November 2010 Archives by author

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Starting: Mon Nov 1 05:06:42 CDT 2010
Ending: Tue Nov 30 20:09:20 CST 2010
Messages: 409

• [asterisk-dev] AGI playback has bug   Ashik Ali
• [asterisk-dev] Trying to debug Fork failed: Cannot allocate memory   Benny Amorsen
• [asterisk-dev] AMI sippeers output in 1.8   Örn Arnarson
• [asterisk-dev] Asterisk 1.8 and character sets   Örn Arnarson
• [asterisk-dev] Asterisk 1.8 and character sets   Örn Arnarson
• [asterisk-dev] Asterisk 1.8 and character sets   Örn Arnarson
• [asterisk-dev] doc/ -> wiki.asterisk.org   Paul Belanger
• [asterisk-dev] AMI sippeers output in 1.8   Paul Belanger
• [asterisk-dev] doc/ -> wiki.asterisk.org   Paul Belanger
• [asterisk-dev] follow up to IPv6 services   Paul Belanger
• [asterisk-dev] follow up to IPv6 services   Paul Belanger
• [asterisk-dev] Is there some standard way modules indicate their dependencies in the source code or config files?   Paul Belanger
• [asterisk-dev] VoicemailMain changed recently   Paul Belanger
• [asterisk-dev] Asterisk 1.6.20 / CDR issue with app-dial / bug or feature?   Paul Belanger
• [asterisk-dev] [Code Review] when code coverage is enabled, do not set DONT_OPTIMIZE   Paul Belanger
• [asterisk-dev] [Code Review] when code coverage is enabled, do not set DONT_OPTIMIZE   Paul Belanger
• [asterisk-dev] [Code Review] Update tests to use install_configs() and create a default logger.conf   Paul Belanger
• [asterisk-dev] [Code Review] Update tests to use install_configs() and create a default logger.conf   Paul Belanger
• [asterisk-dev] [Code Review] Only install asterisk and samples once, not each test.   Paul Belanger
• [asterisk-dev] [Code Review] Only install asterisk and samples once, not each test.   Paul Belanger
• [asterisk-dev] [Code Review] Global debug and verbose settings   Paul Belanger
• [asterisk-dev] [Code Review] Global debug and verbose settings   Paul Belanger
• [asterisk-dev] [Code Review] Global debug and verbose settings   Paul Belanger
• [asterisk-dev] Asterisk build plans (bamboo)   Paul Belanger
• [asterisk-dev] Asterisk build plans (bamboo)   Paul Belanger
• [asterisk-dev] [Code Review] [regression] update testsuite to test issue 18185   Paul Belanger
• [asterisk-dev] [Code Review] [regression] update testsuite to test issue 18185   Paul Belanger
• [asterisk-dev] [Code Review] Create svn:externals for starpy and Makefile   Paul Belanger
• [asterisk-dev] [Code Review] Have fastagi pass DTMF before starting tests   Paul Belanger
• [asterisk-dev] [Code Review] Have fastagi pass DTMF before starting tests   Paul Belanger
• [asterisk-dev] [Code Review] recursive tests.yaml   Paul Belanger
• [asterisk-dev] Asterisk with 3G / 2G GSM Modem   Paul Belanger
• [asterisk-dev] Developer Required   Paul Belanger
• [asterisk-dev] [Code Review] [17949] Error that mutex 'dialog' has been free'd more than it's been locked in add_header_max_forwards   Brett Bryant
• [asterisk-dev] [Code Review] [17949] Error that mutex 'dialog' has been free'd more than it's been locked in add_header_max_forwards   Brett Bryant
• [asterisk-dev] [Code Review] [17949] Error that mutex 'dialog' has been free'd more than it's been locked in add_header_max_forwards   Brett Bryant
• [asterisk-dev] [Code Review] [17949] Error that mutex 'dialog' has been free'd more than it's been locked in add_header_max_forwards   Brett Bryant
• [asterisk-dev] [Code Review] [18031] Patch for deadlock locking order issue between channel/queue during set_queue_variables   Brett Bryant
• [asterisk-dev] reviewboard problems   Russell Bryant
• [asterisk-dev] doc/ -> wiki.asterisk.org   Russell Bryant
• [asterisk-dev] doxygen does not work   Russell Bryant
• [asterisk-dev] doc/ -> wiki.asterisk.org   Russell Bryant
• [asterisk-dev] [Code Review] Call parking and retrieval test   Russell Bryant
• [asterisk-dev] [Code Review] CLI command: gtalk show settings   Russell Bryant
• [asterisk-dev] AstriDevCon 2010 Recap   Russell Bryant
• [asterisk-dev] doc/ -> wiki.asterisk.org   Russell Bryant
• [asterisk-dev] doc/ -> wiki.asterisk.org   Russell Bryant
• [asterisk-dev] Help needed for main/pbx.c   Russell Bryant
• [asterisk-dev] doc/ -> wiki.asterisk.org   Russell Bryant
• [asterisk-dev] Help needed for main/pbx.c   Russell Bryant
• [asterisk-dev] Help needed for main/pbx.c   Russell Bryant
• [asterisk-dev] doc/ -> wiki.asterisk.org   Russell Bryant
• [asterisk-dev] [Code Review] [17949] Error that mutex 'dialog' has been free'd more than it's been locked in add_header_max_forwards   Russell Bryant
• [asterisk-dev] doc/ -> wiki.asterisk.org   Russell Bryant
• [asterisk-dev] doc/ -> wiki.asterisk.org   Russell Bryant
• [asterisk-dev] doc/ -> wiki.asterisk.org   Russell Bryant
• [asterisk-dev] doc/ -> wiki.asterisk.org   Russell Bryant
• [asterisk-dev] doc/ -> wiki.asterisk.org   Russell Bryant
• [asterisk-dev] [Code Review] [17949] Error that mutex 'dialog' has been free'd more than it's been locked in add_header_max_forwards   Russell Bryant
• [asterisk-dev] doc/ -> wiki.asterisk.org   Russell Bryant
• [asterisk-dev] doc/ -> wiki.asterisk.org   Russell Bryant
• [asterisk-dev] CFP - Open Source Telephony Devroom - FOSDEM 2011   Russell Bryant
• [asterisk-dev] [Code Review] [17949] Error that mutex 'dialog' has been free'd more than it's been locked in add_header_max_forwards   Russell Bryant
• [asterisk-dev] [Code Review] [17949] Error that mutex 'dialog' has been free'd more than it's been locked in add_header_max_forwards   Russell Bryant
• [asterisk-dev] device_state distribution issues   Russell Bryant
• [asterisk-dev] [Code Review] [17949] Error that mutex 'dialog' has been free'd more than it's been locked in add_header_max_forwards   Russell Bryant
• [asterisk-dev] device_state distribution issues   Russell Bryant
• [asterisk-dev] device_state distribution issues   Russell Bryant
• [asterisk-dev] reviewboard down for upgrade   Russell Bryant
• [asterisk-dev] reviewboard down for upgrade   Russell Bryant
• [asterisk-dev] device_state distribution issues   Russell Bryant
• [asterisk-dev] device_state distribution issues   Russell Bryant
• [asterisk-dev] device_state distribution issues   Russell Bryant
• [asterisk-dev] need help debugging a deadlock in 1.6.2.13   Russell Bryant
• [asterisk-dev] Is there some standard way modules indicate their dependencies in the source code or config files?   Russell Bryant
• [asterisk-dev] Asterisk 1.4.37 Released   Russell Bryant
• [asterisk-dev] Unknown block - not fact or data   Russell Bryant
• [asterisk-dev] device_state distribution issues   Russell Bryant
• [asterisk-dev] What happened to the documentation of queue_log entries?   Russell Bryant
• [asterisk-dev] What happened to the documentation of queue_log entries?   Russell Bryant
• [asterisk-dev] What happened to the documentation of queue_log entries?   Russell Bryant
• [asterisk-dev] What happened to the documentation of queue_log entries?   Russell Bryant
• [asterisk-dev] [Code Review] res_jabber warning when using openssl 1.0.0+   Russell Bryant
• [asterisk-dev] [Code Review] [18031] Patch for deadlock locking order issue between channel/queue during set_queue_variables   Russell Bryant
• [asterisk-dev] [Code Review] when code coverage is enabled, do not set DONT_OPTIMIZE   Russell Bryant
• [asterisk-dev] [Code Review] Update tests to use install_configs() and create a default logger.conf   Russell Bryant
• [asterisk-dev] [Code Review] Only install asterisk and samples once, not each test.   Russell Bryant
• [asterisk-dev] [Code Review] Scheduler API Cleanup and Improvements   Russell Bryant
• [asterisk-dev] [Code Review] Announce to user that they have been muted when muting is done via AMI   Russell Bryant
• [asterisk-dev] [Code Review] Global debug and verbose settings   Russell Bryant
• [asterisk-dev] [Code Review] app_meetme.c conf_run() leaks refs   Russell Bryant
• [asterisk-dev] [Code Review] app_meetme.c conf_run() leaks refs   Russell Bryant
• [asterisk-dev] [Code Review] MeetMe option for caching join/leave announce names   Russell Bryant
• [asterisk-dev] [Code Review] The channel redirect function (CLI or AMI) hangs up the call instead of redirecting the call.   Russell Bryant
• [asterisk-dev] [Code Review] Ensure that peers are not reloaded when looking for a user.   Russell Bryant
• [asterisk-dev] [Code Review] Fix issue that precludes distributed devicestate via XMPP from working   Russell Bryant
• [asterisk-dev] [Code Review] Global debug and verbose settings   Russell Bryant
• [asterisk-dev] [Code Review] Global debug and verbose settings   Russell Bryant
• [asterisk-dev] [Code Review] MeetMe option for caching join/leave announce names   Russell Bryant
• [asterisk-dev] [Code Review] saynumber(1, n) doesn't work with language = SE   Russell Bryant
• [asterisk-dev] Please remove me from this list   Russell Bryant
• [asterisk-dev] Please remove me from this list   Russell Bryant
• [asterisk-dev] [Code Review] Have fastagi pass DTMF before starting tests   Russell Bryant
• [asterisk-dev] [Code Review] recursive tests.yaml   Russell Bryant
• [asterisk-dev] [Code Review] Create svn:externals for starpy and Makefile   Russell Bryant
• [asterisk-dev] [Code Review] Hints and devices from hints moved to ao2_container   Russell Bryant
• [asterisk-dev] [Code Review] Adding CLI Function sip remove subscribes and sip remove subscribe <peer>   Russell Bryant
• [asterisk-dev] [Code Review] Adding CLI Function sip remove subscribes and sip remove subscribe <peer>   Russell Bryant
• [asterisk-dev] [Code Review] Adding CLI Function sip remove subscribes and sip remove subscribe <peer>   Russell Bryant
• [asterisk-dev] missing CDRs with 1.6.2.9   CDR
• [asterisk-dev] Supported: ms-early-media   CDR
• [asterisk-dev] Supported: ms-early-media   CDR
• [asterisk-dev] Is there some standard way modules indicate their dependencies in the source code or config files?   Frank Church
• [asterisk-dev] rhino dahdi drivers and DAHDI_VER   Tzafrir Cohen
• [asterisk-dev] rhino dahdi drivers and DAHDI_VER   Tzafrir Cohen
• [asterisk-dev] doc/ -> wiki.asterisk.org   Tzafrir Cohen
• [asterisk-dev] doc/ -> wiki.asterisk.org   Tzafrir Cohen
• [asterisk-dev] E1 CAS dchan=16 [was: Re: [svn-commits] kmoore: tools/trunk r9485 - /tools/trunk/xpp/perl_modules/Dahdi/Config/Gen/]   Tzafrir Cohen
• [asterisk-dev] Multiple codecs on SDP answer if SIP_CODEC variable is used   Saúl Ibarra Corretgé
• [asterisk-dev] Multiple codecs on SDP answer if SIP_CODEC variable is used   Saúl Ibarra Corretgé
• [asterisk-dev] reviewboard problems   Klaus Darilion
• [asterisk-dev] Switching unrestricted digital calls   Klaus Darilion
• [asterisk-dev] missing CDRs with 1.6.2.9   Klaus Darilion
• [asterisk-dev] doc/ -> wiki.asterisk.org   Klaus Darilion
• [asterisk-dev] doc/ -> wiki.asterisk.org   Klaus Darilion
• [asterisk-dev] [Code Review] Add Path header support to chan_sip   Klaus Darilion
• [asterisk-dev] [Code Review] Add Path header support to chan_sip   Klaus Darilion
• [asterisk-dev] Switching unrestricted digital calls   Klaus Darilion
• [asterisk-dev] doc/ -> wiki.asterisk.org   Klaus Darilion
• [asterisk-dev] doc/ -> wiki.asterisk.org   Klaus Darilion
• [asterisk-dev] device_state distribution issues   Klaus Darilion
• [asterisk-dev] device_state distribution issues   Klaus Darilion
• [asterisk-dev] device_state distribution issues   Klaus Darilion
• [asterisk-dev] device_state distribution issues   Klaus Darilion
• [asterisk-dev] device_state distribution issues   Klaus Darilion
• [asterisk-dev] device_state distribution issues   Klaus Darilion
• [asterisk-dev] device_state distribution issues   Klaus Darilion
• [asterisk-dev] device_state distribution issues   Klaus Darilion
• [asterisk-dev] CDR issue with dial-app / generating extra cdr   Klaus Darilion
• [asterisk-dev] Asterisk 1.8 and character sets   Klaus Darilion
• [asterisk-dev] Asterisk 1.8 and character sets   Klaus Darilion
• [asterisk-dev] Supported: ms-early-media   Klaus Darilion
• [asterisk-dev] Supported: ms-early-media   Klaus Darilion
• [asterisk-dev] [Code Review] Adding CLI Function sip remove subscribes and sip remove subscribe <peer>   Klaus Darilion
• [asterisk-dev] [Code Review] Adding CLI Function sip remove subscribes and sip remove subscribe <peer>   Klaus Darilion
• [asterisk-dev] resending cause codes   Klaus Darilion
• [asterisk-dev] need help debugging a deadlock in 1.6.2.13   Steve Davies
• [asterisk-dev] need help debugging a deadlock in 1.6.2.13   Steve Davies
• [asterisk-dev] need help debugging a deadlock in 1.6.2.13   Steve Davies
• [asterisk-dev] [Code Review] Prevent DTMF incorrectly triggering during SAS+CAS (CallWaiting) signalling to an FXS port   Alec Davis
• [asterisk-dev] [Code Review] Prevent DTMF incorrectly triggering during SAS+CAS (CallWaiting) signalling to an FXS port   Alec Davis
• [asterisk-dev] Double tones sometimes detected with RFC2833   Bryan Field-Elliot
• [asterisk-dev] Double tones sometimes detected with RFC2833   Bryan Field-Elliot
• [asterisk-dev] rhino dahdi drivers and DAHDI_VER   James Finstrom
• [asterisk-dev] doc/ -> wiki.asterisk.org   Kevin P. Fleming
• [asterisk-dev] Asterisk community services powered by Atlassian tools   Kevin P. Fleming
• [asterisk-dev] doc/ -> wiki.asterisk.org   Kevin P. Fleming
• [asterisk-dev] follow up to IPv6 services   Kevin P. Fleming
• [asterisk-dev] SCF Licencing   Kevin P. Fleming
• [asterisk-dev] SCF Licencing   Kevin P. Fleming
• [asterisk-dev] device_state distribution issues   Kevin P. Fleming
• [asterisk-dev] Double tones sometimes detected with RFC2833   Kevin P. Fleming
• [asterisk-dev] SIT detection in pre-answer audio - guidance for rewriting for trunk   Kevin P. Fleming
• [asterisk-dev] Need testing: chan_unistim improvements   Igor Goncharovsky
• [asterisk-dev] Need testing: chan_unistim improvements   Igor Goncharovsky
• [asterisk-dev] CDR issue with dial-app / generating extra cdr   Thorsten Göllner
• [asterisk-dev] Asterisk 1.6.20 / CDR issue with app-dial / bug or feature?   Thorsten Göllner
• [asterisk-dev] VoicemailMain changed recently   Freddi Hansen
• [asterisk-dev] VoicemailMain changed recently   Freddi Hansen
• [asterisk-dev] Unknown block - not fact or data   Jakob Hirsch
• [asterisk-dev] Unknown block - not fact or data   Jakob Hirsch
• [asterisk-dev] Please remove me from this list   Steve Howes
• [asterisk-dev] Please remove me from this list   Steve Howes
• [asterisk-dev] Developer Required   Kate Jeans
• [asterisk-dev] libpri - q931.c assumes only T1 uses slot maps   Abhinav Jha
• [asterisk-dev] libpri - q931.c assumes only T1 uses slot maps   Abhinav Jha
• [asterisk-dev] libpri - q931.c assumes only T1 uses slot maps   Abhinav Jha
• [asterisk-dev] Libpri - Wrong hangup cause ( should read from B-channel but reads from D-channel ) ?   Abhinav Jha
• [asterisk-dev] Libpri - Wrong hangup cause ( should read from B-channel but reads from D-channel ) ?   Abhinav Jha
• [asterisk-dev] [Code Review] [17949] Error that mutex 'dialog' has been free'd more than it's been locked in add_header_max_forwards   Olle E Johansson
• [asterisk-dev] [Code Review] Ensure SIP responses only have one Via header   Olle E Johansson
• [asterisk-dev] [Code Review] Add Path header support to chan_sip   Olle E Johansson
• [asterisk-dev] [Code Review] Add Path header support to chan_sip   Olle E Johansson
• [asterisk-dev] [Code Review] saynumber(1, n) doesn't work with language = SE   Olle E Johansson
• [asterisk-dev] [Code Review] saynumber(1, n) doesn't work with language = SE   Olle E Johansson
• [asterisk-dev] [Code Review] saynumber(1, n) doesn't work with language = SE   Olle E Johansson
• [asterisk-dev] [Code Review] Swedish saynumber (again :-) ) - correctly say thousands and millions   Olle E Johansson
• [asterisk-dev] [Code Review] res_config_pgsql needs db reconnection support   Olle E Johansson
• [asterisk-dev] [Code Review] Add Path header support to chan_sip   Olle E. Johansson
• [asterisk-dev] AMI sippeers output in 1.8   Olle E. Johansson
• [asterisk-dev] doc/ -> wiki.asterisk.org   Olle E. Johansson
• [asterisk-dev] doc/ -> wiki.asterisk.org   Olle E. Johansson
• [asterisk-dev] doc/ -> wiki.asterisk.org   Olle E. Johansson
• [asterisk-dev] doc/ -> wiki.asterisk.org   Olle E. Johansson
• [asterisk-dev] [Code Review] [17949] Error that mutex 'dialog' has been free'd more than it's been locked in add_header_max_forwards   Olle E. Johansson
• [asterisk-dev] Is there some standard way modules indicate their dependencies in the source code or config files?   Olle E. Johansson
• [asterisk-dev] device_state distribution issues   Olle E. Johansson
• [asterisk-dev] device_state distribution issues   Olle E. Johansson
• [asterisk-dev] What happened to the documentation of queue_log entries?   Olle E. Johansson
• [asterisk-dev] Supported: ms-early-media   Olle E. Johansson
• [asterisk-dev] Supported: ms-early-media   Olle E. Johansson
• [asterisk-dev] resending cause codes   Olle E. Johansson
• [asterisk-dev] resending cause codes   Olle E. Johansson
• [asterisk-dev] asterisk 1.8 fax woes   Jeremy Kister
• [asterisk-dev] doc/ -> wiki.asterisk.org   Andrew Latham
• [asterisk-dev] doc/ -> wiki.asterisk.org   Andrew Latham
• [asterisk-dev] doc/ -> wiki.asterisk.org   Andrew Latham
• [asterisk-dev] doc/ -> wiki.asterisk.org   Andrew Latham
• [asterisk-dev] doc/ -> wiki.asterisk.org   Andrew Latham
• [asterisk-dev] Is there some standard way modules indicate their dependencies in the source code or config files?   Andrew Latham
• [asterisk-dev] What happened to the documentation of queue_log entries?   Andrew Latham
• [asterisk-dev] What happened to the documentation of queue_log entries?   Andrew Latham
• [asterisk-dev] What happened to the documentation of queue_log entries?   Andrew Latham
• [asterisk-dev] Trying to debug Fork failed: Cannot allocate memory   Tilghman Lesher
• [asterisk-dev] [Code Review] Call parking and retrieval test   Tilghman Lesher
• [asterisk-dev] [Code Review] Call parking and retrieval test   Tilghman Lesher
• [asterisk-dev] [Code Review] Call parking and retrieval test   Tilghman Lesher
• [asterisk-dev] Channel locking mechanism in Asterisk   Tilghman Lesher
• [asterisk-dev] doc/ -> wiki.asterisk.org   Tilghman Lesher
• [asterisk-dev] Help needed for main/pbx.c   Tilghman Lesher
• [asterisk-dev] [Code Review] Hints and devices from hints moved to ao2_container   Tilghman Lesher
• [asterisk-dev] [Code Review] Hints and devices from hints moved to ao2_container   Tilghman Lesher
• [asterisk-dev] [Code Review] Fix how mutexes are created/destroyed on platforms which need constructors   Tilghman Lesher
• [asterisk-dev] [Code Review] Fix how mutexes are created/destroyed on platforms which need constructors   Tilghman Lesher
• [asterisk-dev] Is there some standard way modules indicate their dependencies in the source code or config files?   Tilghman Lesher
• [asterisk-dev] Asterisk v1.8.0-RC5 MOH and generic timing source   Tilghman Lesher
• [asterisk-dev] Asterisk build plans (bamboo)   Tilghman Lesher
• [asterisk-dev] [Code Review] [regression] update testsuite to test issue 18185   Tilghman Lesher
• [asterisk-dev] [Code Review] [regression] update testsuite to test issue 18185   Tilghman Lesher
• [asterisk-dev] [Code Review] Swedish saynumber (again :-) ) - correctly say thousands and millions   Tilghman Lesher
• [asterisk-dev] [Code Review] res_config_pgsql needs db reconnection support   Tilghman Lesher
• [asterisk-dev] App_queue doesn't sends correct AMI events to EXITWITHTIMEOUT, EXITWITHKEY, SYSCOMPAT, EXITEMPTY   Tilghman Lesher
• [asterisk-dev] resending cause codes   Tilghman Lesher
• [asterisk-dev] [Code Review] Use non-deprecated CoreAudio apis on Darwin   Tilghman Lesher
• [asterisk-dev] [Code Review] Use non-deprecated CoreAudio apis on Darwin   Tilghman Lesher
• [asterisk-dev] [Code Review] Use non-deprecated CoreAudio apis on Darwin   Tilghman Lesher
• [asterisk-dev] AGI playback has bug   Leif Madsen
• [asterisk-dev] doc/ -> wiki.asterisk.org   Leif Madsen
• [asterisk-dev] What happened to the documentation of queue_log entries?   Leif Madsen
• [asterisk-dev] [Code Review] Fix caching of device state changes for multiple servers   Marquis
• [asterisk-dev] [Code Review] Ensure that peers are not reloaded when looking for a user.   Marquis
• [asterisk-dev] [Code Review] Fix issue that precludes distributed devicestate via XMPP from working   Marquis
• [asterisk-dev] Asterisk v1.8.0-RC5 MOH and generic timing source   Bruce McAlister
• [asterisk-dev] resending cause codes   Bruce McAlister
• [asterisk-dev] Please remove me from this list   Scott McCrea
• [asterisk-dev] asterisk-dev Digest, Vol 76, Issue 77   Scott McCrea
• [asterisk-dev] Channel locking mechanism in Asterisk   Mark Michelson
• [asterisk-dev] Double tones sometimes detected with RFC2833   Mark Michelson
• [asterisk-dev] [Code Review] Ensure SIP responses only have one Via header   Mark Michelson
• [asterisk-dev] libpri - q931.c assumes only T1 uses slot maps   Richard Mudgett
• [asterisk-dev] libpri - q931.c assumes only T1 uses slot maps   Richard Mudgett
• [asterisk-dev] phone_read and phone_write not working on subchannels   Richard Mudgett
• [asterisk-dev] Libpri - Wrong hangup cause ( should read from B-channel but reads from D-channel ) ?   Richard Mudgett
• [asterisk-dev] Dear Friends: nmnn m.D   Koch Máté
• [asterisk-dev] What happened to the documentation of queue_log entries?   Håkon Nessjøen
• [asterisk-dev] What happened to the documentation of queue_log entries?   Håkon Nessjøen
• [asterisk-dev] What happened to the documentation of queue_log entries?   Håkon Nessjøen
• [asterisk-dev] What happened to the documentation of queue_log entries?   Håkon Nessjøen
• [asterisk-dev] AGI playback has bug   Håkon Nessjøen
• [asterisk-dev] [Code Review] Call parking and retrieval test   Matthew Nicholson
• [asterisk-dev] [Code Review] Call parking and retrieval test   Matthew Nicholson
• [asterisk-dev] [Code Review] SayUnixTime always tryes to reach the extension as digit pressed.   Matthew Nicholson
• [asterisk-dev] Proposal: "Diving into Asterisk" for FOSDEM   Steve Oualline
• [asterisk-dev] [Code Review] MeetMe option for caching join/leave announce names   Andrew Parisio
• [asterisk-dev] [Code Review] MeetMe option for caching join/leave announce names   Andrew Parisio
• [asterisk-dev] [Code Review] ConfBridge User Announce Feature Set   Andrew Parisio
• [asterisk-dev] [Code Review] queue reload members does not work   Andrew Parisio
• [asterisk-dev] Help needed for main/pbx.c   Jeff Peeler
• [asterisk-dev] need help debugging a deadlock in 1.6.2.13   Jeff Peeler
• [asterisk-dev] Need testing: chan_unistim improvements   Ian Penney
• [asterisk-dev] [Code Review] Regression: chan_iax2 registration issues.   Simon Perreault
• [asterisk-dev] [Code Review] Regression: chan_iax2 registration issues.   Simon Perreault
• [asterisk-dev] [Code Review] Regression: chan_iax2 registration issues.   Simon Perreault
• [asterisk-dev] follow up to IPv6 services   Simon Perreault
• [asterisk-dev] [Code Review] Ensure SIP responses only have one Via header   Simon Perreault
• [asterisk-dev] [Code Review] Ensure SIP responses only have one Via header   Simon Perreault
• [asterisk-dev] AST_DATA_IPADDR is broken   Simon Perreault
• [asterisk-dev] AST_DATA_IPADDR is broken   Simon Perreault
• [asterisk-dev] phone_read and phone_write not working on subchannels   Alok Prasad
• [asterisk-dev] phone_read and phone_write not working on subchannels   Alok Prasad
• [asterisk-dev] Generating Re-invite from asterisk   Alok Prasad
• [asterisk-dev] ReviewBoard Certificate error   Matt Riddell
• [asterisk-dev] reviewboard problems   Shaun Ruffell
• [asterisk-dev] rhino dahdi drivers and DAHDI_VER   Shaun Ruffell
• [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.)   Shaun Ruffell
• [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.)   Shaun Ruffell
• [asterisk-dev] Asterisk with 3G / 2G GSM Modem   Anshu Sah
• [asterisk-dev] AST_DATA_IPADDR is broken   Eliel Sardañons
• [asterisk-dev] AST_DATA_IPADDR is broken   Eliel Sardañons
• [asterisk-dev] doxygen does not work   Stefan Schmidt
• [asterisk-dev] doxygen does not work   Stefan Schmidt
• [asterisk-dev] Help needed for main/pbx.c   Stefan Schmidt
• [asterisk-dev] Help needed for main/pbx.c   Stefan Schmidt
• [asterisk-dev] Help needed for main/pbx.c   Stefan Schmidt
• [asterisk-dev] using apps in other apps   Stefan Schmidt
• [asterisk-dev] need help debugging a deadlock in 1.6.2.13   Stefan Schmidt
• [asterisk-dev] [Asterisk-Dev] Asterisk-1.8.0 compilation error   Stefan Schmidt
• [asterisk-dev] App_queue doesn't sends correct AMI events to EXITWITHTIMEOUT, EXITWITHKEY, SYSCOMPAT, EXITEMPTY   Marcos Setim
• [asterisk-dev] App_queue doesn't sends correct AMI events to EXITWITHTIMEOUT, EXITWITHKEY, SYSCOMPAT, EXITEMPTY   Marcos Setim
• [asterisk-dev] using apps in other apps   M Shokuie
• [asterisk-dev] using apps in other apps   M Shokuie
• [asterisk-dev] E1 CAS dchan=16 [was: Re: [svn-commits] kmoore: tools/trunk r9485 - /tools/trunk/xpp/perl_modules/Dahdi/Config/Gen/]   Moises Silva
• [asterisk-dev] Trying to debug Fork failed: Cannot allocate memory   Antonio Goméz Soto
• [asterisk-dev] Trying to debug Fork failed: Cannot allocate memory   Antonio Goméz Soto
• [asterisk-dev] [Code Review] Have fastagi pass DTMF before starting tests   Erin Spiceland
• [asterisk-dev] AJAM XML inconsistency ?   Emil Stoyanov
• [asterisk-dev] Regarding Text based Chat Implementation in Iax2 ! Suggestions ?   Kumar Subramanian
• [asterisk-dev] Asterisk/Asterisk SCF Project Wiki   Asterisk Development Team
• [asterisk-dev] Asterisk 1.6.2.14 Released   Asterisk Development Team
• [asterisk-dev] Scheduled maintenance for various Asterisk community services   Asterisk Development Team
• [asterisk-dev] EXTENDED: Scheduled maintenance for various Asterisk community services   Asterisk Development Team
• [asterisk-dev] libpri 1.4.11.5 Now Available   Asterisk Development Team
• [asterisk-dev] libpri 1.4.12-beta3 Now Available   Asterisk Development Team
• [asterisk-dev] Asterisk 1.4.38-rc1 Now Available   Asterisk Development Team
• [asterisk-dev] Asterisk 1.6.1.15-rc1 Now Available   Asterisk Development Team
• [asterisk-dev] Asterisk 1.8.1-rc1 Now Available   Asterisk Development Team
• [asterisk-dev] doc/ -> wiki.asterisk.org   Chris Tooley
• [asterisk-dev] Switching unrestricted digital calls   Pavel Troller
• [asterisk-dev] Switching unrestricted digital calls   Pavel Troller
• [asterisk-dev] SIT detection in pre-answer audio - guidance for rewriting for trunk   D Tucny
• [asterisk-dev] [Asterisk-Dev] Asterisk-1.8.0 compilation error   RAJNIKANT VANZA
• [asterisk-dev] [Code Review] [17949] Error that mutex 'dialog' has been free'd more than it's been locked in add_header_max_forwards   David Vossel
• [asterisk-dev] [Code Review] Ensure SIP responses only have one Via header   David Vossel
• [asterisk-dev] device_state distribution issues   Brad Watkins
• [asterisk-dev] device_state distribution issues   Brad Watkins
• [asterisk-dev] E1 CAS dchan=16 [was: Re: [svn-commits] kmoore: tools/trunk r9485 - /tools/trunk/xpp/perl_modules/Dahdi/Config/Gen/]   Will
• [asterisk-dev] [Code Review] Avoid valgrind warnings for ast_rtp_instance_get_xxx_address   Terry Wilson
• [asterisk-dev] [Code Review] SIP: proper handling of forked outbound INVITE requests.   Terry Wilson
• [asterisk-dev] [Code Review] Ensure SIP responses only have one Via header   Terry Wilson
• [asterisk-dev] [Code Review] Ensure SIP responses only have one Via header   Terry Wilson
• [asterisk-dev] [Code Review] Ensure SIP responses only have one Via header   Terry Wilson
• [asterisk-dev] [Code Review] Ensure SIP responses only have one Via header   Terry Wilson
• [asterisk-dev] doc/ -> wiki.asterisk.org   Hans Witvliet
• [asterisk-dev] [Code Review] chan_ooh323 Ipv6 support   reviewboard at asterisk.org
• [asterisk-dev] [Code Review] Hints and devices from hints moved to ao2_container   reviewboard at asterisk.org
• [asterisk-dev] [Code Review] Hints and devices from hints moved to ao2_container   reviewboard at asterisk.org
• [asterisk-dev] [Code Review] Don't re-auth subscriptions that are just re-transmissions (not re-subscriptions)   reviewboard at asterisk.org
• [asterisk-dev] [Code Review] Don't re-auth subscriptions that are just re-transmissions (not re-subscriptions)   reviewboard at asterisk.org
• [asterisk-dev] [Code Review] Don't re-auth subscriptions that are just re-transmissions (not re-subscriptions)   reviewboard at asterisk.org
• [asterisk-dev] [Code Review] Don't re-auth subscriptions that are just re-transmissions (not re-subscriptions)   reviewboard at asterisk.org
• [asterisk-dev] [Code Review] Don't re-auth subscriptions that are just re-transmissions (not re-subscriptions)   reviewboard at asterisk.org
• [asterisk-dev] [Code Review] Don't re-auth subscriptions that are just re-transmissions (not re-subscriptions)   reviewboard at asterisk.org
• [asterisk-dev] [Code Review] Don't re-auth subscriptions that are just re-transmissions (not re-subscriptions)   reviewboard at asterisk.org
• [asterisk-dev] [Code Review] Don't re-auth subscriptions that are just re-transmissions (not re-subscriptions)   reviewboard at asterisk.org
• [asterisk-dev] [Code Review] Don't re-auth subscriptions that are just re-transmissions (not re-subscriptions)   reviewboard at asterisk.org
• [asterisk-dev] [Code Review] Don't miss control frames if a call is answered and hung up very quickly   reviewboard at asterisk.org
• [asterisk-dev] [Code Review] Don't miss control frames if a call is answered and hung up very quickly   reviewboard at asterisk.org
• [asterisk-dev] [Code Review] Don't re-auth subscriptions that are just re-transmissions (not re-subscriptions)   reviewboard at asterisk.org
• [asterisk-dev] [Code Review] Hints and devices from hints moved to ao2_container   reviewboard at asterisk.org
• [asterisk-dev] [Code Review] Hints and devices from hints moved to ao2_container   reviewboard at asterisk.org
• [asterisk-dev] [Code Review] Wrap OpenSSL library initialization to make it safe for loaded modules to also use OpenSSL.   reviewboard at asterisk.org
• [asterisk-dev] [Code Review] Scheduler API Cleanup and Improvements   reviewboard at asterisk.org
• [asterisk-dev] [Code Review] MeetMe option for caching join/leave announce names   reviewboard at asterisk.org
• [asterisk-dev] [Code Review] MeetMe option for caching join/leave announce names   reviewboard at asterisk.org
• [asterisk-dev] [Code Review] Announce to user that they have been muted when muting is done via AMI   reviewboard at asterisk.org
• [asterisk-dev] [Code Review] ConfBridge User Announce Feature Set   reviewboard at asterisk.org
• [asterisk-dev] [Code Review] Announce to user that they have been muted when muting is done via AMI   reviewboard at asterisk.org
• [asterisk-dev] [Code Review] Announce to user that they have been muted when muting is done via AMI   reviewboard at asterisk.org
• [asterisk-dev] [Code Review] Announce to user that they have been muted when muting is done via AMI   reviewboard at asterisk.org
• [asterisk-dev] [Code Review] Announce to user that they have been muted when muting is done via AMI   reviewboard at asterisk.org
• [asterisk-dev] [Code Review] Hints and devices from hints moved to ao2_container   reviewboard at asterisk.org
• [asterisk-dev] [Code Review] Fix caching of device state changes for multiple servers   reviewboard at asterisk.org
• [asterisk-dev] [Code Review] Announce to user that they have been muted when muting is done via AMI   reviewboard at asterisk.org
• [asterisk-dev] [Code Review] Announce to user that they have been muted when muting is done via AMI   reviewboard at asterisk.org
• [asterisk-dev] [Code Review] MeetMe option for caching join/leave announce names   reviewboard at asterisk.org
• [asterisk-dev] [Code Review] Hints and devices from hints moved to ao2_container   reviewboard at asterisk.org
• [asterisk-dev] [Code Review] ConfBridge User Announce Feature Set   reviewboard at asterisk.org
• [asterisk-dev] [Code Review] Wrap OpenSSL library initialization to make it safe for loaded modules to also use OpenSSL.   reviewboard at asterisk.org
• [asterisk-dev] [Code Review] res_config_pgsql needs db reconnection support   reviewboard at asterisk.org
• [asterisk-dev] [Code Review] The channel redirect function (CLI or AMI) hangs up the call instead of redirecting the call.   reviewboard at asterisk.org
• [asterisk-dev] [Code Review] The channel redirect function (CLI or AMI) hangs up the call instead of redirecting the call.   reviewboard at asterisk.org
• [asterisk-dev] [Code Review] app_queue: Log failed attempts to call members   reviewboard at asterisk.org
• [asterisk-dev] [Code Review] Hints and devices from hints moved to ao2_container   reviewboard at asterisk.org
• [asterisk-dev] [Code Review] SayUnixTime always tryes to reach the extension as digit pressed.   reviewboard at asterisk.org
• [asterisk-dev] [Code Review] app_queue: Log failed attempts to call members   reviewboard at asterisk.org
• [asterisk-dev] [Code Review] Use full path for Asterisk binary   reviewboard at asterisk.org
• [asterisk-dev] [Code Review] Use full path for Asterisk binary   reviewboard at asterisk.org
• [asterisk-dev] [Code Review] Use full path for Asterisk binary   reviewboard at asterisk.org
• [asterisk-dev] [Code Review] Use full path for Asterisk binary   reviewboard at asterisk.org
• [asterisk-dev] [Code Review] MeetMe option for caching join/leave announce names   reviewboard at asterisk.org
• [asterisk-dev] [Code Review] The channel redirect function (CLI or AMI) hangs up the call instead of redirecting the call.   reviewboard at asterisk.org
• [asterisk-dev] [Code Review] The channel redirect function (CLI or AMI) hangs up the call instead of redirecting the call.   reviewboard at asterisk.org
• [asterisk-dev] [Code Review] res_jabber warning when using openssl 1.0.0+   reviewboard at asterisk.org
• [asterisk-dev] [Code Review] res_jabber warning when using openssl 1.0.0+   reviewboard at asterisk.org
• [asterisk-dev] [Code Review] SayUnixTime always tryes to reach the extension as digit pressed.   reviewboard at asterisk.org
• [asterisk-dev] res_config_pgsql.c can't reconnect to database, but if you ..   german aracil boned
• [asterisk-dev] resending cause codes   marek cervenka
• [asterisk-dev] resending cause codes   marek cervenka
• [asterisk-dev] [Code Review] Avoid valgrind warnings for ast_rtp_instance_get_xxx_address   rmudgett at digium.com
• [asterisk-dev] [Code Review] Connected line update on sig_analog, sig_pri, and chan_misdn call transfers   rmudgett at digium.com
• [asterisk-dev] [Code Review] Prevent DTMF incorrectly triggering during SAS+CAS (CallWaiting) signalling to an FXS port   rmudgett at digium.com
• [asterisk-dev] [Code Review] app_queue: Log failed attempts to call members   haakon
• [asterisk-dev] [Code Review] Announce to user that they have been muted when muting is done via AMI   kobaz
• [asterisk-dev] [Code Review] app_meetme.c conf_run() leaks refs   kobaz
• [asterisk-dev] [Code Review] app_meetme.c conf_run() leaks refs   kobaz
• [asterisk-dev] [Code Review] app_meetme.c conf_run() leaks refs   kobaz
• [asterisk-dev] Channel locking mechanism in Asterisk   chaitra.bhat at lakecommunications.com
• [asterisk-dev] Channel locking mechanism in Asterisk   chaitra.bhat at lakecommunications.com
• [asterisk-dev] [Code Review] Regression: chan_iax2 registration issues.   paul.belanger at polybeacon.com
• [asterisk-dev] [Code Review] Regression: chan_iax2 registration issues.   paul.belanger at polybeacon.com
• [asterisk-dev] [Code Review] CLI command: gtalk show settings   paul.belanger at polybeacon.com
• [asterisk-dev] [Code Review] CLI command: gtalk show settings   paul.belanger at polybeacon.com
• [asterisk-dev] Quick Guide for Asterisk development   shiva prasad
• [asterisk-dev] [Code Review] [regression] update testsuite to test issue 18185   rmudgett
• [asterisk-dev] [Code Review] Adding CLI Function sip remove subscribes and sip remove subscribe <peer>   schmidts
• [asterisk-dev] [Code Review] Adding CLI Function sip remove subscribes and sip remove subscribe <peer>   schmidts
• [asterisk-dev] [Code Review] Adding CLI Function sip remove subscribes and sip remove subscribe <peer>   schmidts
• [asterisk-dev] [Code Review] Hints and devices from hints moved to ao2_container   sst at sil.at
• [asterisk-dev] [Code Review] Hints and devices from hints moved to ao2_container   sst at sil.at

Last message date: Tue Nov 30 20:09:20 CST 2010
Archived on: Tue Nov 30 20:12:00 CST 2010

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