August 2010 Archives by thread
Starting: Sun Aug 1 05:36:49 CDT 2010
Ending: Tue Aug 31 16:48:31 CDT 2010
Messages: 470
- [asterisk-dev] SIP STUN support testing
Simon Perreault
- [asterisk-dev] IPv4 and IPv6 preference
Simon Perreault
- [asterisk-dev] [Code Review] Testsuite: new install_configs() function and ability for backwards compatible configs
paul.belanger at polybeacon.com
- [asterisk-dev] [Code Review] Testsuite: new install_configs() function and ability for backwards compatible configs
paul.belanger at polybeacon.com
- [asterisk-dev] IPv4 and IPv6 preference
Paul Belanger
- [asterisk-dev] [Code Review]
Tilghman Lesher
- [asterisk-dev] [Asterisk 0017686]: Asterisk crashing in ast_readaudio_callback at file.c:762
Dave WOOLLEY
- [asterisk-dev] [Code Review] __ast_play_and_record randomize prepend file
tim.ringenbach at gmail.com
- [asterisk-dev] Using FFmpeg
Jeffrey Ollie
- [asterisk-dev] [Code Review] External test for fastagi.execute()
Erin Spiceland
- [asterisk-dev] [Code Review] Remove MP3 decoder source code from main Asterisk source tree
Russell Bryant
- [asterisk-dev] [Code Review] Handle all possible responses to REFER requests.
Matthew Nicholson
- [asterisk-dev] [Code Review] Fix a problem with channel name tab completion
David Vossel
- [asterisk-dev] [Code Review] improved translation paths for wideband codecs
David Vossel
- [asterisk-dev] [Code Review] more reliable sip STUN support
David Vossel
- [asterisk-dev] Why sip_peers sometimes not removed from the peers_by_ip table?
Kirill 'Big K' Katsnelson
- [asterisk-dev] sip reload and autopeers
Kirill 'Big K' Katsnelson
- [asterisk-dev] [Code Review] External test of fastagi.getData()
Erin Spiceland
- [asterisk-dev] [Code Review] Prevent caller ID set on channel from getting discarded when originate used with local channel (small patch - review quickly, earn rewards!)
Jeff Peeler
- [asterisk-dev] [Code Review] sip: hangup channel on pending bye
David Vossel
- [asterisk-dev] Asterisk 1.8 and T.38 gatewaying
Klaus Darilion
- [asterisk-dev] [svn-commits] dvossel: trunk r278536 - /trunk/channels/chan_sip.c
Klaus Darilion
- [asterisk-dev] STUN support in chan_sip revisited
David Vossel
- [asterisk-dev] [Code Review] Top down test "callparking" GSOC 2010
mnick at digium.com
- [asterisk-dev] what use is ZT_IOMUX_xx
Muhammad Ali
- [asterisk-dev] Simple new feature
Gabriel Ortiz Lour
- [asterisk-dev] [Code Review] chan_sip: fixes provisional keepalive scheduled item crash
David Vossel
- [asterisk-dev] 1.8.0 crashing on startup
sean darcy
- [asterisk-dev] How does mISDN work?
Muhammad Ali
- [asterisk-dev] [asterisk-commits] dvossel: trunk r278619 - /trunk/channels/chan_sip.c
Klaus Darilion
- [asterisk-dev] [asterisk-commits] simon.perreault: trunk r281357 - in /trunk: ./ configs/sip.conf.sample
Marc Blanchet
- [asterisk-dev] [Code Review] Fix a problem with channel name tab completion
Russell Bryant
- [asterisk-dev] [Code Review] Use the correct IP address in the c and o sdp lines when using an externally mapped IP address
Russell Bryant
- [asterisk-dev] Realtime SIP peers becomes 'UNREACHABLE'
Deepesh D
- [asterisk-dev] [Code Review] GSoC2010: ast_storage
Tilghman Lesher
- [asterisk-dev] [Code Review] [branch] Expand select(2) bits to allow for more than FD_SETSIZE file descriptors
Russell Bryant
- [asterisk-dev] Asterisk 1.4.35 Now Available
Asterisk Development Team
- [asterisk-dev] Asterisk 1.6.2.11 Now Available
Asterisk Development Team
- [asterisk-dev] Asterisk 1.8.0-beta3 Now Available
Asterisk Development Team
- [asterisk-dev] Patch for Extend the max number of callgroups/pickupgroups
Thorolf Godawa
- [asterisk-dev] [Code Review] [branch] Expand select(2) bits to allow for more than FD_SETSIZE file descriptors
Tilghman Lesher
- [asterisk-dev] [Code Review] res_stun_monitor module
David Vossel
- [asterisk-dev] alter/implement new functions to add/remove member from some/all queues
Gabriel Ortiz Lour
- [asterisk-dev] [Code Review] Add ability to set Max-Forwards header from dialplan, general and device configuration
Matthew Nicholson
- [asterisk-dev] [Code Review] remove current STUN support from chan_sip.c
David Vossel
- [asterisk-dev] [Code Review] Check to make sure we can successfully link libsrtp in a shared library
Terry Wilson
- [asterisk-dev] [Code Review] support for GNU/kFreeBSD
Tzafrir Cohen
- [asterisk-dev] creating telemarketing system
ultra hold
- [asterisk-dev] [Code Review] GSoC2010: ast_storage
Tilghman Lesher
- [asterisk-dev] [Code Review] Add "core reload" CLI command
Russell Bryant
- [asterisk-dev] [Code Review] Properly handle 200 and unknown responses conatined in NOTIFY requests received in response to REFER requests
Matthew Nicholson
- [asterisk-dev] [Code Review]
Tilghman Lesher
- [asterisk-dev] [Code Review] Send the SRCCHANGE indication when we do a masquerade
Terry Wilson
- [asterisk-dev] DTMF Issues Version 1.6.1.20
Bryant Zimmerman
- [asterisk-dev] DTMF Issues Version 1.6.1.20
Bryant Zimmerman
- [asterisk-dev] 1.8-beta 3: crashes on startup
sean darcy
- [asterisk-dev] [Code Review] Top down test "one step parking" GSOC 2010
mnick at digium.com
- [asterisk-dev] Queue with ip address
Bhrugu Mehta
- [asterisk-dev] downloads.asterisk.org not working on IPv6
Saúl Ibarra Corretgé
- [asterisk-dev] [Code Review] External test for RECORD FILE using FastAGI
Erin Spiceland
- [asterisk-dev] Queue stats resetting on reload. Trying to engage the community...
Ron Arts
- [asterisk-dev] [Code Review] External test for SAY ALPHA using FastAGI
Erin Spiceland
- [asterisk-dev] How to build a monitoring module?
Tom Szabo
- [asterisk-dev] processing q931 via SIP INFO
Jared Mauch
- [asterisk-dev] Patch required for 1.4 to 1.6 hangup "regression"
Alistair Cunningham
- [asterisk-dev] [Code Review] Fix memory leak in manager.c action originate when using channel variables
Olle E Johansson
- [asterisk-dev] [Code Review] RTP Packets Not Set With QOS
Russell Bryant
- [asterisk-dev] Feature Request: Calling Party Category
Andre Valentin
- [asterisk-dev] [Code Review] sip: how to hang up a channel before invite receives a response
David Vossel
- [asterisk-dev] astdb->sync called too often slowing down the system
Stefan Schmidt
- [asterisk-dev] channel stay up when extension unreachable
Anton Raharja
- [asterisk-dev] q.931 dnid-subaddress
Martin Vít
- [asterisk-dev] zaptel - dahdi development documentation
"Alen Ruvi��"
- [asterisk-dev] Asterisk 1.4.36-rc1 Now Available
Asterisk Development Team
- [asterisk-dev] Asterisk 1.6.2.12-rc1 Now Available
Asterisk Development Team
- [asterisk-dev] Asterisk 1.8.0-beta4 Now Available
Asterisk Development Team
- [asterisk-dev] dahdi system.conf update
Russ Meyerriecks
- [asterisk-dev] Attended Transfers
Bryant Zimmerman
- [asterisk-dev] [Code Review] Document CONNECTEDLINE and REDIRECTING useage.
rmudgett at digium.com
- [asterisk-dev] [Code Review] ADSI and Crypto backwards compatibility with 1.6.2
Tilghman Lesher
- [asterisk-dev] Asterisk as a Gateway
Paul Hansen
- [asterisk-dev] Call Pickup Issues v1.6.2.11
Bryant Zimmerman
- [asterisk-dev] Call Pickup Issues v1.6.2.11
Bryant Zimmerman
- [asterisk-dev] [Code Review] Fix SRTP for changing SSRC and multiple a=crypto SDP lines
Terry Wilson
- [asterisk-dev] Asterisk V 1.6.2.12-rc1 Broke in-call DTMF with L3 and Broadvox
Bryant Zimmerman
- [asterisk-dev] Asterisk V 1.6.2.12-rc1 Broke in-call DTMF with L3 and Broadvox
Bryant Zimmerman
- [asterisk-dev] Brief outage of Asterisk services for maintenance, Saturday, August 28, 2010.
Asterisk Development Team
- [asterisk-dev] Brief outage of Asterisk services for maintenance, Saturday, August 28, 2010.
Asterisk Development Team
- [asterisk-dev] Brief outage of Asterisk services for maintenance, Saturday, August 28, 2010.
Paul Belanger
- [asterisk-dev] Brief outage of Asterisk services for maintenance, Saturday, August 28, 2010.
Marc Blanchet
- [asterisk-dev] Brief outage of Asterisk services for maintenance, Saturday, August 28, 2010.
Kevin P. Fleming
- [asterisk-dev] Brief outage of Asterisk services for maintenance, Saturday, August 28, 2010.
Paul Belanger
- [asterisk-dev] Brief outage of Asterisk services for maintenance, Saturday, August 28, 2010.
Kevin P. Fleming
- [asterisk-dev] Brief outage of Asterisk services for maintenance, Saturday, August 28, 2010.
Russell Bryant
- [asterisk-dev] Asterisk V 1.6.2.12-rc1 Broke in-call DTMF with L3 and Broadvox
Bryant Zimmerman
- [asterisk-dev] [Code Review] SIP: session timer behavior in Asterisk
David Vossel
- [asterisk-dev] chan_sip.c (1.6.2.11 & 1.8.0-beta3) vs "buf"
Jared Mauch
- [asterisk-dev] Asterisk V 1.6.2.12-rc1 Broke in-call DTMF with L3 and Broadvox
Bryant Zimmerman
- [asterisk-dev] [Code Review] [regression] Progress in band error (don't send RTP packets)
Terry Wilson
- [asterisk-dev] asterisk RTP processing
sarvesh telang
- [asterisk-dev] dahdi system.conf update
Shaun Ruffell
- [asterisk-dev] [Code Review] SIP: authenticate OPTIONS requests just like we would an INVITE
David Vossel
- [asterisk-dev] Call file errors in Asterisk 1.8beta
Michael Keuter
- [asterisk-dev] Call file errors in Asterisk 1.8beta
Kristijan Vrban
- [asterisk-dev] TimeStamp header in SIP
mustafa rifaee
- [asterisk-dev] segfault within libmysqlclient
Wolfgang Pichler
- [asterisk-dev] [Code Review] pbx_spool unable to load call files in 1.8.0-beta4 and trunk
Brett Bryant
- [asterisk-dev] Create trunk/peer in runtime
Juan Ramírez
- [asterisk-dev] [Code Review] 1.4 manager configuration reset to default values before config files are reloaded
Brett Bryant
- [asterisk-dev] rebuilding kmod-dahdi-linux i686 vs i386
Bob Beers
Last message date:
Tue Aug 31 16:48:31 CDT 2010
Archived on: Tue Aug 31 16:48:34 CDT 2010
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