March 2009 Archives by thread
Starting: Sun Mar 1 02:35:18 CST 2009
Ending: Tue Mar 31 20:55:27 CDT 2009
Messages: 661
- [asterisk-dev] [RFC] Audiohooks
Kaloyan Kovachev
- [asterisk-dev] No rtp activity
michel freiha
- [asterisk-dev] Help T.38
michel freiha
- [asterisk-dev] Help T.38
Dwayne Hubbard
- [asterisk-dev] [Code Review] Convert pbx_spool to use string fields
Mark Michelson
- [asterisk-dev] Feedback requested for future 1.6 API documentation
Jeff Peeler
- [asterisk-dev] [RFC] Audiohooks
Joshua Colp
- [asterisk-dev] garbled audio
Sam Liddicott
- [asterisk-dev] [Code Review] Documentation for the timing modules used in Asterisk
Mark Michelson
- [asterisk-dev] [Code Review] Bridging API for Conference Bridge purposes
Joshua Colp
- [asterisk-dev] following asterisk with git-svn
Tzafrir Cohen
- [asterisk-dev] [Code Review] Documentation for the timing modules used in Asterisk
Mark Michelson
- [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
Kevin P. Fleming
- [asterisk-dev] [IAX2] Question on unrecoverable race condition and VNAK's Oseq
Alex Hermann
- [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] New Bounty
Matt King
- [asterisk-dev] [Code Review] Cleanup bridging and features code
Jeff Peeler
- [asterisk-dev] [IAX2] Question on unrecoverable race condition and VNAK's Oseq
David Vossel
- [asterisk-dev] patlooptest tool on DAHDI
Marco Signorini
- [asterisk-dev] [Code Review] Create API for doing variable substitution with dynamic buffers
Mark Michelson
- [asterisk-dev] sip contact suffix on header
luciano digivoice
- [asterisk-dev] Add Farsi Say Date
Milad Rastian
- [asterisk-dev] [Code Review] Fix problems when RTP packet frame size is changed
Kevin Fleming
- [asterisk-dev] [Code Review] String field test module
Mark Michelson
- [asterisk-dev] Porting applications from 1.4 to 1.6 or trunk
Klaus Darilion
- [asterisk-dev] [Code Review] Make the parameter separator backward compatible, and error messages more consistent.
Kevin Fleming
- [asterisk-dev] [Code Review] make disconnect feature code available outside of bridge
Russell Bryant
- [asterisk-dev] [Code Review]
David Vossel
- [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] Allow a position to be specified when entering a queue.
Mark Michelson
- [asterisk-dev] Which channel answer call in Queue
Milad Rastian
- [asterisk-dev] ENUM and overlapped dialing
Timo Teräs
- [asterisk-dev] add a new queue strategy: SBR
nik600
- [asterisk-dev] Doubt Pseudo channel
john at spectross.com
- [asterisk-dev] race condition on bridge/masquerading?
Guillermo Winkler
- [asterisk-dev] Google Summer of Code 2009
Russell Bryant
- [asterisk-dev] Google Summer of Code 2009
Joshua Colp
- [asterisk-dev] Early Media Before 200 ok
raj kiran
- [asterisk-dev] [Code Review] Doxygen API Changes for 1.6.0 -> 1.6.1
Jeff Peeler
- [asterisk-dev] Developer needed
mark at verbatimtpv.com
- [asterisk-dev] is there half duplex call initialation function on asterisk
anupam bairagi
- [asterisk-dev] App to detect fax/voice
Robert McGilvray
- [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] Developer needed
mark at verbatimtpv.com
- [asterisk-dev] [Code Review] SRTP support for Asterisk
Terry Wilson
- [asterisk-dev] RTP after 183 session progress
raj kiran
- [asterisk-dev] [Code Review] IAX2 retransmit with encryption enabled fix
David Vossel
- [asterisk-dev] Asterisk trunk on Mac OS/X 10.5
Joshua Colp
- [asterisk-dev] [policy] Language specific prompts
David Chappell
- [asterisk-dev] hi
Konstantinos Arvanitis
- [asterisk-dev] an easy way to deal with/without leading "1" ?
sean darcy
- [asterisk-dev] integrate voip into a web based application
Daniel
- [asterisk-dev] [Code Review] IAX2 retransmit with encryption enabled fix
David Vossel
- [asterisk-dev] Issue 14511
Paul Belanger
- [asterisk-dev] Moving H.323 forward by linking to a new library
Jeff Peeler
- [asterisk-dev] H323 hangup cause
Mindaugas Kezys
- [asterisk-dev] request for doc about manager command MeetmeMute
salzh
- [asterisk-dev] [Code Review] Randomize IAX2 encryption padding
David Vossel
- [asterisk-dev] [Code Review] Various fixes for extenpatternmatchnew
Mark Michelson
- [asterisk-dev] dahdi-linux include/ and out-of-tree modules
Tzafrir Cohen
- [asterisk-dev] Control payback bookmarks
Julian Lyndon-Smith
- [asterisk-dev] ACK - wrong URI?
Chris Maciejewski
- [asterisk-dev] E&M Type V support
Norbert Phillipps
- [asterisk-dev] url in dial command
Giorgio Incantalupo
- [asterisk-dev] [Code Review] Allow disconnect feature before a call is bridged
David Vossel
- [asterisk-dev] CID match uses "shortest prefix match"
Klaus Darilion
- [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
Kevin Fleming
- [asterisk-dev] Asterisk 1.4.24 Now Available!
Asterisk Development Team
- [asterisk-dev] [Code Review] Calendaring API for Asterisk
Terry Wilson
- [asterisk-dev] pseudo channel
john at spectross.com
- [asterisk-dev] [Code Review] Check dependecies inside the XML documentation.
Eliel Sardañons
- [asterisk-dev] [Code Review] Ensure internal poll is used when we ask for it
Russell Bryant
- [asterisk-dev] Proposed change to branch revision blocking policy
Sean Bright
- [asterisk-dev] [Code Review] Add name to CHANNEL() function
Russell Bryant
- [asterisk-dev] [Code Review] Change T38 passthrough reinvite to use control frames
Joshua Colp
- [asterisk-dev] [Code Review] Calendaring API for Asterisk
Terry Wilson
- [asterisk-dev] [Code Review] inotify(7) support for pbx_spool
Mark Michelson
- [asterisk-dev] About asterisk development plans
Jon Bonilla (Manwe)
- [asterisk-dev] [Code Review] Merge Called party identification changes into trunk.
Mark Michelson
- [asterisk-dev] CID match uses "shortest prefix match"
Jared Smith
- [asterisk-dev] [svn-commits] kpfleming: branch 1.4 r182802 - in /branches/1.4: ./ apps/ build_tools/ inclu...
asterisk at ntplx.net
- [asterisk-dev] zaphfc flortz patch for dahdi
Timo Teräs
- [asterisk-dev] CID match uses "shortest prefix match"
Jared Smith
- [asterisk-dev] Any specific reason that manager action Hangup has no cause header?
Michael Neuhauser
- [asterisk-dev] Google Summer of Code 2009 - We're in!
Russell Bryant
- [asterisk-dev] Mark Spencer's transcript regarding Skype for Asterisk
Lee S Dryburgh
- [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] Recent changes in chan_mobile need testing!
Matthew Nicholson
- [asterisk-dev] Asterisk crashed!!!
Max Alex
- [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] Bug or not - "no reply to our critical packet" ?
Serge Berney
- [asterisk-dev] Can I tell if a call picked up on PSTN extension... for example?
Michael Higgins
- [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] pseudo channel
john at spectross.com
- [asterisk-dev] [Code Review] Coding Guidelines, API naming convention, Line length limit
Russell Bryant
- [asterisk-dev] Idea for GSoC: Support of PRACK in Asterisk
Smita
- [asterisk-dev] SIP Call: Change number to reach without altering Contact IP
Marc Leurent
- [asterisk-dev] dahdi channel forcing AST_FORMAT_MULAW frames to channels not allowing them.
Caballero Collado, Mª Elena
- [asterisk-dev] Ho to know the connection between 2 channels when one of them is still ringing ?
Iñaki Baz Castillo
- [asterisk-dev] GSoC: Implementing Networking Security Framework for DOS attacks
Smita
- [asterisk-dev] [Code Review] Improve event API cache performance
Russell Bryant
- [asterisk-dev] [Code Review] SIP Prefered codec only
David Vossel
- [asterisk-dev] [Code Review] SIP Preferred codec only
Joshua Colp
- [asterisk-dev] channel.c locking and other
Damien Wedhorn
- [asterisk-dev] Transcoding G711A - GSM
maverick me
- [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] Need to test a call path...
eric weaver
- [asterisk-dev] h extension not considered within a macro for Asterisk 1.6.0.6 and later
Emrah
- [asterisk-dev] some memory leak present in asterisk
Mayank Jain Nawal
- [asterisk-dev] Asterisk crashed randomly
Ngo-Vi Hoai-Anh
- [asterisk-dev] gsoc: verbose logging
Pék Dániel
- [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] [Code Review] DTMF no wait after last digit
David Vossel
- [asterisk-dev] kpfleming: branch 1.4 r184447 - /branches/1.4/sounds/Makefile
Matt Riddell
- [asterisk-dev] [Code Review] Improve timing interface to remember which provider provided a timer
Kevin Fleming
- [asterisk-dev] some memory leak present in asterisk
Mayank Jain Nawal
- [asterisk-dev] unanswered=yes not working in 1.6.1 ?
Chris Maciejewski
- [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] [Code Review] Avoid wasteful frame queue traversals in chan_iax2 network_thread()
Russell Bryant
- [asterisk-dev] SIP and using less-prior codec
Klaus Darilion
- [asterisk-dev] Queue strategy with Ringall and multiple agents answering simultenously.
Atis Lezdins
- [asterisk-dev] Asterisk v1.6 stable or testing ?
Serge Berney
- [asterisk-dev] [Code Review] Create API for doing variable substitution with dynamic buffers
Russell Bryant
- [asterisk-dev] Two version of asterisk on the same server
Serge Berney
- [asterisk-dev] [Code Review] Optimizations to the stringfields API
Mark Michelson
- [asterisk-dev] [Code Review] Create API for doing variable substitution with dynamic buffers
Russell Bryant
- [asterisk-dev] Call waiting the broadsoft way
Kevin Stewart
- [asterisk-dev] Queue agent state #12970 http://bugs.digium.com/view.php?id=12970
Martin Vít
- [asterisk-dev] channel formats and ast_set_read/write_format
Klaus Darilion
- [asterisk-dev] [Code Review] SIP Re-invite Glare and 491 Madness...
David Vossel
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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