April 2009 Archives by date

• Messages sorted by: [ thread ] [ subject ] [ author ]
• More info on this list...

Starting: Wed Apr 1 00:36:51 CDT 2009
Ending: Thu Apr 30 19:35:19 CDT 2009
Messages: 543

• [asterisk-dev] [Code Review] SIP Re-invite Glare and 491 Madness...   Olle E. Johansson
• [asterisk-dev] Two version of asterisk on the same server   Serge Berney
• [asterisk-dev] channel formats and ast_set_read/write_format   Klaus Darilion
• [asterisk-dev] function for Multicust on SIP   anupam bairagi
• [asterisk-dev] function for Multicust on SIP   Russell Bryant
• [asterisk-dev] [Code Review] Add technology independent static control   Russell Bryant
• [asterisk-dev] [Code Review] Modularized RTP stack support   Joshua Colp
• [asterisk-dev] mmichelson: trunk r185704 - in /trunk: CHANGES main/manager.c   Mark Michelson
• [asterisk-dev] [Code Review] Add technology independent static control   Olle E. Johansson
• [asterisk-dev] [Code Review] Add technology independent static control   Mark Michelson
• [asterisk-dev] [Code Review] Add technology independent static control   Russell Bryant
• [asterisk-dev] [Code Review] Add technology independent static control   Russell Bryant
• [asterisk-dev] [Code Review] Add technology independent static control   Jared Smith
• [asterisk-dev] [Code Review] Add technology independent static control   Russell Bryant
• [asterisk-dev] [Code Review] Add technology independent static control   Russell Bryant
• [asterisk-dev] [Code Review] Modularized RTP stack support   Joshua Colp
• [asterisk-dev] [Code Review] Modularized RTP stack support   Russell Bryant
• [asterisk-dev] [Code Review] Modularized RTP stack support   Joshua Colp
• [asterisk-dev] [Code Review] Modularized RTP stack support   Joshua Colp
• [asterisk-dev] [Code Review] SIP Re-invite Glare and 491 Madness...   David Vossel
• [asterisk-dev] [Code Review] Add technology independent static control   John Todd
• [asterisk-dev] SIP and using less-prior codec   Anthony Plack
• [asterisk-dev] [Code Review] SIP Re-invite Glare and 491 Madness...   Mark Michelson
• [asterisk-dev] [Code Review] SIP Re-invite Glare and 491 Madness...   Mark Michelson
• [asterisk-dev] [Code Review] Add technology independent static control   Russell Bryant
• [asterisk-dev] [Code Review] Add technology independent static control   Luigi Rizzo
• [asterisk-dev] SIP and using less-prior codec   Klaus Darilion
• [asterisk-dev] function for Multicust on SIP   Leif Madsen
• [asterisk-dev] [Code Review] SIP Re-invite Glare and 491 Madness...   Olle E. Johansson
• [asterisk-dev] [Code Review] Create API for doing variable substitution with dynamic buffers   Russell Bryant
• [asterisk-dev] [Code Review] Add technology independent static control   Michiel van Baak
• [asterisk-dev] [Code Review] Detect if we need bison to regenerate main/ast_expr2.c during ./configure   Michiel van Baak
• [asterisk-dev] Did I miss an announcement   Anthony Plack
• [asterisk-dev] Did I miss an announcement   Michiel van Baak
• [asterisk-dev] Asterisk 1.6.0.7 Now Available   Asterisk Development Team
• [asterisk-dev] Asterisk-Addons 1.6.2.0-beta1 Now Available   Asterisk Development Team
• [asterisk-dev] [Code Review] Create API for doing variable substitution with dynamic buffers   dimas at dataart.com
• [asterisk-dev] Queue strategy with Ringall and multiple agents answering simultenously.   Atis Lezdins
• [asterisk-dev] Asterisk 1.2.32, 1.4.24.1, and 1.6.0.8 Now Available   Asterisk Development Team
• [asterisk-dev] AST-2009-003: SIP responses expose valid usernames   Asterisk Security Team
• [asterisk-dev] [Code Review] Add technology independent static control   Kaloyan Kovachev
• [asterisk-dev] Is This Thing ON?   James Finstrom
• [asterisk-dev] Is This Thing ON?   Martin
• [asterisk-dev] Is This Thing ON?   Sergey Okhapkin
• [asterisk-dev] asterisk-dev Digest, Vol 47, Issue 17   bilal ghayyad
• [asterisk-dev] Is This Thing ON?   Miguel Molina
• [asterisk-dev] asterisk-dev Digest, Vol 47, Issue 17   Russell Bryant
• [asterisk-dev] asterisk-dev Digest, Vol 47, Issue 17   Fredrik Hansson
• [asterisk-dev] Asterisk compilation & install on different folder - segmentation fault   Serge Berney
• [asterisk-dev] Asterisk compilation & install on different folder - segmentation fault   Martin
• [asterisk-dev] Asterisk compilation & install on differentfolder - segmentation fault   Serge Berney
• [asterisk-dev] Asterisk RPID   Gregory Boehnlein
• [asterisk-dev] Asterisk RPID   Klaus Darilion
• [asterisk-dev] Asterisk 1.6.1.0-rc4 Now Available   Asterisk Development Team
• [asterisk-dev] Asterisk 1.6.0.9 Now Available   Asterisk Development Team
• [asterisk-dev] Asterisk compilation & install on differentfolder - segmentation fault   Martin
• [asterisk-dev] h extension not considered within a macro for Asterisk 1.6.0.6 and later   Emrah KAVUN
• [asterisk-dev] h extension not considered within a macro for Asterisk 1.6.0.6 and later   Alec Davis
• [asterisk-dev] h extension not considered within a macro for Asterisk 1.6.0.6 and later   Tzafrir Cohen
• [asterisk-dev] h extension not considered within a macro forAsterisk 1.6.0.6 and later   Alec Davis
• [asterisk-dev] Asterisk compilation & install ondifferentfolder - segmentation fault   Serge Berney
• [asterisk-dev] Asterisk application "RealTime" not found on 1.6 sources   Serge Berney
• [asterisk-dev] Asterisk 1.6.0.9 Now Available   Klaus Darilion
• [asterisk-dev] RTP trunking - 58% savings on media bandwidth?   mctiew at yahoo.com
• [asterisk-dev] Asterisk application "RealTime" not found on 1.6 sources   Atis Lezdins
• [asterisk-dev] RTP trunking - 58% savings on media bandwidth?   Will
• [asterisk-dev] RTP trunking - 58% savings on media bandwidth?   Steve Underwood
• [asterisk-dev] Grandstream blind transfer issue   Max Alex
• [asterisk-dev] h extension not considered within a macro for Asterisk 1.6.0.6 and later   Atis Lezdins
• [asterisk-dev] Asterisk application "RealTime" not found on 1.6 sources   Philipp Kempgen
• [asterisk-dev] h extension not considered within a macro for Asterisk 1.6.0.6 and later   Tilghman Lesher
• [asterisk-dev] About TRANSFER event   Miguel Molina
• [asterisk-dev] Asterisk 1.6.0.9 Now Available   Mark Michelson
• [asterisk-dev] h extension not considered within a macro for Asterisk 1.6.0.6 and later   Emrah KAVUN
• [asterisk-dev] h extension not considered within a macro for Asterisk 1.6.0.6 and later   Alec Davis
• [asterisk-dev] h extension not considered within a macro for Asterisk 1.6.0.6 and later   Tilghman Lesher
• [asterisk-dev] h extension not considered within a macro for Asterisk 1.6.0.6 and later   Alec Davis
• [asterisk-dev] Asterisk 1.6.0.9 Now Available   Klaus Darilion
• [asterisk-dev] [asterisk-users] Grandstream blind transfer issue   Klaus Darilion
• [asterisk-dev] h extension not considered within a macro for Asterisk 1.6.0.6 and later   Tilghman Lesher
• [asterisk-dev] Asterisk 1.6.0.9 Now Available   Russell Bryant
• [asterisk-dev] Asterisk 1.6.0.9 Now Available   David Hansen
• [asterisk-dev] Asterisk 1.6.0.9 Now Available   Mark Michelson
• [asterisk-dev] h extension not considered within a macro for Asterisk 1.6.0.6 and later   Alec Davis
• [asterisk-dev] h extension not considered within a macro for Asterisk 1.6.0.6 and later   asterisk at ntplx.net
• [asterisk-dev] RTP trunking - 58% savings on media bandwidth?   Ming-Ching Tiew
• [asterisk-dev] h extension not considered within a macro for Asterisk 1.6.0.6 and later   Tilghman Lesher
• [asterisk-dev] =h extension not considered within a macro for_Asterisk...   Alec Davis
• [asterisk-dev] Segfault Asterisk 1.2.31.1 on Debian Lenny AMD64   Stefan Schmidt
• [asterisk-dev] Segfault Asterisk 1.2.31.1 on Debian Lenny AMD64   Stefan Schmidt
• [asterisk-dev] RTP trunking - 58% savings on media bandwidth?   Will
• [asterisk-dev] [asterisk-users] Grandstream blind transfer issue   Max Alex
• [asterisk-dev] the question about fast agi   -Aiphie
• [asterisk-dev] the question about fast agi   Steve Edwards
• [asterisk-dev] [asterisk-users] Grandstream blind transfer issue   Klaus Darilion
• [asterisk-dev] h extension not considered within a macro for Asterisk 1.6.0.6 and later   Klaus Darilion
• [asterisk-dev] call files batching - performance question   John Regal
• [asterisk-dev] call files batching - performance   John Regal
• [asterisk-dev] Centos 5.2 and configure for 1.4   Olle E. Johansson
• [asterisk-dev] Centos 5.2 and configure for 1.4   Chris Tooley
• [asterisk-dev] Centos 5.2 and configure for 1.4   Olle E. Johansson
• [asterisk-dev] Centos 5.2 and configure for 1.4   Jason Parker
• [asterisk-dev] Centos 5.2 and configure for 1.4   Eliel Sardañons
• [asterisk-dev] Centos 5.2 and configure for 1.4   Olle E. Johansson
• [asterisk-dev] Carrier Switch invites every one to Join us.   Carrier Switch
• [asterisk-dev] call files batching - performance   Martin
• [asterisk-dev] [Code Review] Backport of state_interface for app_queue in Asterisk 1.4   Matt Riddell
• [asterisk-dev] [Code Review] Backport of state_interface for app_queue in Asterisk 1.4   Russell Bryant
• [asterisk-dev] transfer problem with Mitel PBX 3300   scott_lvsha
• [asterisk-dev] [Code Review] Makefile now specifies whether or not DAHDI hardware was found   Russell Bryant
• [asterisk-dev] transfer problem with Mitel PBX 3300   Mark Michelson
• [asterisk-dev] Reviewboard Usage Guidelines   Russell Bryant
• [asterisk-dev] tilghman: branch 1.4 r187428 - in /branches/1.4: include/asterisk/ main/   Russell Bryant
• [asterisk-dev] [Code Review] Makefile now specifies whether or not DAHDI hardware was found   Sean Bright
• [asterisk-dev] tilghman: branch 1.4 r187428 - in /branches/1.4: include/asterisk/ main/   Tilghman Lesher
• [asterisk-dev] Reviewboard Usage Guidelines   Michiel van Baak
• [asterisk-dev] Reviewboard Usage Guidelines   Jeff Peeler
• [asterisk-dev] Reviewboard Usage Guidelines   Tzafrir Cohen
• [asterisk-dev] Reviewboard Usage Guidelines   Kevin P. Fleming
• [asterisk-dev] Reviewboard Usage Guidelines   Tilghman Lesher
• [asterisk-dev] Reviewboard Usage Guidelines   Michiel van Baak
• [asterisk-dev] Reviewboard Usage Guidelines   Olle E. Johansson
• [asterisk-dev] tilghman: branch 1.4 r187428 - in /branches/1.4: include/asterisk/ main/   Olle E. Johansson
• [asterisk-dev] tilghman: branch 1.4 r187428 - in /branches/1.4: include/asterisk/ main/   Tilghman Lesher
• [asterisk-dev] tilghman: branch 1.4 r187428 - in /branches/1.4: include/asterisk/ main/   Olle E. Johansson
• [asterisk-dev] tilghman: branch 1.4 r187428 - in /branches/1.4: include/asterisk/ main/   Tilghman Lesher
• [asterisk-dev] tilghman: branch 1.4 r187428 - in /branches/1.4: include/asterisk/ main/   Olle E. Johansson
• [asterisk-dev] Reviewboard Usage Guidelines   Russell Bryant
• [asterisk-dev] Reviewboard Usage Guidelines   Russell Bryant
• [asterisk-dev] [Code Review] Check dependecies inside the XML documentation.   Michiel van Baak
• [asterisk-dev] Reviewboard Usage Guidelines   Russell Bryant
• [asterisk-dev] tilghman: branch 1.4 r187428 - in /branches/1.4: include/asterisk/ main/   Russell Bryant
• [asterisk-dev] tilghman: trunk r187483 - in /trunk: ./ include/asterisk/ main/   Russell Bryant
• [asterisk-dev] [svn-commits] mmichelson: trunk r187560 - in /trunk: channels/ configs/   Olle E. Johansson
• [asterisk-dev] Reviewboard Usage Guidelines   Olle E. Johansson
• [asterisk-dev] [svn-commits] mmichelson: trunk r187560 - in /trunk: channels/ configs/   Klaus Darilion
• [asterisk-dev] [svn-commits] mmichelson: trunk r187560 - in /trunk: channels/ configs/   Mark Michelson
• [asterisk-dev] [svn-commits] mmichelson: trunk r187560 - in /trunk: channels/ configs/   Mark Michelson
• [asterisk-dev] [Code Review] Add CCBS/CCNR and enhanced COLP features to chan_misdn.   rmudgett at digium.com
• [asterisk-dev] [Code Review] Add CCBS/CCNR and enhanced COLP features to chan_misdn.   Russell Bryant
• [asterisk-dev] [Code Review] Add CCBS/CCNR and enhanced COLP features to chan_misdn.   rmudgett at digium.com
• [asterisk-dev] [Code Review] Add CCBS/CCNR and enhanced COLP features to chan_misdn.   rmudgett at digium.com
• [asterisk-dev] BUG: 0014846: undefined symbols - modules can't be loaded app_voicemail res_smdi   Anthony Plack
• [asterisk-dev] BUG: 0014846: undefined symbols - modules can't be loaded app_voicemail res_smdi   Tilghman Lesher
• [asterisk-dev] [Code Review] Convert ast_channel over to astobj2   Mark Michelson
• [asterisk-dev] [Code Review] A chan_local with no busy loop deadlock avoidance (and no deadlocks, too)   Mark Michelson
• [asterisk-dev] [Code Review] A chan_local with no busy loop deadlock avoidance (and no deadlocks, too)   Mark Michelson
• [asterisk-dev] [Code Review] A chan_local with no busy loop deadlock avoidance (and no deadlocks, too)   Mark Michelson
• [asterisk-dev] Something happenned   Venefax
• [asterisk-dev] Queue interface vars not available in membermacro   Grigoriy Puzankin
• [asterisk-dev] MWI Indication while copying file   Chandrakant Solanki
• [asterisk-dev] [Code Review] A chan_local with no busy loop deadlock avoidance (and no deadlocks, too)   Russell Bryant
• [asterisk-dev] [Code Review] Convert ast_channel over to astobj2   Russell Bryant
• [asterisk-dev] Queue interface vars not available in membermacro   Mark Michelson
• [asterisk-dev] Queue interface vars not available in membermacro   Mark Michelson
• [asterisk-dev] [Code Review] Add CCBS/CCNR and enhanced COLP features to chan_misdn.   Kevin Fleming
• [asterisk-dev] [Code Review] A chan_local with no busy loop deadlock avoidance (and no deadlocks, too)   Mark Michelson
• [asterisk-dev] [Code Review] A chan_local with no busy loop deadlock avoidance (and no deadlocks, too)   Russell Bryant
• [asterisk-dev] channel.c locking and other   Mark Michelson
• [asterisk-dev] channel.c locking and other   Russell Bryant
• [asterisk-dev] [Code Review] National prefix inserted even when caller ID not available   David Vossel
• [asterisk-dev] [Code Review] National prefix inserted even when caller ID not available   Russell Bryant
• [asterisk-dev] channel.c locking and other   Mark Michelson
• [asterisk-dev] Compiling Asterisk with LUA   Lahav Savir
• [asterisk-dev] [Code Review] National prefix inserted even when caller ID not available   Jeff Peeler
• [asterisk-dev] Compiling Asterisk with LUA   Russell Bryant
• [asterisk-dev] [Code Review] Makefile now specifies whether or not DAHDI hardware was found   Tzafrir Cohen
• [asterisk-dev] [Code Review] Makefile now specifies whether or not DAHDI hardware was found   Tzafrir Cohen
• [asterisk-dev] [Code Review] Makefile now specifies whether or not DAHDI hardware was found   Tzafrir Cohen
• [asterisk-dev] [Code Review] National prefix inserted even when caller ID not available   dimas at dataart.com
• [asterisk-dev] [Code Review] Makefile now specifies whether or not DAHDI hardware was found   Tzafrir Cohen
• [asterisk-dev] [svn-commits] file: branch 1.6.2 r188070 - in /branches/1.6.2: ./ channels/chan_sip.c   Olle E. Johansson
• [asterisk-dev] Compiling Asterisk with LUA   Lahav Savir
• [asterisk-dev] Max Conference slaves for a master in chan_zap/chan_dahdi.c   amit.sharma at spectross.com
• [asterisk-dev] Compiling Asterisk with LUA   Tzafrir Cohen
• [asterisk-dev] Compiling Asterisk with LUA   Lahav Savir
• [asterisk-dev] Queue interface vars not available in membermacro   Grigoriy Puzankin
• [asterisk-dev] [svn-commits] file: branch 1.6.2 r188070 - in /branches/1.6.2: ./ channels/chan_sip.c   Joshua Colp
• [asterisk-dev] [svn-commits] file: branch 1.6.2 r188070 - in /branches/1.6.2: ./ channels/chan_sip.c   Olle E. Johansson
• [asterisk-dev] Compiling Asterisk with LUA   Tilghman Lesher
• [asterisk-dev] res_smdi.so required by chan_dahdi.so ?   Chris Maciejewski
• [asterisk-dev] Vacation reply   awerflli at hotmail.com
• [asterisk-dev] Vacation reply   awerflli at hotmail.com
• [asterisk-dev] SIP problems similar to bug #0013383   Dan Julius
• [asterisk-dev] ISDN Overlap dial mechanism   Paulo Garcia
• [asterisk-dev] Defining new section type in sip.conf - question about syntax   Olle E. Johansson
• [asterisk-dev] [Code Review] Allow to set the wrapuptime per queue member, and disable the wrapuptime for the next call or increase its value as requested.   Eliel Sardañons
• [asterisk-dev] ISDN Overlap dial mechanism   Moises Silva
• [asterisk-dev] ISDN Overlap dial mechanism   Kevin P. Fleming
• [asterisk-dev] Defining new section type in sip.conf - question about syntax   Kevin P. Fleming
• [asterisk-dev] ISDN Overlap dial mechanism   Martin
• [asterisk-dev] Defining new section type in sip.conf - question about syntax   Russell Bryant
• [asterisk-dev] ISDN Overlap dial mechanism   Moises Silva
• [asterisk-dev] [Code Review] Allow to set the wrapuptime per queue member, and disable the wrapuptime for the next call or increase its value as requested.   Mark Michelson
• [asterisk-dev] [Code Review] HTTP Digest authentication   Tilghman Lesher
• [asterisk-dev] [Code Review] Allow to set the wrapuptime per queue member, and disable the wrapuptime for the next call or increase its value as requested.   Mark Michelson
• [asterisk-dev] [Code Review] HTTP Digest authentication   Tilghman Lesher
• [asterisk-dev] [Code Review] HTTP Digest authentication   Tilghman Lesher
• [asterisk-dev] [Code Review] Backport of state_interface for app_queue in Asterisk 1.4   Matt Riddell
• [asterisk-dev] Defining new section type in sip.conf - question about syntax   Olle E. Johansson
• [asterisk-dev] How to send Custom NOTIFY Packets?   ashish mahamuni
• [asterisk-dev] Defining new section type in sip.conf - question about syntax   Tilghman Lesher
• [asterisk-dev] Defining new section type in sip.conf - question about syntax   Jared Smith
• [asterisk-dev] Defining new section type in sip.conf - question about syntax   Mark Michelson
• [asterisk-dev] Last chance for comments on review 194   Mark Michelson
• [asterisk-dev] Defining new section type in sip.conf - question about syntax   Olle E. Johansson
• [asterisk-dev] How to send Custom NOTIFY Packets?   ashish mahamuni
• [asterisk-dev] [Code Review] Allow to set the wrapuptime per queue member, and disable the wrapuptime for the next call or increase its value as requested.   Eliel Sardañons
• [asterisk-dev] [Code Review] HTTP Digest authentication   Mark Michelson
• [asterisk-dev] Pedantic   David Hansen
• [asterisk-dev] Pedantic   Steve Howes
• [asterisk-dev] Pedantic   Philipp Kempgen
• [asterisk-dev] Pedantic   Mark Michelson
• [asterisk-dev] [Code Review] HTTP Digest authentication   Tilghman Lesher
• [asterisk-dev] [Code Review] HTTP Digest authentication   Tilghman Lesher
• [asterisk-dev] [Code Review] HTTP Digest authentication   Tilghman Lesher
• [asterisk-dev] [Code Review] HTTP Digest authentication   Tilghman Lesher
• [asterisk-dev] Defining new section type in sip.conf - question about syntax   Leif Madsen
• [asterisk-dev] Defining new section type in sip.conf - question about syntax   Michiel van Baak
• [asterisk-dev] Defining new section type in sip.conf - question about syntax   Tilghman Lesher
• [asterisk-dev] Defining new section type in sip.conf - question about syntax   Tzafrir Cohen
• [asterisk-dev] Defining new section type in sip.conf - question about syntax   Jared Smith
• [asterisk-dev] ChanSpy() feature requests   Alistair Cunningham
• [asterisk-dev] ChanSpy() feature requests   Alistair Cunningham
• [asterisk-dev] ChanSpy() feature requests   Mark Michelson
• [asterisk-dev] [Code Review] SIP state notify reorganization   David Vossel
• [asterisk-dev] Defining new section type in sip.conf - question about syntax   Tilghman Lesher
• [asterisk-dev] Defining new section type in sip.conf - question about syntax   Kevin P. Fleming
• [asterisk-dev] Defining new section type in sip.conf - question about syntax   Tzafrir Cohen
• [asterisk-dev] Pedantic   Olle E. Johansson
• [asterisk-dev] Defining new section type in sip.conf - question about syntax   Olle E. Johansson
• [asterisk-dev] Defining new section type in sip.conf - question about syntax   John Todd
• [asterisk-dev] Hi Need Help on using module to access the sip request for validation... Thanks   fofo don
• [asterisk-dev] Defining new section type in sip.conf - question about syntax   Jared Smith
• [asterisk-dev] pthread_getspecific() in app_voicemail   Mark Sirota
• [asterisk-dev] Pedantic   Klaus Darilion
• [asterisk-dev] pthread_getspecific() in app_voicemail   Tilghman Lesher
• [asterisk-dev] pthread_getspecific() in app_voicemail   Kevin Stewart
• [asterisk-dev] pthread_getspecific() in app_voicemail   Mark Sirota
• [asterisk-dev] app_meetme.c bug fix   lvsha
• [asterisk-dev] ChanSpy() feature requests   Clod Patry
• [asterisk-dev] Asterisk Lua Reference   Lahav Savir
• [asterisk-dev] Defining new section type in sip.conf - question about syntax   Kaloyan Kovachev
• [asterisk-dev] Defining new section type in sip.conf - question about syntax   Chris Lee
• [asterisk-dev] Defining new section type in sip.conf - question about syntax   David Hansen
• [asterisk-dev] pthread_getspecific() in app_voicemail   Russell Bryant
• [asterisk-dev] pthread_getspecific() in app_voicemail   Mark Sirota
• [asterisk-dev] app_meetme.c bug fix   Mark Michelson
• [asterisk-dev] [Code Review] A chan_local with no busy loop deadlock avoidance (and no deadlocks, too)   Mark Michelson
• [asterisk-dev] 2 callers id needed on the originate command   Stephen Coles
• [asterisk-dev] bug in app_voicemail.c   Jim Capp
• [asterisk-dev] 2 callers id needed on the originate command   Donny Kavanagh
• [asterisk-dev] bug in app_voicemail.c   Kevin P. Fleming
• [asterisk-dev] 2 callers id needed on the originate command   Mark Michelson
• [asterisk-dev] pthread_getspecific() in app_voicemail   Mark Sirota
• [asterisk-dev] app_meetme.c bug fix   Russell Bryant
• [asterisk-dev] pthread_getspecific() in app_voicemail   Russell Bryant
• [asterisk-dev] app_voicemail.c   Jim Capp
• [asterisk-dev] Defining new section type in sip.conf - question about syntax   Leif Madsen
• [asterisk-dev] app_voicemail.c   Kevin P. Fleming
• [asterisk-dev] ChanSpy() feature requests   David Backeberg
• [asterisk-dev] pthread_getspecific() in app_voicemail   Mark Sirota
• [asterisk-dev] [Code Review] SIP state notify reorganization   Russell Bryant
• [asterisk-dev] [Code Review] HTTP Digest authentication   Russell Bryant
• [asterisk-dev] sending AT commands through the SIP channel to the end device?!   Tamer Higazi
• [asterisk-dev] Asterisk European Developers Meet-Up May 2, 3 Rostock DE   John Todd
• [asterisk-dev] [Code Review] Autologoff when using AgentLogin   David Vossel
• [asterisk-dev] sending AT commands through the SIP channel to theend device?!   Gunnar Hellstrom
• [asterisk-dev] svn login fail   lvsha
• [asterisk-dev] sending AT commands through the SIP channel to theend device?!   Tamer Higazi
• [asterisk-dev] Asterisk European Developers Meet-Up May 2, 3 Rostock DE   Nir Simionovich
• [asterisk-dev] Compiling Asterisk with LUA   Wolfgang Pichler
• [asterisk-dev] [Code Review] Autologoff when using AgentLogin   Russell Bryant
• [asterisk-dev] [Code Review] New application JabberReceive, implement SendText in chan_gtalk and chan_jingle   Philippe Sultan
• [asterisk-dev] [Code Review] Various fixes for extenpatternmatchnew   Russell Bryant
• [asterisk-dev] [Code Review] Various fixes for extenpatternmatchnew   Mark Michelson
• [asterisk-dev] [Code Review] Various fixes for extenpatternmatchnew   Mark Michelson
• [asterisk-dev] svn login fail   Tilghman Lesher
• [asterisk-dev] [Code Review] New application JabberReceive, implement SendText in chan_gtalk and chan_jingle   Philippe Sultan
• [asterisk-dev] [Code Review] Autologoff when using AgentLogin   David Vossel
• [asterisk-dev] Asterisk 1.6.0.9 core dumps   Andre Courchesne - Prival
• [asterisk-dev] Asterisk 1.6.0.9 core dumps   Andre Courchesne - Prival
• [asterisk-dev] Asterisk 1.6.0.9 core dumps   Russell Bryant
• [asterisk-dev] Asterisk 1.6.0.9 core dumps   Andre Courchesne - Prival
• [asterisk-dev] Asterisk 1.6.0.9 core dumps   Andre Courchesne - Prival
• [asterisk-dev] Asterisk 1.6.0.9 core dumps   Mark Michelson
• [asterisk-dev] [Code Review] Autologoff when using AgentLogin   Russell Bryant
• [asterisk-dev] Asterisk 1.6.0.9 core dumps   Andre Courchesne - Prival
• [asterisk-dev] Compiling Asterisk with LUA   Tilghman Lesher
• [asterisk-dev] opening 2 and more channels on 1 SIP account   Tamer Higazi
• [asterisk-dev] Asterisk 1.6.0.9 core dumps   Andre Courchesne - Prival
• [asterisk-dev] [Code Review] Makefile now specifies whether or not DAHDI hardware was found   Russell Bryant
• [asterisk-dev] LibPRI: How to add an IE to an outgoing call?   Alexandre Cavalcante Alencar
• [asterisk-dev] opening 2 and more channels on 1 SIP account   Kevin P. Fleming
• [asterisk-dev] [Code Review] HTTP Digest authentication   Tilghman Lesher
• [asterisk-dev] [Code Review] HTTP Digest authentication   Tilghman Lesher
• [asterisk-dev] [Code Review] HTTP Digest authentication   Tilghman Lesher
• [asterisk-dev] [Code Review] Makefile now specifies whether or not DAHDI hardware was found   Oron Peled
• [asterisk-dev] IM support for asterisk   Pasan Meemaduma
• [asterisk-dev] opening 2 and more channels on 1 SIP account   Tamer Higazi
• [asterisk-dev] MATH function up to 'signed long'   Chris Maciejewski
• [asterisk-dev] [Code Review] Makefile now specifies whether or not DAHDI hardware was found   Tzafrir Cohen
• [asterisk-dev] pthread_rwlock_timed* not available on OpenBSD (bug #14705)   Michiel van Baak
• [asterisk-dev] opening 2 and more channels on 1 SIP account   Jeff LaCoursiere
• [asterisk-dev] pthread_rwlock_timed* not available on OpenBSD (bug #14705)   Tilghman Lesher
• [asterisk-dev] LibPRI: How to add an IE to an outgoing call?   michael ricordeau
• [asterisk-dev] LibPRI: How to add an IE to an outgoing call?   Alexandre Cavalcante Alencar
• [asterisk-dev] Asterisk European Developers Meet-Up May 2, 3 Rostock DE   Philipp Kempgen
• [asterisk-dev] LibPRI: How to add an IE to an outgoing call?   Guilherme Loch Waltrick Góes
• [asterisk-dev] Asterisk European Developers Meet-Up May 2, 3 Rostock DE   Hans Witvliet
• [asterisk-dev] Asterisk European Developers Meet-Up May 2, 3 Rostock DE   Philipp Kempgen
• [asterisk-dev] LibPRI: How to add an IE to an outgoing call?   Alexandre Cavalcante Alencar
• [asterisk-dev] LibPRI: How to add an IE to an outgoing call?   michael ricordeau
• [asterisk-dev] LibPRI: How to add an IE to an outgoing call?   Alexandre Cavalcante Alencar
• [asterisk-dev] [Code Review] Allow to set the wrapuptime per queue member, and disable the wrapuptime for the next call or increase its value as requested.   Eliel Sardañons
• [asterisk-dev] unsubscribe   Pablo Digonzelli
• [asterisk-dev] unsubscribe   Steve Howes
• [asterisk-dev] [Code Review] Add CCBS/CCNR and enhanced COLP features to chan_misdn.   Russell Bryant
• [asterisk-dev] [Code Review] Integrate DAHDIScan into chanspy   David Vossel
• [asterisk-dev] [Code Review] Integrate DAHDIScan into chanspy   Russell Bryant
• [asterisk-dev] streamfile reference counting debug   Paul Albrecht
• [asterisk-dev] streamfile reference counting debug   Mark Michelson
• [asterisk-dev] streamfile reference counting debug   Paul Albrecht
• [asterisk-dev] streamfile reference counting debug   Mark Michelson
• [asterisk-dev] Doxygen Janitor Cleanup Project   James Hunt
• [asterisk-dev] Doxygen Janitor Cleanup Project   Olle E. Johansson
• [asterisk-dev] [Code Review] Convert ast_channel over to astobj2   Russell Bryant
• [asterisk-dev] talker detection meetme   David Hansen
• [asterisk-dev] [Code Review] Convert ast_channel over to astobj2   Russell Bryant
• [asterisk-dev] MeetMe, setting MEETMEUNIQUEID   Grigoriy Puzankin
• [asterisk-dev] [Code Review]   David Vossel
• [asterisk-dev] [Code Review] Segfault when switching upd to tcp in sip.conf after reload   David Vossel
• [asterisk-dev] [Code Review] Segfault when switching upd to tcp in sip.conf after reload   Russell Bryant
• [asterisk-dev] [Code Review] New application JabberReceive, implement SendText in chan_gtalk and chan_jingle   Tilghman Lesher
• [asterisk-dev] [Code Review] Convert ast_channel over to astobj2   Tilghman Lesher
• [asterisk-dev] streamfile reference counting debug   Paul Albrecht
• [asterisk-dev] [Code Review] Convert ast_channel over to astobj2   Sean Bright
• [asterisk-dev] Asterisk 1.6.1.0-rc5 Now Available   Asterisk Development Team
• [asterisk-dev] Issue 14863. Parked call announcement   Mat Murdock
• [asterisk-dev] [Code Review] Convert ast_channel over to astobj2   Mark Michelson
• [asterisk-dev] [Code Review] Convert ast_channel over to astobj2   Mark Michelson
• [asterisk-dev] [Code Review] Convert ast_channel over to astobj2   Tilghman Lesher
• [asterisk-dev] Compiling Asterisk with LUA   Watkins, Bradley
• [asterisk-dev] Symmetric RTP behaviour and 'nat' peer option.   Alex Balashov
• [asterisk-dev] Symmetric RTP behaviour and 'nat' peer option.   Alex Balashov
• [asterisk-dev] Symmetric RTP behaviour and 'nat' peer option.   Alex Balashov
• [asterisk-dev] [Code Review] Convert ast_channel over to astobj2   Sean Bright
• [asterisk-dev] Symmetric RTP behaviour and 'nat' peer option.   Venefax
• [asterisk-dev] [Code Review] Add stack-like functions for manipulating comma-separated lists in the dialplan   Terry Wilson
• [asterisk-dev] Symmetric RTP behaviour and 'nat' peer option.   Alex Balashov
• [asterisk-dev] [Code Review] Add stack-like functions for manipulating comma-separated lists in the dialplan   Tilghman Lesher
• [asterisk-dev] How to get to 10.000 open calls   Venefax
• [asterisk-dev] How to get to 10.000 open calls   Alex Balashov
• [asterisk-dev] How to get to 10.000 open calls   Alex Balashov
• [asterisk-dev] [Code Review] Add stack-like functions for manipulating comma-separated lists in the dialplan   Mark Michelson
• [asterisk-dev] How to get to 10.000 open calls   Venefax
• [asterisk-dev] How to get to 10.000 open calls   Tzafrir Cohen
• [asterisk-dev] How to get to 10.000 open calls   Mark Michelson
• [asterisk-dev] How to get to 10.000 open calls   Venefax
• [asterisk-dev] How to get to 10.000 open calls   Alex Balashov
• [asterisk-dev] How to get to 10.000 open calls   Alex Balashov
• [asterisk-dev] CDR feature not working properly for "failed call attempt"   Vikas
• [asterisk-dev] How to get to 10.000 open calls   alexander.harrowell at stlpartners.com
• [asterisk-dev] How to get to 10.000 open calls   Venefax
• [asterisk-dev] How to get to 10.000 open calls   Alex Balashov
• [asterisk-dev] Symmetric RTP behaviour and 'nat' peer option.   Alistair Cunningham
• [asterisk-dev] How to get to 10.000 open calls   Sergey Okhapkin
• [asterisk-dev] How to get to 10.000 open calls   Sergey Okhapkin
• [asterisk-dev] MeetMe depends on zaptel?   Vadim Lebedev
• [asterisk-dev] Symmetric RTP behaviour and 'nat' peer option.   Gregory Boehnlein
• [asterisk-dev] touch utility and app_spool   Vadim Lebedev
• [asterisk-dev] touch utility and app_spool   Russell Bryant
• [asterisk-dev] [Code Review] Add stack-like functions for manipulating comma-separated lists in the dialplan   Terry Wilson
• [asterisk-dev] touch utility and app_spool   Saúl Ibarra
• [asterisk-dev] Amoocon Delivery Service ;-)   Nir Simionovich
• [asterisk-dev] [Code Review] Add stack-like functions for manipulating comma-separated lists in the dialplan   Terry Wilson
• [asterisk-dev] [Code Review] Add stack-like functions for manipulating comma-separated lists in the dialplan   Terry Wilson
• [asterisk-dev] [Code Review] Add stack-like functions for manipulating comma-separated lists in the dialplan   Terry Wilson
• [asterisk-dev] touch utility and app_spool   Vadim Lebedev
• [asterisk-dev] [Code Review] Add stack-like functions for manipulating comma-separated lists in the dialplan   atis at iq-labs.net
• [asterisk-dev] [Code Review] Convert ast_channel over to astobj2   David Vossel
• [asterisk-dev] CDR feature not working properly for "failed call attempt"   John Todd
• [asterisk-dev] [Code Review] Add stack-like functions for manipulating comma-separated lists in the dialplan   Terry Wilson
• [asterisk-dev] [Code Review] Add stack-like functions for manipulating comma-separated lists in the dialplan   Terry Wilson
• [asterisk-dev] [Code Review] Convert ast_channel over to astobj2   Russell Bryant
• [asterisk-dev] [Code Review] Convert ast_channel over to astobj2   Russell Bryant
• [asterisk-dev] [Code Review] Add stack-like functions for manipulating comma-separated lists in the dialplan   Mark Michelson
• [asterisk-dev] Asterisk handling pri PRI_PROG_CALL_NOT_E2E_ISDN progressmask   Paulo Garcia
• [asterisk-dev] CDR feature not working properly for "failed call attempt"   Matt Riddell
• [asterisk-dev] Shouldn't Asterisk call chdir("/") when becoming a daemon?   Chris Maciejewski
• [asterisk-dev] Symmetric RTP behaviour and 'nat' peer option.   Alex Balashov
• [asterisk-dev] Shouldn't Asterisk call chdir("/") when becoming a daemon?   Michiel van Baak
• [asterisk-dev] Symmetric RTP behaviour and 'nat' peer option.   Venefax
• [asterisk-dev] phoniceq ph_tor3_t1.c   Les - LES.NET (1996) INC.
• [asterisk-dev] Shouldn't Asterisk call chdir("/") when becoming a daemon?   Tzafrir Cohen
• [asterisk-dev] Symmetric RTP behaviour and 'nat' peer option.   Joshua Colp
• [asterisk-dev] Symmetric RTP behaviour and 'nat' peer option.   Alex Balashov
• [asterisk-dev] Symmetric RTP behaviour and 'nat' peer option.   Joshua Colp
• [asterisk-dev] SIP authentication and entity matching   Prince Singh
• [asterisk-dev] Symmetric RTP behaviour and 'nat' peer option.   Olle E. Johansson
• [asterisk-dev] Symmetric RTP behaviour and 'nat' peer option.   Johann Steinwendtner
• [asterisk-dev] Q.SIG Advice-Of-Charge encode/decode routines   Julian Lyndon-Smith
• [asterisk-dev] Shouldn't Asterisk call chdir("/") when becoming a daemon?   Chris Maciejewski
• [asterisk-dev] Symmetric RTP behaviour and 'nat' peer option.   Venefax
• [asterisk-dev] Q.SIG Advice-Of-Charge encode/decode routines   Alec Davis
• [asterisk-dev] Q.SIG Advice-Of-Charge encode/decode routines   Kaloyan Kovachev
• [asterisk-dev] Avoiding RTP flows (new topic)   Olle E. Johansson
• [asterisk-dev] Avoiding RTP flows (new topic)   Alex Balashov
• [asterisk-dev] Avoiding RTP flows (new topic)   Venefax
• [asterisk-dev] Avoiding RTP flows (new topic)   Olle E. Johansson
• [asterisk-dev] Avoiding RTP flows (new topic)   Alex Balashov
• [asterisk-dev] Avoiding RTP flows (new topic)   Venefax
• [asterisk-dev] Avoiding RTP flows (new topic)   Alex Balashov
• [asterisk-dev] How to get to 10.000 open calls   Klaus Darilion
• [asterisk-dev] SIP authentication and entity matching   Klaus Darilion
• [asterisk-dev] Avoiding RTP flows (new topic)   Olle E. Johansson
• [asterisk-dev] How to get to 10.000 open calls   Venefax
• [asterisk-dev] How to get to 10.000 open calls   Klaus Darilion
• [asterisk-dev] Avoiding RTP flows (new topic)   Klaus Darilion
• [asterisk-dev] MeetMe, setting MEETMEUNIQUEID   Tony Mountifield
• [asterisk-dev] The check for pthread_rwlock_timedrdlock   Olle E. Johansson
• [asterisk-dev] Avoiding RTP flows (new topic)   Benny Amorsen
• [asterisk-dev] MeetMe, setting MEETMEUNIQUEID   Mark Michelson
• [asterisk-dev] Avoiding RTP flows (new topic)   Alex Balashov
• [asterisk-dev] Q.SIG Advice-Of-Charge encode/decode routines   Richard Mudgett
• [asterisk-dev] Q.SIG Advice-Of-Charge encode/decode routines   Klaus Darilion
• [asterisk-dev] [Code Review] Make the parameter separator backward compatible, and error messages more consistent.   Eliel Sardañons
• [asterisk-dev] MeetMe, setting MEETMEUNIQUEID   Tony Mountifield
• [asterisk-dev] Google Summer of Code 2009   Russell Bryant
• [asterisk-dev] Google Summer of Code 2009   Jeff LaCoursiere
• [asterisk-dev] Libpri-1.4.10 Released   Asterisk Development Team
• [asterisk-dev] Google Summer of Code 2009   Russell Bryant
• [asterisk-dev] Google Summer of Code 2009   Steve Edwards
• [asterisk-dev] Roadmap for BRI point to multipoint in NT (bri_net_ptmp) for chan_dadhi/lipbri?   Kristijan Vrban
• [asterisk-dev] Google Summer of Code 2009   Clod Patry
• [asterisk-dev] dahdi kernel panic on shutdown   Justin Killen
• [asterisk-dev] [Code Review] HTTP Digest authentication   Russell Bryant
• [asterisk-dev] Google Summer of Code 2009   Steve Edwards
• [asterisk-dev] Google Summer of Code 2009   Tzafrir Cohen
• [asterisk-dev] Google Summer of Code 2009   Andre Courchesne - Prival
• [asterisk-dev] Google Summer of Code 2009   Steve Edwards
• [asterisk-dev] dahdi kernel panic on shutdown   Doug Lytle
• [asterisk-dev] Google Summer of Code 2009   Tzafrir Cohen
• [asterisk-dev] Google Summer of Code 2009   Tzafrir Cohen
• [asterisk-dev] Google Summer of Code 2009   Russell Bryant
• [asterisk-dev] New security log channel   freebetel
• [asterisk-dev] dahdi kernel panic on shutdown   Kaloyan Kovachev
• [asterisk-dev] New security log channel   Klaus Darilion
• [asterisk-dev] Shouldn't Asterisk call chdir("/") when becoming a daemon?   Chris Maciejewski
• [asterisk-dev] External Auth / SIP peer authentication?   Andreas Anderson
• [asterisk-dev] [LibPRI] Question on Charging indicator forwaded in Bridged channel   Bernard Merindol(TNE)
• [asterisk-dev] [LibPRI] Question on Charging indicator forwaded in Bridged channel   Klaus Darilion
• [asterisk-dev] External Auth / SIP peer authentication?   Olle E. Johansson
• [asterisk-dev] New security log channel   Russell Bryant
• [asterisk-dev] New security log channel   Atis Lezdins
• [asterisk-dev] New security log channel   Kevin P. Fleming
• [asterisk-dev] External Auth / SIP peer authentication?   Stephan Monette
• [asterisk-dev] Shouldn't Asterisk call chdir("/") when becoming a daemon?   Tzafrir Cohen
• [asterisk-dev] [Code Review] TLS/SSL private key option   David Vossel
• [asterisk-dev] [Code Review] TLS/SSL private key option   Russell Bryant
• [asterisk-dev] [Code Review] TLS/SSL private key option   Russell Bryant
• [asterisk-dev] [Code Review] TLS/SSL private key option   Russell Bryant
• [asterisk-dev] [Code Review] String field test module   Tilghman Lesher
• [asterisk-dev] [Code Review] TLS/SSL private key option   Kevin Fleming
• [asterisk-dev] following asterisk with git-svn   Tzafrir Cohen
• [asterisk-dev] [Code Review] New application JabberReceive, implement SendText in chan_gtalk and chan_jingle   Russell Bryant
• [asterisk-dev] [Code Review] Allow a position to be specified when entering a queue.   Russell Bryant
• [asterisk-dev] [Code Review] Create API for doing variable substitution with dynamic buffers   Russell Bryant
• [asterisk-dev] [Code Review] Allow a position to be specified when entering a queue.   Mark Michelson
• [asterisk-dev] Capturing SIP 4xx cause in Out Call CDR   Deepak Mittal
• [asterisk-dev] [Code Review] TCP, not finding peer.   David Vossel
• [asterisk-dev] [Code Review] TCP/TLS not finding peer   David Vossel
• [asterisk-dev] [Code Review] SIP: TCP not finding peer   David Vossel
• [asterisk-dev] [Code Review] SIP: TCP not finding peer   David Vossel
• [asterisk-dev] Pedantic   Stephen Davies
• [asterisk-dev] Asterisk -C   Venefax
• [asterisk-dev] Asterisk -C   Michiel van Baak
• [asterisk-dev] European Asterisk Developer's Meet-Up: Last Call   John Todd
• [asterisk-dev] [Code Review] Consistent SSL/TLS options across conf files   David Vossel
• [asterisk-dev] Asterisk Lua Reference   Matthew Nicholson
• [asterisk-dev] European Asterisk Developer's Meet-Up: Last Call   Nir Simionovich
• [asterisk-dev] European Asterisk Developer's Meet-Up: Last Call   Tzafrir Cohen
• [asterisk-dev] [Code Review] [updated] This patch implements CEL in trunk.   Matthew Nicholson
• [asterisk-dev] [Code Review] This patch implements CEL in trunk.   Matthew Nicholson
• [asterisk-dev] European Asterisk Developer's Meet-Up: Last Call   Tilghman Lesher
• [asterisk-dev] European Asterisk Developer's Meet-Up: Last Call   Matthew Nicholson
• [asterisk-dev] Asterisk-Addons 1.4.8 Now Available   Asterisk Development Team
• [asterisk-dev] Asterisk-Addons 1.6.1.0 Now Available   Asterisk Development Team
• [asterisk-dev] Asterisk 1.6.1.0 Now Available   Asterisk Development Team
• [asterisk-dev] European Asterisk Developer's Meet-Up: Last Call   Nir Simionovich
• [asterisk-dev] European Asterisk Developer's Meet-Up: Last Call   Nir Simionovich
• [asterisk-dev] [Code Review] Consistent SSL/TLS options across conf files   Olle E Johansson
• [asterisk-dev] [svn-commits] file: r190860 - symmetric_rtp   Olle E. Johansson
• [asterisk-dev] Asterisk Auto Gain Control   Sunny Du
• [asterisk-dev] Asterisk Auto Gain Control   Russell Bryant
• [asterisk-dev] European Asterisk Developer's Meet-Up: Last Call   Benny Amorsen
• [asterisk-dev] European Asterisk Developer's Meet-Up: Last Call   Atis Lezdins
• [asterisk-dev] European Asterisk Developer's Meet-Up: Last Call   Nir Simionovich
• [asterisk-dev] [svn-commits] file: r190860 - symmetric_rtp   Joshua Colp
• [asterisk-dev] [svn-commits] file: r190860 - symmetric_rtp   Olle E. Johansson
• [asterisk-dev] European Asterisk Developer's Meet-Up: Last Call   Benny Amorsen
• [asterisk-dev] XML-RPC dialplan interface (was: European Asterisk Developer's Meet-Up: Last Call)   Russell Bryant
• [asterisk-dev] European Asterisk Developer's Meet-Up: Last Call   Tilghman Lesher
• [asterisk-dev] European Asterisk Developer's Meet-Up: Last Call   Russell Bryant
• [asterisk-dev] XML-RPC dialplan interface   Nir Simionovich
• [asterisk-dev] European Asterisk Developer's Meet-Up: Last Call   Kevin P. Fleming
• [asterisk-dev] [Code Review] Consistent SSL/TLS options across conf files   David Vossel
• [asterisk-dev] Bounty for parking on <slot>@<context>   Alistair Cunningham
• [asterisk-dev] Sip Domain ACL   David Hansen
• [asterisk-dev] Sip Domain ACL   Olle E. Johansson
• [asterisk-dev] Asterisk Auto Gain Control   Prince Singh
• [asterisk-dev] Asterisk Auto Gain Control   Mark Michelson
• [asterisk-dev] [Code Review] SIP option to specify outbound TLS/SSL client protocol.   David Vossel
• [asterisk-dev] Verifone-Asterisk-AGI   Juan Miguel Quiros Arrieta
• [asterisk-dev] [Code Review] SIP option to specify outbound TLS/SSL client protocol.   David Vossel
• [asterisk-dev] Verifone-Asterisk-AGI   Noe Lozano
• [asterisk-dev] [Code Review] SIP option to specify outbound TLS/SSL client protocol.   Olle E Johansson
• [asterisk-dev] Bounty for parking on <slot>@<context>   Jared Smith
• [asterisk-dev] Verifone-Asterisk-AGI   Noe Lozano
• [asterisk-dev] European Asterisk Developer's Meet-Up: Last Call   Benny Amorsen
• [asterisk-dev] Verifone-Asterisk-AGI   Juan Miguel Quiros Arrieta
• [asterisk-dev] Bounty for parking on <slot>@<context>   Alistair Cunningham
• [asterisk-dev] Bounty for parking on <slot>@<context>   Jeff LaCoursiere
• [asterisk-dev] Verifone-Asterisk-AGI   Noe Lozano
• [asterisk-dev] Bounty for parking on <slot>@<context>   Alistair Cunningham
• [asterisk-dev] Verifone-Asterisk-AGI   Juan Miguel Quiros Arrieta
• [asterisk-dev] [Code Review] SIP: TCP not finding peer   David Vossel
• [asterisk-dev] European Asterisk Developer's Meet-Up: Last Call   Tilghman Lesher
• [asterisk-dev] Bounty for parking on <slot>@<context>   Jared Smith
• [asterisk-dev] Bounty for parking on <slot>@<context>   Alistair Cunningham
• [asterisk-dev] no panic [was: Re: [svn-commits] tilghman: trunk r191283 - /trunk/main/asterisk.c]   Tzafrir Cohen
• [asterisk-dev] sruffell: linux/trunk r6524 - /linux/trunk/drivers/dahdi/dahdi_dummy.c   Tony Mountifield
• [asterisk-dev] sruffell: linux/trunk r6528 - /linux/trunk/drivers/dahdi/wct4xxp/base.c   Tony Mountifield
• [asterisk-dev] no panic [was: Re: [svn-commits] tilghman: trunk r191283 - /trunk/main/asterisk.c]   Tilghman Lesher
• [asterisk-dev] [svn-commits] tilghman: trunk r191283 - /trunk/main/asterisk.c   asterisk at ntplx.net
• [asterisk-dev] sruffell: linux/trunk r6524 - /linux/trunk/drivers/dahdi/dahdi_dummy.c   Shaun Ruffell
• [asterisk-dev] sruffell: linux/trunk r6528 - /linux/trunk/drivers/dahdi/wct4xxp/base.c   Shaun Ruffell
• [asterisk-dev] [svn-commits] tilghman: trunk r191283 - /trunk/main/asterisk.c   Tilghman Lesher
• [asterisk-dev] 16khz wav files   asterisk at ntplx.net
• [asterisk-dev] 16khz wav files   Kevin P. Fleming
• [asterisk-dev] Doing something on SIP registration   sip at xbit.ru
• [asterisk-dev] signed/unsgined problem when dealing with iax frame timestamps   Guillermo Winkler
• [asterisk-dev] Doing something on SIP registration   Martin

Last message date: Thu Apr 30 19:35:19 CDT 2009
Archived on: Thu Apr 30 19:35:25 CDT 2009

• Messages sorted by: [ thread ] [ subject ] [ author ]
• More info on this list...