April 2010 Archives by date

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

Starting: Thu Apr 1 06:56:06 CDT 2010
Ending: Fri Apr 30 18:16:19 CDT 2010
Messages: 509

• [asterisk-dev] [Code Review] Fix descriptor (and structure reference) leak when call is hungup before answer   Russell Bryant
• [asterisk-dev] [Code Review] ETSI Message Waiting Indication (MWI) (Asterisk component)   Russell Bryant
• [asterisk-dev] AST_FLAG_EXCEPTION   Luis Santana
• [asterisk-dev] [Code Review] Fix descriptor (and structure reference) leak when call is hungup before answer   Russell Bryant
• [asterisk-dev] AST_FLAG_EXCEPTION   Russell Bryant
• [asterisk-dev] [Code Review] Prevent NewChannel manager event when channel is created solely for variable substitution purposes   Russell Bryant
• [asterisk-dev] GSoC2010: Complete the ast_storage project for voicemail   John Khvatov
• [asterisk-dev] [Bamboo] Asterisk - trunk - Linux - x86_64 build 356 has FAILED (0 tests failed)   Bamboo
• [asterisk-dev] [Bamboo] Asterisk - 1.4 - Linux - x86_64 build 120 has FAILED   Bamboo
• [asterisk-dev] [Bamboo] Asterisk - 1.6.1 - Linux - x86_64 build 155 has FAILED   Bamboo
• [asterisk-dev] [Bamboo] Asterisk - 1.6.0 - Linux - x86_64 build 147 has FAILED   Bamboo
• [asterisk-dev] [Bamboo] Asterisk - 1.6.2 - Linux - x86_64 build 188 has FAILED   Bamboo
• [asterisk-dev] [Code Review] Prevent NewChannel manager event when channel is created solely for variable substitution purposes   Mark Michelson
• [asterisk-dev] [Code Review] Fix descriptor (and structure reference) leak when call is hungup before answer   santosh chintalwar
• [asterisk-dev] [Bamboo] Asterisk - trunk - Linux - x86_64 build 357 has FAILED (0 tests failed). Change made by Matthew Nicholson   Bamboo
• [asterisk-dev] [Bamboo] Asterisk - trunk - Linux - x86_64 build 358 was SUCCESSFUL (with 42 tests)   Bamboo
• [asterisk-dev] [Bamboo] Asterisk - 1.4 - Linux - x86_64 build 121 was SUCCESSFUL (with 5 tests)   Bamboo
• [asterisk-dev] [Bamboo] Asterisk - 1.6.1 - Linux - x86_64 build 156 was SUCCESSFUL (with 5 tests)   Bamboo
• [asterisk-dev] [Bamboo] Asterisk - 1.6.0 - Linux - x86_64 build 148 was SUCCESSFUL (with 5 tests)   Bamboo
• [asterisk-dev] [Bamboo] Asterisk - 1.6.2 - Linux - x86_64 build 189 was SUCCESSFUL (with 5 tests)   Bamboo
• [asterisk-dev] CDR differences   Matthew Nicholson
• [asterisk-dev] [help] Asterisk App dev buffer problem   Eric Locey
• [asterisk-dev] [Code Review] Fix descriptor (and structure reference) leak when call is hungup before answer   Tilghman Lesher
• [asterisk-dev] [Code Review] Prevent NewChannel manager event when channel is created solely for variable substitution purposes   Russell Bryant
• [asterisk-dev] [Code Review] ETSI Message Waiting Indication (MWI) (Asterisk component)   rmudgett at digium.com
• [asterisk-dev] [Code Review] ETSI Message Waiting Indication (MWI) (Asterisk component)   rmudgett at digium.com
• [asterisk-dev] T.38 Issues   Bryant Zimmerman
• [asterisk-dev] [Code Review] Add an option to restore past broken behavor of the Events manager action   Matthew Nicholson
• [asterisk-dev] [Code Review] Test an option to restore past broken behavor of the Events manager action   Matthew Nicholson
• [asterisk-dev] ExternalIVR doesn't like X-Lite   David Ruggles
• [asterisk-dev] ExternalIVR doesn't like X-Lite   Chris Tooley
• [asterisk-dev] T.38 Issues   Klaus Darilion
• [asterisk-dev] change IP address of AA 50?   Matt Bazan
• [asterisk-dev] change IP address of AA 50?   Russell Bryant
• [asterisk-dev] [Code Review] Add an option to restore past broken behavor of the Events manager action   Russell Bryant
• [asterisk-dev] [Code Review] Allow quoting and escaping within application options with arguments   Tilghman Lesher
• [asterisk-dev] GSoC2010: Complete the ast_storage project for voicemail   Tilghman Lesher
• [asterisk-dev] [Code Review] Add an option to restore past broken behavor of the Events manager action   Matthew Nicholson
• [asterisk-dev] [Code Review] Add an option to restore past broken behavor of the Events manager action   Matthew Nicholson
• [asterisk-dev] [Code Review] Test an option to restore past broken behavor of the Events manager action   Matthew Nicholson
• [asterisk-dev] [Bamboo] Asterisk - 1.6.0 - Linux - x86_64 build 151 has FAILED   Bamboo
• [asterisk-dev] [Bamboo] Asterisk - 1.6.2 - Linux - x86_64 build 192 has FAILED   Bamboo
• [asterisk-dev] [Bamboo] Asterisk - trunk - Linux - x86_64 build 362 has FAILED (0 tests failed). Change made by Michiel van Baak   Bamboo
• [asterisk-dev] [Bamboo] Asterisk - 1.4 - Linux - x86_64 build 124 has FAILED   Bamboo
• [asterisk-dev] [Bamboo] Asterisk - 1.6.1 - Linux - x86_64 build 160 has FAILED   Bamboo
• [asterisk-dev] [Bamboo] Asterisk - 1.6.0 - Linux - x86_64 build 152 has FAILED   Bamboo
• [asterisk-dev] [Bamboo] Asterisk - 1.6.2 - Linux - x86_64 build 193 has FAILED   Bamboo
• [asterisk-dev] [Bamboo] Asterisk - trunk - Linux - x86_64 build 363 has FAILED (0 tests failed). Change made by Kevin P. Fleming   Bamboo
• [asterisk-dev] [Bamboo] Asterisk - 1.4 - Linux - x86_64 build 125 has FAILED   Bamboo
• [asterisk-dev] [Bamboo] Asterisk - 1.6.1 - Linux - x86_64 build 161 has FAILED   Bamboo
• [asterisk-dev] [Bamboo] Asterisk - 1.6.0 - Linux - x86_64 build 153 has FAILED   Bamboo
• [asterisk-dev] [Bamboo] Asterisk - 1.6.2 - Linux - x86_64 build 194 has FAILED   Bamboo
• [asterisk-dev] [Bamboo] Asterisk - trunk - Linux - x86_64 build 364 has FAILED (0 tests failed). Change made by tilghman   Bamboo
• [asterisk-dev] [Bamboo] Asterisk - 1.6.2 - Linux - x86_64 build 195 has FAILED. Change made by tilghman   Bamboo
• [asterisk-dev] [Bamboo] Asterisk - 1.6.0 - Linux - x86_64 build 154 has FAILED. Change made by tilghman   Bamboo
• [asterisk-dev] [Bamboo] Asterisk - 1.6.1 - Linux - x86_64 build 162 has FAILED. Change made by tilghman   Bamboo
• [asterisk-dev] Does Bamboo deserve it's own list?   Steve Edwards
• [asterisk-dev] Does Bamboo deserve it's own list?   Kurt Lidl
• [asterisk-dev] [Bamboo] Asterisk - 1.4 - Linux - x86_64 build 126 has FAILED   Bamboo
• [asterisk-dev] Does Bamboo deserve it's own list?   Russell Bryant
• [asterisk-dev] [Bamboo] Asterisk - trunk - Linux - x86_64 build 365 has FAILED (0 tests failed)   Bamboo
• [asterisk-dev] Does Bamboo deserve it's own list?   Russell Bryant
• [asterisk-dev] [Code Review] A python AsteriskCSVCDR library for testing csv cdrs   Terry Wilson
• [asterisk-dev] [Code Review] A python AsteriskCSVCDR library for testing csv cdrs   Russell Bryant
• [asterisk-dev] [Code Review] A python AsteriskCSVCDR library for testing csv cdrs   Russell Bryant
• [asterisk-dev] [Code Review] A python AsteriskCSVCDR library for testing csv cdrs   Terry Wilson
• [asterisk-dev] [Code Review] A python AsteriskCSVCDR library for testing csv cdrs   Terry Wilson
• [asterisk-dev] [Code Review] A python AsteriskCSVCDR library for testing csv cdrs   Russell Bryant
• [asterisk-dev] [Code Review] Add an option to restore past broken behavor of the Events manager action   Russell Bryant
• [asterisk-dev] [Code Review] Test an option to restore past broken behavor of the Events manager action   Russell Bryant
• [asterisk-dev] [Code Review] Allow quoting and escaping within application options with arguments   Russell Bryant
• [asterisk-dev] [Code Review] rfc 4474 patch   Russell Bryant
• [asterisk-dev] Does Bamboo deserve it's own list?   Steve Totaro
• [asterisk-dev] Does Bamboo deserve it's own list?   asterisk at ntplx.net
• [asterisk-dev] Does Bamboo deserve it's own list?   Russell Bryant
• [asterisk-dev] GSoC 2010: Testing the SDP parser in chan_sip   Loren Sands-Ramshaw
• [asterisk-dev] disable XPP in dahdi-linux   Paco Gil
• [asterisk-dev] disable XPP in dahdi-linux   Tzafrir Cohen
• [asterisk-dev] GSoC 2010: Testing the SDP parser in chan_sip   Tilghman Lesher
• [asterisk-dev] [Code Review] Allow quoting and escaping within application options with arguments   Tilghman Lesher
• [asterisk-dev] [Code Review] Allow quoting and escaping within application options with arguments   Tilghman Lesher
• [asterisk-dev] disable XPP in dahdi-linux   Paco Gil
• [asterisk-dev] AST_FLAG_EXCEPTION   Luis Santana
• [asterisk-dev] [Code Review] A python AsteriskCSVCDR library for testing csv cdrs   Godson Gera
• [asterisk-dev] [Code Review] Allow quoting and escaping within application options with arguments   Sean Bright
• [asterisk-dev] GSoC 2010: Testing the SDP parser in chan_sip   Olle E. Johansson
• [asterisk-dev] [Code Review] Prevent NewChannel manager event when channel is created solely for variable substitution purposes   Mark Michelson
• [asterisk-dev] disable XPP in dahdi-linux   Paco Gil
• [asterisk-dev] [Code Review] Allow quoting and escaping within application options with arguments   Tilghman Lesher
• [asterisk-dev] disable XPP in dahdi-linux   Tzafrir Cohen
• [asterisk-dev] disable XPP in dahdi-linux   Paco Gil
• [asterisk-dev] Please sign Petition - Stop Child Labour   Sarfaraz Chougule
• [asterisk-dev] Please sign Petition - Stop Child Labour   jon pounder
• [asterisk-dev] Asterisk 1.4.31-rc1, 1.6.0.27-rc1, 1.6.1.19-rc1, and 1.6.2.7-rc1 Now Available   Asterisk Development Team
• [asterisk-dev] Please sign Petition - Stop Child Labour   Tzafrir Cohen
• [asterisk-dev] [Code Review] fixes deadlock in chan_sip caused by use of MASTER_CHANNEL() dialplan function   David Vossel
• [asterisk-dev] [Code Review] fixes deadlock in chan_sip caused by use of MASTER_CHANNEL() dialplan function   David Vossel
• [asterisk-dev] [Code Review] fixes deadlock in chan_sip caused by use of MASTER_CHANNEL() dialplan function   Mark Michelson
• [asterisk-dev] [Code Review] fixes deadlock in chan_sip caused by use of MASTER_CHANNEL() dialplan function   David Vossel
• [asterisk-dev] [Code Review] rfc 4474 patch   Klaus Darilion
• [asterisk-dev] [Code Review] fixes deadlock in chan_sip caused by use of MASTER_CHANNEL() dialplan function   David Vossel
• [asterisk-dev] [Code Review] fixes deadlock in chan_sip caused by use of MASTER_CHANNEL() dialplan function   David Vossel
• [asterisk-dev] [Code Review] fixes deadlock in chan_sip caused by use of MASTER_CHANNEL() dialplan function   Mark Michelson
• [asterisk-dev] [Code Review] Allow quoting and escaping within application options with arguments   Tilghman Lesher
• [asterisk-dev] CALLERID(name) being overwritten beneath dialplan   Joshua Elson
• [asterisk-dev] CALLERID(name) being overwritten beneath dialplan   Tilghman Lesher
• [asterisk-dev] [Code Review] Allow quoting and escaping within application options with arguments   Tilghman Lesher
• [asterisk-dev] CALLERID(name) being overwritten beneath dialplan   Joshua Elson
• [asterisk-dev] CALLERID(name) being overwritten beneath dialplan   Tilghman Lesher
• [asterisk-dev] Does Bamboo deserve it's own list?   Philip A. Prindeville
• [asterisk-dev] Amoocon 2010, DE   Nir Simionovich
• [asterisk-dev] [Code Review] func_srv and explicit specification of destination for SIP outgoing INVITEs   Mark Michelson
• [asterisk-dev] [Code Review] External tests for func_srv and explicit SIP outgoing destination   Mark Michelson
• [asterisk-dev] [Code Review] External tests for func_srv and explicit SIP outgoing destination   Mark Michelson
• [asterisk-dev] [Code Review] func_srv and explicit specification of destination for SIP outgoing INVITEs   Mark Michelson
• [asterisk-dev] Durfos Terminator has invited you to Dropbox   Dropbox
• [asterisk-dev] ExternalIVR doesn't like X-Lite   David Ruggles
• [asterisk-dev] [Code Review] func_srv and explicit specification of destination for SIP outgoing INVITEs   Tilghman Lesher
• [asterisk-dev] [Code Review] Prevent NewChannel manager event when channel is created solely for variable substitution purposes   Tilghman Lesher
• [asterisk-dev] [Code Review] IAXpeers output now matches SIPpeers format for manager.   Tilghman Lesher
• [asterisk-dev] [Code Review] func_srv and explicit specification of destination for SIP outgoing INVITEs   Mark Michelson
• [asterisk-dev] [Code Review] func_srv and explicit specification of destination for SIP outgoing INVITEs   Mark Michelson
• [asterisk-dev] [Code Review] func_srv and explicit specification of destination for SIP outgoing INVITEs   Mark Michelson
• [asterisk-dev] [Bounty] Patch for Transfer Channel to properly Inherit Channel Variables   Steve Poirier
• [asterisk-dev] [Code Review] External tests for func_srv and explicit SIP outgoing destination   Mark Michelson
• [asterisk-dev] [Code Review] func_srv and explicit specification of destination for SIP outgoing INVITEs   Mark Michelson
• [asterisk-dev] [Code Review] func_srv and explicit specification of destination for SIP outgoing INVITEs   Tilghman Lesher
• [asterisk-dev] [Code Review] Fix issue 17078: MixMonitor records shorter files than the call duration.   Tilghman Lesher
• [asterisk-dev] GSoC2010: Complete the ast_storage project for voicemail   John Khvatov
• [asterisk-dev] [Code Review] Explicit context set in SIP peer overridden by default domain context   Philip A. Prindeville
• [asterisk-dev] [Code Review] Fix issue 17078: MixMonitor records shorter files than the call duration.   David Vossel
• [asterisk-dev] [Code Review] func_srv and explicit specification of destination for SIP outgoing INVITEs   Mark Michelson
• [asterisk-dev] [Code Review] External tests for func_srv and explicit SIP outgoing destination   Mark Michelson
• [asterisk-dev] non-root for Asterisk Test Suite?   Paul Belanger
• [asterisk-dev] [asterisk-commits] dvossel: trunk r256319 - /trunk/channels/chan_sip.c   Russell Bryant
• [asterisk-dev] non-root for Asterisk Test Suite?   Russell Bryant
• [asterisk-dev] [Code Review] func_srv and explicit specification of destination for SIP outgoing INVITEs   Philip A. Prindeville
• [asterisk-dev] [Code Review] func_srv and explicit specification of destination for SIP outgoing INVITEs   Russell Bryant
• [asterisk-dev] [Code Review] func_srv and explicit specification of destination for SIP outgoing INVITEs   Russell Bryant
• [asterisk-dev] GSoC2010: Complete the ast_storage project for voicemail   Russell Bryant
• [asterisk-dev] SIPFROMDOMAIN   Philip A. Prindeville
• [asterisk-dev] [Code Review] Dynamiclly generate asterisk.conf (new script)   pabelanger at gmail.com
• [asterisk-dev] [Code Review] Dynamiclly generate asterisk.conf (new script)   Tzafrir Cohen
• [asterisk-dev] [Code Review] func_srv and explicit specification of destination for SIP outgoing INVITEs   Tilghman Lesher
• [asterisk-dev] SIPFROMDOMAIN   Olle E. Johansson
• [asterisk-dev] SIPFROMDOMAIN   Philip A. Prindeville
• [asterisk-dev] [Code Review] Dynamiclly generate asterisk.conf (new script)   Paul Belanger
• [asterisk-dev] [Code Review] Dynamiclly generate asterisk.conf (new script)   pabelanger at gmail.com
• [asterisk-dev] [Code Review] Dynamiclly generate asterisk.conf (new script)   pabelanger at gmail.com
• [asterisk-dev] [Code Review] func_srv and explicit specification of destination for SIP outgoing INVITEs   Russell Bryant
• [asterisk-dev] [Code Review] ast_event API - subscription unit test and some API bug fixes   Russell Bryant
• [asterisk-dev] [Code Review] Explicit context set in SIP peer overridden by default domain context   Mark Michelson
• [asterisk-dev] [asterisk-commits] dvossel: trunk r256319 - /trunk/channels/chan_sip.c   David Vossel
• [asterisk-dev] [Code Review] Dynamiclly generate asterisk.conf (new script)   pabelanger at gmail.com
• [asterisk-dev] [asterisk-commits] dvossel: trunk r256319 - /trunk/channels/chan_sip.c   Russell Bryant
• [asterisk-dev] Playing a Message into the ConfBridge App   Timothy Rodriguez
• [asterisk-dev] [Code Review] Dynamiclly generate asterisk.conf (new script)   pabelanger at gmail.com
• [asterisk-dev] Playing a Message into the ConfBridge App   Russell Bryant
• [asterisk-dev] Playing a Message into the ConfBridge App   Jamuel Starkey
• [asterisk-dev] Playing a Message into the ConfBridge App   Timothy Rodriguez
• [asterisk-dev] [Code Review] Out-of-tree modules can fail because of compiler flag differences.   Tilghman Lesher
• [asterisk-dev] [Code Review] Enable 'pri service' commands for DAHDI channels configured for the 'national' switchtype   Russell Bryant
• [asterisk-dev] [Code Review] Out-of-tree modules can fail because of compiler flag differences.   Jason Parker
• [asterisk-dev] Playing a Message into the ConfBridge App   Timothy Rodriguez
• [asterisk-dev] Playing a Message into the ConfBridge App   Joshua Colp
• [asterisk-dev] Playing a Message into the ConfBridge App   Joshua Colp
• [asterisk-dev] [svn-commits] moy: branch moy/dahdi-tap-1.6.2 r257024 - /team/moy/dahdi-tap-1.6.2/channels/   Richard Mudgett
• [asterisk-dev] [Code Review] External tests for func_srv and explicit SIP outgoing destination   Mark Michelson
• [asterisk-dev] [Code Review] func_srv and explicit specification of destination for SIP outgoing INVITEs   Mark Michelson
• [asterisk-dev] [Code Review] func_srv and explicit specification of destination for SIP outgoing INVITEs   Mark Michelson
• [asterisk-dev] DAHDI-Linux and DAHDI-Tools 2.3.0 Released   Asterisk Development Team
• [asterisk-dev] SIPFROMDOMAIN   Philip A. Prindeville
• [asterisk-dev] [Code Review] Generic Advice of Charge   David Vossel
• [asterisk-dev] GSoC2010: Complete the ast_storage project for voicemail   John Khvatov
• [asterisk-dev] [Code Review] Generic Advice of Charge   David Vossel
• [asterisk-dev] [Code Review] LibPRI changes for Generic AOC   David Vossel
• [asterisk-dev] SIPFROMDOMAIN   Olle E. Johansson
• [asterisk-dev] Asterisk 1.8 plans   Olle E. Johansson
• [asterisk-dev] Why are the DAHDI Kernel modules not part of the Kernel?   Klaus Darilion
• [asterisk-dev] Why are the DAHDI Kernel modules not part of the Kernel?   François Delawarde
• [asterisk-dev] Why are the DAHDI Kernel modules not part of the Kernel?   Kevin P. Fleming
• [asterisk-dev] Why are the DAHDI Kernel modules not part of the Kernel?   François Delawarde
• [asterisk-dev] Why are the DAHDI Kernel modules not part of the Kernel?   Benny Amorsen
• [asterisk-dev] [Code Review] ast_event API - subscription unit test and some API bug fixes   David Vossel
• [asterisk-dev] Asterisk 1.8 plans   Tilghman Lesher
• [asterisk-dev] GSoC2010: Complete the ast_storage project for voicemail   Tilghman Lesher
• [asterisk-dev] [Code Review] Allow passing multiple arguments to applications specified in features.conf   David Vossel
• [asterisk-dev] [Code Review] Allow quoting and escaping within application options with arguments   David Vossel
• [asterisk-dev] [Code Review] Allow quoting and escaping within application options with arguments   Tilghman Lesher
• [asterisk-dev] [Code Review] Allow quoting and escaping within application options with arguments   David Vossel
• [asterisk-dev] [Code Review] Allow quoting and escaping within application options with arguments   Tilghman Lesher
• [asterisk-dev] [Code Review] Allow quoting and escaping within application options with arguments   David Vossel
• [asterisk-dev] GSoC2010: Complete the ast_storage project for voicemail   John Khvatov
• [asterisk-dev] Asterisk 1.8 plans   Matt Riddell
• [asterisk-dev] Copying channel variables to another channel   Paul Belanger
• [asterisk-dev] [Code Review] LibPRI changes for Generic AOC   rmudgett at digium.com
• [asterisk-dev] Copying channel variables to another channel   Paul Belanger
• [asterisk-dev] Asterisk 1.8 plans   Ron Arts
• [asterisk-dev] Asterisk 1.8 plans   Hans Witvliet
• [asterisk-dev] [Code Review] Allow passing multiple arguments to applications specified in features.conf   Tilghman Lesher
• [asterisk-dev] [Code Review] Allow passing multiple arguments to applications specified in features.conf   Tilghman Lesher
• [asterisk-dev] [Code Review] Allow passing multiple arguments to applications specified in features.conf   David Vossel
• [asterisk-dev] [Code Review] Allow quoting and escaping within application options with arguments   Tilghman Lesher
• [asterisk-dev] [Code Review] Generic Advice of Charge   rmudgett at digium.com
• [asterisk-dev] SIPFROMDOMAIN   Philip A. Prindeville
• [asterisk-dev] [Code Review] Allow quoting and escaping within application options with arguments   David Vossel
• [asterisk-dev] Voice for callee after hangup   Harel Cohen
• [asterisk-dev] Voice for callee after hangup   Donny Kavanagh
• [asterisk-dev] Full implementation for Call-Waiting   Harel Cohen
• [asterisk-dev] New CA cert for Subversion - please read   Jeff Peeler
• [asterisk-dev] [Code Review] LibPRI changes for Generic AOC   David Vossel
• [asterisk-dev] Asterisk 1.8 plans   Vadim Lebedev
• [asterisk-dev] [Code Review] LibPRI changes for Generic AOC   David Vossel
• [asterisk-dev] Full implementation for Call-Waiting   Kevin P. Fleming
• [asterisk-dev] Asterisk 1.8 plans   Kevin P. Fleming
• [asterisk-dev] [Code Review] LibPRI changes for Generic AOC   Kevin Fleming
• [asterisk-dev] [Code Review] LibPRI changes for Generic AOC   David Vossel
• [asterisk-dev] Asterisk 1.6.0.27-rc2, 1.6.1.19-rc2, and 1.6.2.7-rc2 Now Available   Asterisk Development Team
• [asterisk-dev] Asterisk-Addons 1.4.11, 1.6.0.5, 1.6.1.3, and 1.6.2.1 Now Available   Asterisk Development Team
• [asterisk-dev] Asterisk 1.6.0.27-rc2, 1.6.1.19-rc2, and 1.6.2.7-rc2 Now Available   Jamuel Starkey
• [asterisk-dev] [Code Review] Generic Advice of Charge   rmudgett at digium.com
• [asterisk-dev] [Code Review] Generic Advice of Charge   David Vossel
• [asterisk-dev] Asterisk 1.6.0.27-rc2, 1.6.1.19-rc2, and 1.6.2.7-rc2 Now Available   Leif Madsen
• [asterisk-dev] [Code Review] Generic Advice of Charge   David Vossel
• [asterisk-dev] Bugs/patches 16033 and 16590 ignored forever   Kirill 'Big K' Katsnelson
• [asterisk-dev] Broken SIP dialog if INFO messages received before final OK   Karl H. Putz
• [asterisk-dev] [help] Asterisk App dev buffer problem   Nick Lewis
• [asterisk-dev] Bugs/patches 16033 and 16590 ignored forever   Leif Madsen
• [asterisk-dev] [Code Review] Generic Advice of Charge   rmudgett at digium.com
• [asterisk-dev] [Code Review] Generic Advice of Charge   David Vossel
• [asterisk-dev] Bugs/patches 16033 and 16590 ignored forever   Paul Belanger
• [asterisk-dev] Bugs/patches 16033 and 16590 ignored forever   Steve Murphy
• [asterisk-dev] [Code Review] Generic Advice of Charge   Shaun Ruffell
• [asterisk-dev] [Code Review] Generic Advice of Charge   David Vossel
• [asterisk-dev] [Code Review] Generic Advice of Charge   rmudgett at digium.com
• [asterisk-dev] Bugs/patches 16033 and 16590 ignored forever   Kirill 'Big K' Katsnelson
• [asterisk-dev] CDR edge-case causes different results in 1.6.2   Steve Davies
• [asterisk-dev] [Code Review] Generic Advice of Charge   David Vossel
• [asterisk-dev] [Code Review] Fix issue 17078: MixMonitor records shorter files than the call duration.   David Vossel
• [asterisk-dev] Bugs/patches 16033 and 16590 ignored forever   Kevin P. Fleming
• [asterisk-dev] DAHDI-Linux and DAHDI-Tools 2.3.0 Released   Alex Villací­s Lasso
• [asterisk-dev] [Code Review] Generic Advice of Charge   David Vossel
• [asterisk-dev] Bugs/patches 16033 and 16590 ignored forever   Steve Totaro
• [asterisk-dev] Bugs/patches 16033 and 16590 ignored forever   Kevin P. Fleming
• [asterisk-dev] Bugs/patches 16033 and 16590 ignored forever   Steve Totaro
• [asterisk-dev] [Code Review] Mute / Unmute manager commands for MixMonitor   Julian Lyndon-Smith
• [asterisk-dev] [Code Review] Mute / Unmute manager commands for MixMonitor   Julian Lyndon-Smith
• [asterisk-dev] [Code Review] Mute / Unmute manager commands for MixMonitor   Julian Lyndon-Smith
• [asterisk-dev] Caller busy state in IAX2 Protocol returns with hangup ?   Kumar Subramanian
• [asterisk-dev] (SOLUTION) CDR edge-case causes different results in 1.6.2   Steve Davies
• [asterisk-dev] Caller busy state in IAX2 Protocol returns with hangup ?   Kumar Subramanian
• [asterisk-dev] (SOLUTION) CDR edge-case causes different results in 1.6.2   Ryan Bullock
• [asterisk-dev] SIPFROMDOMAIN   Nick Lewis
• [asterisk-dev] Broken SIP dialog if INFO messages received beforefinal OK   Nick Lewis
• [asterisk-dev] (SOLUTION) CDR edge-case causes different results in 1.6.2   Leif Madsen
• [asterisk-dev] Broken SIP dialog if INFO messages receivedbeforefinal OK - Oops   Nick Lewis
• [asterisk-dev] Broken SIP dialog if INFO messages received beforefinal OK   Karl H. Putz
• [asterisk-dev] (SOLUTION) CDR edge-case causes different results in 1.6.2   Steve Davies
• [asterisk-dev] [Code Review] Generic Advice of Charge   David Vossel
• [asterisk-dev] Exceptionally long voice queuing   Timothy Rodriguez
• [asterisk-dev] Exceptionally long voice queuing   Leif Madsen
• [asterisk-dev] Broken SIP dialog if INFO messages received beforefinal OK   Karl H. Putz
• [asterisk-dev] Bugs/patches 16033 and 16590 ignored forever   Kirill 'Big K' Katsnelson
• [asterisk-dev] Bugs/patches 16033 and 16590 ignored forever   Kirill 'Big K' Katsnelson
• [asterisk-dev] Bugs/patches 16033 and 16590 ignored forever   Jamuel Starkey
• [asterisk-dev] Exceptionally long voice queuing   Ceschia, Marcello
• [asterisk-dev] Broken SIP dialog if INFO messages receivedbeforefinal OK   Nick Lewis
• [asterisk-dev] Broken SIP dialog if INFO messagesreceivedbeforefinal OK   Nick Lewis
• [asterisk-dev] Broken SIP dialog if INFO messagesreceivedbeforefinal OK   Olle E. Johansson
• [asterisk-dev] Broken SIP dialog if INFOmessagesreceivedbeforefinal OK   Nick Lewis
• [asterisk-dev] Broken SIP dialog if INFOmessagesreceivedbeforefinal OK   Karl H. Putz
• [asterisk-dev] Broken SIP dialog if INFOmessagesreceivedbeforefinal OK   Olle E. Johansson
• [asterisk-dev] Bugs/patches 16033 and 16590 ignored forever   Leif Madsen
• [asterisk-dev] Exceptionally long voice queuing   Leif Madsen
• [asterisk-dev] [Code Review] Explicit context set in SIP peer overridden by default domain context   Nick Lewis
• [asterisk-dev] "Digium development team" - not Asterisk dev team   Olle E. Johansson
• [asterisk-dev] "Digium development team" - not Asterisk dev team   Kevin P. Fleming
• [asterisk-dev] Exceptionally long voice queuing   Timothy Rodriguez
• [asterisk-dev] "Digium development team" - not Asterisk dev team   Olle E. Johansson
• [asterisk-dev] "Digium development team" - not Asterisk dev team   Nick Lewis
• [asterisk-dev] "Digium development team" - not Asterisk dev team   Steve Howes
• [asterisk-dev] "Digium development team" - not Asterisk dev team   jon pounder
• [asterisk-dev] "Digium development team" - not Asterisk dev team   Steve Edwards
• [asterisk-dev] "Digium development team" - not Asterisk dev team   Tim Ringenbach
• [asterisk-dev] Bugs/patches 16033 and 16590 ignored forever   Steve Murphy
• [asterisk-dev] Sorry for late response - dialstring changes and SRV support   Olle E. Johansson
• [asterisk-dev] Bugs/patches 16033 and 16590 ignored forever   Tzafrir Cohen
• [asterisk-dev] Exceptionally long voice queuing   Timothy Rodriguez
• [asterisk-dev] [Code Review] LibPRI changes for Generic AOC   David Vossel
• [asterisk-dev] [Code Review] LibPRI changes for Generic AOC   David Vossel
• [asterisk-dev] Sorry for late response - dialstring changes and SRV support   Kevin P. Fleming
• [asterisk-dev] Sorry for late response - dialstring changes and SRV support   Olle E. Johansson
• [asterisk-dev] Bugs/patches 16033 and 16590 ignored forever   Kirill 'Big K' Katsnelson
• [asterisk-dev] [Code Review] Generic Advice of Charge   David Vossel
• [asterisk-dev] [Code Review] Generic Advice of Charge   David Vossel
• [asterisk-dev] [Code Review] Generic Advice of Charge   David Vossel
• [asterisk-dev] Sorry for late response - dialstring changes and SRV support   Kevin P. Fleming
• [asterisk-dev] Sorry for late response - dialstring changes and SRV support   Tim Ringenbach
• [asterisk-dev] Bugs/patches 16033 and 16590 ignored forever   Leif Madsen
• [asterisk-dev] Maintenance at Digium HQ affecting origsvn.digium.com on April 21st   Kevin P. Fleming
• [asterisk-dev] [Code Review] LibPRI changes for Generic AOC   rmudgett at digium.com
• [asterisk-dev] [Code Review] Mute / Unmute manager commands for MixMonitor   Jeff Peeler
• [asterisk-dev] [Code Review] LibPRI changes for Generic AOC   David Vossel
• [asterisk-dev] [Code Review] Generic Advice of Charge   rmudgett at digium.com
• [asterisk-dev] Manipulating audio in asterisk   Slawek Sloma
• [asterisk-dev] Sorry for late response - dialstring changes and SRV support   Klaus Darilion
• [asterisk-dev] "Digium development team" - not Asterisk dev team - Java   Nick Lewis
• [asterisk-dev] say.conf implementation of Indian Languages to play numbers and dates   Amit Patkar | Avhan Technologies Pvt. Ltd.
• [asterisk-dev] Manipulating audio in asterisk   Klaus Darilion
• [asterisk-dev] say.conf implementation of Indian Languages to play numbers and dates   Tzafrir Cohen
• [asterisk-dev] Bugs/patches 16033 and 16590 ignored forever   François Delawarde
• [asterisk-dev] Manipulating audio in asterisk   Nick Lewis
• [asterisk-dev] [Code Review] Mute / Unmute manager commands for MixMonitor   Julian Lyndon-Smith
• [asterisk-dev] [Code Review] Mute / Unmute manager commands for MixMonitor   Julian Lyndon-Smith
• [asterisk-dev] Sorry for late response - dialstring changes and SRV support   Kevin P. Fleming
• [asterisk-dev] "Digium development team" - not Asterisk dev team - Java   jon pounder
• [asterisk-dev] Sorry for late response - dialstring changes and SRV support   Klaus Darilion
• [asterisk-dev] "Digium development team" - not Asterisk dev team - Java   Nick Lewis
• [asterisk-dev] "Digium development team" - not Asterisk dev team - Java   Tzafrir Cohen
• [asterisk-dev] "Digium development team" - not Asterisk dev team - Java   Timothy Rodriguez
• [asterisk-dev] [Code Review] Add some CDR tests   Terry Wilson
• [asterisk-dev] [Code Review] Generic Advice of Charge   David Vossel
• [asterisk-dev] [Code Review] Generic Advice of Charge   David Vossel
• [asterisk-dev] [Code Review] VoicemailMain and VMauthenticate: Like VoiceMail allow escape to the 'a' extension when a single '*' is entered in Mailbox or Password   Tilghman Lesher
• [asterisk-dev] "Digium development team" - not Asterisk dev team - Java   Tim Ringenbach
• [asterisk-dev] [Code Review] ETSI Explicit Call Transfer (ECT) support. (libpri portion)   Tilghman Lesher
• [asterisk-dev] "Digium development team" - not Asterisk dev team - Java   Tzafrir Cohen
• [asterisk-dev] [Code Review] ETSI Explicit Call Transfer (ECT) support. (Asterisk portion)   Tilghman Lesher
• [asterisk-dev] "Digium development team" - not Asterisk dev team - Java   Steve Underwood
• [asterisk-dev] [Code Review] directmediapermit/directmediadeny support to restrict which peers can do directmedia based on ip address   Terry Wilson
• [asterisk-dev] [Code Review] ETSI Explicit Call Transfer (ECT) support. (libpri portion)   David Vossel
• [asterisk-dev] [Code Review] ETSI Explicit Call Transfer (ECT) support. (libpri portion)   David Vossel
• [asterisk-dev] [Code Review] Explicit context set in SIP peer overridden by default domain context   Philip A. Prindeville
• [asterisk-dev] Bugs/patches 16033 and 16590 ignored forever   Leif Madsen
• [asterisk-dev] [Code Review] ETSI Explicit Call Transfer (ECT) support. (Asterisk portion)   rmudgett at digium.com
• [asterisk-dev] [Code Review] Explicit context set in SIP peer overridden by default domain context   Olle E. Johansson
• [asterisk-dev] dahdi-linux compile fails on kernel 2.6.34   Christian
• [asterisk-dev] dahdi-linux compile fails on kernel 2.6.34   Kevin P. Fleming
• [asterisk-dev] dahdi-linux compile fails on kernel 2.6.34   Tzafrir Cohen
• [asterisk-dev] Bugs/patches 16033 and 16590 ignored forever   François Delawarde
• [asterisk-dev] [Code Review] Explicit context set in SIP peeroverridden by default domain context   Nick Lewis
• [asterisk-dev] [Code Review] Explicit context set in SIP peeroverridden by default domain context   Olle E. Johansson
• [asterisk-dev] dahdi-linux compile fails on kernel 2.6.34   Christian
• [asterisk-dev] dahdi-linux compile fails on kernel 2.6.34   Christian
• [asterisk-dev] dahdi-linux compile fails on kernel 2.6.34 - fixed   Christian
• [asterisk-dev] [Code Review] Explicit context set in SIPpeeroverridden by default domain context   Nick Lewis
• [asterisk-dev] Bugs/patches 16033 and 16590 ignored forever   Leif Madsen
• [asterisk-dev] [Code Review] Explicit context set in SIP peer overridden by default domain context   Nick Lewis
• [asterisk-dev] Bugs/patches 16033 and 16590 ignored forever   François Delawarde
• [asterisk-dev] [Code Review] func_srv and explicit specification of destination for SIP outgoing INVITEs   Russell Bryant
• [asterisk-dev] [Code Review] func_srv and explicit specification of destination for SIP outgoing INVITEs   Russell Bryant
• [asterisk-dev] [Code Review] ETSI Explicit Call Transfer (ECT) support. (libpri portion)   rmudgett at digium.com
• [asterisk-dev] [Code Review] ETSI Explicit Call Transfer (ECT) support. (libpri portion)   rmudgett at digium.com
• [asterisk-dev] [Code Review] Explicit context set in SIP peer overridden by default domain context   Nick Lewis
• [asterisk-dev] [Code Review] ETSI Explicit Call Transfer (ECT) support. (libpri portion)   rmudgett at digium.com
• [asterisk-dev] [Code Review] ETSI Explicit Call Transfer (ECT) support. (Asterisk portion)   rmudgett at digium.com
• [asterisk-dev] [Code Review] Out-of-tree modules can fail because of compiler flag differences.   Jeff Peeler
• [asterisk-dev] [Code Review] Add some CDR tests   Matthew Nicholson
• [asterisk-dev] [Code Review] Add some CDR tests   Terry Wilson
• [asterisk-dev] [Code Review] Explicit context set in SIPpeeroverridden by default domain context   Philip A. Prindeville
• [asterisk-dev] [Code Review] Fix transcode_via_sln and increase utility of PLC   Mark Michelson
• [asterisk-dev] [Code Review] Add some CDR tests   Matthew Nicholson
• [asterisk-dev] Bugs/patches 16033 and 16590 ignored forever   Leif Madsen
• [asterisk-dev] [Code Review] Crash in audiohook_write_list   David Vossel
• [asterisk-dev] [Code Review] Fix transcode_via_sln and increase utility of PLC   Mark Michelson
• [asterisk-dev] [Code Review] Crash in audiohook_write_list   Tilghman Lesher
• [asterisk-dev] [Code Review] Crash in audiohook_write_list   David Vossel
• [asterisk-dev] Bugs/patches 16033 and 16590 ignored forever   François Delawarde
• [asterisk-dev] [Code Review] Crash in audiohook_write_list   Kaloyan Kovachev
• [asterisk-dev] [Code Review] Explicit context set in SIP peer overridden by default domain context   Nick Lewis
• [asterisk-dev] [Code Review] Explicit context set in SIP peer overridden by default domain context   Nick Lewis
• [asterisk-dev] [Code Review] Crash in audiohook_write_list   Tilghman Lesher
• [asterisk-dev] [Code Review] Crash in audiohook_write_list   Kaloyan Kovachev
• [asterisk-dev] [Code Review] Crash in audiohook_write_list   Kevin P. Fleming
• [asterisk-dev] [Code Review] Crash in audiohook_write_list   Kaloyan Kovachev
• [asterisk-dev] Bugs/patches 16033 and 16590 ignored forever   Paul Belanger
• [asterisk-dev] [Code Review] Crash in audiohook_write_list   David Vossel
• [asterisk-dev] [Code Review] Fix transcode_via_sln and increase utility of PLC   David Vossel
• [asterisk-dev] [Code Review] Fix transcode_via_sln and increase utility of PLC   David Vossel
• [asterisk-dev] [Code Review] External tests for func_srv and explicit SIP outgoing destination   Russell Bryant
• [asterisk-dev] [Code Review] Generic Advice of Charge   David Vossel
• [asterisk-dev] [Code Review] Fix transcode_via_sln and increase utility of PLC   Mark Michelson
• [asterisk-dev] Advice Of Charge ready for testing!   David Vossel
• [asterisk-dev] [Code Review] LibPRI changes for Generic AOC   David Vossel
• [asterisk-dev] Bugs/patches 16033 and 16590 ignored forever   Steve Totaro
• [asterisk-dev] Diameter Support   Alexandre Westfahl (アレクサンドル　ウエス
• [asterisk-dev] Bugs/patches 16033 and 16590 ignored forever   Philip A. Prindeville
• [asterisk-dev] Bugs/patches 16033 and 16590 ignored forever   Kirill 'Big K' Katsnelson
• [asterisk-dev] Bugs/patches 16033 and 16590 ignored forever   Kirill 'Big K' Katsnelson
• [asterisk-dev] Bugs/patches 16033 and 16590 ignored forever   Kirill 'Big K' Katsnelson
• [asterisk-dev] Andre Vo   ronyasad
• [asterisk-dev] (SOLUTION) CDR edge-case causes different results in 1.6.2   Kirill 'Big K' Katsnelson
• [asterisk-dev] [Code Review] Explicit context set in SIP peer overridden by default domain context   Nick Lewis
• [asterisk-dev] [Code Review] Parsing of reg->username containing @domain   Nick Lewis
• [asterisk-dev] (SOLUTION) CDR edge-case causes different results in 1.6.2   Steve Davies
• [asterisk-dev] [Code Review] Parsing of reg->username containing @domain   Nick Lewis
• [asterisk-dev] [Code Review] Explicit context set in SIP peer overridden by default domain context   Mark Michelson
• [asterisk-dev] Google Summer of Code 2010   Akshay Bhardwaj
• [asterisk-dev] (SOLUTION) CDR edge-case causes different results in 1.6.2   Kirill 'Big K' Katsnelson
• [asterisk-dev] [Code Review] Crash in audiohook_write_list   Mark Michelson
• [asterisk-dev] Question about AMD/WaitforSilence errors on EC2   Jimmy Song
• [asterisk-dev] Question about AMD/WaitforSilence errors on EC2   Jamuel Starkey
• [asterisk-dev] (SOLUTION) CDR edge-case causes different results in 1.6.2   Steve Davies
• [asterisk-dev] [Code Review] Extend pvt->relatedpeer support to INVITES in chan_sip.c   Nick Lewis
• [asterisk-dev] [Code Review] Extend pvt->relatedpeer support to INVITES in chan_sip.c   Olle E. Johansson
• [asterisk-dev] kindly asking for a bit of testing for a DTMF issue   François Delawarde
• [asterisk-dev] Unable to lookup '' - v1.6.x   Serge Berney
• [asterisk-dev] Unable to lookup '' - v1.6.x   Nick Lewis
• [asterisk-dev] Unable to lookup '' - v1.6.x   Paul Belanger
• [asterisk-dev] [Code Review] New manager actions for xmldoc   paul.belanger at polybeacon.com
• [asterisk-dev] [Code Review] Extend pvt->relatedpeer support toINVITES in chan_sip.c   Nick Lewis
• [asterisk-dev] Proposed changes to Originate Action [Issue: 0017221]   Ryan Bullock
• [asterisk-dev] Proposed changes to Originate Action [Issue: 0017221]   Timothy Rodriguez
• [asterisk-dev] Proposed changes to Originate Action [Issue: 0017221]   Ryan Bullock
• [asterisk-dev] Proposal for T.38 transparent gateway design in Asterisk   Kevin P. Fleming
• [asterisk-dev] [Code Review] Extend pvt->relatedpeer support to INVITES in chan_sip.c   Nick Lewis
• [asterisk-dev] Proposed changes to Originate Action [Issue: 0017221]   Timothy Rodriguez
• [asterisk-dev] [Code Review] Extend pvt->relatedpeer support to INVITES in chan_sip.c   Nick Lewis
• [asterisk-dev] Proposed changes to Originate Action [Issue: 0017221]   Ryan Bullock
• [asterisk-dev] [Code Review] Enabling Inband DTMF Passthrough for Appropriate Channel Types   dimas at dataart.com
• [asterisk-dev] [Code Review] Enabling Inband DTMF Passthrough for Appropriate Channel Types   paul.belanger at polybeacon.com
• [asterisk-dev] [Code Review] Add some CDR tests   Terry Wilson
• [asterisk-dev] [Code Review] CLI command 'manager show settings'   paul.belanger at polybeacon.com
• [asterisk-dev] [Code Review] CLI command 'manager show settings'   paul.belanger at polybeacon.com
• [asterisk-dev] [Code Review] CLI command 'manager show settings'   paul.belanger at polybeacon.com
• [asterisk-dev] Unable to lookup '' - v1.6.x   Serge Berney
• [asterisk-dev] Proposal for T.38 transparent gateway design in Asterisk   Klaus Darilion
• [asterisk-dev] Unable to lookup '' - v1.6.x   Tilghman Lesher
• [asterisk-dev] sound structure within Asterisk   Paul Belanger
• [asterisk-dev] [Code Review] CLI command 'manager show settings'   Mark Michelson
• [asterisk-dev] [Code Review] CLI command 'manager show settings'   Mark Michelson
• [asterisk-dev] sound structure within Asterisk   Tzafrir Cohen
• [asterisk-dev] audiohook with G722 codec: wrong number of samples   Simon Sester
• [asterisk-dev] [Code Review] CLI command 'manager show settings'   paul.belanger at polybeacon.com
• [asterisk-dev] [Code Review] CLI command 'manager show settings'   paul.belanger at polybeacon.com
• [asterisk-dev] audiohook with G722 codec: wrong number of samples   David Vossel
• [asterisk-dev] [Code Review] Crash in audiohook_write_list   David Vossel
• [asterisk-dev] [Code Review] CLI command 'manager show settings'   paul.belanger at polybeacon.com
• [asterisk-dev] [Code Review] MixMonitor external test   David Vossel
• [asterisk-dev] [Code Review] Enabling Inband DTMF Passthrough for Appropriate Channel Types   dimas at dataart.com
• [asterisk-dev] [Code Review] MixMonitor external test   David Vossel
• [asterisk-dev] [Code Review] MixMonitor external test   David Vossel
• [asterisk-dev] [Code Review] MixMonitor external test   David Vossel
• [asterisk-dev] [Code Review] Extend pvt->relatedpeer support to INVITES in chan_sip.c   Mark Michelson
• [asterisk-dev] [Code Review] MixMonitor external test   Russell Bryant
• [asterisk-dev] [Code Review] MixMonitor external test   Mark Michelson
• [asterisk-dev] [Code Review] Deadlock fixes for chan_local.c   David Vossel
• [asterisk-dev] [Code Review] SRTP support for Asterisk   Terry Wilson
• [asterisk-dev] [Code Review] MixMonitor external test   David Vossel
• [asterisk-dev] [Code Review] Deadlock fixes for chan_local.c   Russell Bryant
• [asterisk-dev] Proposal for T.38 transparent gateway design in Asterisk   Kevin P. Fleming
• [asterisk-dev] [Code Review] MixMonitor external test   David Vossel
• [asterisk-dev] [Code Review] MixMonitor external test   Russell Bryant
• [asterisk-dev] Proposal for T.38 transparent gateway design in Asterisk   Lee Howard
• [asterisk-dev] Proposal for T.38 transparent gateway design in Asterisk   Kevin P. Fleming
• [asterisk-dev] [Code Review] SRTP support for Asterisk   Mark Michelson
• [asterisk-dev] [Code Review] New manager actions for xmldoc   Russell Bryant
• [asterisk-dev] [Code Review] MixMonitor external test   David Vossel
• [asterisk-dev] [Code Review] CLI command 'manager show settings'   Russell Bryant
• [asterisk-dev] [Code Review] Parsing of reg->username containing @domain   David Vossel
• [asterisk-dev] [Code Review] Crash in audiohook_write_list   Tilghman Lesher
• [asterisk-dev] [Code Review] Crash in audiohook_write_list   David Vossel
• [asterisk-dev] [Code Review] Crash in audiohook_write_list   Tilghman Lesher
• [asterisk-dev] [Code Review] Crash in audiohook_write_list   Tilghman Lesher
• [asterisk-dev] [Asterisk 0017040]: [patch] Explicit context set in SIP peer overridden by default domain context   Philip Prindeville
• [asterisk-dev] [Code Review] SRTP support for Asterisk   Terry Wilson
• [asterisk-dev] [Code Review] SRTP support for Asterisk   Terry Wilson
• [asterisk-dev] Proposal for T.38 transparent gateway design in Asterisk   Klaus Darilion
• [asterisk-dev] Proposal for T.38 transparent gateway design in Asterisk   Klaus Darilion
• [asterisk-dev] [Code Review] New manager actions for xmldoc   Olle E. Johansson
• [asterisk-dev] [Code Review] Parsing of reg->username containing @domain   Nick Lewis
• [asterisk-dev] [Code Review] Explicit context set in SIP peer overridden by default domain context   Nick Lewis
• [asterisk-dev] Unable to lookup '' - v1.6.x   Nick Lewis
• [asterisk-dev] audiohook with G722 codec: wrong number of samples   Simon Sester
• [asterisk-dev] [Code Review] Extend pvt->relatedpeer support to INVITES in chan_sip.c   Nick Lewis
• [asterisk-dev] audiohook with G722 codec: wrong number of samples   Klaus Darilion
• [asterisk-dev] [Code Review] CLI command 'manager show settings'   Olle E. Johansson
• [asterisk-dev] [Code Review] CLI command 'manager show settings'   Russell Bryant
• [asterisk-dev] [Code Review] Crash in audiohook_write_list   David Vossel
• [asterisk-dev] [Code Review] CLI command 'manager show settings'   paul.belanger at polybeacon.com
• [asterisk-dev] [Code Review] Add some CDR tests   Russell Bryant
• [asterisk-dev] [Code Review] Add some CDR tests   Russell Bryant
• [asterisk-dev] [Code Review] Add some CDR tests   Terry Wilson
• [asterisk-dev] [Code Review] DTMF CallerID detection problems   rmudgett at digium.com
• [asterisk-dev] [Code Review] DTMF CallerID detection problems   rmudgett at digium.com
• [asterisk-dev] [Code Review] Parsing of reg->username containing @domain   David Vossel
• [asterisk-dev] [Code Review] DTMF CallerID detection problems   David Vossel
• [asterisk-dev] [Code Review] Add some CDR tests   Terry Wilson
• [asterisk-dev] [Code Review] Fix error handling in the recent freenum-in-a-can commit (issue 17263)   Philip A. Prindeville
• [asterisk-dev] [Code Review] MixMonitor external test   David Vossel
• [asterisk-dev] Proposal for T.38 transparent gateway design in Asterisk   Andreas Sikkema
• [asterisk-dev] Proposal for T.38 transparent gateway design in Asterisk   Andreas Sikkema
• [asterisk-dev] [Code Review] New manager actions for xmldoc   paul.belanger at polybeacon.com
• [asterisk-dev] [Code Review] New python module to facilitate monitoring manager events   Jeff Peeler
• [asterisk-dev] Proposal for T.38 transparent gateway design in Asterisk   Steve Underwood
• [asterisk-dev] [Code Review] Parsing of reg->username containing @domain   Nick Lewis
• [asterisk-dev] [Asterisk 0017040]: [patch] Explicit context set in SIP peer overridden by default domain context   Nick Lewis
• [asterisk-dev] [Code Review] Parsing of reg->username containing @domain   Nick Lewis
• [asterisk-dev] audiohook with G722 codec: wrong number of samples   Simon Sester
• [asterisk-dev] [Code Review] Fix error handling in the recent freenum-in-a-can commit (issue 17263)   Nick Lewis
• [asterisk-dev] get hold event   Bhrugu Mehta
• [asterisk-dev] Asterisk 1.4.31-rc2, 1.6.0.27-rc3, 1.6.1.19-rc3, and 1.6.2.7-rc3 Now Available   Asterisk Development Team
• [asterisk-dev] Unable to lookup '' - v1.6.x   Serge Berney
• [asterisk-dev] [Asterisk 0017040]: [patch] Explicit context set in SIP peer overridden by default domain context   Philip A. Prindeville
• [asterisk-dev] DTMF issues with 1.6.1.18   Bryant Zimmerman
• [asterisk-dev] DTMF issues with 1.6.1.18   Leif Madsen
• [asterisk-dev] DTMF issues with 1.6.1.18   Pavel Troller
• [asterisk-dev] [Code Review] Initial set of Queue tests   Mark Michelson
• [asterisk-dev] DTMF issues with 1.6.1.18   Benny Amorsen

Last message date: Fri Apr 30 18:16:19 CDT 2010
Archived on: Fri Apr 30 18:16:35 CDT 2010

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