apps/app_meetme.c   Olle E. Johansson
• [asterisk-dev] SIP extension from-change for connected line identity not supported?   Olle E. Johansson
• [asterisk-dev] AstriCon 2011 Developer Days - Topics Requested   Olle E. Johansson
• [asterisk-dev] [Code Review] Add #includeif statement   Olle E. Johansson
• [asterisk-dev] AstriCon 2011 Update   Bryan M. Johns
• [asterisk-dev] AstriCon 2011 Open Source Showcase   Bryan M. Johns
• [asterisk-dev] AstriCon 2011 Developer Days - Topics Requested   Bryan M. Johns
• [asterisk-dev] [Bamboo] Matthew Jordan commented on Asterisk - 1.8 - FreeBSD 8.1 275   Matthew Jordan
• [asterisk-dev] Creating a Load Balanced Asterisk Voice Broadcasting Application   Nick Khamis
• [asterisk-dev] Moving Calls to Realtime DB   Nick Khamis
• [asterisk-dev] Outbound Call Implementation   Nick Khamis
• [asterisk-dev] Outbound Call Implementation   Nick Khamis
• [asterisk-dev] Outbound Call Implementation   Nick Khamis
• [asterisk-dev] AstriCon 2011 Developer Days - Topics Requested   Nick Khamis
• [asterisk-dev] CID Problem in Saudi Arabia   Nick Khamis
• [asterisk-dev] Outbound Call Implementation   Nick Khamis
• [asterisk-dev] AstriCon 2011 Developer Days - Topics Requested   Nick Khamis
• [asterisk-dev] Learning To Code for Asterisk   Nick Khamis
• [asterisk-dev] jira queue   Jeremy Kister
• [asterisk-dev] A new kind of SIP attack ?   Jeff LaCoursiere
• [asterisk-dev] A new kind of SIP attack ?   Jeff LaCoursiere
• [asterisk-dev] A new kind of SIP attack ?   Jeff LaCoursiere
• [asterisk-dev] Safe to backport fix for ASTERISK-18101 to 1.6.2.20?   Alex Villací­s Lasso
• [asterisk-dev] Safe to backport fix for ASTERISK-18101 to 1.6.2.20?   Alex Villací­s Lasso
• [asterisk-dev] Asterisk Documentation Update   Andrew Latham
• [asterisk-dev] After a BYE, chan_sip need 32s to destroy a SIP dialog. bug or feature?   Leo Leo
• [asterisk-dev] Something wrong with Set command compatibility mode (1.6 and 1.8)   Tilghman Lesher
• [asterisk-dev] [Code Review] Add SQLite 3 realtime driver   Tilghman Lesher
• [asterisk-dev] [Code Review]: Add SQLite 3 realtime driver   Tilghman Lesher
• [asterisk-dev] [Code Review]: Add SQLite 3 realtime driver   Tilghman Lesher
• [asterisk-dev] unexpected behavior between chan_sip and chan_iax2 configuration files   Tilghman Lesher
• [asterisk-dev] [Code Review] Allow codecs to be negated in allow and disallow lines.   Tilghman Lesher
• [asterisk-dev] [Code Review] Add SQLite 3 realtime driver   Tilghman Lesher
• [asterisk-dev] [Code Review]: Allow codecs to be negated in allow and disallow lines.   Tilghman Lesher
• [asterisk-dev] [Code Review]: Add SQLite 3 realtime driver   Tilghman Lesher
• [asterisk-dev] [Code Review] Add SQLite 3 realtime driver   Tilghman Lesher
• [asterisk-dev] [Code Review]: Add SQLite 3 realtime driver   Tilghman Lesher
• [asterisk-dev] [Code Review]: No review yet - Set channel variables with setvar= when manager user originates calls   Tilghman Lesher
• [asterisk-dev] unexpected behavior between chan_sip and chan_iax2 configuration files   Tilghman Lesher
• [asterisk-dev] unexpected behavior between chan_sip and chan_iax2 configuration files   Tilghman Lesher
• [asterisk-dev] [Code Review] Remove the use of queue_member uniqueid   Tilghman Lesher
• [asterisk-dev] [Code Review]: No review yet - Set channel variables with setvar= when manager user originates calls   Tilghman Lesher
• [asterisk-dev] [Code Review] Iterate though cdr.conf setting   Tilghman Lesher
• [asterisk-dev] [Code Review] Iterate though cdr.conf setting   Tilghman Lesher
• [asterisk-dev] [Code Review] fix so func_curl hashcompat can handle comma's in keys/values   Tilghman Lesher
• [asterisk-dev] asterisk 1.4 developer needed   Tilghman Lesher
• [asterisk-dev] [Code Review]: Clean up dnsmgr.conf parsing   Tilghman Lesher
• [asterisk-dev] [Code Review]: Clean up dnsmgr.conf parsing   Tilghman Lesher
• [asterisk-dev] [Code Review] Clean up dnsmgr.conf parsing   Tilghman Lesher
• [asterisk-dev] [svn-commits] mnicholson: branch 1.8 r335618 - in /branches/1.8/main: manager.c pbx.c   Tilghman Lesher
• [asterisk-dev] [Code Review]: Fix crashes in ast_rtcp_write()   Tilghman Lesher
• [asterisk-dev] [Code Review] Make pbx_start NOT default to s at default if extension not found   Tilghman Lesher
• [asterisk-dev] [Code Review] Manager originate: Don't fall back to "s" when given extension does not exist   Tilghman Lesher
• [asterisk-dev] [Code Review] Manager originate: Don't fall back to "s" when given extension does not exist   Tilghman Lesher
• [asterisk-dev] [Code Review] Set Write format if it does not match native format or current write format.   Tilghman Lesher
• [asterisk-dev] [Code Review]: Make pbx_start NOT default to s at default if extension not found   Tilghman Lesher
• [asterisk-dev] [Code Review] Make pbx_start NOT default to s at default if extension not found   Tilghman Lesher
• [asterisk-dev] [Code Review]: Manager originate: Don't fall back to "s" when given extension does not exist   Tilghman Lesher
• [asterisk-dev] [Code Review] Make pbx_start NOT default to s at default if extension not found   Tilghman Lesher
• [asterisk-dev] [Code Review] Plenty of review done - Set channel variables with setvar= when manager user originates calls   Tilghman Lesher
• [asterisk-dev] [Code Review]: Plenty of review done - Set channel variables with setvar= when manager user originates calls   Tilghman Lesher
• [asterisk-dev] [Code Review] Modify merge message so commit log message comes first.   Tilghman Lesher
• [asterisk-dev] [Code Review] Modify merge message so commit log message comes first.   Tilghman Lesher
• [asterisk-dev] [Code Review]: Modify merge message so commit log message comes first.   Tilghman Lesher
• [asterisk-dev] [Code Review]: Modify merge message so commit log message comes first.   Tilghman Lesher
• [asterisk-dev] [Code Review] Modify merge message so commit log message comes first.   Tilghman Lesher
• [asterisk-dev] [Code Review] Clean up asterisk.conf parsing   Tilghman Lesher
• [asterisk-dev] [Code Review] Clean up manager.conf parsing   Tilghman Lesher
• [asterisk-dev] [Code Review]: Clean up asterisk.conf parsing   Tilghman Lesher
• [asterisk-dev] Outbound Call Implementation   Tilghman Lesher
• [asterisk-dev] Outbound Call Implementation   Tilghman Lesher
• [asterisk-dev] [Code Review]: Clean up manager.conf parsing   Tilghman Lesher
• [asterisk-dev] [Code Review] [regression] Non-portable SQL added to app_voicemail   Leif Madsen
• [asterisk-dev] [Code Review] AMI :: Debug manager actions in the CLI   Leif Madsen
• [asterisk-dev] [Code Review] No review yet - Set channel variables with setvar= when manager user originates calls   Leif Madsen
• [asterisk-dev] [Code Review]: No review yet - Set channel variables with setvar= when manager user originates calls   Leif Madsen
• [asterisk-dev] [Code Review] New DUNDi cli commands to list cache entries   Leif Madsen
• [asterisk-dev] Enhancement patch for app_voicemail   Leif Madsen
• [asterisk-dev] Channel lock in "core show channel" CLI command ?   Leif Madsen
• [asterisk-dev] [Code Review] Change default strictrtp to "yes" - enabled - in rtp.conf.sample and res_rtp_asterisk.c   Leif Madsen
• [asterisk-dev] [Code Review] Change default strictrtp to "yes" - enabled - in rtp.conf.sample and res_rtp_asterisk.c   Leif Madsen
• [asterisk-dev] Time for a new beta of version 10?   Leif Madsen
• [asterisk-dev] [Code Review] New DUNDi cli commands to list cache entries   Leif Madsen
• [asterisk-dev] AstriCon 2011 Developer Days - Topics Requested   Leif Madsen
• [asterisk-dev] Bounty   Leif Madsen
• [asterisk-dev] Bounty   Leif Madsen
• [asterisk-dev] Asterisk 1.8 SIP_CAUSE performance regression   Bernard Merindol(F)
• [asterisk-dev] CID Problem in Saudi Arabia   Russ Meyerriecks
• [asterisk-dev] CID Problem in Saudi Arabia (Russ Meyerriecks)   Russ Meyerriecks
• [asterisk-dev] [Code Review] automagically set a pointer to null after it is freed with ast_free   Tony Mountifield
• [asterisk-dev] [Code Review] automagically set a pointer to null after it is freed with ast_free   Tony Mountifield
• [asterisk-dev] [Code Review] automagically set a pointer to null after it is freed with ast_free   Tony Mountifield
• [asterisk-dev] addons/format_mp3.c - why is OUTSCALE 4096?   Tony Mountifield
• [asterisk-dev] Asterisk segfault bug   Richard Mudgett
• [asterisk-dev] [Code Review]: Add a SIPpeerstatus command to chan_sip's manager interface.   Matthew Nicholson
• [asterisk-dev] [Code Review] Add a SIPpeerstatus command to chan_sip's manager interface.   Matthew Nicholson
• [asterisk-dev] [Code Review]: Add a SIPpeerstatus command to chan_sip's manager interface.   Matthew Nicholson
• [asterisk-dev] [Code Review] Asterisk 1.8 Deadlock in app_queue   Matthew Nicholson
• [asterisk-dev] [Code Review] Add manager event for early media   Matthew Nicholson
• [asterisk-dev] [Code Review] Fix a crash in res_ais   Matthew Nicholson
• [asterisk-dev] [svn-commits] mnicholson: branch 1.8 r335618 - in /branches/1.8/main: manager.c pbx.c   Matthew Nicholson
• [asterisk-dev] [Code Review] Asterisk 1.8 Deadlock in app_queue   Matthew Nicholson
• [asterisk-dev] Asterisk 1.8 SIP_CAUSE performance regression   Matthew Nicholson
• [asterisk-dev] [Code Review] Asterisk 1.8 Deadlock in app_queue   Matthew Nicholson
• [asterisk-dev] [Code Review] automagically set a pointer to null after it is freed with ast_free   Matthew Nicholson
• [asterisk-dev] [Code Review] Add test execution modes to the Asterisk Testsuite   Matthew Nicholson
• [asterisk-dev] [Code Review]: Add test execution modes to the Asterisk Testsuite   Matthew Nicholson
• [asterisk-dev] [Code Review] Allow fine grained control over test version asterisk compatibility specification   Matthew Nicholson
• [asterisk-dev] [Code Review] Allow fine grained control over test version asterisk compatibility specification   Matthew Nicholson
• [asterisk-dev] [Code Review] Allow fine grained control over test version asterisk compatibility specification   Matthew Nicholson
• [asterisk-dev] [Code Review] Allow fine grained control over test version asterisk compatibility specification   Matthew Nicholson
• [asterisk-dev] [Code Review] Include test output in results XML file   Matthew Nicholson
• [asterisk-dev] [Code Review] rework cdr_csv parsing (2nd attempt)   Matthew Nicholson
• [asterisk-dev] [Bamboo] Matthew Nicholson commented on Asterisk - 1.6.2 - Ubuntu Lucid (10.04) - amd64 406   Matthew Nicholson
• [asterisk-dev] Use of restart_monitor function in chan_sip module   Nikhil
• [asterisk-dev] Time for a new beta of version 10?   Jeffrey Ollie
• [asterisk-dev] [Code Review] Change default strictrtp to "yes" - enabled - in rtp.conf.sample and res_rtp_asterisk.c   Jason Parker
• [asterisk-dev] modification for QUEUE_WAITING_COUNT   Clod Patry
• [asterisk-dev] AstriCon 2011 Developer Days - Topics Requested   Clod Patry
• [asterisk-dev] NAT mapping while binding to IPv6   Simon Perreault
• [asterisk-dev] NAT mapping while binding to IPv6   Simon Perreault
• [asterisk-dev] NAT mapping while binding to IPv6   Simon Perreault
• [asterisk-dev] [Code Review] automagically set a pointer to null after it is freed with ast_free   Simon Perreault
• [asterisk-dev] [Code Review] automagically set a pointer to null after it is freed with ast_free   Simon Perreault
• [asterisk-dev] [Code Review]: automagically set a pointer to null after it is freed with ast_free   Simon Perreault
• [asterisk-dev] SIP pedantic and multiline headers   Matt Riddell
• [asterisk-dev] Moving Calls to Realtime DB   Matt Riddell
• [asterisk-dev] Learning To Code for Asterisk   Jonathan Rose
• [asterisk-dev] dahdi_speed   Shaun Ruffell
• [asterisk-dev] dahdi-linux 'make test' fails   Shaun Ruffell
• [asterisk-dev] [svn-commits] transnexus: branch 1.8 r338137 - /branches/1.8/apps/app_osplookup.c   Di-Shi Sun
• [asterisk-dev] [svn-commits] transnexus: branch 1.8 r338137 - /branches/1.8/apps/app_osplookup.c   Di-Shi Sun
• [asterisk-dev] [svn-commits] transnexus: branch 1.8 r338137 - /branches/1.8/apps/app_osplookup.c   Di-Shi Sun
• [asterisk-dev] Scheduled Maintenance for Asterisk Issue Tracker (JIRA)   Asterisk Development Team
• [asterisk-dev] Asterisk 1.8.7.0-rc1 Now Available   Asterisk Development Team
• [asterisk-dev] DAHDI-Linux 2.5.0.1 and DAHDI-Tools 2.5.0.1 Released   Asterisk Development Team
• [asterisk-dev] iLBC support in Asterisk after Google's acquisition of GIPS   Asterisk Development Team
• [asterisk-dev] Asterisk 1.8.7.0-rc2 Now Available   Asterisk Development Team
• [asterisk-dev] Asterisk 1.8.7.0 Now Available   Asterisk Development Team
• [asterisk-dev] Scheduled Maintenance for Asterisk Project community services   Asterisk Development Team
• [asterisk-dev] Something wrong with Set command compatibility mode (1.6 and 1.8)   Pavel Troller
• [asterisk-dev] chan_sip 1.8 & 1.6: Can't CANCEL a call attempt if MESSAGE has been sent back   Pavel Troller
• [asterisk-dev] A new kind of SIP attack ?   Pavel Troller
• [asterisk-dev] How to pass a parameter to the DAHDI kernel driver ?   Pavel Troller
• [asterisk-dev] How to pass a parameter to the DAHDI kernel driver ?   Pavel Troller
• [asterisk-dev] How to pass a parameter to the DAHDI kernel driver ?   Pavel Troller
• [asterisk-dev] Channel lock in "core show channel" CLI command ?   Pavel Troller
• [asterisk-dev] Channel lock in "core show channel" CLI command ?   Pavel Troller
• [asterisk-dev] Channel lock in "core show channel" CLI command ?   Pavel Troller
• [asterisk-dev] [Code Review] SIP 484 handling fix and new Incomplete app control frame   Pavel Troller
• [asterisk-dev] NAT mapping while binding to IPv6   Jeremy Visser
• [asterisk-dev] NAT mapping while binding to IPv6   Jeremy Visser
• [asterisk-dev] NAT mapping while binding to IPv6   Jeremy Visser
• [asterisk-dev] [Code Review]: automagically set a pointer to null after it is freed with ast_free   David Vossel
• [asterisk-dev] [Code Review]: automagically set a pointer to null after it is freed with ast_free   David Vossel
• [asterisk-dev] [Code Review] Fix crashes in ast_rtcp_write()   David Vossel
• [asterisk-dev] [Code Review] Fix crashes in ast_rtcp_write()   David Vossel
• [asterisk-dev] [Code Review] Adjust formats of chan_local when channel we proxying changes   David Vossel
• [asterisk-dev] After a BYE, chan_sip need 32s to destroy a SIP dialog. bug or feature?   Kristijan Vrban
• [asterisk-dev] NAT mapping while binding to IPv6   Ryan Wagoner
• [asterisk-dev] AstriCon 2011 Developer Days - Topics Requested   Watkins, Bradley
• [asterisk-dev] [Code Review] Set Write format if it does not match native format or current write format.   Eric Wieling
• [asterisk-dev] [Code Review] Add SQLite 3 realtime driver   Terry Wilson
• [asterisk-dev] [Code Review]: Add SQLite 3 realtime driver   Terry Wilson
• [asterisk-dev] [Code Review]: Add SQLite 3 realtime driver   Terry Wilson
• [asterisk-dev] [Code Review]: Add SQLite 3 realtime driver   Terry Wilson
• [asterisk-dev] [Code Review] Allow codecs to be negated in allow and disallow lines.   Terry Wilson
• [asterisk-dev] [Code Review]: Add SQLite 3 realtime driver   Terry Wilson
• [asterisk-dev] [Code Review] Add SQLite 3 realtime driver   Terry Wilson
• [asterisk-dev] [Code Review]: Add SQLite 3 realtime driver   Terry Wilson
• [asterisk-dev] [Code Review]: Add SQLite 3 realtime driver   Terry Wilson
• [asterisk-dev] [Code Review] Add SQLite 3 realtime driver   Terry Wilson
• [asterisk-dev] [Code Review] Add option to automatically pause a busy queue member   Terry Wilson
• [asterisk-dev] [Code Review] Fix possible memory leak with SIP realtime regs   Terry Wilson
• [asterisk-dev] [Code Review] Fix possible memory leak with SIP realtime regs   Terry Wilson
• [asterisk-dev] [Code Review] rewrite of matching code in chan_sip.c realtime_peer()   Terry Wilson
• [asterisk-dev] [Code Review]: Fix possible memory leak with SIP realtime regs   Terry Wilson
• [asterisk-dev] [Code Review] Fix possible memory leak with SIP realtime regs   Terry Wilson
• [asterisk-dev] [Code Review] rewrite of matching code in chan_sip.c realtime_peer()   Terry Wilson
• [asterisk-dev] Asterisk 1.8.7.0-rc1 Now Available   sean darcy
• [asterisk-dev] [Code Review] No review yet - Set channel variables with setvar= when manager user originates calls   davidw
• [asterisk-dev] [Code Review] Generate security events in chan_sip using new Security Events Framework   elguero
• [asterisk-dev] [Code Review] Generate security events in chan_sip using new Security Events Framework   elguero
• [asterisk-dev] [Code Review]: Generate security events in chan_sip using new Security Events Framework   elguero
• [asterisk-dev] [Code Review] Generate security events in chan_sip using new Security Events Framework   elguero
• [asterisk-dev] [Code Review]: Generate security events in chan_sip using new Security Events Framework   elguero
• [asterisk-dev] [Code Review] Generate security events in chan_sip using new Security Events Framework   elguero
• [asterisk-dev] [Code Review]: Generate security events in chan_sip using new Security Events Framework   elguero
• [asterisk-dev] [Code Review]: Generate security events in chan_sip using new Security Events Framework   elguero
• [asterisk-dev] [Code Review] Add PAUSED tag to queue log when adding queue member as paused   garlew
• [asterisk-dev] [Code Review] Add PAUSED tag to queue log when adding queue member as paused   garlew
• [asterisk-dev] [Code Review]: Add PAUSED tag to queue log when adding queue member as paused   garlew
• [asterisk-dev] [Code Review] chan_misdn has a most broken round robin routiune   irroot
• [asterisk-dev] [Code Review] chan_misdn has a most broken round robin routiune   irroot
• [asterisk-dev] [Code Review] chan_misdn has a most broken round robin routiune   irroot
• [asterisk-dev] [Code Review] chan_misdn has a most broken round robin routiune   irroot
• [asterisk-dev] [Code Review]: Asterisk 1.8 Deadlock in app_queue   irroot
• [asterisk-dev] [Code Review] Asterisk 1.8 Deadlock in app_queue   irroot
• [asterisk-dev] [Code Review] Prevent segfault when asterisk restarts. Happens if call arrives before fully booted.   irroot
• [asterisk-dev] [Code Review] Lock the peer->mwipvt to avoid crashed in sip history management when overflowing history entries   irroot
• [asterisk-dev] [Code Review] chan_iax2: authdebug is now disabled by default   irroot
• [asterisk-dev] [Code Review] [regression] Non-portable SQL added to app_voicemail   irroot
• [asterisk-dev] [Code Review] Add option to automatically pause a busy queue member   irroot
• [asterisk-dev] [Code Review] log Asterisk Version number, Build etc into log file   irroot
• [asterisk-dev] [Code Review] move function VALID_EXTEN to DIALPLAN_EXISTS   irroot
• [asterisk-dev] [Code Review] Asterisk 1.8 Deadlock in app_queue   irroot
• [asterisk-dev] [Code Review] Asterisk 1.8 Deadlock in app_queue   irroot
• [asterisk-dev] [Code Review] Iterate though cdr.conf setting   irroot
• [asterisk-dev] [Code Review] build_peer doesnt unlink a peer object from peers_by_ip container which leads to a wrong refcounter value   irroot
• [asterisk-dev] [Code Review] Asterisk 1.8 Deadlock in app_queue   irroot
• [asterisk-dev] [Code Review] Asterisk 1.8 Deadlock in app_queue   irroot
• [asterisk-dev] [Code Review] Asterisk 1.8 Deadlock in app_queue   irroot
• [asterisk-dev] [Code Review]: Asterisk 1.8 Deadlock in app_queue   irroot
• [asterisk-dev] [Code Review] WaitFAX application   irroot
• [asterisk-dev] [Code Review] WaitFAX application   irroot
• [asterisk-dev] [Code Review] Set Write format if it does not match native format or current write format.   irroot
• [asterisk-dev] [Code Review] Set Write format if it does not match native format or current write format.   irroot
• [asterisk-dev] [Code Review] Allow Setting Auth Tag Bit length Based on invite or config option [BUG]   irroot
• [asterisk-dev] [Code Review] app_queue.c updatecdr appendcdr if needed only update last cdr on stack   irroot
• [asterisk-dev] [Code Review]: automagically set a pointer to null after it is freed with ast_free   irroot
• [asterisk-dev] [Code Review] Set Write format if it does not match native format or current write format.   irroot
• [asterisk-dev] [Code Review] Changes to h323 to allow use with h323plus > 1.20.   irroot
• [asterisk-dev] [Code Review] WaitFAX application   irroot
• [asterisk-dev] [Code Review] Make pbx_start NOT default to s at default if extension not found   irroot
• [asterisk-dev] [Code Review] Set Write format if it does not match native format or current write format.   irroot
• [asterisk-dev] [Code Review]: Set Write format if it does not match native format or current write format.   irroot
• [asterisk-dev] [Code Review] Change default strictrtp to "yes" - enabled - in rtp.conf.sample and res_rtp_asterisk.c   irroot
• [asterisk-dev] [Code Review] Set Write format if it does not match native format or current write format.   irroot
• [asterisk-dev] [Code Review] rtptimeout not working per peer   irroot
• [asterisk-dev] [Code Review] rtptimeout not working per peer   irroot
• [asterisk-dev] [Code Review] rtptimeout not working per peer   irroot
• [asterisk-dev] [Code Review] Adjust formats of chan_local when channel we proxying changes   irroot
• [asterisk-dev] [Code Review]: Adjust formats of chan_local when channel we proxying changes   irroot
• [asterisk-dev] [Code Review] Adjust formats of chan_local when channel we proxying changes   irroot
• [asterisk-dev] [Code Review] rtptimeout not working per peer   irroot
• [asterisk-dev] [Code Review] Preserve DTMF length in main/features.c   irroot
• [asterisk-dev] [Code Review] Fix authentication regression in res_jabber in component mode brought on by r333265.   irroot
• [asterisk-dev] [Code Review] Adjust formats of chan_local when channel we proxying changes   irroot
• [asterisk-dev] [Code Review] Run a macro on picked up channel when bridging   irroot
• [asterisk-dev] [Code Review]: Adjust formats of chan_local when channel we proxying changes   irroot
• [asterisk-dev] [Code Review] Enable support for early media in AMI originate action   irroot
• [asterisk-dev] [Code Review] Adjust formats of chan_local when channel we proxying changes   irroot
• [asterisk-dev] [Code Review] Generate security events in chan_sip using new Security Events Framework   jrose
• [asterisk-dev] [Code Review]: Generate security events in chan_sip using new Security Events Framework   jrose
• [asterisk-dev] [Code Review]: Generate security events in chan_sip using new Security Events Framework   jrose
• [asterisk-dev] [Code Review] Generate security events in chan_sip using new Security Events Framework   jrose
• [asterisk-dev] [Code Review] Generate security events in chan_sip using new Security Events Framework   jrose
• [asterisk-dev] [Code Review] Fix authentication regression in res_jabber in component mode brought on by r333265.   jrose
• [asterisk-dev] [Code Review] res_musiconhold misuse of refs Take 2   kobaz
• [asterisk-dev] [Code Review] log Asterisk Version number, Build etc into log file   lathama
• [asterisk-dev] [Code Review]: log Asterisk Version number, Build etc into log file   lathama
• [asterisk-dev] [Code Review] AMI :: Debug manager actions in the CLI   lathama
• [asterisk-dev] [Code Review]: Additional tests for app_voicemail   mjordan
• [asterisk-dev] [Code Review] SIP 484 handling fix and new Incomplete app control frame   mjordan
• [asterisk-dev] [Code Review] Tests for SIP handling of 484 Address Incomplete; Incomplete application   mjordan
• [asterisk-dev] [Code Review]: SIP 484 handling fix and new Incomplete app control frame   mjordan
• [asterisk-dev] [Code Review] Enable manager by default for the testsuite   mjordan
• [asterisk-dev] [Code Review] SIP 484 handling fix and new Incomplete app control frame   mjordan
• [asterisk-dev] [Code Review]: SIP 484 handling fix and new Incomplete app control frame   mjordan
• [asterisk-dev] [Code Review] SIP 484 handling fix and new Incomplete app control frame   mjordan
• [asterisk-dev] [Code Review]: SIP 484 handling fix and new Incomplete app control frame   mjordan
• [asterisk-dev] [Code Review] SIP 484 handling fix and new Incomplete app control frame   mjordan
• [asterisk-dev] [Code Review] No review yet - Set channel variables with setvar= when manager user originates calls   mjordan
• [asterisk-dev] [Code Review]: No review yet - Set channel variables with setvar= when manager user originates calls   mjordan
• [asterisk-dev] [Code Review]: SIP 484 handling fix and new Incomplete app control frame   mjordan
• [asterisk-dev] [Code Review] SIP 484 handling fix and new Incomplete app control frame   mjordan
• [asterisk-dev] [Code Review]: No review yet - Set channel variables with setvar= when manager user originates calls   mjordan
• [asterisk-dev] [Code Review] Additional tests for app_voicemail   mjordan
• [asterisk-dev] [Code Review]: Additional tests for app_voicemail   mjordan
• [asterisk-dev] [Code Review]: Initial tests for app_voicemail   mjordan
• [asterisk-dev] [Code Review] SIP 484 handling fix and new Incomplete app control frame   mjordan
• [asterisk-dev] [Code Review] SIP 484 handling fix and new Incomplete app control frame   mjordan
• [asterisk-dev] [Code Review] SIP 484 handling fix and new Incomplete app control frame   mjordan
• [asterisk-dev] [Code Review] TestCase.ami_connect clean up   mjordan
• [asterisk-dev] [Code Review] TestSuite pre- and post-test condition check framework   mjordan
• [asterisk-dev] [Code Review]: TestSuite pre- and post-test condition check framework   mjordan
• [asterisk-dev] [Code Review] TestSuite archiving of Asterisk logs on test run failure   mjordan
• [asterisk-dev] [Code Review]: TestSuite archiving of Asterisk logs on test run failure   mjordan
• [asterisk-dev] [Code Review]: TestSuite archiving of Asterisk logs on test run failure   mjordan
• [asterisk-dev] [Code Review]: TestSuite archiving of Asterisk logs on test run failure   mjordan
• [asterisk-dev] [Code Review]: TestSuite archiving of Asterisk logs on test run failure   mjordan
• [asterisk-dev] [Code Review] TestSuite archiving of Asterisk logs on test run failure   mjordan
• [asterisk-dev] [Code Review] TestSuite pre- and post-test condition to check for locks   mjordan
• [asterisk-dev] [Code Review]: TestSuite pre- and post-test condition check framework   mjordan
• [asterisk-dev] [Code Review]: TestSuite pre- and post-test condition check framework   mjordan
• [asterisk-dev] [Code Review] SIP Dialog pre- / post-test condition checking   mjordan
• [asterisk-dev] [Code Review] SIP Dialog pre- / post-test condition checking   mjordan
• [asterisk-dev] [Code Review]: TestSuite pre- and post-test condition check framework   mjordan
• [asterisk-dev] [Code Review]: SIP Dialog pre- / post-test condition checking   mjordan
• [asterisk-dev] [Code Review] SIP Dialog pre- / post-test condition checking   mjordan
• [asterisk-dev] [Code Review] Incorrect Voicemail duration reported in messages containing silence (take 2)   mjordan
• [asterisk-dev] [Code Review] Plenty of review done - Set channel variables with setvar= when manager user originates calls   mjordan
• [asterisk-dev] [Code Review] Add test execution modes to the Asterisk Testsuite   mjordan
• [asterisk-dev] [Code Review] Add test execution modes to the Asterisk Testsuite   mjordan
• [asterisk-dev] [Code Review]: Tests for SIP handling of 484 Address Incomplete; Incomplete application   mjordan
• [asterisk-dev] [Code Review] extenpatternmatchnew breaks ability for caller to exit queue   mjordan
• [asterisk-dev] [Code Review]: extenpatternmatchnew breaks ability for caller to exit queue   mjordan
• [asterisk-dev] [Code Review]: Add test execution modes to the Asterisk Testsuite   mjordan
• [asterisk-dev] [Code Review]: Add test execution modes to the Asterisk Testsuite   mjordan
• [asterisk-dev] [Code Review]: Add test execution modes to the Asterisk Testsuite   mjordan
• [asterisk-dev] [Code Review]: Add test execution modes to the Asterisk Testsuite   mjordan
• [asterisk-dev] [Code Review]: Incorrect Voicemail duration reported in messages containing silence (take 2)   mjordan
• [asterisk-dev] [Code Review]: Incorrect Voicemail duration reported in messages containing silence (take 2)   mjordan
• [asterisk-dev] [Code Review] Incorrect Voicemail duration reported in messages containing silence (take 2)   mjordan
• [asterisk-dev] [Code Review] Incorrect Voicemail duration reported in messages containing silence (take 2)   mjordan
• [asterisk-dev] [Code Review] Tests for SIP handling of 484 Address Incomplete; Incomplete application   opticron
• [asterisk-dev] [Code Review] Prevent segfault when asterisk restarts. Happens if call arrives before fully booted.   rmudgett
• [asterisk-dev] [Code Review] Prevent segfault when asterisk restarts. Happens if call arrives before fully booted.   rmudgett
• [asterisk-dev] [Code Review] Prevent segfault when asterisk restarts. Happens if call arrives before fully booted.   rmudgett
• [asterisk-dev] [Code Review] SIP 484 handling fix and new Incomplete app control frame   rmudgett
• [asterisk-dev] [Code Review] SIP 484 handling fix and new Incomplete app control frame   rmudgett
• [asterisk-dev] [Code Review] SIP 484 handling fix and new Incomplete app control frame   rmudgett
• [asterisk-dev] [Code Review] SIP 484 handling fix and new Incomplete app control frame   rmudgett
• [asterisk-dev] [Code Review] Fix a crash in res_ais   rmudgett
• [asterisk-dev] [Code Review]: Adding the PICKUPSTATUS variable   rmudgett
• [asterisk-dev] [Code Review] Clean up dnsmgr.conf parsing   rmudgett
• [asterisk-dev] [Code Review]: Clean up dnsmgr.conf parsing   rmudgett
• [asterisk-dev] [Code Review]: Clean up cel.conf parsing   rmudgett
• [asterisk-dev] [Code Review] automagically set a pointer to null after it is freed with ast_free   rmudgett
• [asterisk-dev] [Code Review] automagically set a pointer to null after it is freed with ast_free   rmudgett
• [asterisk-dev] [Code Review]: Make pbx_start NOT default to s at default if extension not found   rmudgett
• [asterisk-dev] [Code Review]: Make pbx_start NOT default to s at default if extension not found   rmudgett
• [asterisk-dev] [Code Review] Incorrect Voicemail duration reported in messages containing silence (take 2)   rmudgett
• [asterisk-dev] [Code Review] Incorrect Voicemail duration reported in messages containing silence (take 2)   rmudgett
• [asterisk-dev] [Code Review] Modify merge message so commit log message comes first.   rmudgett
• [asterisk-dev] [Code Review] Modify merge message so commit log message comes first.   rmudgett
• [asterisk-dev] [Code Review] Modify merge message so commit log message comes first.   rmudgett
• [asterisk-dev] [Code Review]: Modify merge message so commit log message comes first.   rmudgett
• [asterisk-dev] [Code Review] Add the Cancel Reason Header if a pickup occurs   schmidts
• [asterisk-dev] [Code Review] Add the Cancel Reason Header if a pickup occurs   schmidts
• [asterisk-dev] [Code Review] Add the Cancel Reason Header if a pickup occurs   schmidts
• [asterisk-dev] [Code Review] build_peer doesnt unlink a peer object from peers_by_ip container which leads to a wrong refcounter value   schmidts
• [asterisk-dev] [Code Review] automagically set a pointer to null after it is freed with ast_free   schmidts
• [asterisk-dev] [Code Review]: automagically set a pointer to null after it is freed with ast_free   schmidts
• [asterisk-dev] [Code Review] automagically set a pointer to null after it is freed with ast_free   schmidts
• [asterisk-dev] keep the dahdi line of server 1 and servers between iax trunk(not hangup)   tsengeljab tsogoo
• [asterisk-dev] [Code Review] Allow codecs to be negated in allow and disallow lines.   wdoekes
• [asterisk-dev] [Code Review] Iterate though cdr.conf setting   wdoekes
• [asterisk-dev] [Code Review] Iterate over the [csv] section of cdr.conf   wdoekes
• [asterisk-dev] [Code Review] Clean up dnsmgr.conf parsing   wdoekes
• [asterisk-dev] [Code Review] Iterate over the [csv] section of cdr.conf   wdoekes
• [asterisk-dev] [Code Review] fix so func_curl hashcompat can handle comma's in keys/values   wdoekes
• [asterisk-dev] [Code Review] Clean up dsp.conf parsing   wdoekes
• [asterisk-dev] [Code Review]: automagically set a pointer to null after it is freed with ast_free   wdoekes
• [asterisk-dev] [Code Review] automagically set a pointer to null after it is freed with ast_free   wdoekes
• [asterisk-dev] [Code Review]: automagically set a pointer to null after it is freed with ast_free   wdoekes
• [asterisk-dev] [Code Review]: automagically set a pointer to null after it is freed with ast_free   wdoekes
• [asterisk-dev] [Code Review]: Clean up udptl.conf parsing   wdoekes
• [asterisk-dev] [Code Review] Clean up udptl.conf parsing   wdoekes
• [asterisk-dev] [Code Review] Clean up asterisk.conf parsing   wdoekes
• [asterisk-dev] [Code Review] Clean up logger.conf parsing   wdoekes
• [asterisk-dev] [Code Review] Fix possible memory leak with SIP realtime regs   wdoekes
• [asterisk-dev] [Code Review] Fix possible memory leak with SIP realtime regs   wdoekes
• [asterisk-dev] [Code Review]: Fix possible memory leak with SIP realtime regs   wdoekes

Last message date: Fri Sep 30 22:51:34 CDT 2011
Archived on: Wed Dec 19 10:10:19 CST 2012

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