October 2009 Archives by author

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Starting: Thu Oct 1 02:05:00 CDT 2009
Ending: Sat Oct 31 16:27:55 CDT 2009
Messages: 375

• [asterisk-dev] Segfault Asterisk 1.2.31.1 on Debian Lenny AMD64   Jon Bonilla (Manwe)
• [asterisk-dev] Segfault Asterisk 1.2.31.1 on Debian Lenny AMD64   Jon Bonilla (Manwe)
• [asterisk-dev] select() and friends in Asterisk limit fd usage to 1024   Benny Amorsen
• [asterisk-dev] Synchronize DND with as-feature-event   Benny Amorsen
• [asterisk-dev] [Asterisk-video] Re: Réponse en cas d'absence   Andreas Anderson
• [asterisk-dev] Asterisk Release Schedule   Michiel van Baak
• [asterisk-dev] [Code Review] SIP: peer matching by callbackextension   Michiel van Baak
• [asterisk-dev] [Code Review] SIP: Pineapple   Michiel van Baak
• [asterisk-dev] [Code Review] SIP: Pineapple   Michiel van Baak
• [asterisk-dev] [svn-commits] lmadsen: trunk r225483 - in /trunk/include/asterisk: ./ doxygen/   Michiel van Baak
• [asterisk-dev] [Code Review] SIP: Pineapple   Michiel van Baak
• [asterisk-dev] [Code Review] SIP: Pineapple   Michiel van Baak
• [asterisk-dev] upstart script [was: Re: [svn-commits] seanbright: trunk r200428 - in /trunk/contrib/upstart: ./ asterisk.upstart-0.3.9]   Michiel van Baak
• [asterisk-dev] Linked lists   Alex Balashov
• [asterisk-dev] trying to port t38 gateway patch to svn revision 222535   Niccolò Belli
• [asterisk-dev] trying to port t38 gateway patch to svn revision 222535   Niccolò Belli
• [asterisk-dev] trying to port t38 gateway patch to svn revision 222535   Niccolò Belli
• [asterisk-dev] trying to port t38 gateway patch to svn revision 222535   Niccolò Belli
• [asterisk-dev] trying to port t38 gateway patch to svn revision 222535   Niccolò Belli
• [asterisk-dev] Asterisk Release Schedule   Gregory Boehnlein
• [asterisk-dev] [Code Review] Finish implementaton of astobj2 OBJ_MULTIPLE, and convert ast_channel_iterator to use it.   Sean Bright
• [asterisk-dev] [Code Review] Finish implementaton of astobj2 OBJ_MULTIPLE, and convert ast_channel_iterator to use it.   Sean Bright
• [asterisk-dev] Ways of invoking channel read function   Rus V. Brushkoff
• [asterisk-dev] Ways of invoking channel read function   Rus V. Brushkoff
• [asterisk-dev] Ways of invoking channel read function   Rus V. Brushkoff
• [asterisk-dev] Ways of invoking channel read function   Rus V. Brushkoff
• [asterisk-dev] How to generate ast_frame with RINGBACK tone in * ?   Rus V. Brushkoff
• [asterisk-dev] How to generate ast_frame with RINGBACK tone in * ?   Rus V. Brushkoff
• [asterisk-dev] Run extension script on SIP peer registration   Russell Bryant
• [asterisk-dev] Q.SIG support   Russell Bryant
• [asterisk-dev] dvossel: trunk r222398 - in /trunk: channels/chan_sip.c configs/sip.conf.sample   Russell Bryant
• [asterisk-dev] Ways of invoking channel read function   Russell Bryant
• [asterisk-dev] Simple chan_jack.so where jack_read and jack_write are never called...   Russell Bryant
• [asterisk-dev] Asterisk Release Schedule   Russell Bryant
• [asterisk-dev] Asterisk Release Schedule   Russell Bryant
• [asterisk-dev] qwell: trunk r222548 - /trunk/configs/queues.conf.sample   Russell Bryant
• [asterisk-dev] [Code Review] Deadlock in channel masquerade handling   Russell Bryant
• [asterisk-dev] [Code Review] ast_netsock_list memory leak   Russell Bryant
• [asterisk-dev] [Code Review] SIP/IAX2 'shrinkcallerid' option   Russell Bryant
• [asterisk-dev] [Code Review] SIP TCP/TLS: move client connection setup/write into tcp helper thread, various related locking/memory fixes.   Russell Bryant
• [asterisk-dev] [Code Review] SIP: peer matching by callbackextension   Russell Bryant
• [asterisk-dev] [Code Review] SIP: peer matching by callbackextension   Russell Bryant
• [asterisk-dev] Release schedule change   Russell Bryant
• [asterisk-dev] [Code Review] IAX2: VNAK loop caused by signaling frames with no destination call number   Russell Bryant
• [asterisk-dev] [Code Review] SIP TCP/TLS: move client connection setup/write into tcp helper thread, various related locking/memory fixes.   Russell Bryant
• [asterisk-dev] Choppy Audio with JACK (app_jack)   Russell Bryant
• [asterisk-dev] New Community Developer: Max Khon   Russell Bryant
• [asterisk-dev] Choppy Audio with JACK (app_jack)   Russell Bryant
• [asterisk-dev] [Code Review] Expand availability of codec bits from 32 to 64   Russell Bryant
• [asterisk-dev] Chan_H323 vs GateKeeper Cisco   Dario Busso
• [asterisk-dev] Asterisk and Jack   Fabien COMTE
• [asterisk-dev] Questions about app_jack.c   Fabien COMTE
• [asterisk-dev] Questions about app_jack.c [solved]   Fabien COMTE
• [asterisk-dev] Questions about app_jack.c [solved]   Fabien COMTE
• [asterisk-dev] Answering a call with ALSA (or OSS)   Fabien COMTE
• [asterisk-dev] Using dmix plugin with chan_alsa.so   Fabien COMTE
• [asterisk-dev] Simple chan_jack.so where jack_read and jack_write are never called...   Fabien COMTE
• [asterisk-dev] Answering a call with ALSA (or OSS)   Fabien COMTE
• [asterisk-dev] Asterisk 1.6.2-rc2 suddenly restart   Citiwave
• [asterisk-dev] Asterisk 1.6.2-rc2 suddenly restart   Citiwave
• [asterisk-dev] Asterisk 1.6.2-rc2 suddenly restart   Citiwave
• [asterisk-dev] DAHDI_CHECK_HOOKSTATE and setting rxisoffhook   Tzafrir Cohen
• [asterisk-dev] DAHDI_CHECK_HOOKSTATE and setting rxisoffhook   Tzafrir Cohen
• [asterisk-dev] [Code Review] RTP monitoring branch: initial review   Tzafrir Cohen
• [asterisk-dev] [Code Review] RTP monitoring branch: initial review   Tzafrir Cohen
• [asterisk-dev] Channel support for GSM devices   Tzafrir Cohen
• [asterisk-dev] [Code Review] RTP monitoring branch: initial review   Tzafrir Cohen
• [asterisk-dev] make fails: dependency was previously satisfied   Tzafrir Cohen
• [asterisk-dev] dahdi svn out of sync?   Tzafrir Cohen
• [asterisk-dev] sip_write   Tzafrir Cohen
• [asterisk-dev] extra dahdi dialing format? [was: Re: [svn-commits] rmudgett: branch rmudgett/dahdi_deflection r224027 - /team/rmudgett/dahdi_def...]   Tzafrir Cohen
• [asterisk-dev] extra dahdi dialing format? [was: Re: [svn-commits] rmudgett: branch rmudgett/dahdi_deflection r224027 - /team/rmudgett/dahdi_def...]   Tzafrir Cohen
• [asterisk-dev] [Code Review] RTP monitoring branch: initial review   Tzafrir Cohen
• [asterisk-dev] [Code Review] RTP monitoring branch: initial review   Tzafrir Cohen
• [asterisk-dev] [Code Review] SIP: Pineapple   Tzafrir Cohen
• [asterisk-dev] [svn-commits] lmadsen: trunk r225483 - in /trunk/include/asterisk: ./ doxygen/   Tzafrir Cohen
• [asterisk-dev] MAX_CHANNELS in chan_dahdi   Tzafrir Cohen
• [asterisk-dev] upstart script [was: Re: [svn-commits] seanbright: trunk r200428 - in /trunk/contrib/upstart: ./ asterisk.upstart-0.3.9]   Tzafrir Cohen
• [asterisk-dev] New Community Developer: Max Khon   Tzafrir Cohen
• [asterisk-dev] [svn-commits] media_address in sip.conf   Joshua Colp
• [asterisk-dev] [Code Review] SIP: Pineapple   Klaus Darilion
• [asterisk-dev] [Code Review] SIP: Pineapple   Klaus Darilion
• [asterisk-dev] [Code Review] SIP: Pineapple   Klaus Darilion
• [asterisk-dev] [Code Review] SIP: Pineapple   Klaus Darilion
• [asterisk-dev] [Code Review] SIP: Frog   Klaus Darilion
• [asterisk-dev] [Code Review] SIP: Pineapple   Klaus Darilion
• [asterisk-dev] [Code Review] SIP: Pineapple   Klaus Darilion
• [asterisk-dev] [Code Review] SIP: Frog   Klaus Darilion
• [asterisk-dev] [Code Review] SIP: Pineapple   Klaus Darilion
• [asterisk-dev] asterisk.org - where is doxygen?   Klaus Darilion
• [asterisk-dev] asterisk.org - where is doxygen?   Klaus Darilion
• [asterisk-dev] How does Digium make their .g722 prompts   Stephen Davies
• [asterisk-dev] How does Digium make their .g722 prompts   Stephen Davies
• [asterisk-dev] How does Digium make their .g722 prompts   Stephen Davies
• [asterisk-dev] Is there an existing Function to between convert hex string and BCD data array?   Alec Davis
• [asterisk-dev] Reviewboard project titled "Add Calling and Called Subaddress support to Libpri" has error   Alec Davis
• [asterisk-dev] Reviewboard: How do you request a code review?   Alec Davis
• [asterisk-dev] Reviewboard project titled "Add Calling and Called Subaddress support to Libpri" has error   Alec Davis
• [asterisk-dev] [Code Review] Add ISDN Calling and Called Subaddress support functions to LIBPRI   Alec Davis
• [asterisk-dev] [Code Review] Add ISDN Calling and Called Subaddress support functions to LIBPRI   Alec Davis
• [asterisk-dev] [Code Review] Add Calling and Called subaddress support for Asterisk apps and funcs   Alec Davis
• [asterisk-dev] [Code Review] Add Calling and Called subaddress support for Asterisk apps and funcs   Alec Davis
• [asterisk-dev] [Code Review] Add ISDN Calling and Called Subaddress support functions to LIBPRI   Alec Davis
• [asterisk-dev] [Code Review] Add Calling and Called subaddress support for Asterisk apps and funcs   Alec Davis
• [asterisk-dev] Asterisk Release Schedule   Rod Dorman
• [asterisk-dev] Asterisk Release Schedule   Rod Dorman
• [asterisk-dev] Asterisk Release Schedule   Rod Dorman
• [asterisk-dev] Asterisk Release Schedule   Rod Dorman
• [asterisk-dev] [Code Review] Finish implementaton of astobj2 OBJ_MULTIPLE, and convert ast_channel_iterator to use it.   Kevin Fleming
• [asterisk-dev] [Code Review] Finish implementaton of astobj2 OBJ_MULTIPLE, and convert ast_channel_iterator to use it.   Kevin Fleming
• [asterisk-dev] [Code Review] Finish implementaton of astobj2 OBJ_MULTIPLE, and convert ast_channel_iterator to use it.   Kevin Fleming
• [asterisk-dev] [Code Review] Finish implementaton of astobj2 OBJ_MULTIPLE, and convert ast_channel_iterator to use it.   Kevin Fleming
• [asterisk-dev] [Code Review] Finish implementaton of astobj2 OBJ_MULTIPLE, and convert ast_channel_iterator to use it.   Kevin Fleming
• [asterisk-dev] [Code Review] Optionally build apps in utils/ directory   Kevin Fleming
• [asterisk-dev] [Code Review] SIP Contact ACL's return bad error code   Kevin Fleming
• [asterisk-dev] [asterisk-commits] tilghman: trunk r221920 - /trunk/main/logger.c   Kevin P. Fleming
• [asterisk-dev] [asterisk-commits] tilghman: trunk r221920 - /trunk/main/logger.c   Kevin P. Fleming
• [asterisk-dev] Ways of invoking channel read function   Kevin P. Fleming
• [asterisk-dev] [Code Review] RTP monitoring branch: initial review   Kevin P. Fleming
• [asterisk-dev] make fails: dependency was previously satisfied   Kevin P. Fleming
• [asterisk-dev] dahdi svn out of sync?   Kevin P. Fleming
• [asterisk-dev] 16 speech codecs only ???   Kevin P. Fleming
• [asterisk-dev] How does Digium make their .g722 prompts   Kevin P. Fleming
• [asterisk-dev] select() and friends in Asterisk limit fd usage to 1024   Kevin P. Fleming
• [asterisk-dev] [asterisk-commits] tilghman: trunk r224225 - in /trunk: include/asterisk/ main/   Kevin P. Fleming
• [asterisk-dev] How does Digium make their .g722 prompts   Kevin P. Fleming
• [asterisk-dev] [asterisk-commits] tilghman: trunk r224225 - in /trunk: include/asterisk/ main/   Kevin P. Fleming
• [asterisk-dev] [Code Review] SIP: Pineapple   Kevin P. Fleming
• [asterisk-dev] [Code Review] SIP: Pineapple   Kevin P. Fleming
• [asterisk-dev] [Code Review] SIP: Pineapple   Kevin P. Fleming
• [asterisk-dev] [Code Review] SIP: Frog   Kevin P. Fleming
• [asterisk-dev] Asterisk Release Schedule   Kevin P. Fleming
• [asterisk-dev] Confused with release numbers, how to manage incremental patching ?   Kevin P. Fleming
• [asterisk-dev] New Community Developer: Max Khon   Kevin P. Fleming
• [asterisk-dev] New Community Developer: Max Khon   Kevin P. Fleming
• [asterisk-dev] channel destruction after a transfer call   Gianpietro Germi
• [asterisk-dev] [Code Review] SIP: Pineapple   Alexander Harrowell
• [asterisk-dev] [Code Review] SIP: Pineapple   Alexander Harrowell
• [asterisk-dev] chan_datacard / GSM channel feedback   Alexander Heinz
• [asterisk-dev] [Code Review] SIP: Pineapple   Alex Hermann
• [asterisk-dev] [Code Review] SIP: Pineapple   Alex Hermann
• [asterisk-dev] [Code Review] SIP: Pineapple   Alex Hermann
• [asterisk-dev] Questions about app_jack.c [solved]   Kai Hoerner
• [asterisk-dev] extra dahdi dialing format? [was: Re: [svn-commits] rmudgett: branch rmudgett/dahdi_deflection r224027 - /team/rmudgett/dahdi_def...]   Kai Hoerner
• [asterisk-dev] Release schedule change   Kai Hoerner
• [asterisk-dev] Synchronize DND with as-feature-event   Kai Hoerner
• [asterisk-dev] Asterisk Release Schedule   Saúl Ibarra
• [asterisk-dev] Asterisk MOH playing old audio for first 30 to 60 seconds   OrangeCell Center Inc.
• [asterisk-dev] Asterisk MOH playing old audio for first 30 to 60 seconds   OrangeCell Center Inc.
• [asterisk-dev] [Code Review] Deadlock in channel masquerade handling   Olle E Johansson
• [asterisk-dev] [Code Review] SIP: peer matching by callbackextension   Olle E Johansson
• [asterisk-dev] [Code Review] AST-33: Create a list of channel variables to be posted within AMI call events   Olle E Johansson
• [asterisk-dev] [Code Review] SIP Contact ACL's return bad error code   Olle E Johansson
• [asterisk-dev] [svn-commits] mnick: branch 1.4 r221157 - in /branches/1.4: configs/ funcs/   Olle E. Johansson
• [asterisk-dev] SIP headers not available on REFER   Olle E. Johansson
• [asterisk-dev] [svn-commits] mnick: branch 1.4 r221157 - in /branches/1.4: configs/ funcs/   Olle E. Johansson
• [asterisk-dev] [Code Review] SIP: peer matching by callbackextension   Olle E. Johansson
• [asterisk-dev] Asterisk Release Schedule   Olle E. Johansson
• [asterisk-dev] Asterisk Release Schedule * suggestion for LTS   Olle E. Johansson
• [asterisk-dev] [svn-commits] media_address in sip.conf   Olle E. Johansson
• [asterisk-dev] [Code Review] SIP: peer matching by callbackextension   Olle E. Johansson
• [asterisk-dev] Release schedule change   Olle E. Johansson
• [asterisk-dev] [svn-commits] media_address in sip.conf   Olle E. Johansson
• [asterisk-dev] [Code Review] SIP: peer matching bycallbackextension   Olle E. Johansson
• [asterisk-dev] Release schedule change   Olle E. Johansson
• [asterisk-dev] [Code Review] SIP: peer matching by callbackextension   Olle E. Johansson
• [asterisk-dev] Building trunk on OS/X SNow leopard   Olle E. Johansson
• [asterisk-dev] [Code Review] SIP: Pineapple   Olle E. Johansson
• [asterisk-dev] [svn-commits] dvossel: trunk r225445 -SIP TCP/TLS: move client connection setup/write into tcp helper thread, various related locking/memory fixes   Olle E. Johansson
• [asterisk-dev] [svn-commits] lmadsen: trunk r225483 - in /trunk/include/asterisk: ./ doxygen/   Olle E. Johansson
• [asterisk-dev] [Code Review] SIP: Pineapple   Olle E. Johansson
• [asterisk-dev] [Code Review] SIP: Pineapple   Olle E. Johansson
• [asterisk-dev] [Code Review] SIP: Frog   Olle E. Johansson
• [asterisk-dev] [svn-commits] lmadsen: trunk r225483 - in /trunk/include/asterisk: ./ doxygen/   Olle E. Johansson
• [asterisk-dev] [Code Review] SIP: Pineapple   Olle E. Johansson
• [asterisk-dev] Answering a call with ALSA (or OSS)   Philipp Kempgen
• [asterisk-dev] New Community Developer: Max Khon   Max Khon
• [asterisk-dev] New Community Developer: Max Khon   Max Khon
• [asterisk-dev] New Community Developer: Max Khon   Max Khon
• [asterisk-dev] app_page code   Jeremy Kister
• [asterisk-dev] app_page code   Jeremy Kister
• [asterisk-dev] app_page code   Jeremy Kister
• [asterisk-dev] Asterisk MOH playing old audio for first 30 to 60 seconds   Jeff LaCoursiere
• [asterisk-dev] Asterisk MOH playing old audio for first 30 to 60 seconds   Jeff LaCoursiere
• [asterisk-dev] Q.SIG support   Vadim Lebedev
• [asterisk-dev] New Community Developer: Max Khon   Vadim Lebedev
• [asterisk-dev] [asterisk-commits] tilghman: trunk r221920 - /trunk/main/logger.c   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] qwell: trunk r222548 - /trunk/configs/queues.conf.sample   Tilghman Lesher
• [asterisk-dev] Asterisk Release Schedule   Tilghman Lesher
• [asterisk-dev] Asterisk Release Schedule   Tilghman Lesher
• [asterisk-dev] 16 speech codecs only ???   Tilghman Lesher
• [asterisk-dev] [asterisk-commits] tilghman: trunk r224225 - in /trunk: include/asterisk/ main/   Tilghman Lesher
• [asterisk-dev] trying to port t38 gateway patch to svn revision 222535   Tilghman Lesher
• [asterisk-dev] [Code Review] SIP/IAX2 'shrinkcallerid' option   Tilghman Lesher
• [asterisk-dev] Building trunk on OS/X SNow leopard   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] 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] Answering a call with ALSA (or OSS)   Nick Lewis
• [asterisk-dev] Using dmix plugin with chan_alsa.so   Nick Lewis
• [asterisk-dev] Channel support for GSM devices   Nick Lewis
• [asterisk-dev] Channel support for GSM devices   Nick Lewis
• [asterisk-dev] [Code Review] SIP: peer matching bycallbackextension   Nick Lewis
• [asterisk-dev] [Code Review] SIP: Pineapple   Nick Lewis
• [asterisk-dev] [Code Review] SIP: Pineapple   Nick Lewis
• [asterisk-dev] [Code Review] SIP: Pineapple   Nick Lewis
• [asterisk-dev] [Code Review] SIP: Pineapple   Nick Lewis
• [asterisk-dev] [Code Review] SIP: Frog   Nick Lewis
• [asterisk-dev] [Code Review] SIP: Frog   Nick Lewis
• [asterisk-dev] app_page code   Nick Lewis
• [asterisk-dev] app_page code   Nick Lewis
• [asterisk-dev] valgrind errors   Atis Lezdins
• [asterisk-dev] valgrind errors   Atis Lezdins
• [asterisk-dev] [asterisk-commits] tilghman: trunk r221920 - /trunk/main/logger.c   Atis Lezdins
• [asterisk-dev] select() and friends in Asterisk limit fd usage to 1024   Atis Lezdins
• [asterisk-dev] Asterisk Release Schedule   Faidon Liambotis
• [asterisk-dev] Asterisk 1.6.2-rc2 suddenly restart   Leif Madsen
• [asterisk-dev] Request for Testing: 0015757: [branch] Add support for distributing device state and MWI via XMPP PubSub   Leif Madsen
• [asterisk-dev] Channel support for GSM devices   Artem Makhutov
• [asterisk-dev] Channel support for GSM devices   Artem Makhutov
• [asterisk-dev] Channel support for GSM devices   Artem Makhutov
• [asterisk-dev] valgrind errors   Martin
• [asterisk-dev] valgrind errors   Martin
• [asterisk-dev] asterisk 2bct/rlt calling   Steve Mathers
• [asterisk-dev] asterisk 2bct/rlt calling   Steve Mathers
• [asterisk-dev] valgrind errors   Mark Michelson
• [asterisk-dev] 1.6.1: segfault ... error 6 - how to debug?   Mark Michelson
• [asterisk-dev] Linked lists   Mark Michelson
• [asterisk-dev] [Code Review] Initial review request for CCSS architecture and generic implementations   Mark Michelson
• [asterisk-dev] [Code Review] Initial review request for CCSS architecture and generic implementations   Mark Michelson
• [asterisk-dev] [Code Review] Initial review request for CCSS architecture and generic implementations   Mark Michelson
• [asterisk-dev] [Code Review] Initial review request for CCSS architecture and generic implementations   Mark Michelson
• [asterisk-dev] Release schedule change   Mark Michelson
• [asterisk-dev] [Code Review] Initial review request for CCSS architecture and generic implementations   Mark Michelson
• [asterisk-dev] [Code Review] Initial review request for CCSS architecture and generic implementations   Mark Michelson
• [asterisk-dev] Asterisk Release Schedule   Miguel Molina
• [asterisk-dev] Call Failed   Miguel Molina
• [asterisk-dev] Asterisk 1.6.2-rc2 suddenly restart   Eric Ruvalcaba Montes
• [asterisk-dev] Asterisk 1.6.2-rc2 suddenly restart   Eric Ruvalcaba Montes
• [asterisk-dev] extra dahdi dialing format? [was: Re: [svn-commits] rmudgett: branch rmudgett/dahdi_deflection r224027 - /team/rmudgett/dahdi_def...]   Richard Mudgett
• [asterisk-dev] extra dahdi dialing format? [was: Re: [svn-commits] rmudgett: branch rmudgett/dahdi_deflection r224027 - /team/rmudgett/dahdi_def...]   Richard Mudgett
• [asterisk-dev] Reviewboard project titled "Add Calling and Called Subaddress support to Libpri" has error   Richard Mudgett
• [asterisk-dev] Problem with *1.6.0.5; "Skipping dialing interface since it has already been dialed"   Håkon Nessjøen
• [asterisk-dev] asterisk-dev Digest, Vol 63, Issue 52   Justin Newman
• [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] CLI command 'manager logout <username> [from <ipaddress>]'   Matthew Nicholson
• [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 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 1.4   Matthew Nicholson
• [asterisk-dev] [Code Review] better SDP parsing algorithm for 1.4   Matthew Nicholson
• [asterisk-dev] [Code Review] Hangup callee when caller hangs up during announcements in app_dial   Matthew Nicholson
• [asterisk-dev] [Code Review] Hangup callee when caller hangs up during announcements in app_dial   Matthew Nicholson
• [asterisk-dev] [Code Review] better SDP parsing algorithm for 1.4   Matthew Nicholson
• [asterisk-dev] [Code Review] SIP/IAX2 'shrinkcallerid' option   Matthew Nicholson
• [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 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 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] better SDP parsing algorithm for trunk   Matthew Nicholson
• [asterisk-dev] [Code Review] better SDP parsing algorithm for trunk   Matthew Nicholson
• [asterisk-dev] Channel support for GSM devices   Odicha
• [asterisk-dev] Channel support for GSM devices   Odicha
• [asterisk-dev] Channel support for GSM devices   Odicha
• [asterisk-dev] Channel support for GSM devices   Odicha
• [asterisk-dev] Channel support for GSM devices   Odicha
• [asterisk-dev] Channel support for GSM devices   Odicha
• [asterisk-dev] Channel support for GSM devices   Odicha
• [asterisk-dev] Channel support for GSM devices   Odicha
• [asterisk-dev] Channel support for GSM devices   Odicha
• [asterisk-dev] New Community Developer: Max Khon   Odicha
• [asterisk-dev] New Community Developer: Max Khon   Clod Patry
• [asterisk-dev] [Code Review] reworked chan_ooh323   Jeff Peeler
• [asterisk-dev] New Community Developer: Max Khon   Philip A. Prindeville
• [asterisk-dev] Call Failed   Kelvin Quemuel
• [asterisk-dev] Call Failed   Kelvin Quemuel
• [asterisk-dev] Directory Problem on * or 0   Eric Osvaldo R
• [asterisk-dev] Directory Problem on * or 0   Eric Osvaldo R
• [asterisk-dev] Questions about app_jack.c [solved]   Matt Riddell
• [asterisk-dev] Call Failed   Matt Riddell
• [asterisk-dev] Linked lists   Luis Santana
• [asterisk-dev] Linked lists   Luis Santana
• [asterisk-dev] [Code Review] CLI command 'manager logout <username> [from <ipaddress>]'   Eliel Sardañons
• [asterisk-dev] Segfault Asterisk 1.2.31.1 on Debian Lenny AMD64   Stefan Schmidt
• [asterisk-dev] Non-existing function defined in dsp.h (Asterisk 1.6.1)   Hans Petter Selasky
• [asterisk-dev] [Code Review] RTP monitoring branch: initial review   Moises Silva
• [asterisk-dev] [Code Review] RTP monitoring branch: initial review   Moises Silva
• [asterisk-dev] [Code Review] RTP monitoring branch: initial review   Moises Silva
• [asterisk-dev] asterisk 2bct/rlt calling   Moises Silva
• [asterisk-dev] Channel support for GSM devices   Moises Silva
• [asterisk-dev] select() and friends in Asterisk limit fd usage to 1024   Moises Silva
• [asterisk-dev] Linked lists   Moises Silva
• [asterisk-dev] asterisk 2bct/rlt calling   Jared Smith
• [asterisk-dev] Fix to libss7 SAM construction bug   Jared Smith
• [asterisk-dev] AsteriskForge Now Open   Steve Sokol
• [asterisk-dev] Action registered or not..??   Chandrakant Solanki
• [asterisk-dev] sip_write   Chandrakant Solanki
• [asterisk-dev] Asterisk MOH playing old audio for first 30 to60 seconds   Jamuel P. Starkey
• [asterisk-dev] select() and friends in Asterisk limit fd usage to 1024   Kevin Stewart
• [asterisk-dev] Choppy Audio with JACK (app_jack)   Esben Stien
• [asterisk-dev] Choppy Audio with JACK (app_jack)   Esben Stien
• [asterisk-dev] Choppy Audio with JACK (app_jack)   Esben Stien
• [asterisk-dev] [Code Review] RFC5389 (STUN), basic ICE, and Jingle support   Philippe Sultan
• [asterisk-dev] Asterisk 1.4.27-rc2, 1.6.0.16-rc2, 1.6.1.7-rc2, and 1.6.2.0-rc3 Now Available   Asterisk Development Team
• [asterisk-dev] Libpri-1.4.10.2 Released   Asterisk Development Team
• [asterisk-dev] Asterisk 1.6.1.8 Now Available   Asterisk Development Team
• [asterisk-dev] AST-2009-007: ACL not respected on SIP INVITE   Asterisk Security Team
• [asterisk-dev] SIP headers not available on REFER   Brent Thomson
• [asterisk-dev] SIP headers not available on REFER   Brent Thomson
• [asterisk-dev] Question about end mark of the Called partey number   Tian
• [asterisk-dev] A bug in libss7 ISUP message construction!   Tian
• [asterisk-dev] Fix to libss7 SAM construction bug   Tian
• [asterisk-dev] Questions about app_jack.c [solved]   John Todd
• [asterisk-dev] Asterisk Release Schedule   John Todd
• [asterisk-dev] asterisk.org - where is doxygen?   John Todd
• [asterisk-dev] asterisk.org - where is doxygen?   John Todd
• [asterisk-dev] [Code Review] RTP monitoring branch: initial review   Steve Totaro
• [asterisk-dev] Question about end mark of the Called partey number   Pavel Troller
• [asterisk-dev] 16 speech codecs only ???   Pavel Troller
• [asterisk-dev] Confused with release numbers, how to manage incremental patching ?   Pavel Troller
• [asterisk-dev] [Code Review] Deadlock in channel masquerade handling   David Vossel
• [asterisk-dev] [Code Review] Deadlock in channel masquerade handling   David Vossel
• [asterisk-dev] [Code Review] SIP TCP/TLS: move client connection setup/write into tcp helper thread, various related locking/memory fixes.   David Vossel
• [asterisk-dev] [Code Review] SIP: peer matching by callbackextension   David Vossel
• [asterisk-dev] [Code Review] Deadlock in channel masquerade handling   David Vossel
• [asterisk-dev] [Code Review] SIP TCP/TLS: ensure that the contact header properly supports TLS/improved support for PAT/port redirection   David Vossel
• [asterisk-dev] [Code Review] SIP TCP/TLS: ensure that the contact header properly supports TLS/improved support for PAT/port redirection   David Vossel
• [asterisk-dev] [Code Review] Deadlock in channel masquerade handling   David Vossel
• [asterisk-dev] [Code Review] SIP TCP/TLS: ensure that the contact header properly supports TLS/improved support for PAT/port redirection   David Vossel
• [asterisk-dev] [Code Review] SIP TCP/TLS: ensure that the contact header properly supports TLS/improved support for PAT/port redirection   David Vossel
• [asterisk-dev] [Code Review] SIP: peer matching by callbackextension   David Vossel
• [asterisk-dev] [Code Review] ast_netsock_list memory leak   David Vossel
• [asterisk-dev] [Code Review] SIP: peer matching by callbackextension   David Vossel
• [asterisk-dev] [Code Review] SIP/IAX2 'shrinkcallerid' option   David Vossel
• [asterisk-dev] [Code Review] IAX2: VNAK loop caused by signaling frames with no destination call number   David Vossel
• [asterisk-dev] [Code Review] SIP TCP/TLS: move client connection setup/write into tcp helper thread, various related locking/memory fixes.   David Vossel
• [asterisk-dev] [Code Review] SIP TCP/TLS: move client connection setup/write into tcp helper thread, various related locking/memory fixes.   David Vossel
• [asterisk-dev] [Code Review] SIP: peer matching by callbackextension   David Vossel
• [asterisk-dev] Q.SIG support   Ryan Wagoner
• [asterisk-dev] Asterisk Release Schedule   Johan Wilfer
• [asterisk-dev] extra dahdi dialing format? [was: Re: [svn-commits] rmudgett: branch rmudgett/dahdi_deflection r224027 - /team/rmudgett/dahdi_def...]   Will
• [asterisk-dev] Using the digium board drive. What's the meaning of num and order in struct t4?   Will
• [asterisk-dev] Asterisk Release Schedule   Hans Witvliet
• [asterisk-dev] trying to port t38 gateway patch to svn revision 222535   Tobias Wolf
• [asterisk-dev] 1.6.1: segfault ... error 6 - how to debug?   sean darcy
• [asterisk-dev] 1.6.1: segfault ... error 6 - how to debug?   sean darcy
• [asterisk-dev] Asterisk MOH playing old audio for first 30 to 60 seconds   dimas at dataart.com
• [asterisk-dev] [Code Review] SIP/IAX2 'shrinkcallerid' option   dimas at dataart.com
• [asterisk-dev] [Code Review] Add Calling and Called subaddress support for Asterisk apps and funcs   rmudgett at digium.com
• [asterisk-dev] [Code Review] Add Calling and Called subaddress support for Asterisk apps and funcs   rmudgett at digium.com
• [asterisk-dev] [Code Review] Add ISDN Calling and Called Subaddress support functions to LIBPRI   rmudgett at digium.com
• [asterisk-dev] [Code Review] Add ISDN Calling and Called Subaddress support functions to LIBPRI   rmudgett at digium.com
• [asterisk-dev] [Code Review] Add Calling and Called subaddress support for Asterisk apps and funcs   rmudgett at digium.com
• [asterisk-dev] [Code Review] Add Calling and Called subaddress support for Asterisk apps and funcs   rmudgett at digium.com
• [asterisk-dev] [Code Review] Add ISDN Calling and Called Subaddress support functions to LIBPRI   rmudgett at digium.com
• [asterisk-dev] [Code Review] Add ISDN Calling and Called Subaddress support functions to LIBPRI   rmudgett at digium.com
• [asterisk-dev] [Code Review] Add Calling and Called subaddress support for Asterisk apps and funcs   rmudgett at digium.com
• [asterisk-dev] [Code Review] Add Calling and Called subaddress support for Asterisk apps and funcs   rmudgett at digium.com
• [asterisk-dev] [Code Review] Initial review request for CCSS architecture and generic implementations   rmudgett at digium.com
• [asterisk-dev] [Code Review] Initial review request for CCSS architecture and generic implementations   rmudgett at digium.com
• [asterisk-dev] Choppy Audio with JACK (app_jack)   russell at digium.com
• [asterisk-dev] Question about end mark of the Called partey number   sky earth
• [asterisk-dev] Fix to libss7 SAM construction bug   sky earth
• [asterisk-dev] [Code Review] RFC5389 (STUN), basic ICE, and Jingle support   vadim at mbdsys.com
• [asterisk-dev] [Code Review] Expand availability of codec bits from 32 to 64   vadim at mbdsys.com
• [asterisk-dev] [svn-commits] twilson: trunk r223874 - in /trunk: apps/ include/asterisk/ res/   asterisk at ntplx.net
• [asterisk-dev] CLI Permissions Asterisk 1.4   thiago.fernandes
• [asterisk-dev] Using the digium board drive. What's the meaning of num and order in struct t4?   张Zhang

Last message date: Sat Oct 31 16:27:55 CDT 2009
Archived on: Sat Oct 31 16:27:45 CDT 2009

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