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[asterisk-dev] Implementation of Broadsoft Sip Access in Asterisk to enable SLA for Sipura/Linksys
From
: Eugene Grossi
Re: [asterisk-dev] [asterisk-commits] file: trunk r95648 - /trunk/codecs/Makefile
From
: Luigi Rizzo
Re: [asterisk-dev] [asterisk-commits] file: trunk r95648 - /trunk/codecs/Makefile
From
: Joshua Colp
Re: [asterisk-dev] [asterisk-commits] file: trunk r95648 - /trunk/codecs/Makefile
From
: Russell Bryant
[asterisk-dev] gentone.c
From
: Bhrugu Mehta
Re: [asterisk-dev] Implementation of Broadsoft Sip Access in Asterisk to enable SLA for Sipura/Linksys
From
: Johansson Olle E
[asterisk-dev] SVN: slin16 bugaboo? "core show translations" runs amok
From
: Brian Capouch
Re: [asterisk-dev] dialog matching
From
: Klaus Darilion
Re: [asterisk-dev] dialog matching
From
: Klaus Darilion
Re: [asterisk-dev] Implementation of Broadsoft Sip Access in Asterisk to enable SLA for Sipura/Linksys
From
: Eugene Grossi
Re: [asterisk-dev] [asterisk-commits] file: trunk r95648 - /trunk/codecs/Makefile
From
: Kevin P. Fleming
Re: [asterisk-dev] gentone.c
From
: Kevin P. Fleming
Re: [asterisk-dev] [asterisk-commits] file: trunk r95648 - /trunk/codecs/Makefile
From
: Kevin P. Fleming
Re: [asterisk-dev] Implementation of Broadsoft Sip Access in Asterisk to enable SLA for Sipura/Linksys
From
: Raj Jain
Re: [asterisk-dev] SVN: slin16 bugaboo? "core show translations" runs amok
From
: Jared Smith
Re: [asterisk-dev] SVN: slin16 bugaboo? "core show translations" runs amok
From
: Jason Parker
Re: [asterisk-dev] [asterisk-commits] file: trunk r95648 - /trunk/codecs/Makefile
From
: Russell Bryant
Re: [asterisk-dev] gentone.c
From
: Russell Bryant
Re: [asterisk-dev] gentone.c
From
: Kevin P. Fleming
Re: [asterisk-dev] [asterisk-commits] file: trunk r95648 - /trunk/codecs/Makefile
From
: Kevin P. Fleming
Re: [asterisk-dev] Implementation of Broadsoft Sip Access in Asterisk to enable SLA for Sipura/Linksys
From
: Raj Jain
Re: [asterisk-dev] gentone.c
From
: Russell Bryant
Re: [asterisk-dev] [asterisk-commits] file: trunk r95648 - /trunk/codecs/Makefile
From
: Russell Bryant
Re: [asterisk-dev] Another module for testing: chan_console
From
: Adrià Vidal
Re: [asterisk-dev] Another module for testing: chan_console
From
: Kevin P. Fleming
Re: [asterisk-dev] gentone.c
From
: Kevin P. Fleming
[asterisk-dev] AST-2008-001: Crash from transfer using BYE with Also header
From
: Asterisk Security Team
[asterisk-announce] Asterisk 1.4.17 Released
From
: The Asterisk Development Team
[asterisk-announce] AST-2008-001: Crash from transfer using BYE with Also header
From
: Asterisk Security Team
Re: [asterisk-dev] Another module for testing: chan_console
From
: Adrià Vidal
Re: [asterisk-dev] Another module for testing: chan_console
From
: Russell Bryant
Re: [asterisk-dev] Implementation of Broadsoft Sip Access in Asterisk to enable SLA for Sipura/Linksys
From
: Eugene Grossi
Re: [asterisk-dev] Implementation of Broadsoft Sip Access in Asterisk to enable SLA for Sipura/Linksys
From
: Jared Smith
Re: [asterisk-dev] Another module for testing: chan_console
From
: Adrià Vidal
[asterisk-dev] astobj2 and chan_sip; first results... wanna test drive it?
From
: Steve Murphy
Re: [asterisk-dev] Implementation of Broadsoft Sip Access in Asterisk to enable SLA for Sipura/Linksys
From
: Johansson Olle E
Re: [asterisk-dev] Implementation of Broadsoft Sip Access in Asterisk to enable SLA for Sipura/Linksys
From
: Johansson Olle E
Re: [asterisk-dev] astobj2 and chan_sip; first results... wanna test drive it?
From
: Johansson Olle E
Re: [asterisk-dev] astobj2 and chan_sip; first results... wanna test drive it?
From
: Gregory Boehnlein
Re: [asterisk-dev] Implementation of Broadsoft Sip Access in Asterisk to enable SLA for Sipura/Linksys
From
: Raj Jain
Re: [asterisk-dev] Implementation of Broadsoft Sip Access in Asterisk to enable SLA for Sipura/Linksys
From
: Johansson Olle E
Re: [asterisk-dev] Implementation of Broadsoft Sip Access in Asterisk to enable SLA for Sipura/Linksys
From
: Philipp Kempgen
Re: [asterisk-dev] Another module for testing: chan_console
From
: Russell Bryant
Re: [asterisk-dev] Implementation of Broadsoft Sip Access in Asterisk to enable SLA for Sipura/Linksys
From
: Eugene Grossi
[asterisk-dev] Number of digit to trunk??
From
: Francois
Re: [asterisk-dev] astobj2 and chan_sip; first results... wanna test drive it?
From
: Steve Murphy
Re: [asterisk-dev] astobj2 and chan_sip; first results... wanna test drive it?
From
: Russell Bryant
[asterisk-dev] Call files using Zap
From
: S. M. Faisal Abbas
Re: [asterisk-dev] Call files using Zap
From
: Doug Lytle
Re: [asterisk-dev] Call files using Zap
From
: Steven S. Critchfield
Re: [asterisk-dev] Another module for testing: chan_console
From
: Adrià Vidal
Re: [asterisk-dev] astobj2 and chan_sip; first results... wanna test drive it?
From
: Johansson Olle E
Re: [asterisk-dev] Implementation of Broadsoft Sip Access in Asterisk to enable SLA for Sipura/Linksys
From
: Russell Bryant
Re: [asterisk-dev] Implementation of Broadsoft Sip Access in Asterisk to enable SLA for Sipura/Linksys
From
: John Lange
Re: [asterisk-dev] Implementation of Broadsoft Sip Access in Asterisk to enable SLA for Sipura/Linksys
From
: Russell Bryant
[asterisk-dev] zaptel programming
From
: Bhrugu Mehta
Re: [asterisk-dev] astobj2 and chan_sip; first results... wanna test drive it?
From
: Klaus Darilion
Re: [asterisk-dev] zaptel programming
From
: Matthew Fredrickson
Re: [asterisk-dev] astobj2 and chan_sip; first results... wanna test drive it?
From
: Steve Murphy
Re: [asterisk-dev] zaptel programming
From
: Steven S. Critchfield
[asterisk-dev] trunk code cleanup: pbx_kdeconsole & pbx_gtkconsole
From
: Caio Begotti
Re: [asterisk-dev] trunk code cleanup: pbx_kdeconsole & pbx_gtkconsole
From
: Caio Begotti
[asterisk-dev] Concurrency threading problem in chan_sip
From
: Nick Gorham
Re: [asterisk-dev] astobj2 and chan_sip; first results... wanna test drive it?
From
: Johansson Olle E
Re: [asterisk-dev] astobj2 and chan_sip; first results... wanna test drive it?
From
: Maxim Sobolev
Re: [asterisk-dev] Concurrency threading problem in chan_sip
From
: Russell Bryant
Re: [asterisk-dev] Concurrency threading problem in chan_sip
From
: Tilghman Lesher
[asterisk-dev] MixMonitor doesn't work right with SIP and Zap/Flash transfers
From
: Vinicius Fontes
[asterisk-dev] CED detection
From
: Dmitry Andrianov
[asterisk-dev] Simulation of radios with something like MeetMe()
From
: Holger Wirtz
Re: [asterisk-dev] Simulation of radios with something like MeetMe()
From
: Luigi Rizzo
Re: [asterisk-dev] Simulation of radios with something like MeetMe()
From
: Steven S. Critchfield
Re: [asterisk-dev] Simulation of radios with something like MeetMe()
From
: Alexander Lopez
[asterisk-dev] compiling trunk under cygwin
From
: Clod Patry
Re: [asterisk-dev] Simulation of radios with something like MeetMe()
From
: Clive Nicolson
Re: [asterisk-dev] [asterisk-commits] russell: trunk r97643 - /trunk/configure.ac
From
: Luigi Rizzo
Re: [asterisk-dev] compiling trunk under cygwin
From
: Luigi Rizzo
Re: [asterisk-dev] compiling trunk under cygwin
From
: Clod Patry
Re: [asterisk-dev] Simulation of radios with something like MeetMe()
From
: Holger Wirtz
Re: [asterisk-dev] compiling trunk under cygwin
From
: Russell Bryant
Re: [asterisk-dev] [asterisk-commits] russell: trunk r97643 - /trunk/configure.ac
From
: Russell Bryant
Re: [asterisk-dev] Simulation of radios with something like MeetMe()
From
: Holger Wirtz
Re: [asterisk-dev] [asterisk-commits] russell: trunk r97643 - /trunk/configure.ac
From
: Luigi Rizzo
Re: [asterisk-dev] Simulation of radios with something like MeetMe()
From
: Holger Wirtz
Re: [asterisk-dev] Simulation of radios with something like MeetMe()
From
: Holger Wirtz
Re: [asterisk-dev] compiling trunk under cygwin
From
: Caio Begotti
[asterisk-dev] q931 decoding question
From
: Klaus Darilion
[asterisk-dev] cdr_odbc.c is broken in trunk
From
: Nick Gorham
Re: [asterisk-dev] cdr_odbc.c is broken in trunk
From
: Michiel van Baak
Re: [asterisk-dev] cdr_odbc.c is broken in trunk
From
: Nick Gorham
Re: [asterisk-dev] Simulation of radios with something like MeetMe()
From
: David Van Ginneken
Re: [asterisk-dev] [asterisk-commits] russell: trunk r97643 - /trunk/configure.ac
From
: Russell Bryant
Re: [asterisk-dev] cdr_odbc.c is broken in trunk
From
: Michiel van Baak
Re: [asterisk-dev] MixMonitor doesn't work right with SIP and Zap/Flash transfers
From
: Michael Cargile
Re: [asterisk-dev] cdr_odbc.c is broken in trunk
From
: Tilghman Lesher
[asterisk-dev] PRI issue
From
: astdev
Re: [asterisk-dev] cdr_odbc.c is broken in trunk
From
: Nick Gorham
Re: [asterisk-dev] cdr_odbc.c is broken in trunk
From
: Nick Gorham
Re: [asterisk-dev] Simulation of radios with something like MeetMe()
From
: Steven S. Critchfield
Re: [asterisk-dev] q931 decoding question
From
: Hans Petter Selasky
Re: [asterisk-dev] cdr_odbc.c is broken in trunk
From
: Tilghman Lesher
Re: [asterisk-dev] [asterisk-commits] murf: trunk r97656 - in /trunk: channels/ include/ utils/
From
: Kevin P. Fleming
Re: [asterisk-dev] cdr_odbc.c is broken in trunk
From
: Kevin P. Fleming
Re: [asterisk-dev] compiling trunk under cygwin
From
: Clod Patry
Re: [asterisk-dev] compiling trunk under cygwin
From
: Caio Begotti
Re: [asterisk-dev] q931 decoding question
From
: Hans Petter Selasky
Re: [asterisk-dev] cdr_odbc.c is broken in trunk
From
: Nick Gorham
Re: [asterisk-dev] compiling trunk under cygwin
From
: Luigi Rizzo
Re: [asterisk-dev] MixMonitor doesn't work right with SIP and Zap/Flash transfers
From
: Vinicius Fontes
Re: [asterisk-dev] cdr_odbc.c is broken in trunk
From
: Tilghman Lesher
Re: [asterisk-dev] [asterisk-commits] murf: trunk r97656 - in /trunk: channels/ include/ utils/
From
: Jason Parker
Re: [asterisk-dev] q931 decoding question
From
: Klaus Darilion
Re: [asterisk-dev] [asterisk-commits] murf: trunk r97656 - in /trunk: channels/ include/ utils/
From
: Kevin P. Fleming
[asterisk-dev] chan_sip.c rev 97077 1.4 branch problem with hold on polycom
From
: asterisk
Re: [asterisk-dev] [asterisk-commits] russell: branch 1.4 r97976 - /branches/1.4/main/translate.c
From
: Luigi Rizzo
Re: [asterisk-dev] Simulation of radios with something like MeetMe()
From
: Holger Wirtz
Re: [asterisk-dev] Simulation of radios with something like MeetMe()
From
: Holger Wirtz
Re: [asterisk-dev] [svn-commits] twilson: trunk r97634 - in /trunk: ./ configs/ doc/tex/ funcs/ include/asteri...
From
: Johansson Olle E
Re: [asterisk-dev] [svn-commits] twilson: trunk r97634 - in /trunk: ./ configs/ doc/tex/ funcs/ include/asteri...
From
: Johansson Olle E
Re: [asterisk-dev] [svn-commits] twilson: trunk r97634 - in /trunk: ./ configs/ doc/tex/ funcs/ include/asteri...
From
: Tzafrir Cohen
Re: [asterisk-dev] cdr_odbc.c is broken in trunk
From
: Nick Gorham
[asterisk-dev] Developing Help
From
: Bhrugu Mehta
Re: [asterisk-dev] [svn-commits] tilghman: trunk r97651 - in /trunk: ./ apps/ channels/ configs/ include/aster...
From
: Johansson Olle E
Re: [asterisk-dev] [svn-commits] twilson: trunk r97634 - in /trunk: ./ configs/ doc/tex/ funcs/ include/asteri...
From
: Kevin P. Fleming
Re: [asterisk-dev] [svn-commits] kpfleming: trunk r98124 - /trunk/channels/chan_sip.c
From
: Johansson Olle E
Re: [asterisk-dev] [svn-commits] twilson: trunk r97634 - in /trunk: ./ configs/ doc/tex/ funcs/ include/asteri...
From
: Russell Bryant
Re: [asterisk-dev] Developing Help
From
: Jeff Gehlbach
Re: [asterisk-dev] cdr_odbc.c is broken in trunk
From
: Tilghman Lesher
Re: [asterisk-dev] cdr_odbc.c is broken in trunk
From
: Nick Gorham
Re: [asterisk-dev] [svn-commits] twilson: trunk r97634 - in /trunk: ./ configs/ doc/tex/ funcs/ include/asteri...
From
: Johansson Olle E
Re: [asterisk-dev] q931 decoding question
From
: Matthew Fredrickson
Re: [asterisk-dev] PRI issue
From
: Matthew Fredrickson
Re: [asterisk-dev] [svn-commits] twilson: trunk r97634 - in /trunk: ./ configs/ doc/tex/ funcs/ include/asteri...
From
: Kevin P. Fleming
Re: [asterisk-dev] [svn-commits] twilson: trunk r97634 - in /trunk: ./ configs/ doc/tex/ funcs/ include/asteri...
From
: Russell Bryant
Re: [asterisk-dev] [svn-commits] twilson: trunk r97634 - in /trunk: ./ configs/ doc/tex/ funcs/ include/asteri...
From
: Kevin P. Fleming
[asterisk-dev] Asked to transmit frame type 64, while native formats is 0x4
From
: Norman Franke
Re: [asterisk-dev] Asked to transmit frame type 64, while native formats is 0x4
From
: Russell Bryant
[asterisk-dev] Unstable releases lately
From
: Bob
Re: [asterisk-dev] [svn-commits] twilson: trunk r97634 - in /trunk: ./ configs/ doc/tex/ funcs/ include/asteri...
From
: Johansson Olle E
[asterisk-dev] request for integration of new dialplan application into trunk - howto proceed?
From
: Andreas Brodmann
Re: [asterisk-dev] request for integration of new dialplan application into trunk - howto proceed?
From
: Johansson Olle E
Re: [asterisk-dev] [zaptel-commits] tzafrir: branch 1.4 r3677 - /branches/1.4/zaptel-base.c
From
: Kevin P. Fleming
Re: [asterisk-dev] Unstable releases lately
From
: Kevin P. Fleming
Re: [asterisk-dev] [zaptel-commits] tzafrir: branch 1.4 r3677 - /branches/1.4/zaptel-base.c
From
: Tzafrir Cohen
Re: [asterisk-dev] [zaptel-commits] tzafrir: branch 1.4 r3677 - /branches/1.4/zaptel-base.c
From
: Kevin P. Fleming
Re: [asterisk-dev] Unstable releases lately
From
: Bob
Re: [asterisk-dev] request for integration of new dialplan application into trunk - howto proceed?
From
: Matt Riddell
[asterisk-dev] janitor job: ast_exists_extension() and direct setting of chan->priority to ast_goto_if_exists()
From
: Caio Begotti
[asterisk-dev] indications and chan_oss
From
: Luigi Rizzo
[asterisk-dev] Including port number on SDP Contact
From
: Robert Moskowitz
[asterisk-dev] zaptel arm wrestling
From
: mark spowage
[asterisk-dev] T38 Gateway As A Codec Translator (#11761)
From
: Gregory Nietsky
Re: [asterisk-dev] Including port number on SDP Contact
From
: Johansson Olle E
[asterisk-dev] T38 Gateway As A Codec Translator (#11761)
From
: Gregory Nietsky
Re: [asterisk-dev] Unstable releases lately
From
: Vinicius Fontes
Re: [asterisk-dev] q931 decoding question
From
: Klaus Darilion
Re: [asterisk-dev] Including port number on SDP Contact
From
: Robert Moskowitz
Re: [asterisk-dev] Unstable releases lately
From
: David Boyd
Re: [asterisk-dev] Unstable releases lately
From
: David Boyd
Re: [asterisk-dev] Unstable releases lately
From
: Dan Evans
Re: [asterisk-dev] Unstable releases lately
From
: Russell Bryant
Re: [asterisk-dev] indications and chan_oss
From
: Russell Bryant
Re: [asterisk-dev] [svn-commits] twilson: trunk r97634 - in /trunk: ./ configs/ doc/tex/ funcs/ include/asteri...
From
: Terry Wilson
Re: [asterisk-dev] [svn-commits] twilson: trunk r97634 - in /trunk: ./ configs/ doc/tex/ funcs/ include/asteri...
From
: Russell Bryant
Re: [asterisk-dev] Unstable releases lately
From
: Jared Smith
[asterisk-dev] G.729 pre-compiled binaries and Asterisk 1.2.x.
From
: Alex Balashov
Re: [asterisk-dev] G.729 pre-compiled binaries and Asterisk 1.2.x.
From
: Steven S. Critchfield
Re: [asterisk-dev] G.729 pre-compiled binaries and Asterisk 1.2.x.
From
: Alex Balashov
[asterisk-dev] SVN servers down for maintenance
From
: Russell Bryant
[asterisk-announce] Zaptel 1.2.23 and 1.4.8 released
From
: The Asterisk Development Team
Re: [asterisk-dev] Zaptel 1.2.23 and 1.4.8 released
From
: Jeffrey Ollie
Re: [asterisk-dev] Zaptel 1.2.23 and 1.4.8 released
From
: Russell Bryant
Re: [asterisk-dev] Asked to transmit frame type 64, while native formats is 0x4
From
: Norman Franke
Re: [asterisk-dev] Unstable releases lately
From
: Bob
Re: [asterisk-dev] TOUPPER/TOLOWER
From
: Johansson Olle E
Re: [asterisk-dev] Unstable releases lately
From
: Tzafrir Cohen
Re: [asterisk-dev] ...asterisk project dead...
From
: Johansson Olle E
Re: [asterisk-dev] Unstable releases lately
From
: John Lange
Re: [asterisk-dev] Unstable releases lately
From
: Kevin P. Fleming
Re: [asterisk-dev] Unstable releases lately
From
: Simon Perreault
Re: [asterisk-dev] ...asterisk project dead...
From
: Russell Bryant
Re: [asterisk-dev] Unstable releases lately
From
: John Lange
Re: [asterisk-dev] Unstable releases lately
From
: Tzafrir Cohen
Re: [asterisk-dev] Unstable releases lately
From
: Tzafrir Cohen
Re: [asterisk-dev] Unstable releases lately
From
: Michael Grigoni
[asterisk-dev] anonymous svn access?
From
: Jeff Knighton
Re: [asterisk-dev] G.729 pre-compiled binaries and Asterisk 1.2.x.
From
: Steven S. Critchfield
Re: [asterisk-dev] anonymous svn access?
From
: Russell Bryant
Re: [asterisk-dev] Unstable releases lately
From
: Maxim Sobolev
Re: [asterisk-dev] Unstable releases lately
From
: John Lange
Re: [asterisk-dev] Unstable releases lately
From
: Tzafrir Cohen
Re: [asterisk-dev] Unstable releases lately
From
: John Lange
Re: [asterisk-dev] Unstable releases lately
From
: Benny Amorsen
Re: [asterisk-dev] Unstable releases lately
From
: Kevin P. Fleming
Re: [asterisk-dev] Unstable releases lately
From
: Benny Amorsen
[asterisk-dev] Easy packaging (Was unstable releases)
From
: Daniel Hazelbaker
Re: [asterisk-dev] Unstable releases lately
From
: Michiel van Baak
[asterisk-dev] devicestate
From
: Clod Patry
Re: [asterisk-dev] Unstable releases lately
From
: Paul Hewlett
[asterisk-dev] SVN reverted 21 months back...
From
: Johansson Olle E
Re: [asterisk-dev] Easy packaging (Was unstable releases)
From
: Tzafrir Cohen
Re: [asterisk-dev] Unstable releases lately
From
: Bob
Re: [asterisk-dev] Unstable releases lately
From
: Tzafrir Cohen
Re: [asterisk-dev] SVN reverted 21 months back...
From
: Edwin Groothuis
Re: [asterisk-dev] Unstable releases lately
From
: Bob
Re: [asterisk-dev] Unstable releases lately
From
: Paul Hewlett
Re: [asterisk-dev] Unstable releases lately
From
: Tzafrir Cohen
Re: [asterisk-dev] SVN reverted 21 months back...
From
: Kevin P. Fleming
Re: [asterisk-dev] devicestate
From
: Kevin P. Fleming
[asterisk-dev] Query about CHECK_BLOCKING(chan) in ast_write()
From
: Tony Mountifield
Re: [asterisk-dev] devicestate
From
: Atis Lezdins
Re: [asterisk-dev] devicestate
From
: Clod Patry
Re: [asterisk-dev] devicestate
From
: Leif Madsen
Re: [asterisk-dev] devicestate
From
: Atis Lezdins
Re: [asterisk-dev] devicestate
From
: Kevin P. Fleming
Re: [asterisk-dev] devicestate
From
: Kevin P. Fleming
Re: [asterisk-dev] Query about CHECK_BLOCKING(chan) in ast_write()
From
: Tony Mountifield
Re: [asterisk-dev] devicestate
From
: Clod Patry
Re: [asterisk-dev] devicestate
From
: Johansson Olle E
Re: [asterisk-dev] devicestate
From
: Joel Vandal
Re: [asterisk-dev] devicestate
From
: Mark Michelson
[asterisk-dev] Zaptel-1.4.8; Zap trunk reverts to dialtone instead of processing call
From
: syd wonder
Re: [asterisk-dev] devicestate
From
: Clod Patry
Re: [asterisk-dev] Easy packaging (Was unstable releases)
From
: Daniel Hazelbaker
Re: [asterisk-dev] Zaptel-1.4.8; Zap trunk reverts to dialtone instead of processing call
From
: Tzafrir Cohen
Re: [asterisk-dev] devicestate
From
: Joel Vandal
Re: [asterisk-dev] Easy packaging (Was unstable releases)
From
: Jared Smith
Re: [asterisk-dev] Easy packaging (Was unstable releases)
From
: Joel Vandal
Re: [asterisk-dev] Unstable releases lately
From
: Michiel van Baak
Re: [asterisk-dev] Easy packaging (Was unstable releases)
From
: Daniel Hazelbaker
Re: [asterisk-dev] Easy packaging (Was unstable releases)
From
: Jeffrey Ollie
Re: [asterisk-dev] devicestate
From
: Atis Lezdins
Re: [asterisk-dev] Easy packaging (Was unstable releases)
From
: Patrick
Re: [asterisk-dev] Easy packaging (Was unstable releases)
From
: Joel Vandal
Re: [asterisk-dev] [svn-commits] qwell: trunk r98971 - in /trunk: ./ include/asterisk/
From
: Russell Bryant
Re: [asterisk-dev] Easy packaging (Was unstable releases)
From
: Axel Thimm
Re: [asterisk-dev] Easy packaging (Was unstable releases)
From
: Jared Smith
Re: [asterisk-dev] Easy packaging (Was unstable releases)
From
: Joel Vandal
[asterisk-dev] Asterisk crashes due to non-atomic check on chan_iax.c:schedule_delivery
From
: Guillermo Winkler
Re: [asterisk-dev] Easy packaging (Was unstable releases)
From
: Daniel Hazelbaker
Re: [asterisk-dev] Easy packaging (Was unstable releases)
From
: Hans Witvliet
Re: [asterisk-dev] Unstable releases lately
From
: Atis Lezdins
Re: [asterisk-dev] rizzo: trunk r95069 - in /trunk/apps: app_ices.c app_queue.c app_voicemail.c
From
: Tony Mountifield
[asterisk-dev] Should SVN be giving good trunk by now?
From
: Brian Capouch
Re: [asterisk-dev] Should SVN be giving good trunk by now?
From
: igi-go
[asterisk-dev] Asterisk start handle a call after 3 seconds
From
: asterisk
Re: [asterisk-dev] devicestate
From
: Atis Lezdins
Re: [asterisk-dev] [svn-commits] tilghman: trunk r98984 - /trunk/CHANGES
From
: Johansson Olle E
Re: [asterisk-dev] [svn-commits] russell: trunk r98986 - in /trunk: CHANGES main/asterisk.c
From
: Johansson Olle E
Re: [asterisk-dev] Unstable releases lately
From
: Kevin P. Fleming
Re: [asterisk-dev] [svn-commits] tilghman: trunk r98984 - /trunk/CHANGES
From
: Tilghman Lesher
Re: [asterisk-dev] Asterisk start handle a call after 3 seconds
From
: Kevin P. Fleming
Re: [asterisk-dev] [svn-commits] russell: trunk r98986 - in /trunk: CHANGES main/asterisk.c
From
: Jared Smith
Re: [asterisk-dev] Asterisk start handle a call after 3 seconds
From
: asterisk
Re: [asterisk-dev] Asterisk start handle a call after 3 seconds
From
: Kevin P. Fleming
Re: [asterisk-dev] [svn-commits] russell: trunk r98986 - in /trunk: CHANGES main/asterisk.c
From
: Tzafrir Cohen
Re: [asterisk-dev] rizzo: trunk r95069 - in /trunk/apps: app_ices.c app_queue.c app_voicemail.c
From
: Mark Michelson
Re: [asterisk-dev] Asterisk start handle a call after 3 seconds
From
: asterisk
Re: [asterisk-dev] devicestate
From
: Mark Michelson
Re: [asterisk-dev] devicestate
From
: Mark Michelson
Re: [asterisk-dev] Asterisk start handle a call after 3 seconds
From
: Kevin P. Fleming
Re: [asterisk-dev] Unstable releases lately
From
: Russell Bryant
[asterisk-dev] Asterisk SVN mirror back up to date
From
: Russell Bryant
Re: [asterisk-dev] Unstable releases lately
From
: Kevin P. Fleming
[asterisk-dev] qsig and TE121P
From
: ruben buron
Re: [asterisk-dev] qsig and TE121P
From
: Steven S. Critchfield
Re: [asterisk-dev] [svn-commits] russell: trunk r98986 - in /trunk: CHANGES main/asterisk.c
From
: James Golovich
Re: [asterisk-dev] Asterisk SVN mirror back up to date
From
: Maxim Sobolev
Re: [asterisk-dev] Asterisk SVN mirror back up to date
From
: Jeffrey Ollie
[asterisk-dev] Revision 87427 not merged to trunk
From
: Dan Austin
Re: [asterisk-dev] Asterisk SVN mirror back up to date
From
: Maxim Sobolev
Re: [asterisk-dev] [svn-commits] russell: trunk r98986 - in /trunk: CHANGES main/asterisk.c
From
: Simon Lockhart
Re: [asterisk-dev] [svn-commits] russell: trunk r98986 - in /trunk: CHANGES main/asterisk.c
From
: Caio Begotti
Re: [asterisk-dev] [svn-commits] russell: trunk r98986 - in /trunk: CHANGES main/asterisk.c
From
: Russell Bryant
Re: [asterisk-dev] [svn-commits] russell: trunk r98986 - in /trunk: CHANGES main/asterisk.c
From
: Kevin P. Fleming
Re: [asterisk-dev] Revision 87427 not merged to trunk
From
: Mark Michelson
Re: [asterisk-dev] Revision 87427 not merged to trunk
From
: Dan Austin
[asterisk-dev] SVN trunk under openWRT: only one build issue remains
From
: Brian Capouch
Re: [asterisk-dev] SVN trunk under openWRT: only one build issue remains
From
: Atis Lezdins
Re: [asterisk-dev] Zaptel-1.4.8; Zap trunk reverts to dialtone instead of processing call
From
: syd wonder
[asterisk-dev] Storing Voicemail messages in multiple formats ...
From
: sushma gupta
Re: [asterisk-dev] Storing Voicemail messages in multiple formats ...
From
: Steve Edwards
Re: [asterisk-dev] Asterisk start handle a call after 3 seconds
From
: asterisk
[asterisk-dev] iax encryption client side
From
: Cavalera Claudio Luigi
Re: [asterisk-dev] Zaptel-1.4.8; Zap trunk reverts to dialtone instead of processing call
From
: Tzafrir Cohen
[asterisk-dev] problems during compling
From
: Theo Belder
Re: [asterisk-dev] problems during compling
From
: Jim Capp
[asterisk-dev] SIP call-limit and Realtime
From
: Atis Lezdins
[asterisk-dev] Probably a simple question. Dial a call.
From
: LWATCDR
Re: [asterisk-dev] Probably a simple question. Dial a call.
From
: Steven S. Critchfield
Re: [asterisk-dev] Probably a simple question. Dial a call.
From
: LWATCDR
Re: [asterisk-dev] Probably a simple question. Dial a call.
From
: Steve Totaro
[asterisk-dev] [ot] Re: Probably a simple question. Dial a call.
From
: Steven S. Critchfield
[asterisk-dev] Number of active calls
From
: sushma gupta
Re: [asterisk-dev] Number of active calls
From
: Steve Totaro
Re: [asterisk-dev] Number of active calls
From
: Steven S. Critchfield
Re: [asterisk-dev] Number of active calls
From
: Dmitry Andrianov
Re: [asterisk-dev] Number of active calls
From
: Joel Vandal
Re: [asterisk-dev] Zaptel-1.4.8; Zap trunk reverts to dialtone instead of processing call
From
: syd wonder
Re: [asterisk-dev] SIP call-limit and Realtime
From
: Russell Bryant
Re: [asterisk-dev] Zaptel-1.4.8; Zap trunk reverts to dialtone instead of processing call
From
: Tzafrir Cohen
Re: [asterisk-dev] SIP call-limit and Realtime
From
: Atis Lezdins
Re: [asterisk-dev] SIP call-limit and Realtime
From
: Russell Bryant
Re: [asterisk-dev] SIP call-limit and Realtime
From
: Atis Lezdins
[asterisk-announce] Asterisk 1.6.0-beta1 released
From
: The Asterisk Development Team
[asterisk-dev] Release Candidates and Nightly Builds
From
: Russell Bryant
Re: [asterisk-dev] Release Candidates and Nightly Builds
From
: Ryan Burke
Re: [asterisk-dev] Release Candidates and Nightly Builds
From
: Daniel Hazelbaker
Re: [asterisk-dev] Release Candidates and Nightly Builds
From
: Maxim Sobolev
Re: [asterisk-dev] Release Candidates and Nightly Builds
From
: Bob
Re: [asterisk-dev] request for integration of new dialplan application into trunk - howto proceed?
From
: Andreas Brodmann
Re: [asterisk-dev] Asterisk Performance as a B2BUA
From
: Steve Murphy
Re: [asterisk-dev] Zaptel-1.4.8; Zap trunk reverts to dialtone instead of processing call
From
: syd wonder
Re: [asterisk-dev] Release Candidates and Nightly Builds
From
: BJ Weschke
Re: [asterisk-dev] Release Candidates and Nightly Builds
From
: Russell Bryant
Re: [asterisk-dev] Release Candidates and Nightly Builds
From
: Russell Bryant
Re: [asterisk-dev] Release Candidates and Nightly Builds
From
: Russell Bryant
Re: [asterisk-dev] Release Candidates and Nightly Builds
From
: Gregory Boehnlein
Re: [asterisk-dev] SIP call-limit and Realtime
From
: Johansson Olle E
Re: [asterisk-dev] iax encryption client side
From
: Tim H. Panton
[asterisk-dev] [Asterisk-Dev] 3rd party call control / CSTA ,
From
: James Montague
Re: [asterisk-dev] Release Candidates and Nightly Builds
From
: Matt Riddell
Re: [asterisk-dev] Release Candidates and Nightly Builds
From
: Jean-Denis Girard
[asterisk-dev] Message type INFORMATION problem
From
: Miloš Kocbek
[asterisk-dev] Rgd Zaptel code for Asterisk
From
: Lavanya Chelikam
Re: [asterisk-dev] Rgd Zaptel code for Asterisk
From
: Philipp Kempgen
Re: [asterisk-dev] Rgd Zaptel code for Asterisk
From
: Lavanya Chelikam
Re: [asterisk-dev] Rgd Zaptel code for Asterisk
From
: Tony Mountifield
Re: [asterisk-dev] Zaptel-1.4.8; Zap trunk reverts to dialtone instead of processing call
From
: MENEAULT Maxime
Re: [asterisk-dev] Zaptel-1.4.8; Zap trunk reverts to dialtone instead of processing call
From
: MENEAULT Maxime
Re: [asterisk-dev] DSP_DIGITMODE_MUTECONF
From
: Dmitry Andrianov
[asterisk-dev] Changes to Video in Asterisk 1.6.0-beta1 ?
From
: Klaus Darilion
Re: [asterisk-dev] Rgd Zaptel code for Asterisk
From
: Lavanya Chelikam
Re: [asterisk-dev] Rgd Zaptel code for Asterisk
From
: Tzafrir Cohen
[asterisk-dev] Iax Bug in Asterisk 1.2.26?
From
: Stefan Schmidt
[asterisk-dev] Rgd Zaptel driver for ATA
From
: Lavanya Chelikam
Re: [asterisk-biz] Operator Console design
From
: Luciano Vaccarella
Re: [asterisk-dev] Iax Bug in Asterisk 1.2.26?
From
: Stefan Schmidt
Re: [asterisk-dev] SIP call-limit and Realtime
From
: Atis Lezdins
Re: [asterisk-dev] SIP call-limit and Realtime
From
: Johansson Olle E
Re: [asterisk-dev] SIP call-limit and Realtime
From
: Atis Lezdins
[asterisk-dev] Asterisk forwards Audio without early session
From
: Sebastian Damm
Re: [asterisk-dev] [zaptel-commits] tzafrir: branch 1.4 r3708 - in /branches/1.4: zaptel-base.c zaptel.h zconfig.h
From
: Kevin P. Fleming
Re: [asterisk-dev] SIP call-limit and Realtime
From
: Johansson Olle E
Re: [asterisk-dev] [zaptel-commits] tzafrir: branch 1.4 r3708 - in /branches/1.4: zaptel-base.c zaptel.h zconfig.h
From
: Tzafrir Cohen
Re: [asterisk-dev] Asterisk forwards Audio without early session
From
: Kevin P. Fleming
Re: [asterisk-dev] [zaptel-commits] tzafrir: branch 1.4 r3708 - in /branches/1.4: zaptel-base.c zaptel.h zconfig.h
From
: Kevin P. Fleming
Re: [asterisk-dev] Asterisk forwards Audio without early session
From
: Maksym Sobolyev
Re: [asterisk-dev] Asterisk forwards Audio without early session
From
: Johansson Olle E
Re: [asterisk-dev] Asterisk forwards Audio without early session
From
: Kevin P. Fleming
Re: [asterisk-dev] Iax Bug in Asterisk 1.2.26?
From
: Nic Bellamy
Re: [asterisk-dev] Iax Bug in Asterisk 1.2.26?
From
: Russell Bryant
[asterisk-dev] Call parking bug
From
: Dan Austin
Re: [asterisk-dev] Call parking bug
From
: Michiel van Baak
Re: [asterisk-dev] SIP call-limit and Realtime
From
: Atis Lezdins
Re: [asterisk-dev] Call parking bug
From
: Antonio Gallo
Re: [asterisk-dev] Asterisk forwards Audio without early session
From
: Sebastian Damm
Re: [asterisk-dev] Asterisk forwards Audio without early session
From
: Sebastian Damm
[asterisk-dev] How to develop a new channel driver for asterisk
From
: MOSBAH ABDELKADER
Re: [asterisk-dev] SIP call-limit and Realtime
From
: Benny Amorsen
[asterisk-dev] XXX in app_dial in 1.6
From
: Klaus Darilion
[asterisk-dev] A question about the nonce generation and checking
From
: Isaac Lee
Re: [asterisk-dev] SIP call-limit and Realtime
From
: Johansson Olle E
Re: [asterisk-dev] A question about the nonce generation and checking
From
: Klaus Darilion
Re: [asterisk-dev] A question about the nonce generation and checking
From
: Johansson Olle E
Re: [asterisk-dev] SIP call-limit and Realtime
From
: Johansson Olle E
Re: [asterisk-dev] How to develop a new channel driver for asterisk
From
: Kevin P. Fleming
Re: [asterisk-dev] [zaptel-commits] tzafrir: branch 1.4 r3708 - in /branches/1.4: zaptel-base.c zaptel.h zconfig.h
From
: Kevin P. Fleming
Re: [asterisk-dev] SIP call-limit and Realtime
From
: Atis Lezdins
Re: [asterisk-dev] Asterisk start handle a call after 3 seconds
From
: asterisk
[asterisk-dev] my Asterisk is missing time...
From
: Wolfgang Pichler
Re: [asterisk-dev] Asterisk start handle a call after 3 seconds
From
: Steven S. Critchfield
Re: [asterisk-dev] my Asterisk is missing time...
From
: Steven S. Critchfield
Re: [asterisk-dev] Asterisk start handle a call after 3 seconds
From
: asterisk
[asterisk-dev] Call transfer
From
: Maxim Telegin
Re: [asterisk-dev] my Asterisk is missing time...
From
: Wolfgang Pichler
Re: [asterisk-dev] Call transfer
From
: Kevin P. Fleming
Re: [asterisk-dev] my Asterisk is missing time...
From
: Steven S. Critchfield
Re: [asterisk-dev] my Asterisk is missing time...
From
: Atis Lezdins
Re: [asterisk-dev] my Asterisk is missing time...
From
: David Boyd
Re: [asterisk-dev] Asterisk start handle a call after 3 seconds
From
: Michael Neuhauser
[asterisk-dev] Attn Olle: closed bug #9239 - "xfersound"
From
: Nic Bellamy
Re: [asterisk-dev] my Asterisk is missing time...
From
: Wolfgang Pichler
Re: [asterisk-dev] my Asterisk is missing time...
From
: Matt Riddell
Re: [asterisk-dev] A question about the nonce generation and checking
From
: Isaac Lee
Re: [asterisk-dev] my Asterisk is missing time...
From
: Tzafrir Cohen
Re: [asterisk-dev] my Asterisk is missing time...
From
: Tilghman Lesher
Re: [asterisk-dev] my Asterisk is missing time...
From
: Matt Riddell
Re: [asterisk-dev] my Asterisk is missing time...
From
: Wolfgang Pichler
Re: [asterisk-dev] channel datastore question (repost: digium mail server problems?)
From
: Klaus Darilion
Re: [asterisk-dev] Asterisk start handle a call after 3 seconds
From
: asterisk
Re: [asterisk-dev] Asterisk start handle a call after 3 seconds
From
: Michael Neuhauser
[asterisk-dev] asterisk optimalizations
From
: marek cervenka
Re: [asterisk-dev] Asterisk start handle a call after 3 seconds
From
: asterisk
Re: [asterisk-dev] Asterisk Performance as a B2BUA]
From
: Di-Shi Sun
Re: [asterisk-dev] my Asterisk is missing time...
From
: Wolfgang Pichler
Re: [asterisk-dev] my Asterisk is missing time...
From
: Atis Lezdins
Re: [asterisk-dev] Asterisk start handle a call after 3 seconds
From
: Kevin P. Fleming
Re: [asterisk-dev] asterisk optimalizations
From
: Atis Lezdins
Re: [asterisk-dev] Asked to transmit frame type 64, while native formats is 0x4
From
: Sergey Tamkovich
Re: [asterisk-dev] How to develop a new channel driver for asterisk
From
: Moises Silva
Re: [asterisk-dev] How to develop a new channel driver for asterisk
From
: Moises Silva
Re: [asterisk-dev] Asterisk start handle a call after 3 seconds
From
: Michael Neuhauser
Re: [asterisk-dev] Iax Bug in Asterisk 1.2.26?
From
: Leonardo Gomes Figueira
[asterisk-dev] Attend transfer * 1.4.17, Zap channels keeps in music on hold after transfer completes
From
: Matheus Rossato
Re: [asterisk-dev] Iax Bug in Asterisk 1.2.26?
From
: Tilghman Lesher
Re: [asterisk-dev] How to develop a new channel driver for asterisk
From
: Russell Bryant
[asterisk-dev] Trunk configure bombs, but the files are there and 1.4.x builds fine
From
: Brian Capouch
Re: [asterisk-dev] Trunk configure bombs, but the files are there and 1.4.x builds fine
From
: Tilghman Lesher
Re: [asterisk-dev] Including port number on SDP Contact
From
: Sergey Tamkovich
[asterisk-dev] help with testing needed
From
: Dmitry Andrianov
Re: [asterisk-dev] [asterisk-commits] oej: trunk r99644 - in /trunk: channels/ include/asterisk/ main/
From
: Kevin P. Fleming
Re: [asterisk-dev] Zaptel-1.4.8; Zap trunk reverts to dialtone instead of processing call
From
: Kevin P. Fleming
[asterisk-dev] Rhino incorporating into zaptel tree
From
: Bob
Re: [asterisk-dev] Rhino incorporating into zaptel tree
From
: Tilghman Lesher
Re: [asterisk-dev] Rhino incorporating into zaptel tree
From
: Oron Peled
Re: [asterisk-dev] Rhino incorporating into zaptel tree
From
: Jared Smith
Re: [asterisk-dev] Rhino incorporating into zaptel tree
From
: Jared Smith
Re: [asterisk-dev] Rhino incorporating into zaptel tree
From
: Brian Capouch
Re: [asterisk-dev] Rhino incorporating into zaptel tree
From
: Russell Bryant
Re: [asterisk-dev] Rhino incorporating into zaptel tree
From
: James Finstrom
Re: [asterisk-dev] Zaptel-1.4.8; Zap trunk reverts to dialtone instead of processing call
From
: MENEAULT Maxime
[asterisk-biz] How to make TE220B work well?
From
: sunxiujun26
[asterisk-dev] Challenging a send-only INVITE
From
: SCG
[asterisk-dev] Challenging a sendonly INVITE
From
: SCG2
[asterisk-dev] Asterisk 1.4.18 and 1.6.0-beta2 coming soon
From
: Russell Bryant
Re: [asterisk-dev] Asterisk 1.4.18 and 1.6.0-beta2 coming soon
From
: Dan Austin
Re: [asterisk-dev] ChannelRedirect improvments (was: Asterisk 1.4.18 and 1.6.0-beta2 coming soon)
From
: Johan Wilfer
[asterisk-dev] (((Getting debug level)))
From
: Ed Greenberg
Re: [asterisk-dev] (((Getting debug level)))
From
: Jim Capp
Re: [asterisk-dev] (((Getting debug level)))
From
: Dmitry Andrianov
Re: [asterisk-dev] ChannelRedirect improvments
From
: Russell Bryant
Re: [asterisk-dev] Asterisk 1.4.18 and 1.6.0-beta2 coming soon
From
: Russell Bryant
[asterisk-dev] measure call quality for performance test
From
: Di-Shi Sun
Re: [asterisk-dev] measure call quality for performance test
From
: Alex Balashov
Re: [asterisk-dev] measure call quality for performance test
From
: Jared Smith
[asterisk-dev] When can I AIG?
From
: Evan Ruff
Re: [asterisk-dev] measure call quality for performance test
From
: Benny Amorsen
Re: [asterisk-dev] When can I AIG?
From
: Steven S. Critchfield
[asterisk-dev] New Team Member: Jeff Peeler
From
: Russell Bryant
Re: [asterisk-dev] New Team Member: Jeff Peeler
From
: jcapp
[asterisk-announce] Asterisk 1.6.0-beta2 and 1.4.18-rc1 Now Available
From
: The Asterisk Development Team
Re: [asterisk-dev] Asterisk 1.4.18 and 1.6.0-beta2 coming soon
From
: Johansson Olle E
Re: [asterisk-dev] Challenging a sendonly INVITE
From
: Johansson Olle E
Re: [asterisk-dev] Asterisk 1.4.18 and 1.6.0-beta2 coming soon
From
: Benny Amorsen
Re: [asterisk-dev] Challenging a sendonly INVITE
From
: Maxim Sobolev
Re: [asterisk-biz] How to make TE220B work well?
From
: sunxiujun26
[asterisk-biz] asterisk-biz]RE: How to make TE220B work well?
From
: sunxiujun26
Re: [asterisk-dev] [svn-commits] oej: branch 1.4 r100740 - /branches/1.4/channels/chan_sip.c
From
: Russell Bryant
Re: [asterisk-dev] Asterisk 1.6.0-beta2 and 1.4.18-rc1 Now Available
From
: Jim Capp
[asterisk-dev] Asterisk 1.4.18-rc2 Now Available
From
: The Asterisk Development Team
Re: [asterisk-dev] Asterisk 1.6.0-beta2 and 1.4.18-rc1 Now Available
From
: Russell Bryant
[asterisk-dev] Patch: SIP: choosing common codec and select "free" codec for transcoding
From
: Andrey Sofronov
Re: [asterisk-dev] Patch: SIP: choosing common codec and select "free" codec for transcoding
From
: Jared Smith
Re: [asterisk-dev] Patch: SIP: choosing common codec and select "free"codec for transcoding
From
: asterisk
Re: [asterisk-dev] Asterisk 1.4.18-rc2 Now Available
From
: Atis Lezdins
Re: [asterisk-dev] Asterisk 1.4.18-rc2 Now Available
From
: bkruse
Re: [asterisk-dev] Asterisk 1.4.18-rc2 Now Available
From
: Atis Lezdins
Re: [asterisk-dev] SOA & Web Services for Asterisk
From
: Kevin Bouchard
[asterisk-dev] Dependencies on zaptel
From
: Andre Courchesne - Prival
Re: [asterisk-dev] Dependencies on zaptel
From
: Tzafrir Cohen
Re: [asterisk-dev] [asterisk-commits] murf: branch group/CDRfix5 r101034 - in /team/group/CDRfix5: ./ agi/ apps/ bu...
From
: Sean Bright
Re: [asterisk-dev] Zaptel-1.4.8; Zap trunk reverts to dialtone instead of processing call
From
: syd wonder
Re: [asterisk-dev] Asterisk 1.4.18-rc2 Now Available
From
: Namal Gunawardene
Re: [asterisk-dev] Asterisk 1.4.18-rc2 Now Available
From
: Brandon Kruse
Re: [asterisk-dev] Asterisk 1.4.18-rc2 Now Available
From
: Brandon Kruse
Re: [asterisk-dev] Zaptel-1.4.8; Zap trunk reverts to dialtone instead of processing call
From
: MENEAULT Maxime
Re: [asterisk-dev] SOA & Web Services for Asterisk
From
: Stefan Reuter
Re: [asterisk-dev] Asterisk 1.4.18-rc2 Now Available
From
: Atis Lezdins
Re: [asterisk-dev] Asterisk 1.4.18-rc2 Now Available
From
: Russell Bryant
Re: [asterisk-dev] Asterisk 1.4.18-rc2 Now Available
From
: Russell Bryant
Re: [asterisk-dev] Asterisk 1.4.18-rc2 Now Available
From
: Joel Vandal
Re: [asterisk-dev] Asterisk 1.4.18-rc2 Now Available
From
: Matt Florell
Re: [asterisk-dev] Asterisk 1.4.18-rc2 Now Available
From
: Atis Lezdins
Re: [asterisk-dev] Asterisk 1.4.18-rc2 Now Available
From
: Joshua Colp
Re: [asterisk-dev] Asterisk 1.4.18-rc2 Now Available
From
: Atis Lezdins
Re: [asterisk-dev] Asterisk 1.6.0-beta2 and 1.4.18-rc1 Now Available
From
: Jim Capp
[asterisk-dev] Kill the user - a murder that needs testing
From
: Johansson Olle E
Re: [asterisk-dev] Asterisk 1.6.0-beta2 and 1.4.18-rc1 Now Available
From
: James Finstrom
[asterisk-dev] minor (one line) patch for app_voicemail
From
: Kurt Lidl
[asterisk-dev] Asterisk 1.4.18-rc3 Now Available
From
: The Asterisk Development Team
[asterisk-dev] Request for Comment Analog states.
From
: James Finstrom
Re: [asterisk-dev] Asterisk 1.6.0-beta2 and 1.4.18-rc1 Now Available
From
: Russell Bryant
Re: [asterisk-dev] Request for Comment Analog states.
From
: James Finstrom
Re: [asterisk-dev] Asterisk 1.4.18-rc2 Now Available
From
: Matt Riddell
[asterisk-dev] ztscan checks __ZT_SIG_DACS for a digital span
From
: Tzafrir Cohen
Re: [asterisk-dev] Asterisk 1.4.18-rc2 Now Available
From
: Matt Riddell
Re: [asterisk-dev] Request for Comment Analog states.
From
: Matt Riddell
Re: [asterisk-dev] Request for Comment Analog states.
From
: Matt Riddell
Re: [asterisk-dev] Asterisk 1.4.18-rc2 Now Available
From
: Russell Bryant
Re: [asterisk-dev] Patch: SIP: choosing common codec and select "free"codec for transcoding
From
: Andrey Sofronov
Re: [asterisk-dev] Attn Olle: closed bug #9239 - "xfersound"
From
: Johansson Olle E
Re: [asterisk-dev] Request for Comment Analog states.
From
: MENEAULT Maxime
Re: [asterisk-dev] Attn Olle: closed bug #9239 - "xfersound"
From
: Steve Davies
Re: [asterisk-dev] Attn Olle: closed bug #9239 - "xfersound"
From
: Johansson Olle E
Re: [asterisk-dev] ztscan checks __ZT_SIG_DACS for a digital span
From
: Kevin P. Fleming
[asterisk-dev] Asterisk mishandling user busy isdn releases
From
: Ken Leland III
[asterisk-dev] problem using Cagi
From
: Yelson Vivas
[asterisk-dev] Asterisk 1.4.18-rc4 Now Available
From
: The Asterisk Development Team
[asterisk-dev] DEBUG_THREADS and unreleased locks
From
: Norman Franke
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