November 2009 Archives by date

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Starting: Sun Nov 1 04:14:08 CST 2009
Ending: Mon Nov 30 18:37:30 CST 2009
Messages: 634

• [asterisk-dev] asterisk.org - where is doxygen?   Olle E. Johansson
• [asterisk-dev] [asterisk-users] IVR ..sip.conf:allowguest=yes   Alec Davis
• [asterisk-dev] [asterisk-users] IVR ..sip.conf:allowguest=yes   Klaus Darilion
• [asterisk-dev] New Community Developer: Max Khon   Nick Lewis
• [asterisk-dev] New Community Developer: Max Khon   Sean Bright
• [asterisk-dev] [Code Review] better SDP parsing algorithm for 1.4   Joshua Colp
• [asterisk-dev] [svn-commits] mnicholson: trunk r226687 - in /trunk: ./ channels/ configs/   Olle E. Johansson
• [asterisk-dev] Adding an option to the Dial app in 1.4   Matthew Nicholson
• [asterisk-dev] Adding an option to the Dial app in 1.4   Olle E. Johansson
• [asterisk-dev] Asterisk 1.6.2-rc2 suddenly restart   Eric Osvaldo R
• [asterisk-dev] [Code Review] better SDP parsing algorithm for 1.4   Matthew Nicholson
• [asterisk-dev] [Code Review] better SDP parsing algorithm for 1.4   Joshua Colp
• [asterisk-dev] [Code Review] better SDP parsing algorithm for 1.4   Matthew Nicholson
• [asterisk-dev] [Code Review] better SDP parsing algorithm for 1.4   Matthew Nicholson
• [asterisk-dev] [Code Review] better SDP parsing algorithm for trunk   Matthew Nicholson
• [asterisk-dev] [Code Review] better SDP parsing algorithm for 1.4   Matthew Nicholson
• [asterisk-dev] [Code Review] better SDP parsing algorithm for trunk   Matthew Nicholson
• [asterisk-dev] [Code Review] better SDP parsing algorithm for trunk   Matthew Nicholson
• [asterisk-dev] [Code Review] reworked chan_ooh323   Jeff Peeler
• [asterisk-dev] Adding an option to the Dial app in 1.4   Matthew Nicholson
• [asterisk-dev] [Code Review] better SDP parsing algorithm for trunk   Joshua Colp
• [asterisk-dev] Sound path   Philip A. Prindeville
• [asterisk-dev] Sound path   Nick Lewis
• [asterisk-dev] Sound path   Kevin P. Fleming
• [asterisk-dev] [Code Review] better SDP parsing algorithm for trunk   Matthew Nicholson
• [asterisk-dev] [Code Review] better SDP parsing algorithm for trunk   Matthew Nicholson
• [asterisk-dev] [Code Review] HELP - Find the bug :: Dialplan function for manager account checks - AMI_CLIENT()   Olle E Johansson
• [asterisk-dev] [Code Review] HELP - Find the bug :: Dialplan function for manager account checks - AMI_CLIENT()   Joshua Colp
• [asterisk-dev] [Code Review] Dialplan function for manager account checks - AMI_CLIENT()   Olle E Johansson
• [asterisk-dev] [Code Review] Dialplan function for manager account checks - AMI_CLIENT()   Olle E Johansson
• [asterisk-dev] New Community Developer: Max Khon   Philip A. Prindeville
• [asterisk-dev] Sound path   Philip A. Prindeville
• [asterisk-dev] New Community Developer: Max Khon   Kevin P. Fleming
• [asterisk-dev] Sound path   Kevin P. Fleming
• [asterisk-dev] [Code Review] Dialplan function for manager account checks - AMI_CLIENT()   Russell Bryant
• [asterisk-dev] New Community Developer: Max Khon   Odicha
• [asterisk-dev] New Community Developer: Max Khon   Kevin P. Fleming
• [asterisk-dev] Adding an option to the Dial app in 1.4   Russell Bryant
• [asterisk-dev] New Community Developer: Max Khon   Tzafrir Cohen
• [asterisk-dev] [Code Review] Expand availability of codec bits from 32 to 64   Tilghman Lesher
• [asterisk-dev] [Code Review] Expand availability of codec bits from 32 to 64   Tilghman Lesher
• [asterisk-dev] [Code Review] Expand availability of codec bits from 32 to 64   Tilghman Lesher
• [asterisk-dev] [Code Review] Initial review request for CCSS architecture and generic implementations   Mark Michelson
• [asterisk-dev] [Code Review] Expand availability of codec bits from 32 to 64   Mark Michelson
• [asterisk-dev] New Community Developer: Max Khon   Odicha
• [asterisk-dev] slinear at dahdi   Odicha
• [asterisk-dev] [Code Review] Dialplan function for manager account checks - AMI_CLIENT()   Olle E Johansson
• [asterisk-dev] res_pktccops   Olle E. Johansson
• [asterisk-dev] Sound path   Benny Amorsen
• [asterisk-dev] Sound path   Philipp Kempgen
• [asterisk-dev] Adding an option to the Dial app in 1.4   Matthew Nicholson
• [asterisk-dev] Sound path   Olle E. Johansson
• [asterisk-dev] Sound path   Tzafrir Cohen
• [asterisk-dev] Sound path   Nick Lewis
• [asterisk-dev] memory leak with static users   Gianpietro Germi
• [asterisk-dev] Sound path   Tzafrir Cohen
• [asterisk-dev] Sound path   Nick Lewis
• [asterisk-dev] Sound path   Tzafrir Cohen
• [asterisk-dev] [Code Review] Initial review request for CCSS architecture and generic implementations   Mark Michelson
• [asterisk-dev] [Code Review] reworked chan_ooh323   Jeff Peeler
• [asterisk-dev] [Code Review] reworked chan_ooh323   Jeff Peeler
• [asterisk-dev] AST-2009-008: SIP responses expose valid usernames   Asterisk Security Team
• [asterisk-dev] AST-2009-009: Cross-site AJAX request vulnerability   Asterisk Security Team
• [asterisk-dev] Asterisk 1.2.36, 1.4.26.3, 1.6.0.17, and 1.6.1.9 Now Available   Asterisk Development Team
• [asterisk-dev] res_pktccops   Nahuel Greco
• [asterisk-dev] Wideband and DSP   Dmitry Andrianov
• [asterisk-dev] [Asterisk:queue] Hold Time report (reportholdtime) issue Testing   matthieu Nicaise
• [asterisk-dev] memory leak with static users   stefania magistrali
• [asterisk-dev] channel destruction after a transfer call   stefania magistrali
• [asterisk-dev] memory leak with static users   Tzafrir Cohen
• [asterisk-dev] modules.conf - loading modules   Olle E. Johansson
• [asterisk-dev] modules.conf - loading modules   Kevin P. Fleming
• [asterisk-dev] modules.conf - loading modules   Russell Bryant
• [asterisk-dev] modules.conf - loading modules   Chris Tooley
• [asterisk-dev] modules.conf - loading modules   Nick Lewis
• [asterisk-dev] modules.conf - loading modules   Olle E. Johansson
• [asterisk-dev] modules.conf - loading modules   Olle E. Johansson
• [asterisk-dev] modules.conf - loading modules   Tzafrir Cohen
• [asterisk-dev] modules.conf - loading modules   Olle E. Johansson
• [asterisk-dev] modules.conf - loading modules   Kevin P. Fleming
• [asterisk-dev] modules.conf - loading modules   Nick Lewis
• [asterisk-dev] modules.conf - loading modules   Tzafrir Cohen
• [asterisk-dev] modules.conf - loading modules   Tilghman Lesher
• [asterisk-dev] modules.conf - loading modules   Nick Lewis
• [asterisk-dev] modules.conf - loading modules   Nick Lewis
• [asterisk-dev] modules.conf - loading modules   Kevin P. Fleming
• [asterisk-dev] modules.conf - loading modules   Olle E. Johansson
• [asterisk-dev] modules.conf - loading modules   Kevin P. Fleming
• [asterisk-dev] memory leak with static users   Gianpietro Germi
• [asterisk-dev] AMI command for checking SIP channel QoS (proposal)   Olle E. Johansson
• [asterisk-dev] AMI command for checking SIP channel QoS (proposal)   Olle E. Johansson
• [asterisk-dev] BLF Issues   Bryant Zimmerman
• [asterisk-dev] AMI command for checking SIP channel QoS (proposal)   Klaus Darilion
• [asterisk-dev] AMI command for checking SIP channel QoS (proposal)   Olle E. Johansson
• [asterisk-dev] AMI command for checking SIP channel QoS (proposal)   Klaus Darilion
• [asterisk-dev] [Code Review] data api gsoc2009   Tilghman Lesher
• [asterisk-dev] Asterisk 1.4.27-rc3, 1.6.0.18-rc1, 1.6.1.10-rc1, and 1.6.2.0-rc4 Now Available   Asterisk Development Team
• [asterisk-dev] Fixing bug #14718   Faidon Liambotis
• [asterisk-dev] Confused with release numbers, how to manage incremental patching ?   Pavel Troller
• [asterisk-dev] 1.6.1.10-rc1 crash on a VOLUME() func   Pavel Troller
• [asterisk-dev] [svn-commits] rmudgett: trunk r228691 - in /trunk: ./ channels/   Tzafrir Cohen
• [asterisk-dev] dahdi_dynamic_ethmf was: [Re: [svn-commits] fjoe: freebsd/trunk r7533 - /freebsd/trunk/drivers/dahdi/dahdi_dynamic_ethmf.c]   Tzafrir Cohen
• [asterisk-dev] [svn-commits] rmudgett: trunk r228691 - in /trunk: ./ channels/   Kevin P. Fleming
• [asterisk-dev] extra DAHDI dial options [was: Re: [svn-commits] rmudgett: trunk r228691 - in /trunk: ./ channels/]   Tzafrir Cohen
• [asterisk-dev] extra DAHDI dial options [was: Re: [svn-commits] rmudgett: trunk r228691 - in /trunk: ./ channels/]   Tilghman Lesher
• [asterisk-dev] ast_cdr_set_var() doesn't work in 1.4.X   Peter
• [asterisk-dev] 1.6.1.10-rc1 crash on a VOLUME() func   Kaloyan Kovachev
• [asterisk-dev] 1.6.1.10-rc1 crash on a VOLUME() func   Kaloyan Kovachev
• [asterisk-dev] 1.6.1.10-rc1 crash on a VOLUME() func   Pavel Troller
• [asterisk-dev] STANDALONE define magic ?   Pavel Troller
• [asterisk-dev] STANDALONE define magic ?   Jonathan Thurman
• [asterisk-dev] STANDALONE define magic ?   Tilghman Lesher
• [asterisk-dev] STANDALONE define magic ?   Pavel Troller
• [asterisk-dev] SIP Session-timers and bug 15621   Morten Isaksen
• [asterisk-dev] [asterisk-commits] tilghman: trunk r228798 - in /trunk: addons/channels/ main/ res/   Alec Davis
• [asterisk-dev] AMI command for checking SIP channel QoS (proposal)   Floimair Florian
• [asterisk-dev] Problem with *1.6.0.5; "Skipping dialing interface since it has already been dialed"   Håkon Nessjøen
• [asterisk-dev] 1.6.1.10-rc1 crash on a VOLUME() func   Kaloyan Kovachev
• [asterisk-dev] 1.6.1.10-rc1 crash on a VOLUME() func   Pavel Troller
• [asterisk-dev] Dear asterisk-dev at lists.digium.com 77% 0FF on Pfizer.   © VIAGRA ® Official Site
• [asterisk-dev] 1.6.1.10-rc1 crash on a VOLUME() func   Pavel Troller
• [asterisk-dev] 1.6.1.10-rc1 crash on a VOLUME() func   Russell Bryant
• [asterisk-dev] SIP Session-timers and bug 15621   Russell Bryant
• [asterisk-dev] [asterisk-commits] tilghman: trunk r228798 - in /trunk: addons/channels/ main/ res/   Tilghman Lesher
• [asterisk-dev] 1.6.1.10-rc1 crash on a VOLUME() func   Pavel Troller
• [asterisk-dev] [asterisk-commits] tilghman: trunk r228798 - in /trunk: addons/channels/ main/ res/   Mark Michelson
• [asterisk-dev] 1.6.1.10-rc1 crash on a VOLUME() func   Tzafrir Cohen
• [asterisk-dev] 1.6.1.10-rc1 crash on a VOLUME() func   Tzafrir Cohen
• [asterisk-dev] Creating a FastAGI equivalent for realtime configs   Alistair Cunningham
• [asterisk-dev] Creating a FastAGI equivalent for realtime configs   Alistair Cunningham
• [asterisk-dev] Short Introduction   David Ruggles
• [asterisk-dev] Creating a FastAGI equivalent for realtime configs   Tilghman Lesher
• [asterisk-dev] 1.6.1.10-rc1 crash on a VOLUME() func   Russell Bryant
• [asterisk-dev] Creating a FastAGI equivalent for realtime configs   Alistair Cunningham
• [asterisk-dev] Asterisk 1.4.27-rc4, 1.6.0.18-rc2, 1.6.1.10-rc2, and 1.6.2.0-rc5 Now Available   Asterisk Development Team
• [asterisk-dev] 1.6.1.10-rc1 crash on a VOLUME() func   Leif Madsen
• [asterisk-dev] Short Introduction   Clod Patry
• [asterisk-dev] SIP Session-timers and bug 15621   Leif Madsen
• [asterisk-dev] Short Introduction   Leif Madsen
• [asterisk-dev] 0016211: [patch] Event collision in ExternalIVR resolved by documenting issue   David Ruggles
• [asterisk-dev] Authentication required after successful login   Stephan van der Walt
• [asterisk-dev] why do we need develop 1.6.0, x and 1.6.1.x at the same time?   lvsha
• [asterisk-dev] why do we need develop 1.6.0, x and 1.6.1.x at the same time?   Saúl Ibarra
• [asterisk-dev] Collecting and reporting bridged dialling information   Pavel Troller
• [asterisk-dev] Dear asterisk-dev at lists.digium.com 73% 0FF on Pfizer.   © VIAGRA ® Official Site
• [asterisk-dev] Collecting and reporting bridged dialling information   Nic Colledge
• [asterisk-dev] why does Dial() not inhert the language setting to the outgoing channel?   Klaus Darilion
• [asterisk-dev] why does Dial() not inhert the language setting to the outgoing channel?   Russell Bryant
• [asterisk-dev] Master subversion server planned maintenance   Russell Bryant
• [asterisk-dev] Collecting and reporting bridged dialling information   Pavel Troller
• [asterisk-dev] why does Dial() not inhert the language setting to the outgoing channel?   Klaus Darilion
• [asterisk-dev] 0016123: [patch] ExternalIVR trapping non-existent files does not work   David Ruggles
• [asterisk-dev] why does Dial() not inhert the language setting to the outgoing channel?   Russell Bryant
• [asterisk-dev] 0015504: [patch] G726 Codec has choppy audio on Version 1.6.1   Dmitry Andrianov
• [asterisk-dev] 0015504: [patch] G726 Codec has choppy audio on Version 1.6.1   Joshua Colp
• [asterisk-dev] ghenry: trunk r229050 - /trunk/contrib/scripts/asterisk.ldap-schema   Russell Bryant
• [asterisk-dev] QUEUE_MEMBER(queue, free) ignores agents in wrap-up   Kirill 'Big K' Katsnelson
• [asterisk-dev] 0015504: [patch] G726 Codec has choppy audio on Version 1.6.1   Johann Steinwendtner
• [asterisk-dev] QUEUE_MEMBER(queue, free) ignores agents in wrap-up   Atis Lezdins
• [asterisk-dev] QUEUE_MEMBER(queue, free) ignores agents in wrap-up   Kirill 'Big K' Katsnelson
• [asterisk-dev] ghenry: trunk r229050 - /trunk/contrib/scripts/asterisk.ldap-schema   Leif Madsen
• [asterisk-dev] strange "new-callerid" on DAHDI   Klaus Darilion
• [asterisk-dev] [Code Review] StopMixMonitor not stopping mixmonitor thread   David Vossel
• [asterisk-dev] [asterisk-users] IVR ..sip.conf:allowguest=yes   Tilghman Lesher
• [asterisk-dev] Security Request for discussion: Should sip.conf allowguest=yes be the default   Alec Davis
• [asterisk-dev] [asterisk-users] IVR ..sip.conf:allowguest=yes   Alec Davis
• [asterisk-dev] [Asterisk 0013405]: [patch] T38 gateway (fwd)   marek cervenka
• [asterisk-dev] Security Request for discussion: Should sip.conf allowguest=yes be the default   Olle E. Johansson
• [asterisk-dev] Security Request for discussion: Should sip.conf allowguest=yes be the default   Alec Davis
• [asterisk-dev] [asterisk-users] IVR ..sip.conf:allowguest=yes   Olle E. Johansson
• [asterisk-dev] strange "new-callerid" on DAHDI   Tony Mountifield
• [asterisk-dev] Security Request for discussion: Should sip.conf allowguest=yes be the default   Tzafrir Cohen
• [asterisk-dev] Security Request for discussion: Should sip.conf allowguest=yes be the default   Olle E. Johansson
• [asterisk-dev] How we can use BV Codec in Asterisk?   alaa_fahham at yahoo.com
• [asterisk-dev] How we can use BV Codec in Asterisk?   Olle E. Johansson
• [asterisk-dev] Security Request for discussion: Should sip.conf allowguest=yes be the default   Michiel van Baak
• [asterisk-dev] Security Request for discussion: Should sip.conf allowguest=yes be the default   Kai Hoerner
• [asterisk-dev] How we can use BV Codec in Asterisk?   Tzafrir Cohen
• [asterisk-dev] Security Request for discussion: Should sip.conf allowguest=yes be the default   Kai Hoerner
• [asterisk-dev] Security Request for discussion: Should sip.conf allowguest=yes be the default   Michiel van Baak
• [asterisk-dev] Security Request for discussion: Should sip.conf allowguest=yes be the default   Kaloyan Kovachev
• [asterisk-dev] [Code Review] Add a new option "require" to modules.conf to make Asterisk fail if a module does not load   Olle E Johansson
• [asterisk-dev] Security Request for discussion: Should sip.conf allowguest=yes be the default   Jared Smith
• [asterisk-dev] Security Request for discussion: Should sip.conf allowguest=yes be the default   Leif Madsen
• [asterisk-dev] Security Request for discussion: Should sip.conf allowguest=yes be the default   Michiel van Baak
• [asterisk-dev] Security Request for discussion: Should sip.conf allowguest=yes be the default   Tilghman Lesher
• [asterisk-dev] Security Request for discussion: Should sip.conf allowguest=yes be the default   Jared Smith
• [asterisk-dev] How we can use BV Codec in Asterisk?   Tilghman Lesher
• [asterisk-dev] RES: [Code Review] Add a new option "require" to modules.conf to make Asterisk fail if a module does not load   Olle E. Johansson
• [asterisk-dev] RES: [Code Review] Add a new option "require" to modules.conf to make Asterisk fail if a module does not load   Rafael Prado Rocchi
• [asterisk-dev] [Code Review] Add a new option "require" to modules.conf to make Asterisk fail if a module does not load   Mark Michelson
• [asterisk-dev] [Code Review] Add a new option "require" to modules.conf to make Asterisk fail if a module does not load   Olle E. Johansson
• [asterisk-dev] How we can use BV Codec in Asterisk?   John Todd
• [asterisk-dev] sdpversion handling   Freddi Hansen
• [asterisk-dev] asterisk log file analyses   Wolfgang Pichler
• [asterisk-dev] Security Request for discussion: Should sip.conf allowguest=yes be the default   Michiel van Baak
• [asterisk-dev] Security Request for discussion: Should sip.conf allowguest=yes be the default   Alexandre Cavalcante Alencar
• [asterisk-dev] Security Request for discussion: Should sip.conf allowguest=yes be the default   Alexander Harrowell
• [asterisk-dev] Security Request for discussion: Should sip.conf allowguest=yes be the default   Atis Lezdins
• [asterisk-dev] asterisk log file analyses   Atis Lezdins
• [asterisk-dev] Security Request for discussion: Should sip.conf allowguest=yes be the default   Tzafrir Cohen
• [asterisk-dev] Security Request for discussion: Should sip.conf allowguest=yes be the default   Tilghman Lesher
• [asterisk-dev] [Code Review] AST-33: Create a list of channel variables to be posted within AMI call events   Tilghman Lesher
• [asterisk-dev] [Code Review] AST-33: Create a list of channel variables to be posted within AMI call events   Tilghman Lesher
• [asterisk-dev] asterisk log file analyses   Wolfgang Pichler
• [asterisk-dev] Security Request for discussion: Should sip.conf allowguest=yes be the default   Atis Lezdins
• [asterisk-dev] Security Request for discussion: Should sip.conf allowguest=yes be the default   Jeff LaCoursiere
• [asterisk-dev] asterisk log file analyses   Gregory Boehnlein
• [asterisk-dev] Security Request for discussion: Should sip.conf allowguest=yes be the default   Tilghman Lesher
• [asterisk-dev] asterisk log file analyses   Wolfgang Pichler
• [asterisk-dev] asterisk log file analyses   Gregory Boehnlein
• [asterisk-dev] Security Request for discussion: Should sip.conf allowguest=yes be the default   Michiel van Baak
• [asterisk-dev] [Code Review] app_queue: Makes call_queue objects ao2 objects so ref counting can be done   David Vossel
• [asterisk-dev] asterisk log file analyses   Russell Bryant
• [asterisk-dev] 0015504: [patch] G726 Codec has choppy audio on Version 1.6.1   Dmitry Andrianov
• [asterisk-dev] compiling Dahdi on blackfin   Dan Amarandei
• [asterisk-dev] How we can use BV Codec in Asterisk?   Vincent Nonnenmacher
• [asterisk-dev] compiling Dahdi on blackfin   Tzafrir Cohen
• [asterisk-dev] sdpversion handling   Olle E. Johansson
• [asterisk-dev] Security Request for discussion: Should sip.conf allowguest=yes be the default   Kaloyan Kovachev
• [asterisk-dev] How we can use BV Codec in Asterisk?   Nick Lewis
• [asterisk-dev] Security Request for discussion: Should sip.conf allowguest=yes be the default   Kai Hoerner
• [asterisk-dev] Security Request for discussion: Should sip.conf allowguest=yes be the default   Kai Hoerner
• [asterisk-dev] asterisk log file analyses   Kai Hoerner
• [asterisk-dev] Security Request for discussion: Should sip.conf allowguest=yes be the default   Tzafrir Cohen
• [asterisk-dev] Security Request for discussion: Should sip.conf allowguest=yes be the default   Kaloyan Kovachev
• [asterisk-dev] Call answer problem   Rafael Seste
• [asterisk-dev] 1.6.2 beta: Realtime SIP and periodic database queries   Gunnar Schaller
• [asterisk-dev] 1.6.2 beta: Realtime SIP and periodic database queries   Tilghman Lesher
• [asterisk-dev] Review ExternalIVR cleanup   David Ruggles
• [asterisk-dev] [Code Review] AST-33: Create a list of channel variables to be posted within AMI call events   Russell Bryant
• [asterisk-dev] Asterisk Release Candidates 1.4.27-rc5, 1.6.0.18-rc3, 1.6.1.10-rc3, and 1.6.2.0-rc5 Now Available   Asterisk Development Team
• [asterisk-dev] [Code Review] AST-33: Create a list of channel variables to be posted within AMI call events   Tilghman Lesher
• [asterisk-dev] How to get SIP packet source ip address in asterisk application   ast guy
• [asterisk-dev] [Code Review] AST-33: Create a list of channel variables to be posted within AMI call events   Russell Bryant
• [asterisk-dev] 0015504: [patch] G726 Codec has choppy audio on Version 1.6.1   Johann Steinwendtner
• [asterisk-dev] Channel support for GSM devices   Odicha
• [asterisk-dev] Dear asterisk-dev at lists.digium.com 77% 0FF on Pfizer.   © VIAGRA ® Official Site
• [asterisk-dev] 0015504: [patch] G726 Codec has choppy audio on Version 1.6.1   Dmitry Andrianov
• [asterisk-dev] compiling Dahdi on blackfin   Dan Amarandei
• [asterisk-dev] How to get SIP packet source ip address in asterisk application   Kai Hoerner
• [asterisk-dev] compiling Dahdi on blackfin   Kai Hoerner
• [asterisk-dev] How we can use BV Codec in Asterisk?   Wolfgang Pichler
• [asterisk-dev] Security Request for discussion: Should sip.conf allowguest=yes be the default   Kai Hoerner
• [asterisk-dev] Security Request for discussion: Should sip.conf allowguest=yes be the default   Olle E. Johansson
• [asterisk-dev] Security Request for discussion: Should sip.conf allowguest=yes be the default   Kai Hoerner
• [asterisk-dev] [Code Review] Basic app_externalivr clean up   thedavidfactor at gmail.com
• [asterisk-dev] [Code Review] Basic app_externalivr clean up   Russell Bryant
• [asterisk-dev] [Code Review] Basic app_externalivr clean up   thedavidfactor at gmail.com
• [asterisk-dev] Hints failing ot update for caller side of the channel   Bryant Zimmerman
• [asterisk-dev] [Code Review] StopMixMonitor not stopping mixmonitor thread   Russell Bryant
• [asterisk-dev] How we can use BV Codec in Asterisk?   Tilghman Lesher
• [asterisk-dev] Conflict of cs/cz language codes   Pavel Troller
• [asterisk-dev] How we can use BV Codec in Asterisk?   Nick Lewis
• [asterisk-dev] How we can use BV Codec in Asterisk?   Steve Edwards
• [asterisk-dev] How we can use BV Codec in Asterisk?   Nick Lewis
• [asterisk-dev] [Code Review] Basic app_externalivr clean up   thedavidfactor at gmail.com
• [asterisk-dev] How we can use BV Codec in Asterisk?   Steve Howes
• [asterisk-dev] Conflict of cs/cz language codes   Steve Murphy
• [asterisk-dev] Hints failing ot update for caller side of the channel   John Todd
• [asterisk-dev] Conflict of language codes   John Todd
• [asterisk-dev] [Code Review] Prevent duplicate entries in format strings in app_voicemail.   Matthew Nicholson
• [asterisk-dev] [Code Review] Prevent duplicate entries in format strings used in app_voicemail   Matthew Nicholson
• [asterisk-dev] Conflict of language codes   Steve Murphy
• [asterisk-dev] Security Request for discussion: Should sip.conf allowguest=yes be the default   Alec Davis
• [asterisk-dev] Security Request for discussion: Should sip.conf allowguest=yes be the default   Atis Lezdins
• [asterisk-dev] why do we need develop 1.6.0, x and 1.6.1.x at the same time?   lvsha
• [asterisk-dev] How to get SIP packet source ip address in asterisk application   Alex Massover
• [asterisk-dev] Hints failing ot update for caller side of the channel   Olle E. Johansson
• [asterisk-dev] How to get SIP packet source ip address in asterisk application   santoshchintalwar at gmail.com
• [asterisk-dev] How we can use BV Codec in Asterisk?   Nick Lewis
• [asterisk-dev] How to get SIP packet source ip address in asterisk application   Olle E. Johansson
• [asterisk-dev] Security Request for discussion: Should sip.confallowguest=yes be the default   Nick Lewis
• [asterisk-dev] Security Request for discussion: Should sip.confallowguest=yes be the default   Olle E. Johansson
• [asterisk-dev] why do we need develop 1.6.0, x and 1.6.1.x at the same time?   Olle E. Johansson
• [asterisk-dev] "matched" versus "authenticated"   Nick Lewis
• [asterisk-dev] Security Request for discussion: Should sip.conf allowguest=yes be the default   Tzafrir Cohen
• [asterisk-dev] compiling Dahdi on blackfin   Tzafrir Cohen
• [asterisk-dev] compiling Dahdi on blackfin   Tzafrir Cohen
• [asterisk-dev] Security Request for discussion: Should sip.conf allowguest=yes be the default   Atis Lezdins
• [asterisk-dev] Security Request for discussion: Should sip.conf allowguest=yes be the default   Tzafrir Cohen
• [asterisk-dev] Security Request for discussion: Should sip.conf allowguest=yes be the default   Atis Lezdins
• [asterisk-dev] Dear asterisk-dev at lists.digium.com 72% 0FF on Pfizer.   © VIAGRA ® Official Site
• [asterisk-dev] Security Request for discussion: Should sip.conf allowguest=yes be the default   Chris Lee
• [asterisk-dev] why do we need develop 1.6.0, x and 1.6.1.x at the same time?   Russell Bryant
• [asterisk-dev] Security Request for discussion: Should sip.conf allowguest=yes be the default   Tzafrir Cohen
• [asterisk-dev] Security Request for discussion: Should sip.conf allowguest=yes be the default   Kai Hoerner
• [asterisk-dev] Security Request for discussion: Should sip.conf allowguest=yes be the default   Tzafrir Cohen
• [asterisk-dev] Security Request for discussion: Should sip.conf allowguest=yes be the default   Kai Hoerner
• [asterisk-dev] Security Request for discussion: Should sip.conf allowguest=yes be the default   Atis Lezdins
• [asterisk-dev] Security Request for discussion: Should sip.conf allowguest=yes be the default   Alec Davis
• [asterisk-dev] Security Request for discussion: Should sip.conf allowguest=yes be the default   Alexandre Cavalcante Alencar
• [asterisk-dev] Security Request for discussion: Should sip.conf allowguest=yes be the default   Tzafrir Cohen
• [asterisk-dev] Security Request for discussion: Should sip.conf allowguest=yes be the default   Olle E. Johansson
• [asterisk-dev] Security Request for discussion: Should sip.conf allowguest=yes be the default   Olle E. Johansson
• [asterisk-dev] IAX Connections Between 1.4, 1.6 and 1.2   Wolfgang Pichler
• [asterisk-dev] compiling Dahdi on blackfin   Dan Amarandei
• [asterisk-dev] Mantis bugs seen as Feature request, when it may only be an issue.   Alec Davis
• [asterisk-dev] Mantis bugs seen as Feature request, when it may only be an issue.   Olle E. Johansson
• [asterisk-dev] 1.6.2 beta: Realtime SIP and periodic database queries   Gunnar Schaller
• [asterisk-dev] Security Request for discussion: Should sip.confallowguest=yes be the default   Alec Davis
• [asterisk-dev] Can asterisk PRI/BRI support redirect calls   Alec Davis
• [asterisk-dev] Security Request for discussion: Should sip.conf allowguest=yes be the default   Kai Hoerner
• [asterisk-dev] Can asterisk PRI/BRI support redirect calls   Saúl Ibarra
• [asterisk-dev] Missing CDR   Daniel A. Veiga
• [asterisk-dev] [asterisk-users] asterisk cdr - remote ip address (fwd)   marek cervenka
• [asterisk-dev] [asterisk-users] asterisk cdr - remote ip address (fwd)   Olle E. Johansson
• [asterisk-dev] Missing CDR   Kai Hoerner
• [asterisk-dev] Changing internal_timing to default to yes?   Leif Madsen
• [asterisk-dev] Clearmode CODEC   jarmenio
• [asterisk-dev] r227824 breaks chan_iax2   Michiel van Baak
• [asterisk-dev] Security Request for discussion: Should sip.conf allowguest=yes be the default   Olle E. Johansson
• [asterisk-dev] Security Request for discussion: Should sip.conf allowguest=yes be the default   Jared Smith
• [asterisk-dev] Security Request for discussion: Should sip.conf allowguest=yes be the default   Miguel Molina
• [asterisk-dev] [asterisk-commits] ghenry: trunk r229050 - /trunk/contrib/scripts/asterisk.ldap-schema   Gavin Henry
• [asterisk-dev] Troubles with Asterisk and chan_h323   Mark Best
• [asterisk-dev] Troubles with Asterisk and chan_h323   Kevin P. Fleming
• [asterisk-dev] Troubles with Asterisk and chan_h323   Mark Best
• [asterisk-dev] r227824 breaks chan_iax2   Russell Bryant
• [asterisk-dev] Changing internal_timing to default to yes?   Russell Bryant
• [asterisk-dev] Missing CDR   Daniel A. Veiga
• [asterisk-dev] IAX Connections Between 1.4, 1.6 and 1.2   Russell Bryant
• [asterisk-dev] Missing CDR   Russell Bryant
• [asterisk-dev] [Code Review] Basic app_externalivr clean up   Russell Bryant
• [asterisk-dev] compiling Dahdi on blackfin   Dan Amarandei
• [asterisk-dev] compiling Dahdi on blackfin   Tzafrir Cohen
• [asterisk-dev] r227824 breaks chan_iax2   Michiel van Baak
• [asterisk-dev] Can asterisk PRI/BRI support redirect calls   Alec Davis
• [asterisk-dev] snuffy: branch snuffy/func_memcache r230433 - in /team/snuffy/func_memcache: ...   Tony Mountifield
• [asterisk-dev] snuffy: branch snuffy/func_memcache r230433 - in /team/snuffy/func_memcache: ...   Olle E. Johansson
• [asterisk-dev] GSM trunks & Dahdi - New funcionality Request for discussion   Jose A. Deniz
• [asterisk-dev] GSM trunks & Dahdi - New funcionality Request for discussion   Kai Hoerner
• [asterisk-dev] CNG fax detection on RTP   Kevin P. Fleming
• [asterisk-dev] Can asterisk PRI/BRI support redirect calls   Don Kelly
• [asterisk-dev] Noise generation   Olle E. Johansson
• [asterisk-dev] GSM trunks & Dahdi - New funcionality Request for discussion   Tzafrir Cohen
• [asterisk-dev] Noise generation   Kevin P. Fleming
• [asterisk-dev] Noise generation   Steve Underwood
• [asterisk-dev] GSM trunks & Dahdi - New funcionality Request for discussion   Odicha
• [asterisk-dev] GSM trunks & Dahdi - New funcionality Request for discussion   Odicha
• [asterisk-dev] GSM trunks & Dahdi - New funcionality Request for discussion   Saúl Ibarra
• [asterisk-dev] GSM trunks & Dahdi - New funcionality Request for discussion   Kai Hoerner
• [asterisk-dev] Asterisk 1.4.27, 1.6.0.18, and 1.6.1.10 Now Available   Asterisk Development Team
• [asterisk-dev] GSM trunks & Dahdi - New funcionality Request for discussion   Odicha
• [asterisk-dev] GSM trunks & Dahdi - New funcionality Request for discussion   Tzafrir Cohen
• [asterisk-dev] GSM trunks & Dahdi - New funcionality Request for discussion   Odicha
• [asterisk-dev] Confused with release numbers, how to manage incremental patching ?   Pavel Troller
• [asterisk-dev] GSM trunks & Dahdi - New funcionality Request for discussion   Kai Hoerner
• [asterisk-dev] [Code Review] Implement error events for non-existant files   thedavidfactor at gmail.com
• [asterisk-dev] GSM trunks & Dahdi - New funcionality Request for discussion   Tzafrir Cohen
• [asterisk-dev] ConfBridge pin auth   Brendan Martens
• [asterisk-dev] ConfBridge pin auth   Jeffrey Ollie
• [asterisk-dev] ConfBridge pin auth   Dmitry Andrianov
• [asterisk-dev] [asterisk-users] asterisk cdr - remote ip address (fwd)   marek cervenka
• [asterisk-dev] [asterisk-users] asterisk cdr - remote ip address (fwd)   Olle E. Johansson
• [asterisk-dev] [Code Review] Avoid deadlocks in chan_sip - 1.6.0   Olle E Johansson
• [asterisk-dev] GSM trunks & Dahdi - New funcionality Request for discussion   Artem Makhutov
• [asterisk-dev] GSM trunks & Dahdi - New funcionality Request for discussion   Odicha
• [asterisk-dev] GSM trunks & Dahdi - New funcionality Request for discussion   Odicha
• [asterisk-dev] GSM trunks & Dahdi - New funcionality Request for discussion   Odicha
• [asterisk-dev] GSM trunks & Dahdi - New funcionality Request for discussion   Artem Makhutov
• [asterisk-dev] GSM trunks & Dahdi - New funcionality Request for discussion   Odicha
• [asterisk-dev] [Code Review] Avoid deadlocks in chan_sip - 1.6.0   Russell Bryant
• [asterisk-dev] Can asterisk PRI/BRI support redirect calls   Richard Mudgett
• [asterisk-dev] [Code Review] Avoid deadlocks in chan_sip - 1.6.0   Russell Bryant
• [asterisk-dev] [Code Review] Implement error events for non-existant files   Russell Bryant
• [asterisk-dev] Confused with release numbers, how to manage incremental patching ?   Russell Bryant
• [asterisk-dev] New committer - odicha   Russell Bryant
• [asterisk-dev] [Code Review] audiohook: signal to trigger on every status change   David Vossel
• [asterisk-dev] [Code Review] audiohook: signal to trigger on every status change   Russell Bryant
• [asterisk-dev] Can asterisk PRI/BRI support redirect calls   Alec Davis
• [asterisk-dev] Can asterisk PRI/BRI support redirect calls   Richard Mudgett
• [asterisk-dev] Can asterisk PRI/BRI support redirect calls   Don Kelly
• [asterisk-dev] Meetme - talk optimization flag = false   Chandrakant Solanki
• [asterisk-dev] Can asterisk PRI/BRI support redirect calls   Alec Davis
• [asterisk-dev] [Code Review] Avoid deadlocks in chan_sip - 1.6.0   Olle E Johansson
• [asterisk-dev] /team/group/CCSS/channels/chan_sip.c - PUBLISH   Olle E. Johansson
• [asterisk-dev] GSM trunks & Dahdi - New funcionality Request for discussion   Artem Makhutov
• [asterisk-dev] /team/group/CCSS/channels/chan_sip.c - PUBLISH   Nick Lewis
• [asterisk-dev] /team/group/CCSS/channels/chan_sip.c - PUBLISH   Olle E. Johansson
• [asterisk-dev] /team/group/CCSS/channels/chan_sip.c - PUBLISH   Nick Lewis
• [asterisk-dev] GSM trunks & Dahdi - New funcionality Request for discussion   Kai Hoerner
• [asterisk-dev] GSM trunks & Dahdi - New funcionality Request for discussion   Odicha
• [asterisk-dev] GSM trunks & Dahdi - New funcionality Request for discussion   Odicha
• [asterisk-dev] [asterisk-users] asterisk cdr - remote ip address (fwd)   Kai Hoerner
• [asterisk-dev] Can asterisk PRI/BRI support redirect calls   Jared Smith
• [asterisk-dev] Can asterisk PRI/BRI support redirect calls   Richard Mudgett
• [asterisk-dev] Can asterisk PRI/BRI support redirect calls   Richard Mudgett
• [asterisk-dev] Meetme - talk optimization flag = false   Russell Bryant
• [asterisk-dev] [Code Review] Avoid deadlocks in chan_sip - 1.6.0   Russell Bryant
• [asterisk-dev] GSM trunks & Dahdi - New funcionality Request for discussion   Artem Makhutov
• [asterisk-dev] GSM trunks & Dahdi - New funcionality Request for discussion   Odicha
• [asterisk-dev] ASDI User Input Prompt...   Shay Smith
• [asterisk-dev] Can asterisk PRI/BRI support redirect calls   Alec Davis
• [asterisk-dev] 1.6.2 beta: Realtime SIP and periodic database queries   Tilghman Lesher
• [asterisk-dev] Can asterisk PRI/BRI support redirect calls   Jared Smith
• [asterisk-dev] Can asterisk PRI/BRI support redirect calls   Richard Mudgett
• [asterisk-dev] Can asterisk PRI/BRI support redirect calls   Alec Davis
• [asterisk-dev] [Code Review] app_queue: Makes call_queue objects ao2 objects so ref counting can be done   David Vossel
• [asterisk-dev] Can asterisk PRI/BRI support redirect calls   Richard Mudgett
• [asterisk-dev] Can asterisk PRI/BRI support redirect calls   Alec Davis
• [asterisk-dev] Can asterisk PRI/BRI support redirect calls   Richard Mudgett
• [asterisk-dev] Strangeness modifying EXTEN   Philip Prindeville
• [asterisk-dev] [Code Review] app_queue: Makes call_queue objects ao2 objects so ref counting can be done   Russell Bryant
• [asterisk-dev] [Code Review] app_queue: Makes call_queue objects ao2 objects so ref counting can be done   Russell Bryant
• [asterisk-dev] [Code Review] Prevent duplicate entries in format strings used in app_voicemail   Russell Bryant
• [asterisk-dev] [Code Review] Initial review request for CCSS architecture and generic implementations   Russell Bryant
• [asterisk-dev] Strangeness modifying EXTEN   Kevin P. Fleming
• [asterisk-dev] [Code Review] Avoid deadlocks in chan_sip - 1.6.0   Olle E. Johansson
• [asterisk-dev] Can asterisk PRI/BRI support redirect calls   Olle E. Johansson
• [asterisk-dev] CDR variables   Olle E. Johansson
• [asterisk-dev] RTCP work going on :: Project Pinefrog   Olle E. Johansson
• [asterisk-dev] Strangeness modifying EXTEN   Olle E. Johansson
• [asterisk-dev] CDR variables   Tilghman Lesher
• [asterisk-dev] CDR variables   Olle E. Johansson
• [asterisk-dev] CDR variables   Olle E. Johansson
• [asterisk-dev] Strangeness modifying EXTEN   Philip Prindeville
• [asterisk-dev] How to get DAHDI_DUMMY to work for MeetMe ?   Lahav Savir
• [asterisk-dev] compiling Dahdi on blackfin   Dan Amarandei
• [asterisk-dev] [Code Review] Fix EIVR socket client interface   thedavidfactor at gmail.com
• [asterisk-dev] [Code Review] Fix EIVR socket client interface   thedavidfactor at gmail.com
• [asterisk-dev] r227824 breaks chan_iax2   Michiel van Baak
• [asterisk-dev] CDR variables   Benny Amorsen
• [asterisk-dev] CDR variables   Olle E. Johansson
• [asterisk-dev] CDR variables   Tilghman Lesher
• [asterisk-dev] CDR variables   Olle E. Johansson
• [asterisk-dev] CDR variables   Tilghman Lesher
• [asterisk-dev] How to get DAHDI_DUMMY to work for MeetMe ?   Lahav Savir
• [asterisk-dev] CDR variables   Jamuel P. Starkey
• [asterisk-dev] Insulting source code comments in main/channel.c   Floimair Florian
• [asterisk-dev] Insulting source code comments in main/channel.c   Pavel Troller
• [asterisk-dev] Adding a signalling type to DAHDI   Pavel Troller
• [asterisk-dev] Insulting source code comments in main/channel.c   Vlasis Hatzistavrou (KTI)
• [asterisk-dev] Insulting source code comments in main/channel.c   Tilghman Lesher
• [asterisk-dev] Insulting source code comments in main/channel.c   Steve Howes
• [asterisk-dev] Insulting source code comments in main/channel.c   Vlasis Hatzistavrou (KTI)
• [asterisk-dev] Insulting source code comments in main/channel.c   Michiel van Baak
• [asterisk-dev] Insulting source code comments in main/channel.c   Michael Cargile
• [asterisk-dev] Insulting source code comments in main/channel.c   Philippe Sultan
• [asterisk-dev] Insulting source code comments in main/channel.c   Vlasis Hatzistavrou (KTI)
• [asterisk-dev] Insulting source code comments in main/channel.c   David Backeberg
• [asterisk-dev] Insulting source code comments in main/channel.c   Örn Arnarson
• [asterisk-dev] Insulting source code comments in main/channel.c   Michiel van Baak
• [asterisk-dev] Insulting source code comments in main/channel.c   Miguel Molina
• [asterisk-dev] AAL semantic checks   Daniel Ferrer
• [asterisk-dev] Insulting source code comments in main/channel.c   Vlasis Hatzistavrou (KTI)
• [asterisk-dev] Insulting source code comments in main/channel.c   Steve Howes
• [asterisk-dev] Insulting source code comments in main/channel.c   Shay Smith
• [asterisk-dev] [Code Review] Initial review request for CCSS architecture and generic implementations   Mark Michelson
• [asterisk-dev] Insulting source code comments in main/channel.c   Terry Wilson
• [asterisk-dev] AAL semantic checks   Tzafrir Cohen
• [asterisk-dev] Insulting source code comments in main/channel.c   Tilghman Lesher
• [asterisk-dev] Insulting source code comments in main/channel.c   Jeff LaCoursiere
• [asterisk-dev] AAL semantic checks   Steve Murphy
• [asterisk-dev] Insulting source code comments in main/channel.c   Steve Davies
• [asterisk-dev] ADSI...   Shay Smith
• [asterisk-dev] [Code Review] app_queue: Makes call_queue objects ao2 objects so ref counting can be done   David Vossel
• [asterisk-dev] Insulting source code comments in main/channel.c   Steve Murphy
• [asterisk-dev] Insulting source code comments in main/channel.c   Donny Kavanagh
• [asterisk-dev] How to get DAHDI_DUMMY to work for MeetMe ?   David Backeberg
• [asterisk-dev] ADSI...   Tzafrir Cohen
• [asterisk-dev] ADSI...   Tilghman Lesher
• [asterisk-dev] ADSI...   Tilghman Lesher
• [asterisk-dev] Insulting source code comments in main/channel.c   Andrew Kohlsmith (Mailing List Account)
• [asterisk-dev] /team/group/CCSS/channels/chan_sip.c - PUBLISH   Mark Michelson
• [asterisk-dev] [Code Review] Prevent duplicate entries in format strings used in app_voicemail   Matthew Nicholson
• [asterisk-dev] [Code Review] Prevent duplicate entries in format strings used in app_voicemail   Matthew Nicholson
• [asterisk-dev] Insulting source code comments in main/channel.c   Vlasis Hatzistavrou (KTI)
• [asterisk-dev] Insulting source code comments in main/channel.c   Steve Edwards
• [asterisk-dev] Insulting source code comments in main/channel.c   Vlasis Hatzistavrou (KTI)
• [asterisk-dev] Insulting source code comments in main/channel.c   Shay Smith
• [asterisk-dev] Insulting source code comments in main/channel.c   David Backeberg
• [asterisk-dev] Insulting source code comments in main/channel.c   Steve Murphy
• [asterisk-dev] Insulting source code comments in main/channel.c   Steve Edwards
• [asterisk-dev] Insulting source code comments in main/channel.c   Moises Silva
• [asterisk-dev] Insulting source code comments in main/channel.c   Andrew Kohlsmith (Mailing List Account)
• [asterisk-dev] CDR variables   Brent Thomson
• [asterisk-dev] Insulting source code comments in main/channel.c   Vlasis Hatzistavrou (KTI)
• [asterisk-dev] Insulting source code comments in main/channel.c   Vlasis Hatzistavrou (KTI)
• [asterisk-dev] Insulting source code comments in main/channel.c   Vlasis Hatzistavrou (KTI)
• [asterisk-dev] CDR variables   Tilghman Lesher
• [asterisk-dev] Insulting source code comments in main/channel.c   Ron Arts
• [asterisk-dev] Insulting source code comments in main/channel.c   Hans Witvliet
• [asterisk-dev] Insulting source code comments in main/channel.c   Don Kelly
• [asterisk-dev] Insulting source code comments in main/channel.c   Vlasis Hatzistavrou (KTI)
• [asterisk-dev] Insulting source code comments in main/channel.c   Steve Edwards
• [asterisk-dev] Insulting source code comments in main/channel.c   Michael Iedema
• [asterisk-dev] Insulting source code comments in main/channel.c   Terry Wilson
• [asterisk-dev] How to get SIP packet source ip address in asterisk application   ast guy
• [asterisk-dev] How to terminate allocated socket resource when channel hangs up!   ast guy
• [asterisk-dev] Insulting source code comments in main/channel.c   Vlasis Hatzistavrou (KTI)
• [asterisk-dev] Insulting source code comments in main/channel.c   Vlasis Hatzistavrou (KTI)
• [asterisk-dev] Insulting source code comments in main/channel.c   Jeff LaCoursiere
• [asterisk-dev] Insulting source code comments in main/channel.c   Vlasis Hatzistavrou (KTI)
• [asterisk-dev] 1.6.2 beta: Realtime SIP and periodic database queries   Gunnar Schaller
• [asterisk-dev] Insulting source code comments in main/channel.c   Vlasis Hatzistavrou (KTI)
• [asterisk-dev] Insulting source code comments in main/channel.c   Steve Edwards
• [asterisk-dev] Insulting source code comments in main/channel.c   Jeff LaCoursiere
• [asterisk-dev] How to debug DAHDI pseudo timer problem..   Peter Tap
• [asterisk-dev] How to debug DAHDI pseudo timer problem..   Tilghman Lesher
• [asterisk-dev] CDR variables   Olle E. Johansson
• [asterisk-dev] CDR variables   Olle E. Johansson
• [asterisk-dev] we all have our buttons - for asterisk-dev it is political correctness - OT   Nick Lewis
• [asterisk-dev] CDR variables   Kai Hoerner
• [asterisk-dev] Asterist 1.6.0.13 - MeetMe - can_write ..??   Chandrakant Solanki
• [asterisk-dev] Insulting source code comments in main/channel.c   Andrew Kohlsmith (Mailing List Account)
• [asterisk-dev] Insulting source code comments in main/channel.c   Kai Hoerner
• [asterisk-dev] Insulting source code comments in main/channel.c   dimas at dataart.com
• [asterisk-dev] Insulting source code comments in main/channel.c   Russell Bryant
• [asterisk-dev] Insulting source code comments in main/channel.c   Vlasis Hatzistavrou (KTI)
• [asterisk-dev] Insulting source code comments in main/channel.c   Vlasis Hatzistavrou (KTI)
• [asterisk-dev] How to terminate allocated socket resource when channel hangs up!   ast guy
• [asterisk-dev] Insulting source code comments in main/channel.c   Tzafrir Cohen
• [asterisk-dev] How to terminate allocated socket resource when channel hangs up!   Kai Hoerner
• [asterisk-dev] CDR variables   Brent Thomson
• [asterisk-dev] asterisk-dev Digest, Vol 64, Issue 73   Peter Tap
• [asterisk-dev] Calling System() from the "h" extension ?   Pavel Troller
• [asterisk-dev] CDR variables   Matthew Nicholson
• [asterisk-dev] Calling System() from the "h" extension ?   Leif Madsen
• [asterisk-dev] app_fax.c   Chandrakant Solanki
• [asterisk-dev] CDR variables   Olle E. Johansson
• [asterisk-dev] Calling System() from the "h" extension ?   Pavel Troller
• [asterisk-dev] Calling System() from the "h" extension ?   Saúl Ibarra
• [asterisk-dev] app_fax.c   Miguel Molina
• [asterisk-dev] cdr entry does not get written after redirect   Wolfgang Pichler
• [asterisk-dev] app_fax.c   David Backeberg
• [asterisk-dev] CDR variables   Steve Murphy
• [asterisk-dev] CDR variables   Matthew Nicholson
• [asterisk-dev] cdr entry does not get written after redirect   Matthew Nicholson
• [asterisk-dev] CDR variables   Olle E. Johansson
• [asterisk-dev] res_config_ldap   Olle E. Johansson
• [asterisk-dev] Calling System() from the "h" extension ?   Mark Michelson
• [asterisk-dev] cdr entry does not get written after redirect   Wolfgang Pichler
• [asterisk-dev] cdr entry does not get written after redirect   Atis Lezdins
• [asterisk-dev] Calling System() from the "h" extension ?   Leif Madsen
• [asterisk-dev] cdr entry does not get written after redirect   Matthew Nicholson
• [asterisk-dev] Asterisk crash when stop Moh   thiago.fernandes
• [asterisk-dev] CDR variables   Benny Amorsen
• [asterisk-dev] DTMF-initiated transfers   Benny Amorsen
• [asterisk-dev] CDR variables   Mark Michelson
• [asterisk-dev] CDR variables   Steve Murphy
• [asterisk-dev] Dear asterisk-dev at lists.digium.com 79% 0FF on Pfizer.   © VIAGRA ® Official Site
• [asterisk-dev] DTMF-initiated transfers   Örn Arnarson
• [asterisk-dev] DTMF-initiated transfers   Nick Lewis
• [asterisk-dev] DTMF-initiated transfers   Nick Lewis
• [asterisk-dev] DTMF-initiated transfers   Tzafrir Cohen
• [asterisk-dev] cdr entry does not get written after redirect   Wolfgang Pichler
• [asterisk-dev] cdr entry does not get written after redirect   Wolfgang Pichler
• [asterisk-dev] DTMF-initiated transfers   Benny Amorsen
• [asterisk-dev] DTMF-initiated transfers   Benny Amorsen
• [asterisk-dev] DTMF-initiated transfers   Nick Lewis
• [asterisk-dev] ODBC stored voicemail blob object format   Nir Simionovich
• [asterisk-dev] Asterisk 1.6.1.10 MOH problem   Örn Arnarson
• [asterisk-dev] DTMF-initiated transfers   Benny Amorsen
• [asterisk-dev] DTMF-initiated transfers   Tzafrir Cohen
• [asterisk-dev] DTMF-initiated transfers   Benny Amorsen
• [asterisk-dev] ODBC stored voicemail blob object format   Kai Hoerner
• [asterisk-dev] DTMF-initiated transfers   Nick Lewis
• [asterisk-dev] ODBC stored voicemail blob object format   Nir Simionovich
• [asterisk-dev] DTMF-initiated transfers   Kai Hoerner
• [asterisk-dev] DTMF-initiated transfers   Olle E. Johansson
• [asterisk-dev] DTMF-initiated transfers   Nick Lewis
• [asterisk-dev] DTMF-initiated transfers   Benny Amorsen
• [asterisk-dev] Re. ODBC stored voicemail blob object format   Nicolas Chapleau
• [asterisk-dev] DTMF-initiated transfers   Nick Lewis
• [asterisk-dev] ODBC stored voicemail blob object format   Nir Simionovich
• [asterisk-dev] Re. ODBC stored voicemail blob object format   Nir Simionovich
• [asterisk-dev] DTMF-initiated transfers   Olle E. Johansson
• [asterisk-dev] rfc2833 deprecated   Nick Lewis
• [asterisk-dev] ODBC stored voicemail blob object format   Donny Kavanagh
• [asterisk-dev] DTMF-initiated transfers   Nick Lewis
• [asterisk-dev] rfc2833 deprecated   Tzafrir Cohen
• [asterisk-dev] rfc2833 deprecated   Nick Lewis
• [asterisk-dev] DTMF-initiated transfers   Nick Lewis
• [asterisk-dev] Asterisk 1.6.1.10 MOH problem   Kevin P. Fleming
• [asterisk-dev] Asterisk 1.6.1.10 MOH problem   Örn Arnarson
• [asterisk-dev] Asterisk 1.6.1.10 MOH problem   Kevin P. Fleming
• [asterisk-dev] DTMF-initiated transfers   Kaloyan Kovachev
• [asterisk-dev] DTMF-initiated transfers   Kaloyan Kovachev
• [asterisk-dev] DTMF-initiated transfers   Nick Lewis
• [asterisk-dev] DTMF-initiated transfers   Kaloyan Kovachev
• [asterisk-dev] DTMF-initiated transfers   Kai Hoerner
• [asterisk-dev] ODBC stored voicemail blob object format   Tilghman Lesher
• [asterisk-dev] ODBC stored voicemail blob object format   Atis Lezdins
• [asterisk-dev] ODBC stored voicemail blob object format   Atis Lezdins
• [asterisk-dev] DTMF-initiated transfers   Benny Amorsen
• [asterisk-dev] ConfBridge Stuck on playing join message   Lahav Savir
• [asterisk-dev] RES: rfc2833 deprecated   Philipp Kempgen
• [asterisk-dev] RES: rfc2833 deprecated   Olle E. Johansson
• [asterisk-dev] RES: rfc2833 deprecated   Rafael Prado Rocchi
• [asterisk-dev] RFC4733 SRTP option   Nick Lewis
• [asterisk-dev] RFC4733 SRTP option   Olle E. Johansson
• [asterisk-dev] RFC4733 SRTP option   Nick Lewis
• [asterisk-dev] BLF and unreachable phones   Dario Busso
• [asterisk-dev] RFC4733 SRTP option   Nick Lewis
• [asterisk-dev] RFC4733 SRTP option   Olle E. Johansson
• [asterisk-dev] RFC4733 SRTP option   Nick Lewis
• [asterisk-dev] ConfBridge Stuck on playing join message   Nick Lewis
• [asterisk-dev] RFC4733 SRTP option   Olle E. Johansson
• [asterisk-dev] RFC4733 SRTP option   Nick Lewis
• [asterisk-dev] CLARIFICATION   Nick Lewis
• [asterisk-dev] RFC4733 SRTP option   Lawrence Conroy
• [asterisk-dev] system completely hangs after executing an ODBC function   Theo Belder
• [asterisk-dev] Strange code in chan_sip.c   Vadim Lebedev
• [asterisk-dev] Strange code in chan_sip.c   Giuseppe Sucameli
• [asterisk-dev] (no subject)   dimas at dataart.com
• [asterisk-dev] DTMF problem when receiving call   Maxim Samo
• [asterisk-dev] Running asterisk trunk build   Kirill 'Big K' Katsnelson
• [asterisk-dev] Running asterisk trunk build   Tzafrir Cohen
• [asterisk-dev] Running asterisk trunk build   Kirill 'Big K' Katsnelson
• [asterisk-dev] Running asterisk trunk build   Kevin P. Fleming
• [asterisk-dev] Running asterisk trunk build   Julian Lyndon-Smith
• [asterisk-dev] Running asterisk trunk build   Kevin P. Fleming
• [asterisk-dev] Running asterisk trunk build   Kirill 'Big K' Katsnelson
• [asterisk-dev] Failure on multiple malformed SIP URI: a case study, likely bugs   Kirill 'Big K' Katsnelson
• [asterisk-dev] cdr entry does not get written after redirect   Matthew Nicholson
• [asterisk-dev] Stupid question: how to re-open a closed issue?   Ron Arts
• [asterisk-dev] Stupid question: how to re-open a closed issue?   Leif Madsen
• [asterisk-dev] [Code Review] app_queue: Makes call_queue objects ao2 objects so ref counting can be done   David Vossel
• [asterisk-dev] [Code Review] app_queue: Makes call_queue objects ao2 objects so ref counting can be done   Russell Bryant
• [asterisk-dev] Problem with *1.6.0.5; "Skipping dialing interface since it has already been dialed"   Håkon Nessjøen
• [asterisk-dev] [Code Review] fix for double close of file descriptors raised in issue 16305   thedavidfactor at gmail.com
• [asterisk-dev] [Code Review] fix for double close of file descriptors raised in issue 16305   thedavidfactor at gmail.com
• [asterisk-dev] [Code Review] fix for double close of file descriptors raised in issue 16305   Tilghman Lesher
• [asterisk-dev] Meetme AMI actions   Olle E. Johansson
• [asterisk-dev] [Code Review] AMI Setvar: Return error when function does not exist or generate error   Olle E Johansson
• [asterisk-dev] Meetme AMI actions   Jared Smith
• [asterisk-dev] Meetme AMI actions   Matt Florell
• [asterisk-dev] [Code Review] AMI Setvar: Return error when function does not exist or generate error   Russell Bryant
• [asterisk-dev] [Code Review] AMI Setvar: Return error when function does not exist or generate error   Russell Bryant
• [asterisk-dev] Asterisk 1.2.37, 1.4.27.1, 1.6.0.19, and 1.6.1.11 Now Available   Asterisk Development Team
• [asterisk-dev] AST-2009-010: RTP Remote Crash Vulnerability   Asterisk Security Team
• [asterisk-dev] Meetme AMI actions   dimas at dataart.com
• [asterisk-dev] SIP URI checking in chan_sip.c   Kirill 'Big K' Katsnelson

Last message date: Mon Nov 30 18:37:30 CST 2009
Archived on: Mon Nov 30 18:37:38 CST 2009

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