February 2011 Archives by author

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Starting: Tue Feb 1 05:57:59 CST 2011
Ending: Mon Feb 28 15:06:29 CST 2011
Messages: 446

• [asterisk-dev] Asterisk crashes when receiving SIP answer (200OK)   ISABEL ORDAS ARNAL
• [asterisk-dev] return control after ast_bridge_call   ISABEL ORDAS ARNAL
• [asterisk-dev] Asterisk crashes when receiving SIP answer (200OK)   ISABEL ORDAS ARNAL
• [asterisk-dev] Subject: Re: return control after ast_bridge_call   ISABEL ORDAS ARNAL
• [asterisk-dev] Debugging modules for Asterisk   ISABEL ORDAS ARNAL
• [asterisk-dev] question about asterisk 1.4, func_curl.c, and close_wait state   Paul Albrecht
• [asterisk-dev] question about asterisk 1.4, func_curl.c, and close_wait state   Paul Albrecht
• [asterisk-dev] question about asterisk 1.4, func_curl.c, and close_wait state   Paul Albrecht
• [asterisk-dev] question about asterisk 1.4, func_curl.c, and close_wait state   Paul Albrecht
• [asterisk-dev] IAX hardphone   Jay Ashworth
• [asterisk-dev] IAX hardphone   Jay Ashworth
• [asterisk-dev] IAX hardphone   Jay Ashworth
• [asterisk-dev] Asterisk 1.10 Update   Dan Austin
• [asterisk-dev] IAX hardphone   Mihai Balea
• [asterisk-dev] IAX hardphone   Mihai Balea
• [asterisk-dev] IAX hardphone   Mihai Balea
• [asterisk-dev] IAX hardphone   Mihai Balea
• [asterisk-dev] IAX hardphone   Mihai Balea
• [asterisk-dev] IAX hardphone   Mihai Balea
• [asterisk-dev] SRTP configuration and behavior in Asterisk18   Bob Beers
• [asterisk-dev] SRTP configuration and behavior in Asterisk18   Bob Beers
• [asterisk-dev] [asterisk-users] Calling Directory app from AGI   Bob Beers
• [asterisk-dev] [Code Review] Add a SIPpeerstatus command to chan_sip's manager interface.   Bob Beers
• [asterisk-dev] Asterisk software development needed !   Paul Belanger
• [asterisk-dev] [svn-commits] lathama: trunk r305650 - /trunk/configs/sip.conf.sample   Paul Belanger
• [asterisk-dev] [svn-commits] lathama: trunk r305650 - /trunk/configs/sip.conf.sample   Paul Belanger
• [asterisk-dev] [svn-commits] lathama: branch 1.6.2 r305752 - /branches/1.6.2/channels/chan_sip.c   Paul Belanger
• [asterisk-dev] [Bamboo] Paul Belanger commented on Asterisk - Trunk - Ubuntu Lucid (10.04) - i386 309   Paul Belanger
• [asterisk-dev] BOUNTY: Implement application MWIMessageSend to call libpri routine   Paul Belanger
• [asterisk-dev] [Code Review] Announce to user that they have been muted when muting is done via AMI   Paul Belanger
• [asterisk-dev] [Bamboo] Paul Belanger commented on Asterisk - Trunk - Ubuntu Lucid (10.04) - amd64 339   Paul Belanger
• [asterisk-dev] [Bamboo] Paul Belanger commented on Asterisk - 1.8 - FreeBSD 8.1 - i386 122   Paul Belanger
• [asterisk-dev] [asterisk-users] Calling Directory app from AGI   Paul Belanger
• [asterisk-dev] [Code Review] new testcase to test loading / unloading for modules.   Paul Belanger
• [asterisk-dev] Asterisk 1.10 Update   Paul Belanger
• [asterisk-dev] Asterisk 1.10 Update   Paul Belanger
• [asterisk-dev] [Bamboo] Paul Belanger commented on Asterisk - 1.4 - Mac OS X Snow Leopard (10.6) 113   Paul Belanger
• [asterisk-dev] [Bamboo] Paul Belanger commented on Asterisk - 1.4 - Mac OS X Snow Leopard (10.6) 113   Paul Belanger
• [asterisk-dev] [Code Review] CCSS Proposal to add Asterisk DEVSTATEs to generic agents to use with hints   Paul Belanger
• [asterisk-dev] [Code Review] Media Project Phase 2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, Custom format creation in codecs.conf   Paul Belanger
• [asterisk-dev] [Code Review] CCSS Proposal to add Asterisk DEVSTATEs to generic agents to use with hints   Paul Belanger
• [asterisk-dev] [Code Review] decrease verbose messages to debug, and code cleanup for enum.c   Paul Belanger
• [asterisk-dev] [Code Review] Repair UDP port leaks, Memory Leaks, Denial of Service in chan_sip   Paul Belanger
• [asterisk-dev] automatic dialer   Paul Belanger
• [asterisk-dev] [svn-commits] lathama: branch 1.8 r308393 - /branches/1.8/main/http.c   Paul Belanger
• [asterisk-dev] [svn-commits] lathama: branch 1.8 r308393 - /branches/1.8/main/http.c   Paul Belanger
• [asterisk-dev] Introducing the new ConfBridge   Paul Belanger
• [asterisk-dev] Patch needed: https://issues.asterisk.org/view.php?id=18722   Dovid Bender
• [asterisk-dev] Bounty: https://issues.asterisk.org/view.php?id=18722   Dovid Bender
• [asterisk-dev] [asterisk-commits] lathama: trunk r308527 - /trunk/main/http.c   Sean Bright
• [asterisk-dev] [Code Review] [patch] Use of MASTER_CHANNEL causes a race condition ending in a deadlock among channels   Brett Bryant
• [asterisk-dev] [Code Review] [patch] reloading cel_pgsql.so in quick succession crashes asterisk   Russell Bryant
• [asterisk-dev] Documentation updates   Russell Bryant
• [asterisk-dev] Is this locking sequence correct?   Russell Bryant
• [asterisk-dev] [svn-commits] lathama: branch 1.6.2 r305752 - /branches/1.6.2/channels/chan_sip.c   Russell Bryant
• [asterisk-dev] [svn-commits] lathama: branch 1.6.2 r305752 - /branches/1.6.2/channels/chan_sip.c   Russell Bryant
• [asterisk-dev] [svn-commits] lathama: branch 1.6.2 r305752 - /branches/1.6.2/channels/chan_sip.c   Russell Bryant
• [asterisk-dev] [svn-commits] lathama: branch 1.6.2 r305752 - /branches/1.6.2/channels/chan_sip.c   Russell Bryant
• [asterisk-dev] [svn-commits] lathama: branch 1.6.2 r305752 - /branches/1.6.2/channels/chan_sip.c   Russell Bryant
• [asterisk-dev] [asterisk-commits] lathama: branch 1.8 r305987 - in /branches/1.8: ./ configs/ phoneprov/   Russell Bryant
• [asterisk-dev] Media Project Merger   Russell Bryant
• [asterisk-dev] [asterisk-commits] lathama: branch 1.8 r305987 - in /branches/1.8: ./ configs/ phoneprov/   Russell Bryant
• [asterisk-dev] [svn-commits] lathama: branch 1.6.2 r305752 - /branches/1.6.2/channels/chan_sip.c   Russell Bryant
• [asterisk-dev] [asterisk-commits] lathama: branch 1.8 r305987 - in /branches/1.8: ./ configs/ phoneprov/   Russell Bryant
• [asterisk-dev] [svn-commits] lathama: branch 1.6.2 r305752 - /branches/1.6.2/channels/chan_sip.c   Russell Bryant
• [asterisk-dev] [Code Review] Don't allow a REFER with replaces to try to replace the dialog it is a part of   Russell Bryant
• [asterisk-dev] [Code Review] Don't try to pickup a call that is already in the middle of a masquerade   Russell Bryant
• [asterisk-dev] [Code Review] Don't try to pickup a call that is already in the middle of a masquerade   Russell Bryant
• [asterisk-dev] QueueRules   Russell Bryant
• [asterisk-dev] [OT] - Question about dev activity   Russell Bryant
• [asterisk-dev] Asterisk 1.10 Update   Russell Bryant
• [asterisk-dev] IAX2 from 1.8.0 upto 1.8.2 incompatible with 1.8.3 and trunk   Russell Bryant
• [asterisk-dev] Asterisk 1.10 Update   Russell Bryant
• [asterisk-dev] [Code Review] app_directed_pickup Implement the ability to remove a channel group on bridged channel   Russell Bryant
• [asterisk-dev] architecture   Russell Bryant
• [asterisk-dev] [svn-commits] lathama: branch 1.8 r308393 - /branches/1.8/main/http.c   Russell Bryant
• [asterisk-dev] Erroneous email from JIRA   Russell Bryant
• [asterisk-dev] [Code Review] Media Project Phase 2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, Custom format creation in codecs.conf   Russell Bryant
• [asterisk-dev] [asterisk-commits] lathama: trunk r308527 - /trunk/main/http.c   Russell Bryant
• [asterisk-dev] [svn-commits] lathama: branch 1.8 r308393 - /branches/1.8/main/http.c   Russell Bryant
• [asterisk-dev] Subversion doesn't like origsvn   Russell Bryant
• [asterisk-dev] [Code Review] Media Project Phase 2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, Custom format creation in codecs.conf   Russell Bryant
• [asterisk-dev] More awesomeness in trunk for Asterisk 1.10   Russell Bryant
• [asterisk-dev] Default Caller ID   Russell Bryant
• [asterisk-dev] Introducing the new ConfBridge   Russell Bryant
• [asterisk-dev] Introducing the new ConfBridge   Russell Bryant
• [asterisk-dev] [Code Review] [patch] Use of MASTER_CHANNEL causes a race condition ending in a deadlock among channels   Russell Bryant
• [asterisk-dev] [Code Review] [patch] Use of MASTER_CHANNEL causes a race condition ending in a deadlock among channels   Russell Bryant
• [asterisk-dev] [Code Review] [patch] Use of MASTER_CHANNEL causes a race condition ending in a deadlock among channels   Russell Bryant
• [asterisk-dev] Debugging modules for Asterisk   Russell Bryant
• [asterisk-dev] [svn-commits] lathama: branch 1.6.2 r305752 - /branches/1.6.2/channels/chan_sip.c   Tzafrir Cohen
• [asterisk-dev] [svn-commits] lathama: branch 1.6.2 r305752 - /branches/1.6.2/channels/chan_sip.c   Tzafrir Cohen
• [asterisk-dev] Manpages and Wiki Access   Tzafrir Cohen
• [asterisk-dev] [asterisk-users] Calling Directory app from AGI   Tzafrir Cohen
• [asterisk-dev] [Code Review] memory leak in device state callback   Tzafrir Cohen
• [asterisk-dev] [Code Review] a "valgrind" command for live_ast   Tzafrir Cohen
• [asterisk-dev] Subversion doesn't like origsvn   Tzafrir Cohen
• [asterisk-dev] Cross compile   Tzafrir Cohen
• [asterisk-dev] Default Caller ID   Saúl Ibarra Corretgé
• [asterisk-dev] Implementing MWI for Definity PBX served by Asterisk VoiceMail   John R. Covert
• [asterisk-dev] Implementing MWI for Definity PBX served by Asterisk VoiceMail   John R. Covert
• [asterisk-dev] Implementing MWI for Definity PBX served by Asterisk VoiceMail   John R. Covert
• [asterisk-dev] BOUNTY: Implement application MWIMessageSend to call libpri routine   John R. Covert
• [asterisk-dev] BOUNTY: Implement application MWIMessageSend to call libpri routine   John R. Covert
• [asterisk-dev] [svn-commits] lathama: branch 1.6.2 r305752 - /branches/1.6.2/channels/chan_sip.c   Klaus Darilion
• [asterisk-dev] [svn-commits] lathama: branch 1.6.2 r305752 - /branches/1.6.2/channels/chan_sip.c   Klaus Darilion
• [asterisk-dev] IAX hardphone   Klaus Darilion
• [asterisk-dev] Asterisk crashes when receiving SIP answer (200OK)   Klaus Darilion
• [asterisk-dev] [svn-commits] lathama: branch 1.8 r308393 - /branches/1.8/main/http.c   Klaus Darilion
• [asterisk-dev] Default Caller ID   Klaus Darilion
• [asterisk-dev] [Code Review] Media Project Phase 2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, Custom format creation in codecs.conf   Malcolm C. Davenport
• [asterisk-dev] Default Caller ID   Malcolm C. Davenport
• [asterisk-dev] Default Caller ID   Malcolm C. Davenport
• [asterisk-dev] Problems with chan_gtalk in 1.8 branch   Malcolm C. Davenport
• [asterisk-dev] [svn-commits] jpeeler: branch 1.6.2 r303677 - in /branches/1.6.2: ./ apps/app_voicemail.c   Steve Davies
• [asterisk-dev] [svn-commits] jpeeler: branch 1.6.2 r303677 - in /branches/1.6.2: ./ apps/app_voicemail.c   Steve Davies
• [asterisk-dev] [svn-commits] jpeeler: branch 1.6.2 r303677 - in /branches/1.6.2: ./ apps/app_voicemail.c   Steve Davies
• [asterisk-dev] Query of an app_dial diff from SVN   Steve Davies
• [asterisk-dev] Query of an app_dial diff from SVN   Steve Davies
• [asterisk-dev] IAX2 from 1.8.0 upto 1.8.2 incompatible with 1.8.3 and trunk   Alec Davis
• [asterisk-dev] SIP Secure and Annouced Transfert Asterisk 1.8 Trunk.   Alec Davis
• [asterisk-dev] SIP Secure and Annouced Transfert Asterisk 1.8Trunk.   Alec Davis
• [asterisk-dev] [Code Review] fix Deadlock with attended transfers of SIP calls   Alec Davis
• [asterisk-dev] [Code Review] fix Deadlock with attended transfers of SIP calls   Alec Davis
• [asterisk-dev] [Code Review] fix Deadlock with attended transfers of SIP calls   Alec Davis
• [asterisk-dev] [Code Review] fix Deadlock with attended transfers of SIP calls   Alec Davis
• [asterisk-dev] [asterisk-users] Calling Directory app from AGI   Mike Diehl
• [asterisk-dev] [asterisk-users] Calling Directory app from AGI   Mike Diehl
• [asterisk-dev] [asterisk-users] Calling Directory app from AGI   Mike Diehl
• [asterisk-dev] Asterisk software development needed !   Martin Dumas
• [asterisk-dev] IAX hardphone   Steve Edwards
• [asterisk-dev] [asterisk-commits] may: trunk r306499 - /trunk/addons/chan_ooh323.c   Kevin P. Fleming
• [asterisk-dev] [asterisk-commits] may: trunk r306499 - /trunk/addons/chan_ooh323.c   Kevin P. Fleming
• [asterisk-dev] IAX hardphone   Kevin P. Fleming
• [asterisk-dev] IAX hardphone   Kevin P. Fleming
• [asterisk-dev] IAX hardphone   Kevin P. Fleming
• [asterisk-dev] woomera and Woomera vs Unistim   Kevin P. Fleming
• [asterisk-dev] IAX hardphone   Kevin P. Fleming
• [asterisk-dev] IAX hardphone   Kevin P. Fleming
• [asterisk-dev] IAX hardphone   Kevin P. Fleming
• [asterisk-dev] [dahdi] Sync cable code for TE405P   Kevin P. Fleming
• [asterisk-dev] [dahdi] Sync cable code for TE405P   Kevin P. Fleming
• [asterisk-dev] IAX hardphone   Kevin P. Fleming
• [asterisk-dev] return control after ast_bridge_call   Kevin P. Fleming
• [asterisk-dev] SIP Secure and Annouced Transfert Asterisk 1.8 Trunk.   Kevin P. Fleming
• [asterisk-dev] Subversion doesn't like origsvn   Kevin P. Fleming
• [asterisk-dev] Introducing the new ConfBridge   Kevin P. Fleming
• [asterisk-dev] Introducing the new ConfBridge   Kevin P. Fleming
• [asterisk-dev] 1.8.3?   Vinícius Fontes
• [asterisk-dev] 1.8.3?   Vinícius Fontes
• [asterisk-dev] Sip-i   Freddi Hansen
• [asterisk-dev] IAX hardphone   Martin Harriss
• [asterisk-dev] Introducing the new ConfBridge   Jakob Hirsch
• [asterisk-dev] asterisk error   Steve Howes
• [asterisk-dev] [Code Review] Keep RTP DTMF short and sweet   Olle E Johansson
• [asterisk-dev] [Code Review] Keep RTP DTMF short and sweet   Olle E Johansson
• [asterisk-dev] [Code Review] Don't delay DTMF in core bridge while listening for DTMF features   Olle E Johansson
• [asterisk-dev] [Code Review] Don't delay DTMF in core bridge while listening for DTMF features   Olle E Johansson
• [asterisk-dev] [Code Review] Don't delay DTMF in core bridge while listening for DTMF features   Olle E Johansson
• [asterisk-dev] [Code Review] Keep RTP DTMF short and sweet   Olle E. Johansson
• [asterisk-dev] DTMF adventures - DAHDI?   Olle E. Johansson
• [asterisk-dev] DTMF adventures - DAHDI?   Olle E. Johansson
• [asterisk-dev] [svn-commits] lathama: branch 1.6.2 r305752 - /branches/1.6.2/channels/chan_sip.c   Olle E. Johansson
• [asterisk-dev] [svn-commits] lathama: branch 1.6.2 r305752 - /branches/1.6.2/channels/chan_sip.c   Olle E. Johansson
• [asterisk-dev] [svn-commits] lathama: branch 1.6.2 r305752 - /branches/1.6.2/channels/chan_sip.c   Olle E. Johansson
• [asterisk-dev] [svn-commits] lathama: branch 1.6.2 r305752 - /branches/1.6.2/channels/chan_sip.c   Olle E. Johansson
• [asterisk-dev] [svn-commits] lathama: branch 1.6.2 r305752 - /branches/1.6.2/channels/chan_sip.c   Olle E. Johansson
• [asterisk-dev] [svn-commits] lathama: branch 1.6.2 r305752 - /branches/1.6.2/channels/chan_sip.c   Olle E. Johansson
• [asterisk-dev] [svn-commits] lathama: branch 1.6.2 r305752 - /branches/1.6.2/channels/chan_sip.c   Olle E. Johansson
• [asterisk-dev] Media Project Merger   Olle E. Johansson
• [asterisk-dev] [svn-commits] lathama: branch 1.6.2 r305752 - /branches/1.6.2/channels/chan_sip.c   Olle E. Johansson
• [asterisk-dev] Issue 18768: Agi streamfile and MOH   Olle E. Johansson
• [asterisk-dev] Asterisk crashes when receiving SIP answer (200OK)   Olle E. Johansson
• [asterisk-dev] Reminder: Asterisk Developer Call @ 5:00 PM EST Tomorrow   Bryan M. Johns
• [asterisk-dev] Reminder: Asterisk Developer Call Tomorrow at 10:00 AM EST   Bryan M. Johns
• [asterisk-dev] AEL, GoSub and DIALPLAN_EXISTS() do not like each other!   Kirill Katsnelson
• [asterisk-dev] Asterisk freeze and... a catch-22.   Kirill Katsnelson
• [asterisk-dev] Asterisk freeze and... a catch-22.   Kirill Katsnelson
• [asterisk-dev] AEL, GoSub and DIALPLAN_EXISTS() do not like each other!   Kirill Katsnelson
• [asterisk-dev] Asterisk freeze and... a catch-22.   Kirill Katsnelson
• [asterisk-dev] Is this locking sequence correct?   Kirill Katsnelson
• [asterisk-dev] Asterisk vs system thread IDs   Kirill Katsnelson
• [asterisk-dev] Asterisk vs system thread IDs   Kirill Katsnelson
• [asterisk-dev] Asterisk vs system thread IDs   Kirill Katsnelson
• [asterisk-dev] Asterisk vs system thread IDs   Kirill Katsnelson
• [asterisk-dev] Asterisk lock-up with impossible(?) lock backtrace   Kirill Katsnelson
• [asterisk-dev] Default Caller ID   Don Kelly
• [asterisk-dev] [dahdi] Sync cable code for TE405P   Guillaume Knispel
• [asterisk-dev] IAX hardphone   Jeff LaCoursiere
• [asterisk-dev] [svn-commits] lathama: trunk r305650 - /trunk/configs/sip.conf.sample   Andrew Latham
• [asterisk-dev] [svn-commits] lathama: trunk r305650 - /trunk/configs/sip.conf.sample   Andrew Latham
• [asterisk-dev] [svn-commits] lathama: trunk r305650 - /trunk/configs/sip.conf.sample   Andrew Latham
• [asterisk-dev] Documentation Updates   Andrew Latham
• [asterisk-dev] [svn-commits] lathama: branch 1.6.2 r305752 - /branches/1.6.2/channels/chan_sip.c   Andrew Latham
• [asterisk-dev] [svn-commits] lathama: branch 1.6.2 r305752 - /branches/1.6.2/channels/chan_sip.c   Andrew Latham
• [asterisk-dev] [svn-commits] lathama: branch 1.6.2 r305752 - /branches/1.6.2/channels/chan_sip.c   Andrew Latham
• [asterisk-dev] [svn-commits] lathama: branch 1.6.2 r305752 - /branches/1.6.2/channels/chan_sip.c   Andrew Latham
• [asterisk-dev] [svn-commits] lathama: branch 1.6.2 r305752 - /branches/1.6.2/channels/chan_sip.c   Andrew Latham
• [asterisk-dev] [asterisk-commits] lathama: branch 1.8 r305987 - in /branches/1.8: ./ configs/ phoneprov/   Andrew Latham
• [asterisk-dev] [asterisk-commits] lathama: branch 1.8 r305987 - in /branches/1.8: ./ configs/ phoneprov/   Andrew Latham
• [asterisk-dev] [svn-commits] lathama: branch 1.6.2 r305752 - /branches/1.6.2/channels/chan_sip.c   Andrew Latham
• [asterisk-dev] [asterisk-commits] lathama: branch 1.8 r305987 - in /branches/1.8: ./ configs/ phoneprov/   Andrew Latham
• [asterisk-dev] [asterisk-commits] lathama: branch 1.8 r305987 - in /branches/1.8: ./ configs/ phoneprov/   Andrew Latham
• [asterisk-dev] [asterisk-commits] lathama: branch 1.8 r305987 - in /branches/1.8: ./ configs/ phoneprov/   Andrew Latham
• [asterisk-dev] [asterisk-commits] lathama: branch 1.8 r305987 - in /branches/1.8: ./ configs/ phoneprov/   Andrew Latham
• [asterisk-dev] Manpages and Wiki Access   Andrew Latham
• [asterisk-dev] Manpages and Wiki Access   Andrew Latham
• [asterisk-dev] [OT] - Question about dev activity   Andrew Latham
• [asterisk-dev] option_debug, debug, verbose etc...   Andrew Latham
• [asterisk-dev] [svn-commits] lathama: branch 1.8 r308393 - /branches/1.8/main/http.c   Andrew Latham
• [asterisk-dev] [svn-commits] lathama: branch 1.8 r308393 - /branches/1.8/main/http.c   Andrew Latham
• [asterisk-dev] [svn-commits] lathama: branch 1.8 r308393 - /branches/1.8/main/http.c   Andrew Latham
• [asterisk-dev] [asterisk-commits] lathama: trunk r308527 - /trunk/main/http.c   Andrew Latham
• [asterisk-dev] [svn-commits] lathama: branch 1.8 r308393 - /branches/1.8/main/http.c   Andrew Latham
• [asterisk-dev] [svn-commits] lathama: branch 1.8 r308393 - /branches/1.8/main/http.c   Andrew Latham
• [asterisk-dev] [svn-commits] lathama: branch 1.8 r308393 - /branches/1.8/main/http.c   Andrew Latham
• [asterisk-dev] Default Caller ID   Andrew Latham
• [asterisk-dev] [asterisk-users] Calling Directory app from AGI   Tilghman Lesher
• [asterisk-dev] [asterisk-commits] lathama: branch 1.8 r305987 - in /branches/1.8: ./ configs/ phoneprov/   Tilghman Lesher
• [asterisk-dev] [svn-commits] jpeeler: branch 1.6.2 r303677 - in /branches/1.6.2: ./ apps/app_voicemail.c   Tilghman Lesher
• [asterisk-dev] Issue 18768: Agi streamfile and MOH   Tilghman Lesher
• [asterisk-dev] [Code Review] 0018729: [regression] Dial() and Queue() with a macro argument are broken by AEL macro compilation change   Tilghman Lesher
• [asterisk-dev] [Code Review] 0018729: [regression] Dial() and Queue() with a macro argument are broken by AEL macro compilation change   Tilghman Lesher
• [asterisk-dev] [Code Review] Treat integer date fields (and enum fields) properly in cdr_adaptive_odbc   Tilghman Lesher
• [asterisk-dev] IAX hardphone   Tilghman Lesher
• [asterisk-dev] Asterisk vs system thread IDs   Tilghman Lesher
• [asterisk-dev] Asterisk vs system thread IDs   Tilghman Lesher
• [asterisk-dev] Asterisk vs system thread IDs   Tilghman Lesher
• [asterisk-dev] [Code Review] Force module dependencies to load in the correct order   Tilghman Lesher
• [asterisk-dev] [svn-commits] lathama: branch 1.8 r308393 - /branches/1.8/main/http.c   Tilghman Lesher
• [asterisk-dev] Default Caller ID   Tilghman Lesher
• [asterisk-dev] Default Caller ID   Tilghman Lesher
• [asterisk-dev] [Code Review] [patch] Use of MASTER_CHANNEL causes a race condition ending in a deadlock among channels   Tilghman Lesher
• [asterisk-dev] [Code Review] [patch] Use of MASTER_CHANNEL causes a race condition ending in a deadlock among channels   Tilghman Lesher
• [asterisk-dev] question about asterisk 1.4, func_curl.c, and close_wait state   Tilghman Lesher
• [asterisk-dev] CCSS add BLF Device State ability for generic agents   Philippe Lindheimer
• [asterisk-dev] Implementing MWI for Definity PBX served by Asterisk VoiceMail   Doug Lytle
• [asterisk-dev] 1.8.3?   Leif Madsen
• [asterisk-dev] IVR application needed   Leif Madsen
• [asterisk-dev] Default Caller ID   Leif Madsen
• [asterisk-dev] 1.8.3?   Bernard Merindol(F)
• [asterisk-dev] SIP Secure and Annouced Transfert Asterisk 1.8 Trunk.   Bernard Merindol(F)
• [asterisk-dev] SIP Secure and Annouced Transfert Asterisk 1.8 Trunk.   Bernard Merindol(F)
• [asterisk-dev] SIP Secure and Annouced Transfert Asterisk 1.8 Trunk.   Bernard Merindol(F)
• [asterisk-dev] SIP Secure and Annouced Transfert Asterisk 1.8Trunk.   Bernard Merindol(F)
• [asterisk-dev] what is "which" in ast_channel_set_fd   Mark Michelson
• [asterisk-dev] Query of an app_dial diff from SVN   Mark Michelson
• [asterisk-dev] [Code Review] memory leak in device state callback   Mark Michelson
• [asterisk-dev] [svn-commits] lathama: branch 1.8 r308393 - /branches/1.8/main/http.c   Mark Michelson
• [asterisk-dev] IAX hardphone   Débora Moraes
• [asterisk-dev] QueueRules   Débora Moraes
• [asterisk-dev] architecture   Débora Moraes
• [asterisk-dev] architecture   Débora Moraes
• [asterisk-dev] automatic dialer   Débora Moraes
• [asterisk-dev] automatic dialer   Débora Moraes
• [asterisk-dev] How to Change The Caller Position in Queue   Tony Mountifield
• [asterisk-dev] [Code Review] Media Project Phase 2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, Custom format creation in codecs.conf   Tony Mountifield
• [asterisk-dev] DTMF adventures - DAHDI?   Richard Mudgett
• [asterisk-dev] DTMF adventures - DAHDI?   Richard Mudgett
• [asterisk-dev] Implementing MWI for Definity PBX served by Asterisk VoiceMail   Richard Mudgett
• [asterisk-dev] Implementing MWI for Definity PBX served by Asterisk VoiceMail   Richard Mudgett
• [asterisk-dev] [asterisk-commits] may: trunk r306499 - /trunk/addons/chan_ooh323.c   Richard Mudgett
• [asterisk-dev] BOUNTY: Implement application MWIMessageSend to call libpri routine   Richard Mudgett
• [asterisk-dev] IAX hardphone   Mark Murawski
• [asterisk-dev] pbx_findapp() equivalent for functions?   Mark Murawski
• [asterisk-dev] pbx_findapp() equivalent for functions?   Mark Murawski
• [asterisk-dev] IAX hardphone   Mark Murawski
• [asterisk-dev] Asterisk vs system thread IDs   Mark Murawski
• [asterisk-dev] QueueRules   Mark Murawski
• [asterisk-dev] Asterisk vs system thread IDs   Mark Murawski
• [asterisk-dev] AEL, GoSub and DIALPLAN_EXISTS() do not like each other!   Steve Murphy
• [asterisk-dev] AEL, GoSub and DIALPLAN_EXISTS() do not like each other!   Steve Murphy
• [asterisk-dev] Code review of: app_queue: Give members a penalty time for not answering   Håkon Nessjøen
• [asterisk-dev] Code review of: app_queue: Give members a penalty time for not answering   Håkon Nessjøen
• [asterisk-dev] [Code Review] Don't try to pickup a call that is already in the middle of a masquerade   Michael Neuhauser
• [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] Add a SIPpeerstatus command to chan_sip's manager interface.   Matthew Nicholson
• [asterisk-dev] [Code Review] fix Deadlock with attended transfers of SIP calls   Matthew Nicholson
• [asterisk-dev] [Code Review] fix Deadlock with attended transfers of SIP calls   Matthew Nicholson
• [asterisk-dev] [Code Review] fix Deadlock with attended transfers of SIP calls   Matthew Nicholson
• [asterisk-dev] IAX hardphone   Jeffrey Ollie
• [asterisk-dev] IAX hardphone   Jeffrey Ollie
• [asterisk-dev] IAX hardphone   Jeffrey Ollie
• [asterisk-dev] External MWI question   Alejandro Orellana
• [asterisk-dev] [svn-commits] jpeeler: branch 1.6.2 r303677 - in /branches/1.6.2: ./ apps/app_voicemail.c   Jeff Peeler
• [asterisk-dev] shan_sip: Should we zero the Content-Length here?   Dimitar Penev
• [asterisk-dev] [OT] - Question about dev activity   Pablo Perez
• [asterisk-dev] Subversion doesn't like origsvn   Simon Perreault
• [asterisk-dev] Default Caller ID   Simon Perreault
• [asterisk-dev] what is "which" in ast_channel_set_fd   Alok Prasad
• [asterisk-dev] woomera and Woomera vs Unistim   Alok Prasad
• [asterisk-dev] channel_read doesnet work for some fd values in ast_channel_set_fd()   Alok Prasad
• [asterisk-dev] Inband RTP Ringback tone generator function   Alok Prasad
• [asterisk-dev] (no subject)   Abhi Rana
• [asterisk-dev] IVR application needed   Abhi Rana
• [asterisk-dev] Create Multiple Context for many Companies   Amardeep Rana
• [asterisk-dev] svn dahdi: pciradio doesn't build   Shaun Ruffell
• [asterisk-dev] [dahdi] Sync cable code for TE405P   Shaun Ruffell
• [asterisk-dev] [dahdi] Sync cable code for TE405P   Shaun Ruffell
• [asterisk-dev] Create Multiple Context for many Companies   David Ruggles
• [asterisk-dev] IAX hardphone   David Ruggles
• [asterisk-dev] Subversion doesn't like origsvn   David Ruggles
• [asterisk-dev] Subversion doesn't like origsvn   David Ruggles
• [asterisk-dev] Subversion doesn't like origsvn   David Ruggles
• [asterisk-dev] Default Caller ID   David Ruggles
• [asterisk-dev] Cross compile   David Ruggles
• [asterisk-dev] Hi,friends!   David Ruggles
• [asterisk-dev] Spam sent from my gmail account   David Ruggles
• [asterisk-dev] Asterisk 1.10 Update   Eliel Sardañons
• [asterisk-dev] Hold-Unhold information in queue-log   Stefan Schmidt
• [asterisk-dev] Default Caller ID   Stefan Schmidt
• [asterisk-dev] app_sms   Sebastian
• [asterisk-dev] app_sms   Sebastian
• [asterisk-dev] DAHDI timing source cleanup proposal   Pavel Selivanov
• [asterisk-dev] IAX hardphone   Bill Shaw
• [asterisk-dev] IAX hardphone   Bill Shaw
• [asterisk-dev] IAX hardphone   Bill Shaw
• [asterisk-dev] IAX hardphone   Bill Shaw
• [asterisk-dev] IAX hardphone   Bill Shaw
• [asterisk-dev] IAX hardphone   Bill Shaw
• [asterisk-dev] IAX hardphone   Bill Shaw
• [asterisk-dev] IAX hardphone   Bill Shaw
• [asterisk-dev] IAX hardphone   Bill Shaw
• [asterisk-dev] IAX hardphone   Bill Shaw
• [asterisk-dev] IAX hardphone   Bill Shaw
• [asterisk-dev] IAX hardphone   Bill Shaw
• [asterisk-dev] IAX hardphone   Bill Shaw
• [asterisk-dev] IAX hardphone   Bill Shaw
• [asterisk-dev] IAX hardphone   Bill Shaw
• [asterisk-dev] IAX hardphone   Bill Shaw
• [asterisk-dev] IAX hardphone   Andreas Sikkema
• [asterisk-dev] pbx_lua RPM module   Nir Simionovich
• [asterisk-dev] Default Caller ID   Steve Sokol
• [asterisk-dev] Deadlock in chan_dahdi when pbx_builtin_setvar_helper is called on a newly allocated channel. How to trace?   Antonio Goméz Soto
• [asterisk-dev] Scheduled Maintenance: wiki.asterisk.org and code.asterisk.org   Asterisk Development Team
• [asterisk-dev] Asterisk 1.4.40-rc3 Now Available   Asterisk Development Team
• [asterisk-dev] Asterisk 1.6.2.17-rc3 Now Available   Asterisk Development Team
• [asterisk-dev] Asterisk 1.8.3-rc3 Now Available   Asterisk Development Team
• [asterisk-dev] Asterisk 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and 1.8.2.4 Now Available   Asterisk Development Team
• [asterisk-dev] Asterisk 1.4.40 Now Available   Asterisk Development Team
• [asterisk-dev] Asterisk 1.6.2.17 Now Available   Asterisk Development Team
• [asterisk-dev] Asterisk 1.8.3 Now Available   Asterisk Development Team
• [asterisk-dev] Asterisk 1.4.41-rc1 Now Available   Asterisk Development Team
• [asterisk-dev] Asterisk 1.6.2.18-rc1 Now Available   Asterisk Development Team
• [asterisk-dev] Asterisk 1.8.4-rc2 Now Available   Asterisk Development Team
• [asterisk-dev] AST-2011-002: Multiple array overflow and crash vulnerabilities in UDPTL code   Asterisk Security Team
• [asterisk-dev] Problems with chan_gtalk in 1.8 branch   Pavel Troller
• [asterisk-dev] [Code Review] Asterisk media architecture conversion - no more format bitfields   David Vossel
• [asterisk-dev] Media Project Merger   David Vossel
• [asterisk-dev] [Code Review] Don't allow a REFER with replaces to try to replace the dialog it is a part of   David Vossel
• [asterisk-dev] [Code Review] Don't delay DTMF in core bridge while listening for DTMF features   David Vossel
• [asterisk-dev] [Code Review] sip deadlock fix in handle_request_do   David Vossel
• [asterisk-dev] [Code Review] Media Project Phase 2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, Custom format creation in codecs.conf   David Vossel
• [asterisk-dev] [Code Review] Media Project Phase 2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, Custom format creation in codecs.conf   David Vossel
• [asterisk-dev] [Code Review] Media Project Phase 2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, Custom format creation in codecs.conf   David Vossel
• [asterisk-dev] IAX hardphone   David Vossel
• [asterisk-dev] [Code Review] Media Project Phase 2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, Custom format creation in codecs.conf   David Vossel
• [asterisk-dev] [Code Review] Media Project Phase 2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, Custom format creation in codecs.conf   David Vossel
• [asterisk-dev] [Code Review] Media Project Phase 2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, Custom format creation in codecs.conf   David Vossel
• [asterisk-dev] [Code Review] Use 'remotesecret' when it is set instead of 'secret' when authenticating to a remote party   David Vossel
• [asterisk-dev] [Code Review] Media Project Phase 2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, Custom format creation in codecs.conf   David Vossel
• [asterisk-dev] [Code Review] Media Project Phase 2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, Custom format creation in codecs.conf   David Vossel
• [asterisk-dev] [Code Review] Media Project Phase 2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, Custom format creation in codecs.conf   David Vossel
• [asterisk-dev] [Code Review] Media Project Phase 2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, Custom format creation in codecs.conf   David Vossel
• [asterisk-dev] [Code Review] Media Project Phase 2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, Custom format creation in codecs.conf   David Vossel
• [asterisk-dev] [Code Review] Media Project Phase 2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, Custom format creation in codecs.conf   David Vossel
• [asterisk-dev] Introducing the new ConfBridge   David Vossel
• [asterisk-dev] Introducing the new ConfBridge   David Vossel
• [asterisk-dev] Regression with changeset 298596 (Fix improper hangup when doing an attended transfer to queue)   Kristijan Vrban
• [asterisk-dev] Asterisk freeze and... a catch-22.   Ryan Wagoner
• [asterisk-dev] Asterisk freeze and... a catch-22.   Ryan Wagoner
• [asterisk-dev] Introducing the new ConfBridge   Watkins, Bradley
• [asterisk-dev] [Code Review] Asterisk media architecture conversion - no more format bitfields   Terry Wilson
• [asterisk-dev] [Code Review] Asterisk media architecture conversion - no more format bitfields   Terry Wilson
• [asterisk-dev] [Code Review] Don't allow a REFER with replaces to try to replace the dialog it is a part of   Terry Wilson
• [asterisk-dev] [Code Review] Don't try to pickup a call that is already in the middle of a masquerade   Terry Wilson
• [asterisk-dev] [Code Review] Don't allow a REFER with replaces to try to replace the dialog it is a part of   Terry Wilson
• [asterisk-dev] Create Multiple Context for many Companies   Terry Wilson
• [asterisk-dev] SRTP configuration and behavior in Asterisk18   Terry Wilson
• [asterisk-dev] [Code Review] Use 'remotesecret' when it is set instead of 'secret' when authenticating to a remote party   Terry Wilson
• [asterisk-dev] [Code Review] Don't delay DTMF in core bridge while listening for DTMF features   Terry Wilson
• [asterisk-dev] More awesomeness in trunk for Asterisk 1.10   Terry Wilson
• [asterisk-dev] Introducing the new ConfBridge   Michael L. Young
• [asterisk-dev] Introducing the new ConfBridge   Michael L. Young
• [asterisk-dev] Play one audio file to the called part before the Dial() command   Songtao Yu
• [asterisk-dev] automatic dialer   ayodele abejide
• [asterisk-dev] Hold-Unhold information in queue-log   winay chaudhari
• [asterisk-dev] Hold-Unhold information in queue-log   winay chaudhari
• [asterisk-dev] svn dahdi: pciradio doesn't build   sean darcy
• [asterisk-dev] svn dahdi: pciradio doesn't build   sean darcy
• [asterisk-dev] How to Change The Caller Position in Queue   shayne.alone at gmail.com
• [asterisk-dev] How to Change The Caller Position in Queue   shayne.alone at gmail.com
• [asterisk-dev] QueueRules   shayne.alone at gmail.com
• [asterisk-dev] QueueRules   shayne.alone at gmail.com
• [asterisk-dev] [Code Review] Changes to h323 to allow use with h323plus > 1.20.   irroot
• [asterisk-dev] [Code Review] Implemet the ability to remove a channel group on bridged channel   irroot
• [asterisk-dev] [Code Review] app_queue per member ringinuse [ignorebusy] / set chan group on answer / change behaviour of neg penalty   irroot
• [asterisk-dev] [Code Review] app_directed_pickup Implement the ability to remove a channel group on bridged channel   irroot
• [asterisk-dev] [Code Review] Changes to h323 to allow use with h323plus > 1.20.   irroot
• [asterisk-dev] [Code Review] Changes to h323 to allow use with h323plus > 1.20.   irroot
• [asterisk-dev] [Code Review] Changes to h323 to allow use with h323plus > 1.20.   irroot
• [asterisk-dev] [Code Review] app_queue per member ringinuse [ignorebusy] / set chan group on answer / change behaviour of neg penalty   irroot
• [asterisk-dev] [Code Review] app_queue per member ringinuse [ignorebusy] / set chan group on answer / change behaviour of neg penalty   irroot
• [asterisk-dev] [Code Review] app_directed_pickup Implement the ability to remove a channel group on bridged channel   irroot
• [asterisk-dev] [Code Review] app_directed_pickup Implement the ability to remove a channel group on bridged channel   irroot
• [asterisk-dev] [Code Review] Fax Gateway Implementation T30<->T38   irroot
• [asterisk-dev] [Code Review] Fax Gateway Implementation T30<->T38   irroot
• [asterisk-dev] [Code Review] Set T38 State To UNAVAILABLE when we could not negotiate T38   irroot
• [asterisk-dev] [Code Review] Fax Gateway Implementation T30<->T38   irroot
• [asterisk-dev] [Code Review] Fax Gateway Implementation T30<->T38   irroot
• [asterisk-dev] [Code Review] Set T38 State To UNAVAILABLE when we could not negotiate T38   irroot
• [asterisk-dev] [Code Review] app_directed_pickup Implement the ability to remove a channel group on bridged channel   irroot
• [asterisk-dev] [Code Review] ast_dsp_process logs a error about inband DTMF on faxdetect shut it up   irroot
• [asterisk-dev] [Code Review] 0018729: [regression] Dial() and Queue() with a macro argument are broken by AEL macro compilation change   kkm
• [asterisk-dev] [Code Review] 0018729: [regression] Dial() and Queue() with a macro argument are broken by AEL macro compilation change   kkm
• [asterisk-dev] [Code Review] 0018729: [regression] Dial() and Queue() with a macro argument are broken by AEL macro compilation change   kkm
• [asterisk-dev] [Code Review] Announce to user that they have been muted when muting is done via AMI   kobaz
• [asterisk-dev] [Code Review] Announce to user that they have been muted when muting is done via AMI   kobaz
• [asterisk-dev] [Code Review] Announce to user that they have been muted when muting is done via AMI   kobaz
• [asterisk-dev] [Code Review] Auto create context when using dialplan add   kobaz
• [asterisk-dev] [Code Review] Auto create context when using dialplan add   kobaz
• [asterisk-dev] (no subject)   akash mishra
• [asterisk-dev] asterisk error   akash mishra
• [asterisk-dev] [Code Review] CCSS Proposal to add Asterisk DEVSTATEs to generic agents to use with hints   p_lindheimer
• [asterisk-dev] [Code Review] CCSS Proposal to add Asterisk DEVSTATEs to generic agents to use with hints   p_lindheimer
• [asterisk-dev] [Code Review] CCSS Proposal to add Asterisk DEVSTATEs to generic agents to use with hints   p_lindheimer
• [asterisk-dev] [Code Review] CCSS Proposal to add Asterisk DEVSTATEs to generic agents to use with hints   p_lindheimer
• [asterisk-dev] [Code Review] CCSS Proposal to add Asterisk DEVSTATEs to generic agents to use with hints   p_lindheimer
• [asterisk-dev] [Code Review] CCSS Proposal to add Asterisk DEVSTATEs to generic agents to use with hints   p_lindheimer
• [asterisk-dev] [Code Review] CCSS Proposal to add Asterisk DEVSTATEs to generic agents to use with hints   p_lindheimer
• [asterisk-dev] [Code Review] CCSS Proposal to add Asterisk DEVSTATEs to generic agents to use with hints   p_lindheimer
• [asterisk-dev] [Code Review] CCSS Proposal to add Asterisk DEVSTATEs to generic agents to use with hints   p_lindheimer
• [asterisk-dev] [Code Review] CCSS Proposal to add Asterisk DEVSTATEs to generic agents to use with hints   p_lindheimer
• [asterisk-dev] [Code Review] Repair UDP port leaks, Memory Leaks, Denial of Service in chan_sip   rgagnon
• [asterisk-dev] [Code Review] Repair UDP port leaks, Memory Leaks, Denial of Service in chan_sip   rgagnon
• [asterisk-dev] [Code Review] Repair UDP port leaks, Memory Leaks, Denial of Service in chan_sip   rgagnon
• [asterisk-dev] [Code Review] Repair UDP port leaks, Memory Leaks, Denial of Service in chan_sip   rgagnon
• [asterisk-dev] [Code Review] Repair UDP port leaks, Memory Leaks, Denial of Service in chan_sip   rgagnon
• [asterisk-dev] [Code Review] Repair UDP port leaks, Memory Leaks, Denial of Service in chan_sip   rgagnon
• [asterisk-dev] [Code Review] app_queue: skill routing   romain_proformatique
• [asterisk-dev] [Code Review] Repair UDP port leaks, Memory Leaks, Denial of Service in chan_sip   schmidts
• [asterisk-dev] [Code Review] Repair UDP port leaks, Memory Leaks, Denial of Service in chan_sip   schmidts
• [asterisk-dev] [Code Review] Repair UDP port leaks, Memory Leaks, Denial of Service in chan_sip   schmidts
• [asterisk-dev] [Code Review] RFC 3261 compliant ACK retransmission for unreliable final responses   wdoekes

Last message date: Mon Feb 28 15:06:29 CST 2011
Archived on: Mon Feb 28 15:08:32 CST 2011

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