March 2009 Archives by author

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Starting: Sun Mar 1 02:35:18 CST 2009
Ending: Tue Mar 31 20:55:27 CDT 2009
Messages: 661

• [asterisk-dev] Moving H.323 forward by linking to a new library   Vlasis Hatzistavrou (KTI)
• [asterisk-dev] About asterisk development plans   Jon Bonilla (Manwe)
• [asterisk-dev] About asterisk development plans   Jon Bonilla (Manwe)
• [asterisk-dev] About asterisk development plans   Jon Bonilla (Manwe)
• [asterisk-dev] About asterisk development plans   Jon Bonilla (Manwe)
• [asterisk-dev] About asterisk development plans   Jon Bonilla (Manwe)
• [asterisk-dev] Test framework (changed topic)   Jon Bonilla (Manwe)
• [asterisk-dev] About asterisk development plans   Jon Bonilla (Manwe)
• [asterisk-dev] About asterisk development plans   Jon Bonilla (Manwe)
• [asterisk-dev] Asterisk Network Security Idea (using tcp_wrappers)   Andrew Kohlsmith (lists)
• [asterisk-dev] Asterisk Network Security Idea ( using tcp_wrappers )   Andrew Kohlsmith (lists)
• [asterisk-dev] Multiple Agent Login   Shanavaz E A
• [asterisk-dev] **Opportunity Available for Developer in NE Atlanta Area**   Cherri Van Aken
• [asterisk-dev] Asterisk crashed!!!   Max Alex
• [asterisk-dev] Asterisk crashed!!!   Max Alex
• [asterisk-dev] CID match uses "shortest prefix match"   Benny Amorsen
• [asterisk-dev] hi   Konstantinos Arvanitis
• [asterisk-dev] Moving H.323 forward by linking to a new library   Dan Austin
• [asterisk-dev] Moving H.323 forward by linking to a new library   Dan Austin
• [asterisk-dev] Google Summer of Code 2009   Michiel van Baak
• [asterisk-dev] Google Summer of Code 2009   Michiel van Baak
• [asterisk-dev] Google Summer of Code 2009   Michiel van Baak
• [asterisk-dev] Asterisk Network Security Idea (using tcp_wrappers)   Michiel van Baak
• [asterisk-dev] Asterisk Network Security Idea (using tcp_wrappers)   Michiel van Baak
• [asterisk-dev] Asterisk v1.6 stable or testing ?   Michiel van Baak
• [asterisk-dev] About asterisk development plans   David Backeberg
• [asterisk-dev] Early Media Before 200 ok   Alex Balashov
• [asterisk-dev] [IAX2] Question on unrecoverable race condition and VNAK's Oseq   Mihai Balea
• [asterisk-dev] About asterisk development plans   Mihai Balea
• [asterisk-dev] Asterisk trunk on Mac OS/X 10.5   Fredy Alexander Ballesteros
• [asterisk-dev] get out of this lis   Fredy Alexander Ballesteros
• [asterisk-dev] Asterisk 1.4.24-rc1 Now Available   Peter Beckman
• [asterisk-dev] About asterisk development plans   Peter Beckman
• [asterisk-dev] About asterisk development plans   Peter Beckman
• [asterisk-dev] CID match uses "shortest prefix match"   Peter Beckman
• [asterisk-dev] Issue 14511   Paul Belanger
• [asterisk-dev] [IAX2] Question on unrecoverable race condition and VNAK's Oseq   Nic Bellamy
• [asterisk-dev] Idea for GSoC: Support of PRACK in Asterisk   Nic Bellamy
• [asterisk-dev] Idea for GSoC: Support of PRACK in Asterisk   Nic Bellamy
• [asterisk-dev] Idea for GSoC: Support of PRACK in Asterisk   Nic Bellamy
• [asterisk-dev] Asterisk Network Security Idea (using tcp_wrappers)   Joseph Benden
• [asterisk-dev] Asterisk Network Security Idea (using tcp_wrappers)   Joseph Benden
• [asterisk-dev] Asterisk Network Security Idea (using tcp_wrappers)   Joseph Benden
• [asterisk-dev] Asterisk Network Security Idea (using tcp_wrappers)   Joseph Benden
• [asterisk-dev] Asterisk crashed randomly   Felipe Bergo
• [asterisk-dev] Bug or not - "no reply to our critical packet" ?   Serge Berney
• [asterisk-dev] Asterisk 1.6.0.7-rc2, 1.6.1.0-rc3, 1.6.2.0-beta1 & Asterisk-Addons 1.6.0.2-rc1, 1.6.1.0-rc3 Now Available   Serge Berney
• [asterisk-dev] Asterisk v1.6 stable or testing ?   Serge Berney
• [asterisk-dev] Asterisk v1.6 stable or testing ?   Serge Berney
• [asterisk-dev] Two version of asterisk on the same server   Serge Berney
• [asterisk-dev] Google Summer of Code 2009   Gregory Boehnlein
• [asterisk-dev] [Code Review] Change T38 passthrough reinvite to use control frames   Gregory Boehnlein
• [asterisk-dev] Any specific reason that manager action Hangup has no cause header?   Gregory Boehnlein
• [asterisk-dev] About asterisk development plans   Gregory Boehnlein
• [asterisk-dev] [Code Review] Change T38 passthrough reinvite to use control frames   Gregory Boehnlein
• [asterisk-dev] Asterisk Network Security Idea ( using tcp_wrappers )   Gregory Boehnlein
• [asterisk-dev] [Code Review] Allow a position to be specified when entering a queue.   Sean Bright
• [asterisk-dev] [Code Review] Allow a position to be specified when entering a queue.   Sean Bright
• [asterisk-dev] [Code Review] Fix handling of backreferences for ENUM lookups   Sean Bright
• [asterisk-dev] Asterisk 1.4.24-rc1 Now Available   Sean Bright
• [asterisk-dev] Proposed change to branch revision blocking policy   Sean Bright
• [asterisk-dev] get out of this lis   Sean Bright
• [asterisk-dev] Feedback requested for future 1.6 API documentation   Russell Bryant
• [asterisk-dev] [Code Review] Convert pbx_spool to use string fields   Russell Bryant
• [asterisk-dev] [Code Review] Convert pbx_spool to use string fields   Russell Bryant
• [asterisk-dev] [Code Review] Gtalk call from Empathy - no corresponding codecs   Russell Bryant
• [asterisk-dev] [Code Review] app_meetme not setting filename and fileformat correctly when recording   Russell Bryant
• [asterisk-dev] [Code Review] app_read does not break from prompt loop with empty string   Russell Bryant
• [asterisk-dev] [Code Review] app_read does not break from prompt loop with empty string   Russell Bryant
• [asterisk-dev] [asterisk-commits] russell: branch 1.4 r179532 - /branches/1.4/apps/app_meetme.c   Russell Bryant
• [asterisk-dev] [asterisk-commits] russell: branch 1.4 r179532 - /branches/1.4/apps/app_meetme.c   Russell Bryant
• [asterisk-dev] [Code Review] app_read does not break from prompt loop with empty string   Russell Bryant
• [asterisk-dev] [Code Review] app_meetme not setting filename and fileformat correctly when recording   Russell Bryant
• [asterisk-dev] [Code Review] Documentation for the timing modules used in Asterisk   Russell Bryant
• [asterisk-dev] Review Board Problem   Russell Bryant
• [asterisk-dev] [Code Review] Bridging API for Conference Bridge purposes   Russell Bryant
• [asterisk-dev] [Code Review] Fix problems when RTP packet frame size is changed   Russell Bryant
• [asterisk-dev] [Code Review] make disconnect feature code available outside of bridge   Russell Bryant
• [asterisk-dev] [Code Review] make disconnect feature code available outside of bridge   Russell Bryant
• [asterisk-dev] [Code Review] New feature: Add audio announce option to app_page.c (#14365)   Russell Bryant
• [asterisk-dev] [Code Review] Add support in AEL for macro return values and direct assignment of them to variables and functions.   Russell Bryant
• [asterisk-dev] [Code Review] Fix handling of backreferences for ENUM lookups   Russell Bryant
• [asterisk-dev] [Code Review] Fix handling of backreferences for ENUM lookups   Russell Bryant
• [asterisk-dev] [Code Review] Fix handling of backreferences for ENUM lookups   Russell Bryant
• [asterisk-dev] [Code Review] Fix handling of backreferences for ENUM lookups   Russell Bryant
• [asterisk-dev] ENUM and overlapped dialing   Russell Bryant
• [asterisk-dev] race condition on bridge/masquerading?   Russell Bryant
• [asterisk-dev] Google Summer of Code 2009   Russell Bryant
• [asterisk-dev] Google Summer of Code 2009   Russell Bryant
• [asterisk-dev] [Code Review] Doxygen API Changes for 1.6.0 -> 1.6.1   Russell Bryant
• [asterisk-dev] [Code Review] Doxygen API Changes for 1.6.0 -> 1.6.1   Russell Bryant
• [asterisk-dev] [Code Review] Doxygen API Changes for 1.6.0 -> 1.6.1   Russell Bryant
• [asterisk-dev] Google Summer of Code 2009   Russell Bryant
• [asterisk-dev] Google Summer of Code 2009   Russell Bryant
• [asterisk-dev] Google Summer of Code 2009   Russell Bryant
• [asterisk-dev] Google Summer of Code 2009   Russell Bryant
• [asterisk-dev] Google Summer of Code 2009   Russell Bryant
• [asterisk-dev] Google Summer of Code 2009   Russell Bryant
• [asterisk-dev] Asterisk trunk on Mac OS/X 10.5   Russell Bryant
• [asterisk-dev] Asterisk trunk on Mac OS/X 10.5   Russell Bryant
• [asterisk-dev] [Code Review] IAX2 retransmit with encryption enabled fix   Russell Bryant
• [asterisk-dev] [Code Review] IAX2 retransmit with encryption enabled fix   Russell Bryant
• [asterisk-dev] [Code Review] IAX2 retransmit with encryption enabled fix   Russell Bryant
• [asterisk-dev] [Code Review] IAX2 retransmit with encryption enabled fix   Russell Bryant
• [asterisk-dev] request for doc about manager command MeetmeMute   Russell Bryant
• [asterisk-dev] [Code Review] Randomize IAX2 encryption padding   Russell Bryant
• [asterisk-dev] [Code Review] Randomize IAX2 encryption padding   Russell Bryant
• [asterisk-dev] Asterisk trunk on Mac OS/X 10.5   Russell Bryant
• [asterisk-dev] Asterisk trunk on Mac OS/X 10.5   Russell Bryant
• [asterisk-dev] Control payback bookmarks   Russell Bryant
• [asterisk-dev] Asterisk 1.6.2 branch now created   Russell Bryant
• [asterisk-dev] [Code Review] Improve behavior of ast_answer() to not lose incoming frames   Russell Bryant
• [asterisk-dev] [Code Review] Improve behavior of ast_answer() to not lose incoming frames   Russell Bryant
• [asterisk-dev] [Code Review] Improve behavior of ast_answer() to not lose incoming frames   Russell Bryant
• [asterisk-dev] [Code Review] Improve behavior of ast_answer() to not lose incoming frames   Russell Bryant
• [asterisk-dev] [Code Review] Ensure internal poll is used when we ask for it   Russell Bryant
• [asterisk-dev] [Code Review] Improve behavior of ast_answer() to not lose incoming frames   Russell Bryant
• [asterisk-dev] pseudo channel   Russell Bryant
• [asterisk-dev] [Code Review] Add name to CHANNEL() function   Russell Bryant
• [asterisk-dev] Proposed change to branch revision blocking policy   Russell Bryant
• [asterisk-dev] pseudo channel   Russell Bryant
• [asterisk-dev] Any specific reason that manager action Hangup has no cause header?   Russell Bryant
• [asterisk-dev] Google Summer of Code 2009 - We're in!   Russell Bryant
• [asterisk-dev] [Code Review] Merge Called party identification changes into trunk.   Russell Bryant
• [asterisk-dev] [Code Review] Merge Called party identification changes into trunk.   Russell Bryant
• [asterisk-dev] [Code Review] Change T38 passthrough reinvite to use control frames   Russell Bryant
• [asterisk-dev] dvossel: branch 1.4 r183126 - in /branches/1.4: apps/ include/asterisk/ res/   Russell Bryant
• [asterisk-dev] dvossel: branch 1.4 r183126 - in /branches/1.4: apps/ include/asterisk/ res/   Russell Bryant
• [asterisk-dev] [Code Review] Convert ast_channel over to astobj2   Russell Bryant
• [asterisk-dev] [Code Review] Convert ast_channel over to astobj2   Russell Bryant
• [asterisk-dev] [Code Review] Convert ast_channel over to astobj2   Russell Bryant
• [asterisk-dev] [Code Review] Convert ast_channel over to astobj2   Russell Bryant
• [asterisk-dev] [Code Review] Convert ast_channel over to astobj2   Russell Bryant
• [asterisk-dev] [Code Review] Convert ast_channel over to astobj2   Russell Bryant
• [asterisk-dev] [Code Review] Convert ast_channel over to astobj2   Russell Bryant
• [asterisk-dev] [Code Review] Convert ast_channel over to astobj2   Russell Bryant
• [asterisk-dev] [Code Review] Convert ast_channel over to astobj2   Russell Bryant
• [asterisk-dev] [Code Review] Merge Called party identification changes into trunk.   Russell Bryant
• [asterisk-dev] [Code Review] Merge Called party identification changes into trunk.   Russell Bryant
• [asterisk-dev] [Code Review] Merge Called party identification changes into trunk.   Russell Bryant
• [asterisk-dev] [Code Review] Merge Called party identification changes into trunk.   Russell Bryant
• [asterisk-dev] [Code Review] Coding Guidelines, API naming convention, Line length limit   Russell Bryant
• [asterisk-dev] [Code Review] Convert ast_channel over to astobj2   Russell Bryant
• [asterisk-dev] [Code Review] Convert ast_channel over to astobj2   Russell Bryant
• [asterisk-dev] [Code Review] Improve event API cache performance   Russell Bryant
• [asterisk-dev] [Code Review] Improve event API cache performance   Russell Bryant
• [asterisk-dev] channel.c locking and other   Russell Bryant
• [asterisk-dev] [Code Review] Merge Called party identification changes into trunk.   Russell Bryant
• [asterisk-dev] [Code Review] Improve event API cache performance   Russell Bryant
• [asterisk-dev] [Code Review] Improve event API cache performance   Russell Bryant
• [asterisk-dev] [Code Review] Improve event API cache performance   Russell Bryant
• [asterisk-dev] [Code Review] Convert ast_channel over to astobj2   Russell Bryant
• [asterisk-dev] [Code Review] Convert ast_channel over to astobj2   Russell Bryant
• [asterisk-dev] [Code Review] Coding Guidelines, API naming convention, Line length limit   Russell Bryant
• [asterisk-dev] [Code Review] Improve event API cache performance   Russell Bryant
• [asterisk-dev] [Code Review] Improve event API cache performance   Russell Bryant
• [asterisk-dev] [Code Review] DTMF no wait after last digit   Russell Bryant
• [asterisk-dev] [Code Review] DTMF no wait after last digit   Russell Bryant
• [asterisk-dev] gsoc: verbose logging   Russell Bryant
• [asterisk-dev] [Code Review] Improve timing interface to remember which provider provided a timer   Russell Bryant
• [asterisk-dev] [Code Review] Avoid wasteful frame queue traversals in chan_iax2 network_thread()   Russell Bryant
• [asterisk-dev] [Code Review] Avoid wasteful frame queue traversals in chan_iax2 network_thread()   Russell Bryant
• [asterisk-dev] [Code Review] Improve frame_queue handing in chan_iax2   Russell Bryant
• [asterisk-dev] [Code Review] Improve frame_queue handing in chan_iax2   Russell Bryant
• [asterisk-dev] [Code Review] Improve frame_queue handing in chan_iax2   Russell Bryant
• [asterisk-dev] [Code Review] Backport of state_interface for app_queue in Asterisk 1.4   Russell Bryant
• [asterisk-dev] [Code Review] Fix queue weight behavior so that calls in low-weighted queues are not blocked when they should not be.   Russell Bryant
• [asterisk-dev] [Code Review] Modularized RTP stack support   Russell Bryant
• [asterisk-dev] [Code Review] Backport of state_interface for app_queue in Asterisk 1.4   Russell Bryant
• [asterisk-dev] [Code Review] Create API for doing variable substitution with dynamic buffers   Russell Bryant
• [asterisk-dev] [Code Review] Create API for doing variable substitution with dynamic buffers   Russell Bryant
• [asterisk-dev] [Code Review] Modularized RTP stack support   Russell Bryant
• [asterisk-dev] [Code Review] Modularized RTP stack support   Russell Bryant
• [asterisk-dev] [Code Review] Modularized RTP stack support   Russell Bryant
• [asterisk-dev] mmichelson: trunk r185704 - in /trunk: CHANGES main/manager.c   Russell Bryant
• [asterisk-dev] dahdi channel forcing AST_FORMAT_MULAW frames to channels not allowing them.   Caballero Collado, Mª Elena
• [asterisk-dev] About asterisk development plans   Iñaki Baz Castillo
• [asterisk-dev] Ho to know the connection between 2 channels when one of them is still ringing ?   Iñaki Baz Castillo
• [asterisk-dev] Idea for GSoC: Support of PRACK in Asterisk   Iñaki Baz Castillo
• [asterisk-dev] Feedback requested for future 1.6 API   Nicolas Chapleau
• [asterisk-dev] [policy] Language specific prompts   David Chappell
• [asterisk-dev] [policy] Language specific prompts   David Chappell
• [asterisk-dev] Cisco IP Phone 7940G.   Sergio Chersovani
• [asterisk-dev] Feedback requested for future 1.6 API documentation   Tzafrir Cohen
• [asterisk-dev] following asterisk with git-svn   Tzafrir Cohen
• [asterisk-dev] Feedback requested for future 1.6 API documentation   Tzafrir Cohen
• [asterisk-dev] Add Farsi Say Date   Tzafrir Cohen
• [asterisk-dev] Porting applications from 1.4 to 1.6 or trunk   Tzafrir Cohen
• [asterisk-dev] Google Summer of Code 2009   Tzafrir Cohen
• [asterisk-dev] dahdi-linux include/ and out-of-tree modules   Tzafrir Cohen
• [asterisk-dev] dahdi-linux include/ and out-of-tree modules   Tzafrir Cohen
• [asterisk-dev] dahdi-linux include/ and out-of-tree modules   Tzafrir Cohen
• [asterisk-dev] About asterisk development plans   Tzafrir Cohen
• [asterisk-dev] About asterisk development plans   Tzafrir Cohen
• [asterisk-dev] About asterisk development plans   Tzafrir Cohen
• [asterisk-dev] some memory leak present in asterisk   Tzafrir Cohen
• [asterisk-dev] Asterisk Network Security Idea (using tcp_wrappers)   Tzafrir Cohen
• [asterisk-dev] Two version of asterisk on the same server   Tzafrir Cohen
• [asterisk-dev] [Code Review] Coding Guidelines, API naming convention, Line length limit   Tzafrir Cohen
• [asterisk-dev] [Code Review] Coding Guidelines, API naming convention, Line length limit   Tzafrir Cohen
• [asterisk-dev] Two version of asterisk on the same server   Tzafrir Cohen
• [asterisk-dev] Google Summer of Code 2009   Dean Collins
• [asterisk-dev] Google Summer of Code 2009   Dean Collins
• [asterisk-dev] [RFC] Audiohooks   Joshua Colp
• [asterisk-dev] [Code Review] Bridging API for Conference Bridge purposes   Joshua Colp
• [asterisk-dev] [Code Review] Fix problems when RTP packet frame size is changed   Joshua Colp
• [asterisk-dev] [Code Review] Bridging API for Conference Bridge purposes   Joshua Colp
• [asterisk-dev] [Code Review] Bridging API for Conference Bridge purposes   Joshua Colp
• [asterisk-dev] Google Summer of Code 2009   Joshua Colp
• [asterisk-dev] [Code Review] Doxygen API Changes for 1.6.0 -> 1.6.1   Joshua Colp
• [asterisk-dev] Asterisk trunk on Mac OS/X 10.5   Joshua Colp
• [asterisk-dev] Asterisk trunk on Mac OS/X 10.5   Joshua Colp
• [asterisk-dev] Asterisk trunk on Mac OS/X 10.5   Joshua Colp
• [asterisk-dev] [Code Review] Improve behavior of ast_answer() to not lose incoming frames   Joshua Colp
• [asterisk-dev] [Code Review] Ensure internal poll is used when we ask for it   Joshua Colp
• [asterisk-dev] [Code Review] Change T38 passthrough reinvite to use control frames   Joshua Colp
• [asterisk-dev] [Code Review] Change T38 passthrough reinvite to use control frames   Joshua Colp
• [asterisk-dev] [Code Review] Change T38 passthrough reinvite to use control frames   Joshua Colp
• [asterisk-dev] [Code Review] Change T38 passthrough reinvite to use control frames   Joshua Colp
• [asterisk-dev] [Code Review] Change T38 passthrough reinvite to use control frames   Joshua Colp
• [asterisk-dev] [Code Review] SIP Preferred codec only   Joshua Colp
• [asterisk-dev] [Code Review] SIP Preferred codec only   Joshua Colp
• [asterisk-dev] [Code Review] Improved support for T38 on initial INVITE   Joshua Colp
• [asterisk-dev] [Code Review] Improved support for T38 on initial INVITE   Joshua Colp
• [asterisk-dev] [Code Review] Modularized RTP stack support   Joshua Colp
• [asterisk-dev] [Code Review] Modularized RTP stack support   Joshua Colp
• [asterisk-dev] [Code Review] Improve timing interface to remember which provider provided a timer   Joshua Colp
• [asterisk-dev] [Code Review] Modularized RTP stack support   Joshua Colp
• [asterisk-dev] [Code Review] Modularized RTP stack support   Joshua Colp
• [asterisk-dev] [Code Review] Modularized RTP stack support   Joshua Colp
• [asterisk-dev] [Code Review] Modularized RTP stack support   Joshua Colp
• [asterisk-dev] ENUMLOOKUP - regex broken?   Lawrence Conroy
• [asterisk-dev] ENUMLOOKUP - regex broken?   Lawrence Conroy
• [asterisk-dev] integrate voip into a web based application   Gustavo Cordeiro
• [asterisk-dev] integrate voip into a web based application   Daniel
• [asterisk-dev] sip contact suffix on header   Klaus Darilion
• [asterisk-dev] Porting applications from 1.4 to 1.6 or trunk   Klaus Darilion
• [asterisk-dev] sip contact suffix on header   Klaus Darilion
• [asterisk-dev] ENUM and overlapped dialing   Klaus Darilion
• [asterisk-dev] What happened to Defaddr->IP ?   Klaus Darilion
• [asterisk-dev] Moving H.323 forward by linking to a new library   Klaus Darilion
• [asterisk-dev] CID match uses "shortest prefix match"   Klaus Darilion
• [asterisk-dev] CID match uses "shortest prefix match"   Klaus Darilion
• [asterisk-dev] Early Media Before 200 ok   Klaus Darilion
• [asterisk-dev] CID match uses "shortest prefix match"   Klaus Darilion
• [asterisk-dev] CID match uses "shortest prefix match"   Klaus Darilion
• [asterisk-dev] CID match uses "shortest prefix match"   Klaus Darilion
• [asterisk-dev] CID match uses "shortest prefix match"   Klaus Darilion
• [asterisk-dev] CID match uses "shortest prefix match"   Klaus Darilion
• [asterisk-dev] Bug or not - "no reply to our critical packet" ?   Klaus Darilion
• [asterisk-dev] CID match uses "shortest prefix match"   Klaus Darilion
• [asterisk-dev] Idea for GSoC: Support of PRACK in Asterisk   Klaus Darilion
• [asterisk-dev] [Code Review] SIP Preferred codec only   Klaus Darilion
• [asterisk-dev] SIP and using less-prior codec   Klaus Darilion
• [asterisk-dev] channel formats and ast_set_read/write_format   Klaus Darilion
• [asterisk-dev] About asterisk development plans   Stephen Davies
• [asterisk-dev] CID match uses "shortest prefix match"   Steve Davies
• [asterisk-dev] CID match uses "shortest prefix match"   Steve Davies
• [asterisk-dev] pseudo channel   Steve Davies
• [asterisk-dev] **Opportunity Available for Developer in NE Atlanta Area**   Rod Dorman
• [asterisk-dev] Mark Spencer's transcript regarding Skype for Asterisk   Lee S Dryburgh
• [asterisk-dev] gsoc: verbose logging   Pék Dániel
• [asterisk-dev] Asterisk Network Security Idea (using tcp_wrappers)   Steve Edwards
• [asterisk-dev] Asterisk Network Security Idea (using tcp_wrappers)   Steve Edwards
• [asterisk-dev] [Code Review] Coding Guidelines, API naming convention, Line length limit   Steve Edwards
• [asterisk-dev] [Code Review] Coding Guidelines, API naming convention, Line length limit   Steve Edwards
• [asterisk-dev] [Code Review] Coding Guidelines, API naming convention, Line length limit   Steve Edwards
• [asterisk-dev] h extension not considered within a macro for Asterisk 1.6.0.6 and later   Emrah
• [asterisk-dev] h extension not considered within a macro for Asterisk 1.6.0.6 and later   Emrah
• [asterisk-dev] [Code Review] Documentation for the timing modules used in Asterisk   Kevin Fleming
• [asterisk-dev] [Code Review] Fix problems when RTP packet frame size is changed   Kevin Fleming
• [asterisk-dev] [Code Review] Fix problems when RTP packet frame size is changed   Kevin Fleming
• [asterisk-dev] [Code Review] Fix problems when RTP packet frame size is changed   Kevin Fleming
• [asterisk-dev] [Code Review] Fix problems when RTP packet frame size is changed   Kevin Fleming
• [asterisk-dev] [Code Review] Make the parameter separator backward compatible, and error messages more consistent.   Kevin Fleming
• [asterisk-dev] [Code Review] Improve behavior of ast_answer() to not lose incoming frames   Kevin Fleming
• [asterisk-dev] [Code Review] Improve behavior of ast_answer() to not lose incoming frames   Kevin Fleming
• [asterisk-dev] [Code Review] Improve behavior of ast_answer() to not lose incoming frames   Kevin Fleming
• [asterisk-dev] [Code Review] Improve behavior of ast_answer() to not lose incoming frames   Kevin Fleming
• [asterisk-dev] [Code Review] Improve timing interface to remember which provider provided a timer   Kevin Fleming
• [asterisk-dev] [Code Review] Improve timing interface to remember which provider provided a timer   Kevin Fleming
• [asterisk-dev] [Code Review] Documentation for the timing modules used in Asterisk   Kevin P. Fleming
• [asterisk-dev] [asterisk-commits] russell: branch 1.4 r179532 - /branches/1.4/apps/app_meetme.c   Kevin P. Fleming
• [asterisk-dev] [asterisk-commits] russell: branch 1.4 r179532 - /branches/1.4/apps/app_meetme.c   Kevin P. Fleming
• [asterisk-dev] [Code Review] IAX2 retransmit with encryption enabled fix   Kevin P. Fleming
• [asterisk-dev] Moving H.323 forward by linking to a new library   Kevin P. Fleming
• [asterisk-dev] Moving H.323 forward by linking to a new library   Kevin P. Fleming
• [asterisk-dev] [svn-commits] kpfleming: branch 1.4 r182802 - in /branches/1.4: ./ apps/ build_tools/ inclu...   Kevin P. Fleming
• [asterisk-dev] Idea for GSoC: Support of PRACK in Asterisk   Kevin P. Fleming
• [asterisk-dev] Idea for GSoC: Support of PRACK in Asterisk   Kevin P. Fleming
• [asterisk-dev] Idea for GSoC: Support of PRACK in Asterisk   Kevin P. Fleming
• [asterisk-dev] Idea for GSoC: Support of PRACK in Asterisk   Kevin P. Fleming
• [asterisk-dev] kpfleming: branch 1.4 r184447 - /branches/1.4/sounds/Makefile   Kevin P. Fleming
• [asterisk-dev] Asterisk Core Sounds 1.4.15, Extra Sounds 1.4.8, and Freeplay MoH Update Released   Kevin P. Fleming
• [asterisk-dev] Need an IVR system for Banking Transactions   Jeff Gehlbach
• [asterisk-dev] Asterisk Network Security Idea (using tcp_wrappers)   Jeff Gehlbach
• [asterisk-dev] ACK - wrong URI?   Santiago Gimeno
• [asterisk-dev] About asterisk development plans   Gavin Henry
• [asterisk-dev] About asterisk development plans   Gavin Henry
• [asterisk-dev] About asterisk development plans   Gavin Henry
• [asterisk-dev] About asterisk development plans   Gavin Henry
• [asterisk-dev] Test framework (changed topic)   Gavin Henry
• [asterisk-dev] About asterisk development plans   Gavin Henry
• [asterisk-dev] [IAX2] Question on unrecoverable race condition and VNAK's Oseq   Alex Hermann
• [asterisk-dev] [IAX2] Question on unrecoverable race condition and VNAK's Oseq   Alex Hermann
• [asterisk-dev] CID match uses "shortest prefix match"   Alex Hermann
• [asterisk-dev] Can I tell if a call picked up on PSTN extension... for example?   Michael Higgins
• [asterisk-dev] Asterisk crashed randomly   Ngo-Vi Hoai-Anh
• [asterisk-dev] Developer needed   Steve Howes
• [asterisk-dev] Asterisk trunk on Mac OS/X 10.5   Steve Howes
• [asterisk-dev] Can I tell if a call picked up on PSTN extension... for example?   Steve Howes
• [asterisk-dev] Help T.38   Dwayne Hubbard
• [asterisk-dev] About asterisk development plans   Saúl Ibarra
• [asterisk-dev] h extension not considered within a macro for Asterisk 1.6.0.6 and later   Saúl Ibarra
• [asterisk-dev] url in dial command   Giorgio Incantalupo
• [asterisk-dev] sip contact suffix on header   Raj Jain
• [asterisk-dev] Idea for GSoC: Support of PRACK in Asterisk   Raj Jain
• [asterisk-dev] [Code Review] Doxygen API Changes for 1.6.0 -> 1.6.1   Olle E Johansson
• [asterisk-dev] [Code Review] Add name to CHANNEL() function   Olle E Johansson
• [asterisk-dev] BYE to 408 Request Timeout - is this correct?   Olle E. Johansson
• [asterisk-dev] Feedback requested for future 1.6 API documentation   Olle E. Johansson
• [asterisk-dev] sip contact suffix on header   Olle E. Johansson
• [asterisk-dev] sip contact suffix on header   Olle E. Johansson
• [asterisk-dev] Google Summer of Code 2009   Olle E. Johansson
• [asterisk-dev] Google Summer of Code 2009   Olle E. Johansson
• [asterisk-dev] Google Summer of Code 2009   Olle E. Johansson
• [asterisk-dev] What happened to Defaddr->IP ?   Olle E. Johansson
• [asterisk-dev] ACK - wrong URI?   Olle E. Johansson
• [asterisk-dev] ACK - wrong URI?   Olle E. Johansson
• [asterisk-dev] About asterisk development plans   Olle E. Johansson
• [asterisk-dev] [Code Review] Merge Called party identification changes into trunk.   Olle E. Johansson
• [asterisk-dev] Any specific reason that manager action Hangup has no cause header?   Olle E. Johansson
• [asterisk-dev] About asterisk development plans   Olle E. Johansson
• [asterisk-dev] About asterisk development plans   Olle E. Johansson
• [asterisk-dev] Test framework (changed topic)   Olle E. Johansson
• [asterisk-dev] Idea for GSoC: Support of PRACK in Asterisk   Olle E. Johansson
• [asterisk-dev] Asterisk Network Security Idea (using tcp_wrappers)   Olle E. Johansson
• [asterisk-dev] Review Board Problem   Donny Kavanagh
• [asterisk-dev] H323 hangup cause   Mindaugas Kezys
• [asterisk-dev] Asterisk Core Sounds 1.4.15, Extra Sounds 1.4.8, and Freeplay MoH Update Released   Kristian Kielhofner
• [asterisk-dev] New Bounty   Matt King
• [asterisk-dev] [RFC] Audiohooks   Kaloyan Kovachev
• [asterisk-dev] Help T.38   Kaloyan Kovachev
• [asterisk-dev] [RFC] Audiohooks   Kaloyan Kovachev
• [asterisk-dev] Feedback requested for future 1.6 API   Kaloyan Kovachev
• [asterisk-dev] [RFC] Audiohooks   Kaloyan Kovachev
• [asterisk-dev] Any specific reason that manager action Hangup has no cause header?   Kaloyan Kovachev
• [asterisk-dev] - meetme delay (asterisk 1.2.18)   Jaco Labuschagne
• [asterisk-dev] What happened to Defaddr->IP ?   John Lange
• [asterisk-dev] What happened to Defaddr->IP ?   John Lange
• [asterisk-dev] [Code Review] Change T38 passthrough reinvite to use control frames   John Lange
• [asterisk-dev] [Code Review] Change T38 passthrough reinvite to use control frames   John Lange
• [asterisk-dev] About asterisk development plans   John Lange
• [asterisk-dev] About asterisk development plans   John Lange
• [asterisk-dev] About asterisk development plans   John Lange
• [asterisk-dev] About asterisk development plans   John Lange
• [asterisk-dev] Review Board Problem   Vadim Lebedev
• [asterisk-dev] Review Board Problem   Vadim Lebedev
• [asterisk-dev] Invite a person to a conf   Vadim Lebedev
• [asterisk-dev] Asterisk trunk on Mac OS/X 10.5   Vadim Lebedev
• [asterisk-dev] Asterisk trunk on Mac OS/X 10.5   Vadim Lebedev
• [asterisk-dev] Asterisk trunk on Mac OS/X 10.5   Vadim Lebedev
• [asterisk-dev] Asterisk trunk on Mac OS/X 10.5   Vadim Lebedev
• [asterisk-dev] Asterisk trunk on Mac OS/X 10.5   Vadim Lebedev
• [asterisk-dev] Asterisk trunk on Mac OS/X 10.5   Vadim Lebedev
• [asterisk-dev] Asterisk trunk on Mac OS/X 10.5   Vadim Lebedev
• [asterisk-dev] Asterisk trunk on Mac OS/X 10.5   Vadim Lebedev
• [asterisk-dev] [Code Review] Merge Called party identification changes into trunk.   Vadim Lebedev
• [asterisk-dev] [RFC] Audiohooks   Tilghman Lesher
• [asterisk-dev] Feedback requested for future 1.6 API   Tilghman Lesher
• [asterisk-dev] Google Summer of Code 2009   Tilghman Lesher
• [asterisk-dev] What happened to Defaddr->IP ?   Tilghman Lesher
• [asterisk-dev] Moving H.323 forward by linking to a new library   Tilghman Lesher
• [asterisk-dev] Moving H.323 forward by linking to a new library   Tilghman Lesher
• [asterisk-dev] CID match uses "shortest prefix match"   Tilghman Lesher
• [asterisk-dev] CID match uses "shortest prefix match"   Tilghman Lesher
• [asterisk-dev] Proposed change to branch revision blocking policy   Tilghman Lesher
• [asterisk-dev] SIP Call: Change number to reach without altering Contact IP   Marc Leurent
• [asterisk-dev] Queue strategy with Ringall and multiple agents answering simultenously.   Atis Lezdins
• [asterisk-dev] garbled audio   Sam Liddicott
• [asterisk-dev] Feedback requested for future 1.6 API documentation   Sam Liddicott
• [asterisk-dev] add a new queue strategy: SBR   Fredrik Liljegren
• [asterisk-dev] Control payback bookmarks   Julian Lyndon-Smith
• [asterisk-dev] Two version of asterisk on the same server   Julian Lyndon-Smith
• [asterisk-dev] Two version of asterisk on the same server   Julian Lyndon-Smith
• [asterisk-dev] BYE to 408 Request Timeout - is this correct?   Chris Maciejewski
• [asterisk-dev] ENUMLOOKUP - regex broken?   Chris Maciejewski
• [asterisk-dev] ACK - wrong URI?   Chris Maciejewski
• [asterisk-dev] ACK - wrong URI?   Chris Maciejewski
• [asterisk-dev] ACK - wrong URI?   Chris Maciejewski
• [asterisk-dev] unanswered=yes not working in 1.6.1 ?   Chris Maciejewski
• [asterisk-dev] About asterisk development plans   Leif Madsen
• [asterisk-dev] About asterisk development plans   Leif Madsen
• [asterisk-dev] About asterisk development plans   Leif Madsen
• [asterisk-dev] About asterisk development plans   Leif Madsen
• [asterisk-dev] CID match uses "shortest prefix match"   Leif Madsen
• [asterisk-dev] CID match uses "shortest prefix match"   Leif Madsen
• [asterisk-dev] CID match uses "shortest prefix match"   Leif Madsen
• [asterisk-dev] CID match uses "shortest prefix match"   Leif Madsen
• [asterisk-dev] CID match uses "shortest prefix match"   Leif Madsen
• [asterisk-dev] App to detect fax/voice   Robert McGilvray
• [asterisk-dev] [Code Review] Convert pbx_spool to use string fields   Mark Michelson
• [asterisk-dev] Feedback requested for future 1.6 API documentation   Mark Michelson
• [asterisk-dev] [Code Review] Documentation for the timing modules used in Asterisk   Mark Michelson
• [asterisk-dev] [Code Review] Documentation for the timing modules used in Asterisk   Mark Michelson
• [asterisk-dev] [Code Review] Documentation for the timing modules used in Asterisk   Mark Michelson
• [asterisk-dev] Review Board Problem   Mark Michelson
• [asterisk-dev] [Code Review] Create API for doing variable substitution with dynamic buffers   Mark Michelson
• [asterisk-dev] [Code Review] String field test module   Mark Michelson
• [asterisk-dev] [Code Review] String field test module   Mark Michelson
• [asterisk-dev] [Code Review] String field test module   Mark Michelson
• [asterisk-dev] Add Farsi Say Date   Mark Michelson
• [asterisk-dev] [Code Review] Allow a position to be specified when entering a queue.   Mark Michelson
• [asterisk-dev] [Code Review] Allow a position to be specified when entering a queue.   Mark Michelson
• [asterisk-dev] [Code Review] Allow a position to be specified when entering a queue.   Mark Michelson
• [asterisk-dev] Which channel answer call in Queue   Mark Michelson
• [asterisk-dev] Google Summer of Code 2009   Mark Michelson
• [asterisk-dev] Google Summer of Code 2009   Mark Michelson
• [asterisk-dev] [Code Review] SRTP support for Asterisk   Mark Michelson
• [asterisk-dev] [Code Review] SRTP support for Asterisk   Mark Michelson
• [asterisk-dev] Asterisk trunk on Mac OS/X 10.5   Mark Michelson
• [asterisk-dev] [Code Review] Various fixes for extenpatternmatchnew   Mark Michelson
• [asterisk-dev] [Code Review] Allow disconnect feature before a call is bridged   Mark Michelson
• [asterisk-dev] [Code Review] Improve behavior of ast_answer() to not lose incoming frames   Mark Michelson
• [asterisk-dev] [Code Review] Add name to CHANNEL() function   Mark Michelson
• [asterisk-dev] [Code Review] inotify(7) support for pbx_spool   Mark Michelson
• [asterisk-dev] [Code Review] Change T38 passthrough reinvite to use control frames   Mark Michelson
• [asterisk-dev] [Code Review] Merge Called party identification changes into trunk.   Mark Michelson
• [asterisk-dev] [Code Review] Allow disconnect feature before a call is bridged   Mark Michelson
• [asterisk-dev] [Code Review] Merge Called party identification changes into trunk.   Mark Michelson
• [asterisk-dev] [Code Review] Merge Called party identification changes into trunk.   Mark Michelson
• [asterisk-dev] [Code Review] Merge Called party identification changes into trunk.   Mark Michelson
• [asterisk-dev] [Code Review] Fix queue weight behavior so that calls in low-weighted queues are not blocked when they should not be.   Mark Michelson
• [asterisk-dev] [Code Review] Allow disconnect feature before a call is bridged   Mark Michelson
• [asterisk-dev] [Code Review] Allow disconnect feature before a call is bridged   Mark Michelson
• [asterisk-dev] [Code Review] Merge Called party identification changes into trunk.   Mark Michelson
• [asterisk-dev] [Code Review] Merge Called party identification changes into trunk.   Mark Michelson
• [asterisk-dev] [Code Review] Merge Called party identification changes into trunk.   Mark Michelson
• [asterisk-dev] [Code Review] Merge Called party identification changes into trunk.   Mark Michelson
• [asterisk-dev] [Code Review] Convert ast_channel over to astobj2   Mark Michelson
• [asterisk-dev] [Code Review] Convert ast_channel over to astobj2   Mark Michelson
• [asterisk-dev] [Code Review] Convert ast_channel over to astobj2   Mark Michelson
• [asterisk-dev] [Code Review] Convert ast_channel over to astobj2   Mark Michelson
• [asterisk-dev] [Code Review] Convert ast_channel over to astobj2   Mark Michelson
• [asterisk-dev] [Code Review] Merge Called party identification changes into trunk.   Mark Michelson
• [asterisk-dev] [Code Review] Merge Called party identification changes into trunk.   Mark Michelson
• [asterisk-dev] [Code Review] Convert ast_channel over to astobj2   Mark Michelson
• [asterisk-dev] [Code Review] Merge Called party identification changes into trunk.   Mark Michelson
• [asterisk-dev] [Code Review] Merge Called party identification changes into trunk.   Mark Michelson
• [asterisk-dev] [Code Review] Merge Called party identification changes into trunk.   Mark Michelson
• [asterisk-dev] [Code Review] Merge Called party identification changes into trunk.   Mark Michelson
• [asterisk-dev] [Code Review] Merge Called party identification changes into trunk.   Mark Michelson
• [asterisk-dev] [Code Review] Coding Guidelines, API naming convention, Line length limit   Mark Michelson
• [asterisk-dev] [Code Review] Merge Called party identification changes into trunk.   Mark Michelson
• [asterisk-dev] [Code Review] Merge Called party identification changes into trunk.   Mark Michelson
• [asterisk-dev] [Code Review] Merge Called party identification changes into trunk.   Mark Michelson
• [asterisk-dev] [Code Review] Improve event API cache performance   Mark Michelson
• [asterisk-dev] [Code Review] Improve event API cache performance   Mark Michelson
• [asterisk-dev] [Code Review] Improve event API cache performance   Mark Michelson
• [asterisk-dev] [Code Review] Merge Called party identification changes into trunk.   Mark Michelson
• [asterisk-dev] [Code Review] Merge Called party identification changes into trunk.   Mark Michelson
• [asterisk-dev] [Code Review] Coding Guidelines, API naming convention, Line length limit   Mark Michelson
• [asterisk-dev] [Code Review] Convert ast_channel over to astobj2   Mark Michelson
• [asterisk-dev] [Code Review] Convert ast_channel over to astobj2   Mark Michelson
• [asterisk-dev] [Code Review] Improved support for T38 on initial INVITE   Mark Michelson
• [asterisk-dev] [Code Review] Backport of state_interface for app_queue in Asterisk 1.4   Mark Michelson
• [asterisk-dev] [Code Review] Backport of state_interface for app_queue in Asterisk 1.4   Mark Michelson
• [asterisk-dev] Queue strategy with Ringall and multiple agents answering simultenously.   Mark Michelson
• [asterisk-dev] Asterisk v1.6 stable or testing ?   Mark Michelson
• [asterisk-dev] [Code Review] Backport of state_interface for app_queue in Asterisk 1.4   Mark Michelson
• [asterisk-dev] [Code Review] Backport of state_interface for app_queue in Asterisk 1.4   Mark Michelson
• [asterisk-dev] Asterisk v1.6 stable or testing ?   Mark Michelson
• [asterisk-dev] [Code Review] Optimizations to the stringfields API   Mark Michelson
• [asterisk-dev] Queue agent state #12970 http://bugs.digium.com/view.php?id=12970   Mark Michelson
• [asterisk-dev] [Code Review] SIP Re-invite Glare and 491 Madness...   Mark Michelson
• [asterisk-dev] get out of this lis   Miguel Molina
• [asterisk-dev] some memory leak present in asterisk   Miguel Molina
• [asterisk-dev] Add Farsi Say Date   Steve Murphy
• [asterisk-dev] Add Farsi Say Date   Steve Murphy
• [asterisk-dev] About asterisk development plans   Steve Murphy
• [asterisk-dev] About asterisk development plans   Steve Murphy
• [asterisk-dev] About asterisk development plans   Steve Murphy
• [asterisk-dev] some memory leak present in asterisk   Mayank Jain Nawal
• [asterisk-dev] some memory leak present in asterisk   Mayank Jain Nawal
• [asterisk-dev] Any specific reason that manager action Hangup has no cause header?   Michael Neuhauser
• [asterisk-dev] Any specific reason that manager action Hangup has no cause header?   Michael Neuhauser
• [asterisk-dev] Recent changes in chan_mobile need testing!   Matthew Nicholson
• [asterisk-dev] About asterisk development plans   Matthew Nicholson
• [asterisk-dev] Asterisk Network Security Idea (using tcp_wrappers)   Sergey Okhapkin
• [asterisk-dev] [Code Review] IAX2 retransmit with encryption enabled fix   Tim Panton
• [asterisk-dev] [Code Review] IAX2 retransmit with encryption enabled fix   Tim Panton
• [asterisk-dev] [Code Review] IAX2 retransmit with encryption enabled fix   Tim Panton
• [asterisk-dev] [Code Review] IAX2 retransmit with encryption enabled fix   Tim Panton
• [asterisk-dev] [Code Review] IAX2 retransmit with encryption enabled fix   Tim Panton
• [asterisk-dev] Google Summer of Code 2009   Clod Patry
• [asterisk-dev] Feedback requested for future 1.6 API documentation   Jeff Peeler
• [asterisk-dev] Feedback requested for future 1.6 API documentation   Jeff Peeler
• [asterisk-dev] [Code Review] Cleanup bridging and features code   Jeff Peeler
• [asterisk-dev] Feedback requested for future 1.6 API documentation   Jeff Peeler
• [asterisk-dev] [Code Review] Doxygen API Changes for 1.6.0 -> 1.6.1   Jeff Peeler
• [asterisk-dev] Moving H.323 forward by linking to a new library   Jeff Peeler
• [asterisk-dev] Moving H.323 forward by linking to a new library   Jeff Peeler
• [asterisk-dev] Moving H.323 forward by linking to a new library   Jeff Peeler
• [asterisk-dev] E&M Type V support   Norbert Phillipps
• [asterisk-dev] [Code Review] Backport of state_interface for app_queue in Asterisk 1.4   Wolfgang Pichler
• [asterisk-dev] Asterisk Network Security Idea (using tcp_wrappers)   Anthony Plack
• [asterisk-dev] Add Farsi Say Date   Milad Rastian
• [asterisk-dev] Add Farsi Say Date   Milad Rastian
• [asterisk-dev] Which channel answer call in Queue   Milad Rastian
• [asterisk-dev] Add Farsi Say Date   Milad Rastian
• [asterisk-dev] integrate voip into a web based application   Matt Riddell
• [asterisk-dev] Test framework (changed topic)   Matt Riddell
• [asterisk-dev] [asterisk-users] Asterisk crashed!!!   Matt Riddell
• [asterisk-dev] Idea for GSoC: Support of PRACK in Asterisk   Matt Riddell
• [asterisk-dev] kpfleming: branch 1.4 r184447 - /branches/1.4/sounds/Makefile   Matt Riddell
• [asterisk-dev] kpfleming: branch 1.4 r184447 - /branches/1.4/sounds/Makefile   Matt Riddell
• [asterisk-dev] Google Summer of Code 2009   Luigi Rizzo
• [asterisk-dev] patlooptest tool on DAHDI   Shaun Ruffell
• [asterisk-dev] Google Summer of Code 2009   Eliel Sardañons
• [asterisk-dev] [Code Review] Check dependecies inside the XML documentation.   Eliel Sardañons
• [asterisk-dev] [Code Review] Improve event API cache performance   Eliel Sardañons
• [asterisk-dev] [Code Review] Improve timing interface to remember which provider provided a timer   Eliel Sardañons
• [asterisk-dev] Asterisk crashed!!!   Sebastian
• [asterisk-dev] Asterisk crashed!!!   Sebastian
• [asterisk-dev] patlooptest tool on DAHDI   Marco Signorini
• [asterisk-dev] patlooptest tool on DAHDI   Marco Signorini
• [asterisk-dev] Moving H.323 forward by linking to a new library   Andreas Sikkema
• [asterisk-dev] Idea for GSoC: Support of PRACK in Asterisk   Smita
• [asterisk-dev] Idea for GSoC: Support of PRACK in Asterisk   Smita
• [asterisk-dev] Idea for GSoC: Support of PRACK in Asterisk   Smita
• [asterisk-dev] GSoC: Implementing Networking Security Framework for DOS attacks   Smita
• [asterisk-dev] Idea for GSoC: Support of PRACK in Asterisk   Smita
• [asterisk-dev] Early Media Before 200 ok   Jared Smith
• [asterisk-dev] request for doc about manager command MeetmeMute   Jared Smith
• [asterisk-dev] CID match uses "shortest prefix match"   Jared Smith
• [asterisk-dev] CID match uses "shortest prefix match"   Jared Smith
• [asterisk-dev] CID match uses "shortest prefix match"   Jared Smith
• [asterisk-dev] CID match uses "shortest prefix match"   Jared Smith
• [asterisk-dev] CID match uses "shortest prefix match"   Jared Smith
• [asterisk-dev] Idea for GSoC: Support of PRACK in Asterisk   Jared Smith
• [asterisk-dev] Transcoding G711A - GSM   Jared Smith
• [asterisk-dev] [Code Review] Improve event API cache performance   Darwin O. Solano
• [asterisk-dev] Call waiting the broadsoft way   Kevin Stewart
• [asterisk-dev] [Code Review] Gtalk call from Empathy - no corresponding codecs   Philippe Sultan
• [asterisk-dev] Asterisk 1.4.24-rc1 Now Available   Asterisk Development Team
• [asterisk-dev] Asterisk 1.6.0.7-rc1 Now Available   Asterisk Development Team
• [asterisk-dev] Asterisk 1.6.1.0-rc2 Now Available   Asterisk Development Team
• [asterisk-dev] Asterisk 1.4.24 Now Available!   Asterisk Development Team
• [asterisk-dev] Asterisk 1.6.0.7-rc2, 1.6.1.0-rc3, 1.6.2.0-beta1 & Asterisk-Addons 1.6.0.2-rc1, 1.6.1.0-rc3 Now Available   Asterisk Development Team
• [asterisk-dev] Asterisk Core Sounds 1.4.15, Extra Sounds 1.4.8, and Freeplay MoH Update Released   Asterisk Development Team
• [asterisk-dev] UPDATED: Asterisk Core Sounds 1.4.15, Extra Sounds 1.4.9, and Freeplay MoH Update Released   Asterisk Development Team
• [asterisk-dev] AST-2009-002: Remote Crash Vulnerability in SIP channel driver   Asterisk Security Team
• [asterisk-dev] ENUM and overlapped dialing   Timo Teräs
• [asterisk-dev] ENUM and overlapped dialing   Timo Teräs
• [asterisk-dev] zaphfc flortz patch for dahdi   Timo Teräs
• [asterisk-dev] Moving H.323 forward by linking to a new library   Ming-Ching Tiew
• [asterisk-dev] Moving H.323 forward by linking to a new library   Ming-Ching Tiew
• [asterisk-dev] Feedback requested for future 1.6 API   John Todd
• [asterisk-dev] Google Summer of Code 2009   John Todd
• [asterisk-dev] About asterisk development plans   John Todd
• [asterisk-dev] Test framework (changed topic)   John Todd
• [asterisk-dev] Test framework (changed topic)   John Todd
• [asterisk-dev] GSoC: Implementing Networking Security Framework for DOS attacks   John Todd
• [asterisk-dev] GSoC: Implementing Networking Security Framework for DOS attacks   John Todd
• [asterisk-dev] Asterisk Developers European Meet-Up   John Todd
• [asterisk-dev] Idea for GSoC: Support of PRACK in Asterisk   John Todd
• [asterisk-dev] Asterisk crashed randomly   John Todd
• [asterisk-dev] Asterisk Network Security Idea (using tcp_wrappers)   John Todd
• [asterisk-dev] Idea for GSoC: Support of PRACK in Asterisk   Chris Tooley
• [asterisk-dev] Two version of asterisk on the same server   Chris Tooley
• [asterisk-dev] Queue agent state #12970 http://bugs.digium.com/view.php?id=12970   Joël Vandal
• [asterisk-dev] Need an IVR system for Banking Transactions   Venefax
• [asterisk-dev] Help T.38   Venefax
• [asterisk-dev] Moving H.323 forward by linking to a new library   Venefax
• [asterisk-dev] About asterisk development plans   Venefax
• [asterisk-dev] Asterisk Network Security Idea (using tcp_wrappers)   Venefax
• [asterisk-dev] Asterisk Network Security Idea (using tcp_wrappers)   Venefax
• [asterisk-dev] Asterisk Network Security Idea (using tcp_wrappers)   Venefax
• [asterisk-dev] [Code Review] app_meetme not setting filename and fileformat correctly when recording   David Vossel
• [asterisk-dev] [Code Review] app_read does not break from prompt loop with empty string   David Vossel
• [asterisk-dev] [IAX2] Question on unrecoverable race condition and VNAK's Oseq   David Vossel
• [asterisk-dev] [Code Review]   David Vossel
• [asterisk-dev] [Code Review] Fix handling of backreferences for ENUM lookups   David Vossel
• [asterisk-dev] [Code Review] Fix handling of backreferences for ENUM lookups   David Vossel
• [asterisk-dev] [Code Review] IAX2 retransmit with encryption enabled fix   David Vossel
• [asterisk-dev] [Code Review] IAX2 retransmit with encryption enabled fix   David Vossel
• [asterisk-dev] [Code Review] IAX2 retransmit with encryption enabled fix   David Vossel
• [asterisk-dev] [Code Review] IAX2 retransmit with encryption enabled fix   David Vossel
• [asterisk-dev] [Code Review] IAX2 retransmit with encryption enabled fix   David Vossel
• [asterisk-dev] [Code Review] IAX2 retransmit with encryption enabled fix   David Vossel
• [asterisk-dev] [Code Review] Randomize IAX2 encryption padding   David Vossel
• [asterisk-dev] [Code Review] Randomize IAX2 encryption padding   David Vossel
• [asterisk-dev] [Code Review] Randomize IAX2 encryption padding   David Vossel
• [asterisk-dev] [Code Review] Allow disconnect feature before a call is bridged   David Vossel
• [asterisk-dev] [Code Review] Allow disconnect feature before a call is bridged   David Vossel
• [asterisk-dev] [Code Review] Allow disconnect feature before a call is bridged   David Vossel
• [asterisk-dev] [Code Review] Allow disconnect feature before a call is bridged   David Vossel
• [asterisk-dev] [Code Review] SIP Prefered codec only   David Vossel
• [asterisk-dev] [Code Review] SIP Preferred codec only   David Vossel
• [asterisk-dev] [Code Review] SIP Preferred codec only   David Vossel
• [asterisk-dev] [Code Review] SIP Preferred codec only   David Vossel
• [asterisk-dev] [Code Review] SIP Preferred codec only   David Vossel
• [asterisk-dev] [Code Review] DTMF no wait after last digit   David Vossel
• [asterisk-dev] [Code Review] DTMF no wait after last digit   David Vossel
• [asterisk-dev] [Code Review] Improve frame_queue handing in chan_iax2   David Vossel
• [asterisk-dev] [Code Review] SIP Re-invite Glare and 491 Madness...   David Vossel
• [asterisk-dev] [Code Review] SIP Re-invite Glare and 491 Madness...   David Vossel
• [asterisk-dev] Queue agent state #12970 http://bugs.digium.com/view.php?id=12970   Martin Vít
• [asterisk-dev] channel.c locking and other   Damien Wedhorn
• [asterisk-dev] [Code Review] SRTP support for Asterisk   Terry Wilson
• [asterisk-dev] [Code Review] SRTP support for Asterisk   Terry Wilson
• [asterisk-dev] [Code Review] Calendaring API for Asterisk   Terry Wilson
• [asterisk-dev] [Code Review] Calendaring API for Asterisk   Terry Wilson
• [asterisk-dev] Test framework (changed topic)   Terry Wilson
• [asterisk-dev] [Code Review] Coding Guidelines, API naming convention, Line length limit   Terry Wilson
• [asterisk-dev] race condition on bridge/masquerading?   Guillermo Winkler
• [asterisk-dev] race condition on bridge/masquerading?   Guillermo Winkler
• [asterisk-dev] Asterisk Network Security Idea (using tcp_wrappers)   Hans Witvliet
• [asterisk-dev] New Bounty   Tobias Wolf
• [asterisk-dev] Asterisk trunk on Mac OS/X 10.5   atik
• [asterisk-dev] Asterisk trunk on Mac OS/X 10.5   atik
• [asterisk-dev] is there half duplex call initialation function on asterisk   anupam bairagi
• [asterisk-dev] Developer needed   anupam bairagi
• [asterisk-dev] Need an IVR system for Banking Transactions   hpabon at cusoft-pr.com
• [asterisk-dev] an easy way to deal with/without leading "1" ?   sean darcy
• [asterisk-dev] an easy way to deal with/without leading "1" ?   sean darcy
• [asterisk-dev] [Code Review] Change T38 passthrough reinvite to use control frames   dimas at dataart.com
• [asterisk-dev] [Code Review] Merge Called party identification changes into trunk.   rmudgett at digium.com
• [asterisk-dev] [Code Review] Merge Called party identification changes into trunk.   rmudgett at digium.com
• [asterisk-dev] [Code Review] Merge Called party identification changes into trunk.   rmudgett at digium.com
• [asterisk-dev] sip contact suffix on header   luciano digivoice
• [asterisk-dev] No rtp activity   michel freiha
• [asterisk-dev] Help T.38   michel freiha
• [asterisk-dev] Two version of asterisk on the same server   michel freiha
• [asterisk-dev] Google Summer of Code 2009   ast guy
• [asterisk-dev] Early Media Before 200 ok   raj kiran
• [asterisk-dev] RTP after 183 session progress   raj kiran
• [asterisk-dev] Early Media Before 200 ok   raj kiran
• [asterisk-dev] [Code Review] Bridging API for Conference Bridge purposes   vadim at mbdsys.com
• [asterisk-dev] [Code Review] Fix problems when RTP packet frame size is changed   vadim at mbdsys.com
• [asterisk-dev] [Code Review] Improve behavior of ast_answer() to not lose incoming frames   vadim at mbdsys.com
• [asterisk-dev] [Code Review] Ensure internal poll is used when we ask for it   vadim at mbdsys.com
• [asterisk-dev] [Code Review] Convert ast_channel over to astobj2   vadim at mbdsys.com
• [asterisk-dev] [Code Review] Improve frame_queue handing in chan_iax2   vadim at mbdsys.com
• [asterisk-dev] Transcoding G711A - GSM   maverick me
• [asterisk-dev] add a new queue strategy: SBR   nik600
• [asterisk-dev] add a new queue strategy: SBR   nik600
• [asterisk-dev] [svn-commits] kpfleming: branch 1.4 r182802 - in /branches/1.4: ./ apps/ build_tools/ inclu...   asterisk at ntplx.net
• [asterisk-dev] [svn-commits] kpfleming: branch 1.4 r182802 - in /branches/1.4: ./ apps/ build_tools/ inclu...   asterisk at ntplx.net
• [asterisk-dev] Asterisk Core Sounds 1.4.15, Extra Sounds 1.4.8, and Freeplay MoH Update Released   asterisk at ntplx.net
• [asterisk-dev] request for doc about manager command MeetmeMute   salzh
• [asterisk-dev] Bug or not - "no reply to our critical packet" ?   samuel
• [asterisk-dev] Doubt Pseudo channel   john at spectross.com
• [asterisk-dev] pseudo channel   john at spectross.com
• [asterisk-dev] pseudo channel   john at spectross.com
• [asterisk-dev] pseudo channel   john at spectross.com
• [asterisk-dev] pseudo channel   john at spectross.com
• [asterisk-dev] Developer needed   mark at verbatimtpv.com
• [asterisk-dev] Developer needed   mark at verbatimtpv.com
• [asterisk-dev] Need to test a call path...   eric weaver
• [asterisk-dev] a question about zaptel resolution   张Zhang

Last message date: Tue Mar 31 20:55:27 CDT 2009
Archived on: Tue Mar 31 20:55:34 CDT 2009

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