December 2009 Archives by date

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Starting: Tue Dec 1 00:32:05 CST 2009
Ending: Thu Dec 31 23:28:34 CST 2009
Messages: 366

• [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   Olle E. Johansson
• [asterisk-dev] SIP URI checking in chan_sip.c   Olle E. Johansson
• [asterisk-dev] SIP URI checking in chan_sip.c   Kirill 'Big K' Katsnelson
• [asterisk-dev] Insulting source code comments in main/channel.c   Kirill 'Big K' Katsnelson
• [asterisk-dev] New committer - odicha   Odicha
• [asterisk-dev] New committer - odicha   Thomas Kenyon
• [asterisk-dev] SIP URI checking in chan_sip.c   Olle E. Johansson
• [asterisk-dev] New committer - odicha   Odicha
• [asterisk-dev] SIP URI checking in chan_sip.c   Nick Lewis
• [asterisk-dev] SIP URI checking in chan_sip.c   Tilghman Lesher
• [asterisk-dev] SIP URI checking in chan_sip.c   Olle E. Johansson
• [asterisk-dev] New committer - odicha   Thomas Kenyon
• [asterisk-dev] SIP URI checking in chan_sip.c   Nick Lewis
• [asterisk-dev] New committer - odicha   Odicha
• [asterisk-dev] Can asterisk PRI/BRI support redirect calls   Alec Davis
• [asterisk-dev] Can asterisk PRI/BRI support redirect calls   Kevin P. Fleming
• [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] Working in useful examples... and freenum/e.164 dialing in extensions.conf.example   Philip A. Prindeville
• [asterisk-dev] SIP URI checking in chan_sip.c   Kirill 'Big K' Katsnelson
• [asterisk-dev] team creation (novice question)   Odicha
• [asterisk-dev] team creation (novice question)   David Ruggles
• [asterisk-dev] Working in useful examples... and freenum/e.164dialing in extensions.conf.example   Nick Lewis
• [asterisk-dev] [asterisk-commits] tilghman: trunk r232164 - in /trunk: ./ include/asterisk/ main/   Kevin P. Fleming
• [asterisk-dev] Working in useful examples... and freenum/e.164dialing in extensions.conf.example   Philip A. Prindeville
• [asterisk-dev] Asterisk-Addons 1.4.10, 1.6.0.4, and 1.6.1.2 Now Available   Asterisk Development Team
• [asterisk-dev] Asterisk 1.6.2.0-rc7 released, and Asterisk-Addons 1.6.2.0-rc2 re-released   Asterisk Development Team
• [asterisk-dev] [asterisk-commits] tilghman: trunk r232164 - in /trunk: ./ include/asterisk/ main/   Tilghman Lesher
• [asterisk-dev] [Code Review] fix for double close of file descriptors raised in issue 16305   Russell Bryant
• [asterisk-dev] Meetme AMI actions   Dan Austin
• [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] Meetme AMI actions   Nick Lewis
• [asterisk-dev] Meetme AMI actions   Olle E. Johansson
• [asterisk-dev] chan_sip - sip_write error   Chandrakant Solanki
• [asterisk-dev] chan_sip - sip_write error   santoshchintalwar at gmail.com
• [asterisk-dev] [Code Review] Fix minor bugs preventing EIVR socket client from working   thedavidfactor at gmail.com
• [asterisk-dev] [Code Review] Fix minor bugs preventing EIVR socket client from working   thedavidfactor at gmail.com
• [asterisk-dev] State of FAX (primarily T.38) in Asterisk trunk (planning for 1.8 release)   Kevin P. Fleming
• [asterisk-dev] State of FAX (primarily T.38) in Asterisk trunk (planning for 1.8 release)   David Ruggles
• [asterisk-dev] State of FAX (primarily T.38) in Asterisk trunk (planning for 1.8 release)   Lee Howard
• [asterisk-dev] State of FAX (primarily T.38) in Asterisk trunk (planning for 1.8 release)   Kevin P. Fleming
• [asterisk-dev] Meetme AMI actions   Russell Bryant
• [asterisk-dev] State of FAX (primarily T.38) in Asterisk trunk (planning for 1.8 release)   asterisk at ntplx.net
• [asterisk-dev] State of FAX (primarily T.38) in Asterisk trunk (planning for 1.8 release)   Kevin P. Fleming
• [asterisk-dev] State of FAX (primarily T.38) in Asterisk trunk (planning for 1.8 release)   Steve Underwood
• [asterisk-dev] State of FAX (primarily T.38) in Asterisk trunk (planning for 1.8 release)   Steve Underwood
• [asterisk-dev] What's the verdict on the new release plan?   Olle E. Johansson
• [asterisk-dev] * 1.6.1.11 res_config_sqlite No application SQLITE()   Andrea Cristofanini
• [asterisk-dev] diruggles: branch 1.6.0 r232811 - /branches/1.6.0/apps/app_externalivr.c   Tony Mountifield
• [asterisk-dev] What's the verdict on the new release plan?   Russell Bryant
• [asterisk-dev] State of FAX (primarily T.38) in Asterisk trunk (planning for 1.8 release)   Kevin P. Fleming
• [asterisk-dev] New feature in app_queue: Add support for ring indication when calling member   Håkon Nessjøen
• [asterisk-dev] app_meetme call for testing: Roll call, eject all, mute all, record in-conf   Ryan Finnie
• [asterisk-dev] tilghman: trunk r232660 - in /trunk: include/asterisk/ res/   Russell Bryant
• [asterisk-dev] tilghman: trunk r232660 - in /trunk: include/asterisk/ res/   Tilghman Lesher
• [asterisk-dev] tilghman: trunk r232660 - in /trunk: include/asterisk/ res/   Andrew Parisio
• [asterisk-dev] tilghman: trunk r232660 - in /trunk: include/asterisk/ res/   Russell Bryant
• [asterisk-dev] app_meetme call for testing: Roll call, eject all, mute all, record in-conf   John Todd
• [asterisk-dev] app_meetme call for testing: Roll call, eject all, mute all, record in-conf   Michiel van Baak
• [asterisk-dev] app_meetme call for testing: Roll call, eject all, mute all, record in-conf   Ryan Finnie
• [asterisk-dev] app_meetme call for testing: Roll call, eject all, mute all, record in-conf   John Todd
• [asterisk-dev] [svn-commits] mnick: branch 1.6.0 r233236 - in /branches/1.6.0: ./ pbx/pbx_config.c   Olle E. Johansson
• [asterisk-dev] Changing function name of a static function and updating doxygen doc in 1.4   Olle E. Johansson
• [asterisk-dev] bad file prompt play   Chandrakant Solanki
• [asterisk-dev] Asterisk and multiple SIP registrations to the same host   asterisk-dev at eurower.com
• [asterisk-dev] bad file prompt play   santosh chintalwar
• [asterisk-dev] [asterisk-commits] tilghman: trunk r232164 - in /trunk: ./ include/asterisk/ main/   Tilghman Lesher
• [asterisk-dev] Makefile.rules MAKE_DEPS usage   Luigi Rizzo
• [asterisk-dev] app_meetme call for testing: Roll call, eject all, mute all, record in-conf   Olle E. Johansson
• [asterisk-dev] app_meetme call for testing: Roll call, eject all, mute all, record in-conf   Russell Bryant
• [asterisk-dev] Changing function name of a static function and updating doxygen doc in 1.4   Russell Bryant
• [asterisk-dev] bad file prompt play   Chandrakant Solanki
• [asterisk-dev] Can Asterik act as a SIP Proxy   Gayathri G
• [asterisk-dev] Can Asterik act as a SIP Proxy   John Peter Loh
• [asterisk-dev] Can Asterik act as a SIP Proxy   Gayathri G
• [asterisk-dev] app_meetme call for testing: Roll call, eject all, mute all, record in-conf   dimas at dataart.com
• [asterisk-dev] Asterisk and multiple SIP registrations to the samehost   Nick Lewis
• [asterisk-dev] Asterisk and multiple SIP registrations to the samehost   asterisk-dev at eurower.com
• [asterisk-dev] g722 question   Saúl Ibarra
• [asterisk-dev] g722 question   Steve Underwood
• [asterisk-dev] g722 question   Saúl Ibarra
• [asterisk-dev] app_meetme call for testing: Roll call, eject all, mute all, record in-conf   Joshua Colp
• [asterisk-dev] g722 question   Steve Underwood
• [asterisk-dev] [svn-commits] mnick: branch 1.6.0 r233236 - in /branches/1.6.0: ./ pbx/pbx_config.c   Richard Mudgett
• [asterisk-dev] g722 question   Saúl Ibarra
• [asterisk-dev] g722 question   Steve Underwood
• [asterisk-dev] app_meetme call for testing: Roll call, eject all, mute all, record in-conf   dimas at dataart.com
• [asterisk-dev] Asterisk 1.6.2: CDR missing for .call files   Mario Moran
• [asterisk-dev] ChanSpy mod   Gabriel Ortiz Lour
• [asterisk-dev] ChanSpy mod   Mark Michelson
• [asterisk-dev] ChanSpy mod   Gabriel Ortiz Lour
• [asterisk-dev] [Code Review] Fix minor bugs preventing EIVR socket client from working   Russell Bryant
• [asterisk-dev] Asterisk 1.6.2: CDR missing for .call files   Russell Bryant
• [asterisk-dev] [Code Review] [16388] MOH refcount fix   Tilghman Lesher
• [asterisk-dev] [Code Review] [16388] MOH refcount fix   Russell Bryant
• [asterisk-dev] [Code Review] [16388] MOH refcount fix   Tilghman Lesher
• [asterisk-dev] [Code Review] [16388] MOH refcount fix   Tilghman Lesher
• [asterisk-dev] [Code Review] [16388] MOH refcount fix   Russell Bryant
• [asterisk-dev] SIP URI checking in chan_sip.c   Kirill 'Big K' Katsnelson
• [asterisk-dev] Forbidden - wrong password on authentication for REGISTER   Nick Lewis
• [asterisk-dev] SIP URI checking in chan_sip.c   David Vossel
• [asterisk-dev] why reset p->method in chan_sip?   David Vossel
• [asterisk-dev] [Code Review] [16388] MOH refcount fix   Tilghman Lesher
• [asterisk-dev] [Code Review] [16388] MOH refcount fix   Tilghman Lesher
• [asterisk-dev] Add attributes to Asterisk T38 invite for MAX TNT Interop?   JR Richardson
• [asterisk-dev] Add attributes to Asterisk T38 invite for MAX TNT Interop?   Steve Underwood
• [asterisk-dev] Add attributes to Asterisk T38 invite for MAX TNT Interop?   Kevin P. Fleming
• [asterisk-dev] SIP URI checking in chan_sip.c   Kirill 'Big K' Katsnelson
• [asterisk-dev] [Code Review] Resolve format module dependency issue for app_voicemail   Russell Bryant
• [asterisk-dev] [Code Review] Resolve format module dependency issue for app_voicemail   Mark Michelson
• [asterisk-dev] Add attributes to Asterisk T38 invite for MAX TNT Interop?   JR Richardson
• [asterisk-dev] Add attributes to Asterisk T38 invite for MAX TNT Interop?   Alex Balashov
• [asterisk-dev] Add attributes to Asterisk T38 invite for MAX TNT Interop? Update   JR Richardson
• [asterisk-dev] [Code Review] Resolve format module dependency issue for app_voicemail   Russell Bryant
• [asterisk-dev] [Code Review] Cancel retries for all dialogs when the remote host isn't answering   Tilghman Lesher
• [asterisk-dev] How to Add New Language Prompts to Asterisk (Officially to get along with distribution)   Kannaiyan Natesan
• [asterisk-dev] How to Add New Language Prompts to Asterisk (Officially to get along with distribution)   John Todd
• [asterisk-dev] [Code Review] Cancel retries for all dialogs when the remote host isn't answering   David Vossel
• [asterisk-dev] [Code Review] Cancel retries for all dialogs when the remote host isn't answering   David Vossel
• [asterisk-dev] [Code Review] Cancel retries for all dialogs when the remote host isn't answering   Mark Michelson
• [asterisk-dev] [Code Review] Cancel retries for all dialogs when the remote host isn't answering   Tilghman Lesher
• [asterisk-dev] [Code Review] Cancel retries for all dialogs when the remote host isn't answering   Tilghman Lesher
• [asterisk-dev] [Code Review] Cancel retries for all dialogs when the remote host isn't answering   Tilghman Lesher
• [asterisk-dev] why reset p->method in chan_sip?   Olle E. Johansson
• [asterisk-dev] [Code Review] Cancel retries for all dialogs when the remote host isn't answering   Mark Michelson
• [asterisk-dev] why reset p->method in chan_sip?   Mark Michelson
• [asterisk-dev] Asterisk Release Candidates Available (1.4.28-rc1, 1.6.0.20-rc1, 1.6.1.12-rc1, and 1.6.2.0-rc8)   Asterisk Development Team
• [asterisk-dev] How to Add New Language Prompts to Asterisk (Officially to get along with distribution)   Kannaiyan Natesan
• [asterisk-dev] ASTERISK INCOMMING CALLER ID   Dulip Ravindra
• [asterisk-dev] Asterisk and multiple SIP registrations to the same host   Leif Madsen
• [asterisk-dev] Asterisk and multiple SIP registrations to the same host   Nick Lewis
• [asterisk-dev] ASTERISK INCOMMING CALLER ID   Nick Lewis
• [asterisk-dev] TDMOE protocol   Antonio
• [asterisk-dev] TDMOE protocol   Miguel Molina
• [asterisk-dev] fxo ring parameter adjust   晶讯田工
• [asterisk-dev] Music on hold after an attended transfer to a queue.   nico kooijman
• [asterisk-dev] TDMOE protocol   Andrew Latham
• [asterisk-dev] TDMOE protocol   Andrew Kohlsmith (Mailing List Account)
• [asterisk-dev] Asterisk Caller ID Pop Using Dial Event in 5038 manager port.   Dulip Ravindra
• [asterisk-dev] TDMOE protocol   Dulip Ravindra
• [asterisk-dev] TDMOE protocol   Tzafrir Cohen
• [asterisk-dev] Music on hold after an attended transfer to a queue.   Dulip Ravindra
• [asterisk-dev] Music on hold after an attended transfer to a queue.   nico kooijman
• [asterisk-dev] Addons1.6.1.2 compile issue   Bryant Zimmerman
• [asterisk-dev] Addons1.6.1.2 compile issue   Bryant Zimmerman
• [asterisk-dev] Asterisk Caller ID Pop Using Dial Event in 5038manager port.   Nick Lewis
• [asterisk-dev] ISDN remote HOLD DAHDI   Andrea Cristofanini
• [asterisk-dev] Disabling frame cache to workaround #16374 - negative impact ?   Pavel Troller
• [asterisk-dev] Disabling frame cache to workaround #16374 - negative impact ?   Pavel Troller
• [asterisk-dev] Asterisk 1.6.2: CDR missing for .call files   Leif Madsen
• [asterisk-dev] Default audiohook inheritance   Dwayne Hubbard
• [asterisk-dev] Default audiohook inheritance   Mark Michelson
• [asterisk-dev] Asterisk Caller ID Pop Using Dial Event in 5038manager port.   Dulip Ravindra
• [asterisk-dev] [Code Review] parkinglots with different parking extensions   Russell Bryant
• [asterisk-dev] [Code Review] RTP monitoring branch: initial review   Russell Bryant
• [asterisk-dev] Fwd: Inquiry:Asterisk sip server?   hadi motamedi
• [asterisk-dev] AGI question   Ngo-Vi Hoai-Anh
• [asterisk-dev] AGI question   ABBAS SHAKEEL
• [asterisk-dev] AGI question   Ngo-Vi Hoai-Anh
• [asterisk-dev] AGI question   Tilghman Lesher
• [asterisk-dev] Default audiohook inheritance   Russell Bryant
• [asterisk-dev] Fwd: Inquiry:Asterisk sip server?   John Todd
• [asterisk-dev] Make menu select for asterisk   Julian Lyndon-Smith
• [asterisk-dev] Make menu select for asterisk   Julian Lyndon-Smith
• [asterisk-dev] [Code Review] Unit Test Framework   David Vossel
• [asterisk-dev] [Code Review] Unit Test Framework   Russell Bryant
• [asterisk-dev] Fwd: Inquiry:Asterisk sip server?   hadi motamedi
• [asterisk-dev] [svn-commits] extrachannel   Olle E. Johansson
• [asterisk-dev] [svn-commits] extrachannel it is...   Olle E. Johansson
• [asterisk-dev] [Code Review] Allow app_dial to play 'indication tone while ringing' like 'option m' which will provide music on hold while ringing   Alec Davis
• [asterisk-dev] [Code Review] Allow app_dial to play 'indication tone while ringing' like 'option m' which will provide music on hold while ringing   Alec Davis
• [asterisk-dev] Asterisk and multiple SIP registrations to the same host   Klaus Darilion
• [asterisk-dev] Working in useful examples... and freenum/e.164 dialing in extensions.conf.example   Klaus Darilion
• [asterisk-dev] Asterisk and multiple SIP registrations to the same host   Olle E. Johansson
• [asterisk-dev] Asterisk and multiple SIP registrations to thesame host   Nick Lewis
• [asterisk-dev] Asterisk and multiple SIP registrations to thesame host   Olle E. Johansson
• [asterisk-dev] Asterisk and multiple SIP registrations tothesame host   Nick Lewis
• [asterisk-dev] Asterisk and multiple SIP registrations to the same host   Klaus Darilion
• [asterisk-dev] Has VoipJet Support Vanished?   Dave At Greenroom Creations
• [asterisk-dev] [Code Review] Unit Test Framework   Russell Bryant
• [asterisk-dev] [Code Review] Unit Test Framework   Russell Bryant
• [asterisk-dev] [Code Review] Unit Test Framework   Mark Michelson
• [asterisk-dev] [Code Review] Unit Test Framework   Mark Michelson
• [asterisk-dev] [Code Review] Unit Test Framework   Mark Michelson
• [asterisk-dev] [Code Review] Unit Test Framework   David Vossel
• [asterisk-dev] [Code Review] Unit Test Framework   Mark Michelson
• [asterisk-dev] [Code Review] Unit Test Framework   Matthew Nicholson
• [asterisk-dev] [Code Review] Unit Test Framework   Matthew Nicholson
• [asterisk-dev] [Code Review] Unit Test Framework   David Vossel
• [asterisk-dev] [Code Review] Unit Test Framework   Russell Bryant
• [asterisk-dev] [Code Review] Unit Test Framework   David Vossel
• [asterisk-dev] [Code Review] Unit Test Framework   Russell Bryant
• [asterisk-dev] [Code Review] Allow app_dial to play 'indication tone while ringing' like 'option m' which will provide music on hold while ringing   Russell Bryant
• [asterisk-dev] [Code Review] Unit Test Framework   David Vossel
• [asterisk-dev] [Code Review] Unit Test Framework   David Vossel
• [asterisk-dev] Asterisk and multiple SIP registrations to the same host   Olle E. Johansson
• [asterisk-dev] SIP replies to via or to ip/port (according to RFC)   Olle E. Johansson
• [asterisk-dev] Peer matching architecture   Olle E. Johansson
• [asterisk-dev] Asterisk and multiple SIP registrations tothesame host   Olle E. Johansson
• [asterisk-dev] Has VoipJet Support Vanished?   Olle E. Johansson
• [asterisk-dev] [Code Review] Allow app_dial to play 'indication tone while ringing' like 'option m' which will provide music on hold while ringing   Alec Davis
• [asterisk-dev] [Code Review] Allow app_dial to play 'indication tone while ringing' like 'option m' which will provide music on hold while ringing   Alec Davis
• [asterisk-dev] SIP replies to via or to ip/port (according to RFC)   Alex Hermann
• [asterisk-dev] Peer matching architecture   Nick Lewis
• [asterisk-dev] Asterisk and multiple SIP registrationstothesame host   Nick Lewis
• [asterisk-dev] SIP replies to via or to ip/port (according toRFC)   Nick Lewis
• [asterisk-dev] SIP replies to via or to ip/port (accordingtoRFC)   Nick Lewis
• [asterisk-dev] [Code Review] Unit Test Framework   Matthew Nicholson
• [asterisk-dev] [Code Review] Unit Test Framework   Matthew Nicholson
• [asterisk-dev] [Code Review] Unit Test Framework   Matthew Nicholson
• [asterisk-dev] [Code Review] Unit Test Framework   Matthew Nicholson
• [asterisk-dev] [Code Review] Unit Test Framework   Matthew Nicholson
• [asterisk-dev] [Code Review] Unit Test Framework   David Vossel
• [asterisk-dev] [Code Review] Unit Test Framework   David Vossel
• [asterisk-dev] [Code Review] Unit Test Framework   Russell Bryant
• [asterisk-dev] [Code Review] Unit Test Framework   David Vossel
• [asterisk-dev] [Code Review] Unit Test Framework   Matthew Nicholson
• [asterisk-dev] [Code Review] Unit Test Framework   Matthew Nicholson
• [asterisk-dev] [Code Review] Unit Test Framework   David Vossel
• [asterisk-dev] Peer matching architecture   David Vossel
• [asterisk-dev] [Code Review] Unit Test Framework   Russell Bryant
• [asterisk-dev] Peer matching architecture   Olle E. Johansson
• [asterisk-dev] [Code Review] Unit Test Framework   Matthew Nicholson
• [asterisk-dev] [Code Review] Unit Test Framework   Russell Bryant
• [asterisk-dev] Peer matching architecture   Olle E. Johansson
• [asterisk-dev] Working in useful examples... and freenum/e.164 dialing in extensions.conf.example   John Todd
• [asterisk-dev] [Code Review] Unit Test Framework   David Vossel
• [asterisk-dev] 1.6.1.12 Asterisk Crashes after 100 Bridged Calls in SIP?   JR Richardson
• [asterisk-dev] Asterisk Core Dump   postalforall
• [asterisk-dev] 1.6.1.12 Asterisk Crashes after 100 Bridged Calls in SIP?   Olle E. Johansson
• [asterisk-dev] Working in useful examples in sample config (changed topic)   Olle E. Johansson
• [asterisk-dev] Working in useful examples in sample config (changed topic)   Michiel van Baak
• [asterisk-dev] Asterisk Core Dump   nico kooijman
• [asterisk-dev] 1.6.1.12 Asterisk Crashes after 100 Bridged Calls in SIP?   Benny Amorsen
• [asterisk-dev] 1.6.1.12 Asterisk Crashes after 100 Bridged Calls in SIP?   Olle E. Johansson
• [asterisk-dev] Peer matching architecture   Leif Madsen
• [asterisk-dev] Peer matching architecture   Nick Lewis
• [asterisk-dev] Working in useful examples in sample config (changed topic)   Jared Smith
• [asterisk-dev] 1.6.1.12 Asterisk Crashes after 100 Bridged Calls in SIP?   Russell Bryant
• [asterisk-dev] [Code Review] Allow app_dial to play 'indication tone while ringing' like 'option m' which will provide music on hold while ringing   Russell Bryant
• [asterisk-dev] [Code Review] Allow app_dial to play 'indication tone while ringing' like 'option m' which will provide music on hold while ringing   Russell Bryant
• [asterisk-dev] TDMOE protocol   Antonio
• [asterisk-dev] SIP replies to via or to ip/port (according to RFC)   Klaus Darilion
• [asterisk-dev] Working in useful examples... and freenum/e.164 dialing in extensions.conf.example   Klaus Darilion
• [asterisk-dev] Asterisk and multiple SIP registrations tothesame host   Klaus Darilion
• [asterisk-dev] Peer matching architecture   Klaus Darilion
• [asterisk-dev] dahdi-linux: don't mknot on install by default   Tzafrir Cohen
• [asterisk-dev] dahdi-dude   Tzafrir Cohen
• [asterisk-dev] dahdi-dude   Russell Bryant
• [asterisk-dev] DAHDI-Linux 2.2.1-rc2 and DAHDI-Tools 2.2.1-rc2 Now Available   Asterisk Development Team
• [asterisk-dev] Asterisk 1.4.28 Now Available   Asterisk Development Team
• [asterisk-dev] Asterisk 1.6.0.20 Now Available   Asterisk Development Team
• [asterisk-dev] Asterisk 1.6.1.12 Now Available   Asterisk Development Team
• [asterisk-dev] [Code Review] Unit Test Framework   Russell Bryant
• [asterisk-dev] Asterisk 1.6.2.0 Now Available!   Asterisk Development Team
• [asterisk-dev] Asterisk 1.6.2.0 Now Available!   Michael Iedema
• [asterisk-dev] Asterisk and multiple SIP registrations tothesame host   Olle E. Johansson
• [asterisk-dev] Peer matching architecture   Olle E. Johansson
• [asterisk-dev] Peer matching architecture   Olle E. Johansson
• [asterisk-dev] [Code Review] Allow app_dial to play 'indication tone while ringing' like 'option m' which will provide music on hold while ringing   Alec Davis
• [asterisk-dev] Peer matching architecture   Olle E. Johansson
• [asterisk-dev] 1.6.1.12 Asterisk Crashes after 100 Bridged Calls in SIP?   Leif Madsen
• [asterisk-dev] dahdi-linux: don't mknot on install by default   Kevin P. Fleming
• [asterisk-dev] Asterisk and multiple SIP registrations tothesame host   Klaus Darilion
• [asterisk-dev] Compiling asterisk as single binary / minimalist OS   Julian Lyndon-Smith
• [asterisk-dev] Compiling asterisk as single binary / minimalist OS   Steve Howes
• [asterisk-dev] Compiling asterisk as single binary / minimalist OS   Julian Lyndon-Smith
• [asterisk-dev] Compiling asterisk as single binary / minimalist OS   Tzafrir Cohen
• [asterisk-dev] Compiling asterisk as single binary / minimalist OS   Hans Witvliet
• [asterisk-dev] Peer matching architecture   Tilghman Lesher
• [asterisk-dev] SIP replies to via or to ip/port (according to RFC)   Nick Lewis
• [asterisk-dev] SIP replies to via or to ip/port (according to RFC)   Olle E. Johansson
• [asterisk-dev] Peer matching architecture   Nick Lewis
• [asterisk-dev] Peer matching architecture   Nick Lewis
• [asterisk-dev] [Code Review] Add notificiation of intrupted file and fix bug   thedavidfactor at gmail.com
• [asterisk-dev] [Code Review] Add notificiation of intrupted file and fix bug   thedavidfactor at gmail.com
• [asterisk-dev] [Code Review] SIP: decode uri after parsing   David Vossel
• [asterisk-dev] Expression logic problem   Mueller, Alexander
• [asterisk-dev] 900 second call drop - 420 Bad Extension error w/ SIP, IOS GW   Jared Mauch
• [asterisk-dev] [Code Review] SIP: decode uri after parsing   Olle E. Johansson
• [asterisk-dev] 900 second call drop - 420 Bad Extension error w/ SIP, IOS GW   Olle E. Johansson
• [asterisk-dev] Expression logic problem   Steve Edwards
• [asterisk-dev] [Code Review] SIP: decode uri after parsing   Nick Lewis
• [asterisk-dev] 900 second call drop - 420 Bad Extension error w/ SIP, IOS GW   Kevin P. Fleming
• [asterisk-dev] [Code Review] SIP: decode uri after parsing   Kevin P. Fleming
• [asterisk-dev] 900 second call drop - 420 Bad Extension error w/ SIP, IOS GW   Jared Mauch
• [asterisk-dev] [Code Review] SIP: decode uri after parsing   Olle E. Johansson
• [asterisk-dev] Expression logic problem   Mueller, Alexander
• [asterisk-dev] 900 second call drop - 420 Bad Extension error w/ SIP, IOS GW   Jared Mauch
• [asterisk-dev] [Code Review] SIP: decode uri after parsing   Nick Lewis
• [asterisk-dev] [Code Review] SIP: decode uri after parsing   David Vossel
• [asterisk-dev] [Code Review] SIP: decode uri after parsing   Olle E. Johansson
• [asterisk-dev] Peer matching architecture   Olle E. Johansson
• [asterisk-dev] [Code Review] SIP: decode uri after parsing   Olle E. Johansson
• [asterisk-dev] [Code Review] SIP: decode uri after parsing   Nick Lewis
• [asterisk-dev] Version 1.8 or 1.6.3 next?   Olle E. Johansson
• [asterisk-dev] Version 1.8 or 1.6.3 next?   Andrew Latham
• [asterisk-dev] Version 1.8 or 1.6.3 next?   Miguel Molina
• [asterisk-dev] Version 1.8 or 1.6.3 next?   Olle E. Johansson
• [asterisk-dev] Working in useful examples in sample config (changed topic)   Olle E. Johansson
• [asterisk-dev] Peer matching architecture   Tilghman Lesher
• [asterisk-dev] Peer matching architecture   Olle E. Johansson
• [asterisk-dev] Asterisk 1.6.2.0 Now Available!   Faidon Liambotis
• [asterisk-dev] 1.6.1.12 Asterisk Crashes after 100 Bridged Calls in SIP?   JR Richardson
• [asterisk-dev] Asterisk 1.6.2.0 Now Available!   Asterisk Development Team
• [asterisk-dev] [Code Review] parkinglots with different parking extensions   mnick at digium.com
• [asterisk-dev] Release Schedules and plans for Asterisk 1.8   Russell Bryant
• [asterisk-dev] [Code Review] parkinglots with different parking extensions   Russell Bryant
• [asterisk-dev] [Code Review] Unit Test Framework   David Vossel
• [asterisk-dev] Release Schedules and plans for Asterisk 1.8   Russell Bryant
• [asterisk-dev] [Code Review] Unit Test Framework   Russell Bryant
• [asterisk-dev] Release Schedules and plans for Asterisk 1.8   Faidon Liambotis
• [asterisk-dev] Regarding predictive dialing   Shivamurthy
• [asterisk-dev] Release Schedules and plans for Asterisk 1.8   Olle E. Johansson
• [asterisk-dev] Release Schedules and plans for Asterisk 1.8   Russell Bryant
• [asterisk-dev] Unit Test Framework Now Available!   David Vossel
• [asterisk-dev] Unit Test Framework Now Available!   Olle E. Johansson
• [asterisk-dev] [Code Review] ast_uri_encode() behavior change   David Vossel
• [asterisk-dev] asterisk 1.6.2.0 realtime queue   hooble at gmx.net
• [asterisk-dev] [Code Review] ast_uri_encode() behavior change   David Vossel
• [asterisk-dev] Happy Holidays from OpSys Consulting Group   Alexander Lopez
• [asterisk-dev] Failed to record Radius CDR record!   Zhang Shukun
• [asterisk-dev] srtp   Hans Witvliet
• [asterisk-dev] srtp   takesver.thakur at mobilefundas.com
• [asterisk-dev] DNID and RDNIS are not carried across IAX2   Nir Simionovich
• [asterisk-dev] No CDR for non-bridged outgoing calls   Jim Gottlieb
• [asterisk-dev] CallerID issue using DTMF on 1.2   Stelios Koroneos
• [asterisk-dev] DNID and RDNIS are not carried across IAX2   Benny Amorsen
• [asterisk-dev] No CDR for non-bridged outgoing calls   Kaloyan Kovachev
• [asterisk-dev] DNID and RDNIS are not carried across IAX2   Nir Simionovich
• [asterisk-dev] DNID and RDNIS are not carried across IAX2   Tilghman Lesher
• [asterisk-dev] CallerID issue using DTMF on 1.2   Will
• [asterisk-dev] Possible Call Event Logging (CEL) Bugs   Nic Colledge
• [asterisk-dev] CallerID issue using DTMF on 1.2   Stelios Koroneos
• [asterisk-dev] DNID and RDNIS are not carried across IAX2   Nir Simionovich
• [asterisk-dev] No CDR for non-bridged outgoing calls   Jim Gottlieb
• [asterisk-dev] de-latinisation of the web - http://blog.collins.net.pr/2009/12/de-latinisation-of-web.html   Dean Collins
• [asterisk-dev] DNID and RDNIS are not carried across IAX2   Pavel Troller
• [asterisk-dev] Skype-Asterisk DTMF issues with Originate command   srinivas antarvedi
• [asterisk-dev] No CDR for non-bridged outgoing calls   Klaus Darilion
• [asterisk-dev] srtp   John Todd
• [asterisk-dev] Issue related to CDR for calls dialed using Asterisk Manager Originate Command   Amit Patkar | Avhan Technologies Pvt. Ltd.
• [asterisk-dev] Professional Services required   Amit Patkar | Avhan Technologies Pvt. Ltd.
• [asterisk-dev] srtp   Olle E. Johansson
• [asterisk-dev] srtp   Steve Underwood
• [asterisk-dev] Possible Call Event Logging (CEL) Bugs   Steve Murphy
• [asterisk-dev] Professional Services required   Moises Silva
• [asterisk-dev] Possible Call Event Logging (CEL) Bugs   Tilghman Lesher
• [asterisk-dev] Propagation of cidnum/cidname to the called party in Originate cmd (1.6.1.12)   Pavel Troller
• [asterisk-dev] Professional Services required   Andrew Kohlsmith (Mailing List Account)
• [asterisk-dev] [Code Review] [15609] Prevent a stream of warnings about the voice frame queue too long   Tilghman Lesher
• [asterisk-dev] Possible Call Event Logging (CEL) Bugs   Nic Colledge
• [asterisk-dev] Possible Call Event Logging (CEL) Bugs   Tilghman Lesher
• [asterisk-dev] [asterisk-commits] oej: branch oej/deluxepine-1.4 r236897 - in /team/oej/deluxepine-1.4: include...   Kevin P. Fleming
• [asterisk-dev] [Code Review] [15609] Prevent a stream of warnings about the voice frame queue too long   Russell Bryant
• [asterisk-dev] Possible Call Event Logging (CEL) Bugs   Nic Colledge
• [asterisk-dev] Possible Call Event Logging (CEL) Bugs   Nic Colledge
• [asterisk-dev] Propagation of cidnum/cidname to the called party in Originate cmd (1.6.1.12)   Prince Singh
• [asterisk-dev] Propagation of cidnum/cidname to the called party in Originate cmd (1.6.1.12)   Pavel Troller
• [asterisk-dev] [asterisk-commits] oej: branch oej/deluxepine-1.4 r236897 - in /team/oej/deluxepine-1.4: include...   Olle E. Johansson
• [asterisk-dev] [Code Review] Added ability to perform SRV lookups for AGI URIs   Brent Thomson
• [asterisk-dev] Regarding the "progressinband" SIP option   Pavel Troller

Last message date: Thu Dec 31 23:28:34 CST 2009
Archived on: Thu Dec 31 23:28:39 CST 2009

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