July 2008 Archives by thread
Starting: Tue Jul 1 01:24:04 CDT 2008
Ending: Thu Jul 31 21:20:06 CDT 2008
Messages: 423
- [asterisk-dev] mvanbaak: trunk r125096 - /trunk/channels/chan_skinny.c
Michiel van Baak
- [asterisk-dev] Asterisk 1.4.21.1 Released
Jeffrey Ollie
- [asterisk-dev] Lookupcidname and utf8
Gunnar Schaller
- [asterisk-dev] the call fails to complete.
Neha Punia
- [asterisk-dev] (no subject)
Jerome Poggi
- [asterisk-dev] DTMF Volume
Konstantin Bokarius
- [asterisk-dev] DTMF Volume
Konstantin Bokarius
- [asterisk-dev] Locking, coding guidelines addition
Stephen Davies
- [asterisk-dev] Locking, coding guidelines addition
Raj Jain
- [asterisk-dev] Locking, coding guidelines addition
Grey Man
- [asterisk-dev] BUGFIX: main/rtp.c bridge_p2p_rtp_write() fix inbound payload type validity check
Tony Redstone
- [asterisk-dev] Locking, coding guidelines addition
Russell Bryant
- [asterisk-dev] Locking, coding guidelines addition
Simon Perreault
- [asterisk-dev] [asterisk-commits] oej: trunk r128237 - /trunk/configs/sip.conf.sample
Donny Kavanagh
- [asterisk-dev] Zaptel and Solaris X86
Frank Tarczynski
- [asterisk-dev] Small libss7 cleanup patch
Vincent Thomasset
- [asterisk-dev] Voicemail Call Transfer
Richard Laager
- [asterisk-dev] A plan for span
Oron Peled
- [asterisk-dev] SDP version
Dmitry Andrianov
- [asterisk-dev] SIP_HEADER in dialplan from different states of SIP dialog?
Alex Balashov
- [asterisk-dev] A plan for span
Shaun Ruffell
- [asterisk-dev] XML documentation of apps/functions/the_rest_of_the_world
Michiel van Baak
- [asterisk-dev] A plan for span
Steven S. Critchfield
- [asterisk-dev] XML documentation of apps/functions/the_rest_of_the_world
Brandon Kruse
- [asterisk-dev] A plan for span
Oron Peled
- [asterisk-dev] audiohooks - Read factory was pretty quick last time
Wolfgang Pichler
- [asterisk-dev] the latest 1.4 start crashes app_queue
Martin Vít
- [asterisk-dev] XML documentation of apps/functions/the_rest_of_the_world
Brandon Kruse
- [asterisk-dev] [policy] Bug Tracker Workflow Discussion
Leif Madsen
- [asterisk-dev] tilghman: branch 1.4 r129803 - /branches/1.4/channels/chan_iax2.c
Russell Bryant
- [asterisk-dev] chan_zap/chan_dahdi hooktransfer channel name bug
Daniel Ferrer
- [asterisk-dev] murf: trunk r130145 - /trunk/main/pbx.c
Russell Bryant
- [asterisk-dev] XML documentation of apps/functions/the_rest_of_the_world
Brandon Kruse
- [asterisk-dev] Implementing asterisk CLI permissions, feedback needed
Eliel Sardañons
- [asterisk-dev] Porting asterisk to ltib
Vadim Lebedev
- [asterisk-dev] mISDN with crypto
andrea
- [asterisk-dev] Next 1.6 beta release?
W. Michael Petullo
- [asterisk-dev] mmichelson: branch 1.4 r131299 - /branches/1.4/apps/app_queue.c
Russell Bryant
- [asterisk-dev] Reverse Scenario
Ali Karimi
- [asterisk-dev] GMail Support
Marc Smith
- [asterisk-dev] jpeeler: trunk r131868 - /trunk/channels/chan_dahdi.c
Jeff Peeler
- [asterisk-dev] How Register to ONE SIP provider with Multi Accounts
jiangtao
- [asterisk-dev] 1.4 core dump mISDN 1.2 git version
Martin Vít
- [asterisk-dev] "hangup" unanswered SIP-call
Fredrik Hansson
- [asterisk-dev] GMail Support
Marc Smith
- [asterisk-dev] Announcing AstriDevCon 2008!
Asterisk Development Team
- [asterisk-dev] SIP disconnect code
Carles Pina i Estany
- [asterisk-dev] App_bridge in Asterisk 1.6???
Douglas Garstang
- [asterisk-dev] App_bridge in Asterisk 1.6???
Douglas Garstang
- [asterisk-dev] segmentation fault with chan_h323, asterisk 1.4.21.1 , Open H.323 version v1.18.0, PWLib v1.10.0
nik600
- [asterisk-dev] Porting asterisk to i.mx 27
Vadim Lebedev
- [asterisk-dev] [svn-commits] mmichelson: branch 1.4 r132790 - /branches/1.4/channels/chan_sip.c
Mark Michelson
- [asterisk-dev] Asterisk 1.4.21.2 and 1.2.30 Released
The Asterisk Development Team
- [asterisk-dev] func_dialgroup
Pavel Jezek
- [asterisk-dev] kpfleming: branch 1.4 r132872 - in /branches/1.4: include/asterisk/ main/
Russell Bryant
- [asterisk-dev] Does Asterisk 'queue' DTMFs?
Saúl Ibarra
- [asterisk-dev] Queues and voicemail Asterisk
Evelyn Lopez
- [asterisk-dev] Call Center Type Recording
kenny sigafoose
- [asterisk-dev] Does Asterisk 'queue' DTMFs?
Steven S. Critchfield
- [asterisk-dev] Help Solve the Mysterious Slowdown in Sip channel driver in Trunk!
Steve Murphy
- [asterisk-dev] Rejected frames occuring after substantial number of calls made over PRI, related to bug 11189
Jeff Peeler
- [asterisk-dev] Asterisk 1.6, SIP 484 going to invalid extension
Carles Pina i Estany
- [asterisk-dev] iax2-parser.c - Memory
Eric Dantie
- [asterisk-dev] [asterisk-commits] mmichelson: branch 1.4 r132790 - /branches/1.4/channels/chan_sip.c
Brett Bryant
- [asterisk-dev] call parking in one step
rajeeva ranjan
- [asterisk-dev] iax2-parser.c - Memory
Octavio Ascanio Su‡árez
- [asterisk-dev] SRTP and SIP over TLS
Andre Courchesne - Prival
- [asterisk-dev] Does Asterisk 'queue' DTMFs?
Steven S. Critchfield
- [asterisk-dev] Call Intrusion
Vadim Lebedev
- [asterisk-dev] app_dial.c - multiple dials - M option - ANSWEREDTIME
Ruddy Gbaguidi
- [asterisk-dev] Astricon 2008 updates: keynotes, content, contests
John Todd
- [asterisk-dev] ASTERISK / VICIDIAL
Sammy Lakhany
- [asterisk-dev] Asterisk "moh reload" causes music to not work properly until the next "moh reload"
Fernando Urzedo
Last message date:
Thu Jul 31 21:20:06 CDT 2008
Archived on: Thu Jul 31 21:15:30 CDT 2008
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