January 2009 Archives by author

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Starting: Thu Jan 1 09:12:27 CST 2009
Ending: Sat Jan 31 16:56:58 CST 2009
Messages: 389

• [asterisk-dev] Adding Bluetooth Pairing Support to chan_mobile   Vlasis Hatzistavrou (KTI)
• [asterisk-dev] OT: Paying people in faraway (Western) places.   Chris Mason (Lists)
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• [asterisk-dev] Asterisk CLI got freezed!!   Max Alex
• [asterisk-dev] Patch: wcfxo; Failed to initailize DAA, giving up...   Timothy Allen
• [asterisk-dev] No emails from Mantis   Benny Amorsen
• [asterisk-dev] i extension does not match on initial context - bug or not?   Benny Amorsen
• [asterisk-dev] i extension does not match on initial context - bug or not?   Benny Amorsen
• [asterisk-dev] i extension does not match on initial context - bug or not?   Benny Amorsen
• [asterisk-dev] SIPAdd/RemoveHeader apps   Benny Amorsen
• [asterisk-dev] SIPAdd/RemoveHeader apps   Benny Amorsen
• [asterisk-dev] Access to encryption functions from dial plan   Benny Amorsen
• [asterisk-dev] Access to encryption functions from dial plan   Benny Amorsen
• [asterisk-dev] Access to encryption functions from dial plan   Benny Amorsen
• [asterisk-dev] chan_sip SIP Authentication   Benny Amorsen
• [asterisk-dev] chan_sip SIP Authentication   Benny Amorsen
• [asterisk-dev] chan_sip SIP Authentication   Benny Amorsen
• [asterisk-dev] chan_sip.c - no reply to critical packet   Benny Amorsen
• [asterisk-dev] [design] Realtime changes   Dmitry Andrianov
• [asterisk-dev] [Code Review] Queue SIP requests/responses that cannot be immediately processed.   Dmitry Andrianov
• [asterisk-dev] [Code Review] Queue SIP requests/responses that cannot be immediately processed.   Dmitry Andrianov
• [asterisk-dev] i extension does not match on initial context - bug or not?   Dmitry Andrianov
• [asterisk-dev] i extension does not match on initial context - bug or not?   Dmitry Andrianov
• [asterisk-dev] i extension does not match on initial context - bug or not?   Dmitry Andrianov
• [asterisk-dev] i extension does not match on initial context - bug or not?   Dmitry Andrianov
• [asterisk-dev] G.729.1 - any interest?   Dmitry Andrianov
• [asterisk-dev] AEL macro backward compatibility for 1.6   Dmitry Andrianov
• [asterisk-dev] AEL macro backward compatibility for 1.6   Dmitry Andrianov
• [asterisk-dev] How ti let execution continue to the next priorité after callee or caller hangup   Dmitry Andrianov
• [asterisk-dev] BugID 014021, Zap/DAHDI timers, internal timing and packetization   Dan Austin
• [asterisk-dev] [Code Review] unload/load/reload support for chan_skinny   Dan Austin
• [asterisk-dev] [Code Review] unload/load/reload support for chan_skinny   Michiel van Baak
• [asterisk-dev] Need to start in development   Michiel van Baak
• [asterisk-dev] RFC: Capacity Study   Alex Balashov
• [asterisk-dev] OT: Paying people in faraway (Western) places.   Alex Balashov
• [asterisk-dev] OT: Paying people in faraway (Western) places.   Alex Balashov
• [asterisk-dev] GPU Audio Codec Transcoding within Asterisk PBX   Joseph Benden
• [asterisk-dev] GPU Audio Codec Transcoding within Asterisk PBX   Joseph Benden
• [asterisk-dev] GPU Audio Codec Transcoding within Asterisk PBX   Felipe Bergo
• [asterisk-dev] RTP interop with Sonus: hack   Gregory Boehnlein
• [asterisk-dev] chan_sip.c - no reply to critical packet   Gregory Boehnlein
• [asterisk-dev] chan_sip.c - no reply to critical packet   Gregory Boehnlein
• [asterisk-dev] chan_sip.c - no reply to critical packet   Gregory Boehnlein
• [asterisk-dev] chan_sip.c - no reply to critical packet   Gregory Boehnlein
• [asterisk-dev] chan_sip.c - no reply to critical packet   Gregory Boehnlein
• [asterisk-dev] Adding Bluetooth Pairing Support to chan_mobile   David Bowerman
• [asterisk-dev] Adding Bluetooth Pairing Support to chan_mobile   David Bowerman
• [asterisk-dev] seanbright: tools/trunk r5656 - /tools/trunk/dahdi_monitor.c   Sean Bright
• [asterisk-dev] Asterisk 1.4.23 Now Available!   Sean Bright
• [asterisk-dev] [Code Review] Queue SIP requests/responses that cannot be immediately processed.   Russell Bryant
• [asterisk-dev] [Code Review] Forward port of SIP request queueing to trunk   Russell Bryant
• [asterisk-dev] [Code Review] Forward port of SIP request queueing to trunk   Russell Bryant
• [asterisk-dev] 2008 Post Count   Russell Bryant
• [asterisk-dev] Conversion to C++ (groups/asterisk-cpp)?   Russell Bryant
• [asterisk-dev] seanbright: tools/trunk r5656 - /tools/trunk/dahdi_monitor.c   Russell Bryant
• [asterisk-dev] seanbright: tools/trunk r5656 - /tools/trunk/dahdi_monitor.c   Russell Bryant
• [asterisk-dev] SIPAdd/RemoveHeader apps (was: oej: trunk r168638 - /trunk/channels/chan_sip.c)   Russell Bryant
• [asterisk-dev] calls problem   Russell Bryant
• [asterisk-dev] calls problem   Russell Bryant
• [asterisk-dev] SIPAdd/RemoveHeader apps   Russell Bryant
• [asterisk-dev] Minimum packages for building and running asterisk   Russell Bryant
• [asterisk-dev] SIP channel/owner question   Russell Bryant
• [asterisk-dev] [Code Review] properly report ast_func_read errors in getvar AMI action   Russell Bryant
• [asterisk-dev] Access to encryption functions from dial plan   Russell Bryant
• [asterisk-dev] [Code Review] added AES_ENCRYPT and AES_DECRYPT dialplan functions   Russell Bryant
• [asterisk-dev] Welcome to David Vossel   Russell Bryant
• [asterisk-dev] [Code Review] added AES_ENCRYPT and AES_DECRYPT dialplan functions   Russell Bryant
• [asterisk-dev] New AGI manager command: PlaySound   Russell Bryant
• [asterisk-dev] [Code Review] This patch implements CEL in trunk.   Russell Bryant
• [asterisk-dev] [Code Review] Add common implementation for a scheduler context with a dedicated thread   Russell Bryant
• [asterisk-dev] Using SIP REFER with a non bridged channel - is there a workaround?   Russell Bryant
• [asterisk-dev] [Code Review] unload/load/reload support for chan_skinny   Russell Bryant
• [asterisk-dev] [Code Review] Resolve some race conditions in chan_iax2 scheduler handling   Russell Bryant
• [asterisk-dev] [Code Review] added AES_ENCRYPT and AES_DECRYPT dialplan functions   Russell Bryant
• [asterisk-dev] [Code Review] Calendaring API for Asterisk   Russell Bryant
• [asterisk-dev] [Code Review] Calendaring API for Asterisk   Russell Bryant
• [asterisk-dev] [Code Review] added AES_ENCRYPT and AES_DECRYPT dialplan functions   Russell Bryant
• [asterisk-dev] [Code Review] Add common implementation for a scheduler context with a dedicated thread   Russell Bryant
• [asterisk-dev] [Code Review] Add common implementation for a scheduler context with a dedicated thread   Russell Bryant
• [asterisk-dev] [Code Review] Add common implementation for a scheduler context with a dedicated thread   Russell Bryant
• [asterisk-dev] [Code Review] Add common implementation for a scheduler context with a dedicated thread   Russell Bryant
• [asterisk-dev] [Code Review] Add common implementation for a scheduler context with a dedicated thread   Russell Bryant
• [asterisk-dev] DTMF queuing   Russell Bryant
• [asterisk-dev] [Code Review] Keep bridge up if parking attempt fails   Russell Bryant
• [asterisk-dev] [Code Review] properly report ast_func_read errors in getvar AMI action   Russell Bryant
• [asterisk-dev] [Code Review] Bridging API for Conference Bridge purposes   Russell Bryant
• [asterisk-dev] [Code Review] Bridging API for Conference Bridge purposes   Russell Bryant
• [asterisk-dev] [Code Review] This patch implements CEL in trunk.   Russell Bryant
• [asterisk-dev] [Code Review] properly report ast_func_read errors in getvar AMI action   Russell Bryant
• [asterisk-dev] SMDI Function Call Redux   Russell Bryant
• [asterisk-dev] SMDI Function Call Redux   Russell Bryant
• [asterisk-dev] [Code Review] unload/load/reload support for chan_skinny   Russell Bryant
• [asterisk-dev] jpeeler: branch 1.0 r465 - in /branches/1.0: menuselect.c menuselect.h   Russell Bryant
• [asterisk-dev] ast_careful_fwrite: fflush() returned error: Broken pipe   Russell Bryant
• [asterisk-dev] Conversion to C++ (groups/asterisk-cpp)?   David A. Burgess
• [asterisk-dev] Newbie Developer Resource?   Nicolas Chapleau
• [asterisk-dev] dahdi static device files   Tzafrir Cohen
• [asterisk-dev] dahdi static device files   Tzafrir Cohen
• [asterisk-dev] Interesting SVN graphical representations of Asterisk project   Tzafrir Cohen
• [asterisk-dev] i extension does not match on initial context - bug or not?   Tzafrir Cohen
• [asterisk-dev] 1.6.0.3 chan_dahdi - can not dial FXS until call is first recieved - including fix   Tzafrir Cohen
• [asterisk-dev] voicemail: storing vmsecret in /var/spool/asterisk/voicemail   Tzafrir Cohen
• [asterisk-dev] dahdi sysfs branch   Tzafrir Cohen
• [asterisk-dev] dahdi sysfs branch   Tzafrir Cohen
• [asterisk-dev] Dean Collins added you as a business connection on Plaxo   Dean Collins
• [asterisk-dev] Dean Collins added you as a business connection on Plaxo   Dean Collins
• [asterisk-dev] RTP interop with Sonus: hack   Joshua Colp
• [asterisk-dev] RTP interop with Sonus: hack   Joshua Colp
• [asterisk-dev] [Code Review] Queue SIP requests/responses that cannot be immediately processed.   Joshua Colp
• [asterisk-dev] RTP interop with Sonus: hack   Joshua Colp
• [asterisk-dev] RTP interop with Sonus: hack   Joshua Colp
• [asterisk-dev] [Code Review] Bridging API for Conference Bridge purposes   Joshua Colp
• [asterisk-dev] [Code Review] Bridging API for Conference Bridge purposes   Joshua Colp
• [asterisk-dev] [Code Review] Bridging API for Conference Bridge purposes   Joshua Colp
• [asterisk-dev] [Code Review] Bridging API for Conference Bridge purposes   Joshua Colp
• [asterisk-dev] 2008 Post Count   Steven S. Critchfield
• [asterisk-dev] Welcome to David Vossel   Steven S. Critchfield
• [asterisk-dev] [Code Review] Queue SIP requests/responses that cannot be immediately processed.   Klaus Darilion
• [asterisk-dev] [Code Review] Queue SIP requests/responses that cannot be immediately processed.   Klaus Darilion
• [asterisk-dev] i extension does not match on initial context - bug or not?   Klaus Darilion
• [asterisk-dev] i extension does not match on initial context - bug or not?   Klaus Darilion
• [asterisk-dev] i extension does not match on initial context - bug or not?   Klaus Darilion
• [asterisk-dev] reset/remove SipAddHeader() headers   Klaus Darilion
• [asterisk-dev] strange spiral handling in chan_sip   Klaus Darilion
• [asterisk-dev] reset/remove SipAddHeader() headers   Klaus Darilion
• [asterisk-dev] reset/remove SipAddHeader() headers   Klaus Darilion
• [asterisk-dev] SIPAdd/RemoveHeader apps   Klaus Darilion
• [asterisk-dev] reset/remove SipAddHeader() headers   Klaus Darilion
• [asterisk-dev] voicemail: storing vmsecret in /var/spool/asterisk/voicemail   Klaus Darilion
• [asterisk-dev] voicemail: storing vmsecret in /var/spool/asterisk/voicemail   Klaus Darilion
• [asterisk-dev] voicemail: storing vmsecret in /var/spool/asterisk/voicemail   Klaus Darilion
• [asterisk-dev] reset/remove SipAddHeader() headers   Klaus Darilion
• [asterisk-dev] SIPAdd/RemoveHeader apps   Klaus Darilion
• [asterisk-dev] SIP channel/owner question   Klaus Darilion
• [asterisk-dev] X-Asterisk-HangupCause header only added in auto fallthrough mode   Klaus Darilion
• [asterisk-dev] SIP channel/owner question   Klaus Darilion
• [asterisk-dev] SIP channel/owner question   Klaus Darilion
• [asterisk-dev] chan_sip SIP Authentication   Klaus Darilion
• [asterisk-dev] chan_sip SIP Authentication   Klaus Darilion
• [asterisk-dev] chan_sip SIP Authentication   Klaus Darilion
• [asterisk-dev] chan_sip SIP Authentication   Klaus Darilion
• [asterisk-dev] chan_sip SIP Authentication   Klaus Darilion
• [asterisk-dev] chan_sip SIP Authentication   Klaus Darilion
• [asterisk-dev] chan_sip SIP Authentication   Klaus Darilion
• [asterisk-dev] chan_sip SIP Authentication   Klaus Darilion
• [asterisk-dev] chan_sip SIP Authentication   Klaus Darilion
• [asterisk-dev] chan_sip SIP Authentication   Klaus Darilion
• [asterisk-dev] chan_sip SIP Authentication   Klaus Darilion
• [asterisk-dev] wctdm 4/8/24 VMWI generation capabilities   Alec Davis
• [asterisk-dev] Reminder: Emerging Communications Conference Early-Bird   Lee S Dryburgh
• [asterisk-dev] Apple TV and Asterisk - who was it?   Johansson Olle E
• [asterisk-dev] [Code Review] Queue SIP requests/responses that cannot be immediately processed.   Johansson Olle E
• [asterisk-dev] SIPAdd/RemoveHeader apps   Johansson Olle E
• [asterisk-dev] SIPAdd/RemoveHeader apps   Johansson Olle E
• [asterisk-dev] X-Asterisk-HangupCause header only added in auto fallthrough mode   Johansson Olle E
• [asterisk-dev] SIP channel/owner question   Johansson Olle E
• [asterisk-dev] [Code Review] Patch for issue 12312 (DNS SRV lookups causing re-registration problems)   Johansson Olle E
• [asterisk-dev] chan_sip SIP Authentication   Johansson Olle E
• [asterisk-dev] chan_sip SIP Authentication   Johansson Olle E
• [asterisk-dev] chan_sip SIP Authentication   Johansson Olle E
• [asterisk-dev] chan_sip SIP Authentication   Johansson Olle E
• [asterisk-dev] chan_sip SIP Authentication   Johansson Olle E
• [asterisk-dev] chan_sip SIP Authentication   Johansson Olle E
• [asterisk-dev] chan_sip SIP Authentication   Johansson Olle E
• [asterisk-dev] chan_sip SIP Authentication   Johansson Olle E
• [asterisk-dev] chan_sip SIP Authentication   Johansson Olle E
• [asterisk-dev] chan_sip SIP Authentication   Johansson Olle E
• [asterisk-dev] chan_sip SIP Authentication   Johansson Olle E
• [asterisk-dev] chan_sip SIP Authentication   Johansson Olle E
• [asterisk-dev] chan_sip SIP Authentication   Johansson Olle E
• [asterisk-dev] chan_sip SIP Authentication   Johansson Olle E
• [asterisk-dev] chan_sip SIP Authentication   Johansson Olle E
• [asterisk-dev] chan_sip SIP Authentication   Johansson Olle E
• [asterisk-dev] chan_sip SIP Authentication   Johansson Olle E
• [asterisk-dev] Asterisk RFC2833 SDP fmtp 0-16?   Johansson Olle E
• [asterisk-dev] chan_sip.c - no reply to critical packet   Johansson Olle E
• [asterisk-dev] 1.6.0.3 chan_dahdi - can not dial FXS until call is first recieved - including fix   Karl Fife
• [asterisk-dev] [Code Review] Queue SIP requests/responses that cannot be immediately processed.   Kevin Fleming
• [asterisk-dev] [Code Review] Queue SIP requests/responses that cannot be immediately processed.   Kevin Fleming
• [asterisk-dev] [Code Review] Queue SIP requests/responses that cannot be immediately processed.   Kevin Fleming
• [asterisk-dev] [Code Review] Forward port of SIP request queueing to trunk   Kevin Fleming
• [asterisk-dev] [Code Review] Forward port of SIP request queueing to trunk   Kevin Fleming
• [asterisk-dev] [Code Review] Convert character pointers in sip_request structure to be offsets from the beginning of the buffer   Kevin Fleming
• [asterisk-dev] [Code Review] Convert character pointers in sip_request structure to be offsets from the beginning of the buffer   Kevin Fleming
• [asterisk-dev] [design] Realtime changes   Kevin P. Fleming
• [asterisk-dev] [Code Review] Queue SIP requests/responses that cannot be immediately processed.   Kevin P. Fleming
• [asterisk-dev] [Code Review] Queue SIP requests/responses that cannot be immediately processed.   Kevin P. Fleming
• [asterisk-dev] origsvn.digium.com maintenance   Kevin P. Fleming
• [asterisk-dev] origsvn.digium.com maintenance completed   Kevin P. Fleming
• [asterisk-dev] seanbright: tools/trunk r5656 - /tools/trunk/dahdi_monitor.c   Kevin P. Fleming
• [asterisk-dev] G.729.1 - any interest?   Kevin P. Fleming
• [asterisk-dev] G.729.1 - any interest?   Kevin P. Fleming
• [asterisk-dev] chan_sip.c - no reply to critical packet   Kevin P. Fleming
• [asterisk-dev] G.729.1 - any interest?   Stéphane Van Geystelen
• [asterisk-dev] G.729.1 - any interest?   Stéphane Van Geystelen
• [asterisk-dev] chan_sip.c - no reply to critical packet   James Golovich
• [asterisk-dev] calls problem   Ali Haydar GÜMÜŞ
• [asterisk-dev] calls problem   Ali Haydar GÜMÜŞ
• [asterisk-dev] codec preference   David Hansen
• [asterisk-dev] codec preference   David Hansen
• [asterisk-dev] Fall Markdowns, hurry before they’re gone!   ® Men's Health
• [asterisk-dev] development possibilities with existing SW   Steven Hoffmann
• [asterisk-dev] Asterisk CLI got freezed!!   Steve Howes
• [asterisk-dev] [svn-commits] tilghman: trunk r97651 - in /trunk: ./ apps/ channels/ configs/ include/aster...   Marcus Hunger
• [asterisk-dev] How ti let execution continue to the next priorité after callee or caller hangup   Paul Isidore
• [asterisk-dev] Trying to do a transfer   David Jones
• [asterisk-dev] Using SIP REFER with a non bridged channel - is there a workaround?   Dan Julius
• [asterisk-dev] dahdi static device files   Philipp Kempgen
• [asterisk-dev] i extension does not match on initial context - bug or not?   Philipp Kempgen
• [asterisk-dev] make config update-rc.d on Debian   Philipp Kempgen
• [asterisk-dev] Bug 14153 was closed by accident [Fwd: Re: [asterisk-users] bug 14153 and svn checkout.]   Philipp Kempgen
• [asterisk-dev] Answering Machine Detection (was: Re: Call Detection)   Philipp Kempgen
• [asterisk-dev] chan_sip SIP Authentication   Philipp Kempgen
• [asterisk-dev] chan_sip SIP Authentication   Philipp Kempgen
• [asterisk-dev] chan_sip SIP Authentication   Philipp Kempgen
• [asterisk-dev] RTP interop with Sonus: hack   Kristian Kielhofner
• [asterisk-dev] RTP interop with Sonus: hack   Kristian Kielhofner
• [asterisk-dev] RTP interop with Sonus: hack   Kristian Kielhofner
• [asterisk-dev] RTP interop with Sonus: hack   Kristian Kielhofner
• [asterisk-dev] RTP interop with Sonus: hack   Kristian Kielhofner
• [asterisk-dev] RTP interop with Sonus: hack   Kristian Kielhofner
• [asterisk-dev] RFC: Capacity Study   Kristian Kielhofner
• [asterisk-dev] RTP interop with Sonus: hack   Kristian Kielhofner
• [asterisk-dev] RTP interop with Sonus: hack   Kristian Kielhofner
• [asterisk-dev] RTP interop with Sonus: hack   Kristian Kielhofner
• [asterisk-dev] RTP interop with Sonus: hack   Kristian Kielhofner
• [asterisk-dev] RTP interop with Sonus: hack   Kristian Kielhofner
• [asterisk-dev] G.729.1 - any interest?   Kristian Kielhofner
• [asterisk-dev] Asterisk RFC2833 SDP fmtp 0-16?   Kristian Kielhofner
• [asterisk-dev] GPU Audio Codec Transcoding within Asterisk PBX   Stelios Koroneos
• [asterisk-dev] [0014275] Group does not count all channels   Marcin J. Kowalczyk
• [asterisk-dev] DTMF queuing   James Lamanna
• [asterisk-dev] DTMF queuing   James Lamanna
• [asterisk-dev] DTMF queuing   James Lamanna
• [asterisk-dev] Asterisk burns 100% cpu when pressing "#" in Voicemail   John Lange
• [asterisk-dev] Asterisk 1.4.23 Now Available!   John Lange
• [asterisk-dev] Looking for commit that fixed bug #14139, similar to #14173   Alex Villací­s Lasso
• [asterisk-dev] apps/app_page.c: fix buffer overflow and invalid memory access   Alex Villací­s Lasso
• [asterisk-dev] apps/app_page.c: fix buffer overflow and invalid memory access   Alex Villací­s Lasso
• [asterisk-dev] Looking for commit that fixed bug #14139, similar to #14173   Alex Villací­s Lasso
• [asterisk-dev] [design] Realtime changes   Tilghman Lesher
• [asterisk-dev] [design] Realtime changes   Tilghman Lesher
• [asterisk-dev] [design] Realtime changes   Tilghman Lesher
• [asterisk-dev] RFC: Capacity Study   Tilghman Lesher
• [asterisk-dev] 2008 Post Count   Tilghman Lesher
• [asterisk-dev] Building asterisk apps, features, etc. outside of Asterisk tree   Tilghman Lesher
• [asterisk-dev] SIPAdd/RemoveHeader apps   Tilghman Lesher
• [asterisk-dev] AST_LIST_LOCK/UNLOCK   Tilghman Lesher
• [asterisk-dev] SIPAdd/RemoveHeader apps   Tilghman Lesher
• [asterisk-dev] SIPAdd/RemoveHeader apps   Tilghman Lesher
• [asterisk-dev] New AGI manager command: PlaySound   Tilghman Lesher
• [asterisk-dev] [design] Realtime changes   Atis Lezdins
• [asterisk-dev] [design] Realtime changes   Atis Lezdins
• [asterisk-dev] UDPTL crash anyone?   Atis Lezdins
• [asterisk-dev] UDPTL crash anyone?   Atis Lezdins
• [asterisk-dev] [Code Review] Backport of state_interface for app_queue in Asterisk 1.4   Atis Lezdins
• [asterisk-dev] ChannelRedirect and channels on Dial   Gabriel Ortiz Lour
• [asterisk-dev] Minimum packages for building and running asterisk   Julian Lyndon-Smith
• [asterisk-dev] ast_careful_fwrite: fflush() returned error: Broken pipe   Chris Maciejewski
• [asterisk-dev] ast_careful_fwrite: fflush() returned error: Broken pipe   Chris Maciejewski
• [asterisk-dev] No emails from Mantis   Leif Madsen
• [asterisk-dev] i extension does not match on initial context - bug or not?   Leif Madsen
• [asterisk-dev] i extension does not match on initial context - bug or not?   Leif Madsen
• [asterisk-dev] i extension does not match on initial context - bug or not?   Leif Madsen
• [asterisk-dev] Request for Testing: Issue 12312, DNS SRV lookups causing re-registrations problems   Leif Madsen
• [asterisk-dev] apps/app_page.c: fix buffer overflow and invalid memory access   Leif Madsen
• [asterisk-dev] reset/remove SipAddHeader() headers   Leif Madsen
• [asterisk-dev] Access to encryption functions from dial plan   Leif Madsen
• [asterisk-dev] Welcome to David Vossel   Leif Madsen
• [asterisk-dev] Asterisk RFC2833 SDP fmtp 0-16?   Leif Madsen
• [asterisk-dev] Newbie Developer Resource?   Grey Man
• [asterisk-dev] Newbie Developer Resource?   Grey Man
• [asterisk-dev] Asterisk burns 100% cpu when pressing "#" in Voicemail   Steve Mathers
• [asterisk-dev] [Code Review] New queue CLI and Manager commands to facilitate fine-grained reloading   Mark Michelson
• [asterisk-dev] [Code Review] Backport of state_interface for app_queue in Asterisk 1.4   Mark Michelson
• [asterisk-dev] [Code Review] [bug] Asterisk does not choose same SRV record when attempting to re-register   Mark Michelson
• [asterisk-dev] [Code Review] [bug] Asterisk does not choose same SRV record when attempting to re-register   Mark Michelson
• [asterisk-dev] Bug 14153 was closed by accident [Fwd: Re: [asterisk-users] bug 14153 and svn checkout.]   Mark Michelson
• [asterisk-dev] strange spiral handling in chan_sip   Mark Michelson
• [asterisk-dev] [Code Review] Convert character pointers in sip_request structure to be offsets from the beginning of the buffer   Mark Michelson
• [asterisk-dev] [Code Review] Convert character pointers in sip_request structure to be offsets from the beginning of the buffer   Mark Michelson
• [asterisk-dev] [Code Review] Convert character pointers in sip_request structure to be offsets from the beginning of the buffer   Mark Michelson
• [asterisk-dev] app_queue, ringinuse, joinempty combination problems   Mark Michelson
• [asterisk-dev] [Code Review] Resolve some race conditions in chan_iax2 scheduler handling   Mark Michelson
• [asterisk-dev] [Code Review] Add common implementation for a scheduler context with a dedicated thread   Mark Michelson
• [asterisk-dev] [Code Review] Patch for issue 12312 (DNS SRV lookups causing re-registration problems)   Mark Michelson
• [asterisk-dev] [Code Review] Patch for issue 12312 (DNS SRV lookups causing re-registration problems)   Mark Michelson
• [asterisk-dev] [Code Review] Add common implementation for a scheduler context with a dedicated thread   Mark Michelson
• [asterisk-dev] [Code Review] added AES_ENCRYPT and AES_DECRYPT dialplan functions   Mark Michelson
• [asterisk-dev] [Code Review] added AES_ENCRYPT and AES_DECRYPT dialplan functions   Mark Michelson
• [asterisk-dev] [Code Review] Fix feature inheritance when using builtin features   Mark Michelson
• [asterisk-dev] [Code Review] Fix feature inheritance when using builtin features   Mark Michelson
• [asterisk-dev] [Code Review] Bridging API for Conference Bridge purposes   Mark Michelson
• [asterisk-dev] chan_sip SIP Authentication   Mark Michelson
• [asterisk-dev] codec preference   Adam Moffett
• [asterisk-dev] SIP channel/owner question   Miguel Molina
• [asterisk-dev] Conversion to C++ (groups/asterisk-cpp)?   Tony Mountifield
• [asterisk-dev] Conversion to C++ (groups/asterisk-cpp)?   Tony Mountifield
• [asterisk-dev] Conversion to C++ (groups/asterisk-cpp)?   Tony Mountifield
• [asterisk-dev] [Code Review] Add support in AEL for macro return values and direct assignment of them to variables and functions.   Steve Murphy
• [asterisk-dev] i extension does not match on initial context - bug or not?   Steve Murphy
• [asterisk-dev] AEL macro backward compatibility for 1.6   Steve Murphy
• [asterisk-dev] Adding Bluetooth Pairing Support to chan_mobile   Steve Murphy
• [asterisk-dev] Adding Bluetooth Pairing Support to chan_mobile   Steve Murphy
• [asterisk-dev] [Code Review] This patch implements CEL in trunk.   Steve Murphy
• [asterisk-dev] [Code Review] This patch implements CEL in trunk.   Steve Murphy
• [asterisk-dev] [design] Realtime changes   Matthew Nicholson
• [asterisk-dev] Asterisk burns 100% cpu when pressing "#" in Voicemail   Matthew Nicholson
• [asterisk-dev] Adding Bluetooth Pairing Support to chan_mobile   Matthew Nicholson
• [asterisk-dev] Adding Bluetooth Pairing Support to chan_mobile   Matthew Nicholson
• [asterisk-dev] Adding Bluetooth Pairing Support to chan_mobile   Matthew Nicholson
• [asterisk-dev] Newbie Developer Resource?   Sergey Okhapkin
• [asterisk-dev] AES encryption in IAX2   Tim Panton
• [asterisk-dev] Interesting SVN graphical representations of Asterisk project   Jeff Peeler
• [asterisk-dev] [Code Review] Keep bridge up if parking attempt fails   Jeff Peeler
• [asterisk-dev] [Code Review] Keep bridge up if parking attempt fails   Jeff Peeler
• [asterisk-dev] Multiple Server   Jeff Peeler
• [asterisk-dev] Call Detection   David Phillips
• [asterisk-dev] Newbie Developer Resource?   David Phillips
• [asterisk-dev] Newbie Developer Resource?   David Phillips
• [asterisk-dev] Newbie Developer Resource?   David Phillips
• [asterisk-dev] app_queue, ringinuse, joinempty combination problems   Wolfgang Pichler
• [asterisk-dev] app_queue, ringinuse, joinempty combination problems   Wolfgang Pichler
• [asterisk-dev] Building asterisk apps, features, etc. outside of Asterisk tree   Philip Prindeville
• [asterisk-dev] Building asterisk apps, features, etc. outside of Asterisk tree   Philip Prindeville
• [asterisk-dev] Building asterisk apps, features, etc. outside of Asterisk tree   Philip A. Prindeville
• [asterisk-dev] chan_sip SIP Authentication   Jesus Rodriguez
• [asterisk-dev] chan_sip SIP Authentication   Jesus Rodriguez
• [asterisk-dev] chan_sip SIP Authentication   Jesus Rodriguez
• [asterisk-dev] dahdi sysfs branch   Shaun Ruffell
• [asterisk-dev] Looking for commit that fixed bug #14139, similar to #14173   Rajkumar S
• [asterisk-dev] [Code Review] MFC/R2 support for chan_dahdi   Moises Silva
• [asterisk-dev] development possibilities with existing SW   Moises Silva
• [asterisk-dev] New AGI manager command: PlaySound   Moises Silva
• [asterisk-dev] i extension does not match on initial context - bug or not?   Nir Simionovich
• [asterisk-dev] i extension does not match on initial context - bug or not?   Nir Simionovich
• [asterisk-dev] i extension does not match on initial context - bug or not?   Nir Simionovich
• [asterisk-dev] OT: Paying people in faraway (Western) places.   Nir Simionovich
• [asterisk-dev] New AGI manager command: PlaySound   Nir Simionovich
• [asterisk-dev] Call Detection   Nir Simionovich
• [asterisk-dev] SMDI Function Call Redux   Bill Sofko
• [asterisk-dev] SMDI Function Call Redux   Bill Sofko
• [asterisk-dev] AST_LIST_LOCK/UNLOCK   Chandrakant Solanki
• [asterisk-dev] Multiple Server   Chandrakant Solanki
• [asterisk-dev] p2p asterisk/gtalk   Mark Spowage
• [asterisk-dev] p2p asterisk action   Mark Spowage
• [asterisk-dev] Asterisk 1.2.31, 1.4.22.1, and 1.6.0.3 released   Asterisk Development Team
• [asterisk-dev] Asterisk 1.4.23-rc4 Now Available   Asterisk Development Team
• [asterisk-dev] Asterisk 1.4.23 Now Available!   Asterisk Development Team
• [asterisk-dev] Asterisk 1.6.0.4 Release Candidate 1 Now Available   Asterisk Development Team
• [asterisk-dev] Asterisk 1.2.31.1, 1.4.22.2, 1.4.23.1, and 1.6.0.5 released   Asterisk Development Team
• [asterisk-dev] Asterisk 1.6.1 Release Candidate 1 Now Available   Asterisk Development Team
• [asterisk-dev] AST-2009-001: Information leak in IAX2 authentication   Asterisk Security Team
• [asterisk-dev] Events generated by followme   Mirko Tochtermann
• [asterisk-dev] Apple TV and Asterisk - who was it?   John Todd
• [asterisk-dev] RTP interop with Sonus: hack   John Todd
• [asterisk-dev] Interesting SVN graphical representations of Asterisk project   John Todd
• [asterisk-dev] apps/app_page.c: fix buffer overflow and invalid memory access   John Todd
• [asterisk-dev] reset/remove SipAddHeader() headers   John Todd
• [asterisk-dev] G.729.1 - any interest?   John Todd
• [asterisk-dev] [0014275] Group does not count all channels   John Todd
• [asterisk-dev] Newbie Developer Resource?   John Todd
• [asterisk-dev] DTMF queuing   John Todd
• [asterisk-dev] SIPAdd/RemoveHeader apps   Chris Tooley
• [asterisk-dev] SIPAdd/RemoveHeader apps   Chris Tooley
• [asterisk-dev] Multiple Server   Chris Tooley
• [asterisk-dev] Dean Collins added you as a business connection on Plaxo   Steve Totaro
• [asterisk-dev] GPU Audio Codec Transcoding within Asterisk PBX   Steve Underwood
• [asterisk-dev] GPU Audio Codec Transcoding within Asterisk PBX   Steve Underwood
• [asterisk-dev] UDPTL crash anyone?   Steve Underwood
• [asterisk-dev] [Code Review] Queue SIP requests/responses that cannot be immediately processed.   Fernando Urzedo
• [asterisk-dev] [Code Review] Add support in AEL for macro return values and direct assignment of them to variables and functions.   Watkins, Bradley
• [asterisk-dev] Events generated by followme   BJ Weschke
• [asterisk-dev] SRTP and Dialplan control   Terry Wilson
• [asterisk-dev] [Code Review] Fix feature inheritance when using builtin features   Terry Wilson
• [asterisk-dev] [Code Review] Fix feature inheritance when using builtin features   Terry Wilson
• [asterisk-dev] [Code Review] Fix feature inheritance when using builtin features   Terry Wilson
• [asterisk-dev] [Code Review] Bridging API for Conference Bridge purposes   Terry Wilson
• [asterisk-dev] External MWI & IMAP   Jay R. Worthington
• [asterisk-dev] Need to start in development   bilal ghayyad
• [asterisk-dev] New AGI manager command: PlaySound   nik600
• [asterisk-dev] New AGI manager command: PlaySound   nik600
• [asterisk-dev] New AGI manager command: PlaySound   nik600
• [asterisk-dev] chan_sip SIP Authentication   asterisk at ntplx.net
• [asterisk-dev] chan_sip SIP Authentication   asterisk at ntplx.net

Last message date: Sat Jan 31 16:56:58 CST 2009
Archived on: Wed Jun 3 18:20:43 CDT 2009

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