April 2008 Archives by date

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Starting: Tue Apr 1 03:08:51 CDT 2008
Ending: Wed Apr 30 15:27:33 CDT 2008
Messages: 283

• [asterisk-dev] PREDIGITS for WaitExten   Timo Teräs
• [asterisk-dev] PREDIGITS for WaitExten   Timo Teräs
• [asterisk-dev] NCS patch for 1.4.18   Nickola Kolev
• [asterisk-dev] Asterisk 1.4.19-rc4 and 1.6.0-beta7 Now Available   Maxim Sobolev
• [asterisk-dev] app_confbridge   Joshua Colp
• [asterisk-dev] app_confbridge   Joshua Colp
• [asterisk-dev] PREDIGITS for WaitExten   Tilghman Lesher
• [asterisk-dev] PREDIGITS for WaitExten   Timo Teräs
• [asterisk-dev] app_confbridge   Marcus Hunger
• [asterisk-dev] app_confbridge   Dan Austin
• [asterisk-dev] beta 7.1: probably a bug?   sean darcy
• [asterisk-dev] app_confbridge   Marcus Hunger
• [asterisk-dev] PREDIGITS for WaitExten   Tilghman Lesher
• [asterisk-dev] chan-mobile   Hans Witvliet
• [asterisk-dev] Fwd: [svn-commits] russell: trunk r4121 - /trunk/   John Todd
• [asterisk-dev] Fwd: [svn-commits] russell: trunk r4121 - /trunk/   Michiel van Baak
• [asterisk-dev] Fwd: [svn-commits] russell: trunk r4121 - /trunk/   Russell Bryant
• [asterisk-dev] chan-mobile   David Bowerman
• [asterisk-dev] chan-mobile   Steve Totaro
• [asterisk-dev] ilbc codec compile issue   Di-Shi Sun
• [asterisk-dev] PREDIGITS for WaitExten   Timo Teräs
• [asterisk-dev] PREDIGITS for WaitExten   Tilghman Lesher
• [asterisk-dev] PREDIGITS for WaitExten   Timo Teräs
• [asterisk-dev] sandeep.s at briotelecom.com   sandeep.s
• [asterisk-dev] ilbc codec compile issue   Kevin P. Fleming
• [asterisk-dev] ilbc codec compile issue   Jeffrey Ollie
• [asterisk-dev] ilbc codec compile issue   Kevin P. Fleming
• [asterisk-dev] Debugging SIP locking/reload problem   Atlan Fox
• [asterisk-dev] Debugging SIP locking/reload problem   Michiel van Baak
• [asterisk-dev] chan_zap.c question   Steve Davies
• [asterisk-dev] Asterisk 1.4.19 and Asterisk-addons 1.6.0-beta3 Released   The Asterisk Development Team
• [asterisk-dev] chan_zap.c question   Kevin P. Fleming
• [asterisk-dev] SIP to 3G and vice versa   Kévin Bouchard
• [asterisk-dev] ilbc codec compile issue   Faidon Liambotis
• [asterisk-dev] ilbc codec compile issue   Juan Carlos Castro y Castro
• [asterisk-dev] ilbc codec compile issue   Jeffrey Ollie
• [asterisk-dev] ilbc codec compile issue   Tzafrir Cohen
• [asterisk-dev] ilbc codec compile issue   Jeffrey Ollie
• [asterisk-dev] adjusting the playback speed of voicemail messages.   Travis Hein
• [asterisk-dev] Cross compile issue in Zaptel 1.4.9.2   Kristian Kielhofner
• [asterisk-dev] adjusting the playback speed of voicemail messages.   Steven S. Critchfield
• [asterisk-dev] asterisk-1.6.0-beta7.1 chan_sip.c authuser not found when not using deprecated username   Dennis DeDonatis
• [asterisk-dev] Real-time call control for Dial app   Kaloyan Kovachev
• [asterisk-dev] (Refrased question). Does anyone have tips for debugging asterisk threading/reload problems   Atlan Fox
• [asterisk-dev] adjusting the playback speed of voicemail messages.   Johan Wilfer
• [asterisk-dev] chan_zap.c question   Steve Davies
• [asterisk-dev] adjusting the playback speed of voicemail messages.   John Todd
• [asterisk-dev] adjusting the playback speed of voicemail messages.   Steven S. Critchfield
• [asterisk-dev] Tiny addition to app_queue.c   John Peter Loh
• [asterisk-dev] adjusting the playback speed of voicemail messages.   Steven S. Critchfield
• [asterisk-dev] adjusting the playback speed of voicemail messages.   Michael Grigoni
• [asterisk-dev] adjusting the playback speed of voicemail messages.   Dan Mills
• [asterisk-dev] h324m TIFFReverseBits function   Kévin Bouchard
• [asterisk-dev] adjusting the playback speed of voicemail messages.   Michael Grigoni
• [asterisk-dev] adjusting the playback speed of voicemail messages.   Steve Underwood
• [asterisk-dev] Combined patch with queue-state and bug12127 for 1.4.x   Rajkumar S
• [asterisk-dev] question about VoiceMail function   mark morreny
• [asterisk-dev] [svn-commits at lists.digium.com: [svn-commits] dhubbard: branch 1.4 r112689 - /branches/1.4/main/asterisk.c]   Tzafrir Cohen
• [asterisk-dev] question about VoiceMail function   Mayank Mathur
• [asterisk-dev] Custom MOH Generator   Jon Schøpzinsky
• [asterisk-dev] [svn-commits at lists.digium.com: [svn-commits] dhubbard: branch 1.4 r112689 - /branches/1.4/main/asterisk.c]   Kevin P. Fleming
• [asterisk-dev] Custom MOH Generator   Kevin P. Fleming
• [asterisk-dev] Asterisk trunk failure to build with zaptel branches/1.4   Brian Candler
• [asterisk-dev] Combined patch with queue-state and bug12127 for 1.4.x   Joel Vandal
• [asterisk-dev] (no subject)   gilbert saunders
• [asterisk-dev] exchange rtp info between channels   André Abrantes
• [asterisk-dev] question about VoiceMail function   Tilghman Lesher
• [asterisk-dev] (no subject)   Moises Silva
• [asterisk-dev] Combined patch with queue-state and bug12127 for 1.4.x   Rajkumar S
• [asterisk-dev] Asterisk 1.4.19 and Asterisk-addons 1.6.0-beta3 Released   Sergey Okhapkin
• [asterisk-dev] [svn-commits at lists.digium.com: [svn-commits] dhubbard: branch 1.4 r112689 - /branches/1.4/main/asterisk.c]   John Todd
• [asterisk-dev] Asterisk trunk failure to build with zaptel branches/1.4   Brian Candler
• [asterisk-dev] Asterisk 1.4.19 and Asterisk-addons 1.6.0-beta3 Released   Tzafrir Cohen
• [asterisk-dev] Asterisk 1.4.19 and Asterisk-addons 1.6.0-beta3 Released   Sergey Okhapkin
• [asterisk-dev] No Read Events in oh323_read() from fdno 2 -- But from fdno 0 Instead   Robin J. Russell
• [asterisk-dev] [svn-commits at lists.digium.com: [svn-commits] dhubbard: branch 1.4 r112689 - /branches/1.4/main/asterisk.c]   Daniel Hazelbaker
• [asterisk-dev] [svn-commits at lists.digium.com: [svn-commits] dhubbard: branch 1.4 r112689 - /branches/1.4/main/asterisk.c]   John Todd
• [asterisk-dev] Dial tone Problem   Murithi Martin
• [asterisk-dev] chan_sip.c: realtime_peer function make crash on version 1.4.19 and 1.6.0Beta7.1   Germán Aracil Boned
• [asterisk-dev] chan_sip.c: realtime_peer function make crash on version 1.4.19 and 1.6.0Beta7.1   Germán Aracil Boned
• [asterisk-dev] (no subject)   Brian Capouch
• [asterisk-dev] [svn-commits at lists.digium.com: [svn-commits] dhubbard: branch 1.4 r112689 - /branches/1.4/main/asterisk.c]   Daniel Hazelbaker
• [asterisk-dev] [svn-commits at lists.digium.com: [svn-commits] dhubbard: branch 1.4 r112689 - /branches/1.4/main/asterisk.c]   Tzafrir Cohen
• [asterisk-dev] [svn-commits at lists.digium.com: [svn-commits] dhubbard: branch 1.4 r112689 - /branches/1.4/main/asterisk.c]   Steve Edwards
• [asterisk-dev] [svn-commits at lists.digium.com: [svn-commits] dhubbard: branch 1.4 r112689 - /branches/1.4/main/asterisk.c]   Clive Nicolson
• [asterisk-dev] exchange rtp info between channels   Sergey Tamkovich
• [asterisk-dev] chan_sip.c: realtime_peer function make crash on version 1.4.19 and 1.6.0Beta7.1   Adrian A
• [asterisk-dev] [Asterisk-video] SIP to 3G and vice versa   Klaus Darilion
• [asterisk-dev] Tiny addition to app_queue.c   William Moore
• [asterisk-dev] Poopy (ffffffff) on card 1!   Vinícius Fontes
• [asterisk-dev] Poopy (ffffffff) on card 1!   Matthew Fredrickson
• [asterisk-dev] [svn-commits at lists.digium.com: [svn-commits] dhubbard: branch 1.4 r112689 - /branches/1.4/main/asterisk.c]   Daniel Hazelbaker
• [asterisk-dev] [svn-commits at lists.digium.com: [svn-commits] dhubbard: branch 1.4 r112689 - /branches/1.4/main/asterisk.c]   Tzafrir Cohen
• [asterisk-dev] [svn-commits at lists.digium.com: [svn-commits] dhubbard: branch 1.4 r112689 - /branches/1.4/main/asterisk.c]   Kevin P. Fleming
• [asterisk-dev] Real-time call control for Dial app   Kaloyan Kovachev
• [asterisk-dev] Asterisk trunk failure to build with zaptel branches/1.4   Jason Parker
• [asterisk-dev] [svn-commits at lists.digium.com: [svn-commits] dhubbard: branch 1.4 r112689 - /branches/1.4/main/asterisk.c]   Daniel Hazelbaker
• [asterisk-dev] [svn-commits at lists.digium.com: [svn-commits] dhubbard: branch 1.4 r112689 - /branches/1.4/main/asterisk.c]   Tzafrir Cohen
• [asterisk-dev] [asterisk-announce] Zaptel 1.2.25 and 1.4.10 released   Asterisk Development Team
• [asterisk-dev] Message waiting indication(MWI) for voicemail - to H323 endpoints   Anisha Kumar
• [asterisk-dev] Message waiting indication(MWI) for voicemail - to H323 endpoints   Anisha Kumar
• [asterisk-dev] How the code is organized ?   Gianluca Crucillà
• [asterisk-dev] How the code is organized ?   Alexey Shimeshov
• [asterisk-dev] meetme and zaptel mixing algorithm   Takashi Ohashi
• [asterisk-dev] How to syncronise the audio in the local and remote machine   Gianluca Crucillà
• [asterisk-dev] Playback queue agent id when call is connected   Tommaso Calosi
• [asterisk-dev] How the code is organized ?   Terry Wilson
• [asterisk-dev] How to syncronise the audio in the local and remote machine   gincantalupo
• [asterisk-dev] How to syncronise the audio in the local and remote machine   Gianluca Crucillà
• [asterisk-dev] How to syncronise the audio in the local and remote machine   gincantalupo
• [asterisk-dev] Asterisk trunk failure to build with zaptel branches/1.4   Brian Candler
• [asterisk-dev] regular expression issue   Di-Shi Sun
• [asterisk-dev] Play at the world's leading online casino.....   mailman at lists.digium.com
• [asterisk-dev] features.conf Problem with DTMF_sequence   Sergey Shumeyko
• [asterisk-dev] Chat Line APP   Justin Newman
• [asterisk-dev] TouchTone to a text file?   Justin Newman
• [asterisk-dev] Bridging Question   Justin Newman
• [asterisk-dev] [svn-commits at lists.digium.com: [svn-commits] dhubbard: branch 1.4 r112689 - /branches/1.4/main/asterisk.c]   John Todd
• [asterisk-dev] AsteriskNow source?   John Lange
• [asterisk-dev] Chat Line APP   Loic DIDELOT
• [asterisk-dev] agenst.conf   Alex Haugg
• [asterisk-dev] agents.conf   Alex Haugg
• [asterisk-dev] chan_sip.c / sip.conf : need similar option as "promiscredir" but for REFER and not redirect (302)   tloginbr-asteriskdev at yahoo.com.br
• [asterisk-dev] MALLOC_DEBUG for our application   Jim Gottlieb
• [asterisk-dev] agenst.conf   Tzafrir Cohen
• [asterisk-dev] Looking for volunteers to debug chan_h323 (issue 9299)   Sergey Tamkovich
• [asterisk-dev] Jitter buffer problem with queues?   John Lange
• [asterisk-dev] adjusting the playback speed of voicemail messages.   Travis Hein
• [asterisk-dev] [svn-commits at lists.digium.com: [svn-commits] dhubbard: branch 1.4 r112689 - /branches/1.4/main/asterisk.c]   Kevin P. Fleming
• [asterisk-dev] [svn-commits at lists.digium.com: [svn-commits] dhubbard: branch 1.4 r112689 - /branches/1.4/main/asterisk.c]   John Todd
• [asterisk-dev] adjusting the playback speed of voicemail messages.   Steve Underwood
• [asterisk-dev] MWI from Telco on zap channel in 1.6 ?   Jim Duda
• [asterisk-dev] MWI from Telco on zap channel in 1.6 ?   Kevin P. Fleming
• [asterisk-dev] wcfxo and X100P card won't play nice.   Alex Balashov
• [asterisk-dev] wcfxo and X100P card won't play nice.   Bob
• [asterisk-dev] wcfxo and X100P card won't play nice.   syd wonder
• [asterisk-dev] wcfxo and X100P card won't play nice.   Alex Balashov
• [asterisk-dev] wcfxo and X100P card won't play nice.   Alex Balashov
• [asterisk-dev] wcfxo and X100P card won't play nice.   Alex Balashov
• [asterisk-dev] wcfxo and X100P card won't play nice.   Alex Balashov
• [asterisk-dev] wcfxo and X100P card won't play nice.   Tzafrir Cohen
• [asterisk-dev] wcfxo and X100P card won't play nice.   Alex Balashov
• [asterisk-dev] MWI from Telco on zap channel in 1.6 ?   Jim Duda
• [asterisk-dev] wcfxo and X100P card won't play nice.   Jeff Gehlbach
• [asterisk-dev] wcfxo and X100P card won't play nice.   Jay R. Ashworth
• [asterisk-dev] [asterisk-commits] seanbright: trunk r114165 - /trunk/apps/app_chanspy.c   Kevin P. Fleming
• [asterisk-dev] testing zaptel with sparse   Tzafrir Cohen
• [asterisk-dev] Sangoma as Zaptel timing source.   Alex Balashov
• [asterisk-dev] Sangoma as Zaptel timing source.   Kevin P. Fleming
• [asterisk-dev] ast_masq_park_call() - getting the parked slot   Paulo Garcia
• [asterisk-dev] AsteriskNow source?   mohammed ilyas
• [asterisk-dev] AsteriskNow source?   Tzafrir Cohen
• [asterisk-dev] jitterbuffer question   Klaus Darilion
• [asterisk-dev] jitterbuffer question   Russell Bryant
• [asterisk-dev] AsteriskNow source?   John Lange
• [asterisk-dev] jitterbuffer question   Klaus Darilion
• [asterisk-dev] soft hangup and DIALSTATUS   Atis Lezdins
• [asterisk-dev] [asterisk-commits] mmichelson: branch 1.4 r114226 - /branches/1.4/apps/app_chanspy.c   Kevin P. Fleming
• [asterisk-dev] jitterbuffer question   Russell Bryant
• [asterisk-dev] SIP registration and fromdomain patch   Steve Davies
• [asterisk-dev] [asterisk-commits] mmichelson: branch 1.4 r114226 - /branches/1.4/apps/app_chanspy.c   Daniel Hazelbaker
• [asterisk-dev] Tiny little bug   Brett Crapser
• [asterisk-dev] Tiny little bug   Tilghman Lesher
• [asterisk-dev] astcli [was: Re: Tiny little bug]   Tzafrir Cohen
• [asterisk-dev] Asterisk Busy Line Feature: Subscription not persistent across Asterisk restarts in Asterisk 1.4.19   Andreas Brodmann
• [asterisk-dev] Same channel name at the same time?   Ma Zhiyong
• [asterisk-dev] features.conf Problem with DTMF_sequence   Sergey Shumeyko
• [asterisk-dev] Same channel name at the same time?   Jared Smith
• [asterisk-dev] features.conf Problem with DTMF_sequence   William Moore
• [asterisk-dev] Same channel name at the same time?   Tilghman Lesher
• [asterisk-dev] Same channel name at the same time?   Steve Totaro
• [asterisk-dev] Asterisk Busy Line Feature: Subscription not persistent across Asterisk restarts in Asterisk 1.4.19   Russell Bryant
• [asterisk-dev] Strange SIP packet   fadey
• [asterisk-dev] Strange SIP packet   Klaus Darilion
• [asterisk-dev] AST_CONTROL_ANSWER doesn't get copied on bridge   Guillermo Winkler
• [asterisk-dev] AST-2008-006 - 3-way handshake in IAX2 incomplete   Security Officer
• [asterisk-dev] [asterisk-announce] Asterisk 1.2.28, 1.4.19.1, and 1.6.0-beta8 Released   The Asterisk Development Team
• [asterisk-dev] AST_CONTROL_ANSWER doesn't get copied on bridge   Klaus Darilion
• [asterisk-dev] AST_CONTROL_ANSWER doesn't get copied on bridge   Guillermo Winkler
• [asterisk-dev] A method to retrieve sip custom headers, is this a good way?   Eric Dantie
• [asterisk-dev] A method to retrieve sip custom headers, is this a good way?   Johan Wilfer
• [asterisk-dev] AST_CONTROL_ANSWER doesn't get copied on bridge   Atis Lezdins
• [asterisk-dev] AST_CONTROL_ANSWER doesn't get copied on bridge   Guillermo Winkler
• [asterisk-dev] AST_CONTROL_ANSWER doesn't get copied on bridge   Atis Lezdins
• [asterisk-dev] Reception of RFC2833 (DTMF) when dtmfmode=info   Bryan Boatright
• [asterisk-dev] Reception of RFC2833 (DTMF) when dtmfmode=info   Jared Smith
• [asterisk-dev] Reception of RFC2833 (DTMF) when dtmfmode=info   Kevin P. Fleming
• [asterisk-dev] FXS to FXO loopback test   Pawel Pastuszak
• [asterisk-dev] mISDN clear channel   Kévin Bouchard
• [asterisk-dev] FXS to FXO loopback test   Matthew Fredrickson
• [asterisk-dev] FXS to FXO loopback test   Pawel Pastuszak
• [asterisk-dev] MFC/R2 in chan_zap   Moises Silva
• [asterisk-dev] mISDN clear channel   Klaus Darilion
• [asterisk-dev] 1.2.28 patch - a minor point   Paul Hewlett
• [asterisk-dev] (off-list) Re: MFC/R2 in chan_zap   Russell Bryant
• [asterisk-dev] 302 - REDIRECT (Moved Temporaly)   Thiago Maluf
• [asterisk-dev] 302 - REDIRECT (Moved Temporaly)   Johansson Olle E
• [asterisk-dev] h263+ format hardcoded in app_h324m   Kévin Bouchard
• [asterisk-dev] [asterisk-commits] mvanbaak: trunk r114637 - in /trunk: apps/ channels/ include/asterisk/ main/   Kevin P. Fleming
• [asterisk-dev] [asterisk-commits] mvanbaak: trunk r114637 - in /trunk: apps/ channels/ include/asterisk/ main/   Michiel van Baak
• [asterisk-dev] 1.2.28 patch - a minor point   Dinesh Nair
• [asterisk-dev] russell: branch 1.4 r114608 - /branches/1.4/channels/chan_iax2.c   Tony Mountifield
• [asterisk-dev] [asterisk-commits] mvanbaak: trunk r114637 - in /trunk: apps/ channels/ include/asterisk/ main/   Kevin P. Fleming
• [asterisk-dev] These fixes ok?   Patrick
• [asterisk-dev] These fixes ok?   Tony Mountifield
• [asterisk-dev] Overlapped dialling on SIP trunks for outgoing calls   Andreas Brodmann
• [asterisk-dev] These fixes ok?   Jeffrey Ollie
• [asterisk-dev] Overlapped dialling on SIP trunks for outgoing calls   Kevin P. Fleming
• [asterisk-dev] 1.2.28 patch - a minor point   Tzafrir Cohen
• [asterisk-dev] Overlapped dialling on SIP trunks for outgoing calls   Andreas Brodmann
• [asterisk-dev] Overlapped dialling on SIP trunks for outgoing calls   Johansson Olle E
• [asterisk-dev] Overlapped dialling on SIP trunks for outgoing calls   Tzafrir Cohen
• [asterisk-dev] Overlapped dialling on SIP trunks for outgoing calls   Johansson Olle E
• [asterisk-dev] Overlapped dialling on SIP trunks for outgoing calls   Kevin P. Fleming
• [asterisk-dev] Overlapped dialling on SIP trunks for outgoing calls   Tzafrir Cohen
• [asterisk-dev] Overlapped dialling on SIP trunks for outgoing calls   Andreas Brodmann
• [asterisk-dev] Overlapped dialling on SIP trunks for outgoing calls   Andreas Brodmann
• [asterisk-dev] Overlapped dialling on SIP trunks for outgoing calls   Tzafrir Cohen
• [asterisk-dev] These fixes ok?   Patrick
• [asterisk-dev] Overlapped dialling on SIP trunks for outgoing calls   Johansson Olle E
• [asterisk-dev] Overlapped dialling on SIP trunks for outgoing calls   Johansson Olle E
• [asterisk-dev] Overlapped dialling on SIP trunks for outgoing calls   Andreas Brodmann
• [asterisk-dev] Overlapped dialling on SIP trunks for outgoing calls   Raj Jain
• [asterisk-dev] Overlapped dialling on SIP trunks for outgoing calls   Andreas Sikkema
• [asterisk-dev] Overlapped dialling on SIP trunks for outgoing calls   Raj Jain
• [asterisk-dev] zap not coming online on fedora 8   bilal ghayyad
• [asterisk-dev] Ring detect or how to use the asterisk dsp to modify AMD so it ignores rings   Andre Courchesne - Prival
• [asterisk-dev] Overlapped dialling on SIP trunks for outgoing calls   Andreas Brodmann
• [asterisk-dev] Bridging two Channels   ast guy
• [asterisk-dev] Bridging two Channels   Wolfgang Pichler
• [asterisk-dev] Bridging two Channels   ast guy
• [asterisk-dev] Ring detect or how to use the asterisk dsp to modify AMD so it ignores rings   Clod Patry
• [asterisk-dev] Bridging two Channels   Steve Totaro
• [asterisk-dev] Bridging two Channels   ast guy
• [asterisk-dev] Bridging two Channels   Matt Florell
• [asterisk-dev] Bridging two Channels   Steve Totaro
• [asterisk-dev] zap not coming online on fedora 8   Steve Totaro
• [asterisk-dev] Bridging two Channels   Matt Florell
• [asterisk-dev] Bridging two Channels   Steve Totaro
• [asterisk-dev] [svn-commits] tilghman: trunk r114683 - /trunk/channels/chan_sip.c   Russell Bryant
• [asterisk-dev] zap not coming online on fedora 8   Tzafrir Cohen
• [asterisk-dev] Bridging two Channels   Matt Florell
• [asterisk-dev] Ring detect or how to use the asterisk dsp to modify AMD so it ignores rings   Andre Courchesne - Prival
• [asterisk-dev] zap not coming online on fedora 8   bilal ghayyad
• [asterisk-dev] Bridging two Channels   Steve Totaro
• [asterisk-dev] 1.2.28 patch - a minor point   Dinesh Nair
• [asterisk-dev] zap not coming online on fedora 8   bilal ghayyad
• [asterisk-dev] zap not coming online on fedora 8   bilal ghayyad
• [asterisk-dev] Bridging two Channels   ast guy
• [asterisk-dev] Overlapped dialling on SIP trunks for outgoing calls   Klaus Darilion
• [asterisk-dev] Overlapped dialling on SIP trunks for outgoing calls   Klaus Darilion
• [asterisk-dev] can not overwite parameters to wctdm by modprobe loading!   lizhong zhu
• [asterisk-dev] Overlapped dialling on SIP trunks for outgoing calls   Tilghman Lesher
• [asterisk-dev] Overlapped dialling on SIP trunks for outgoing calls   Andreas Brodmann
• [asterisk-dev] Overlapped dialling on SIP trunks for outgoing calls   Klaus Darilion
• [asterisk-dev] Overlapped dialling on SIP trunks for outgoing calls   Tilghman Lesher
• [asterisk-dev] Overlapped dialling on SIP trunks for outgoing calls   Tilghman Lesher
• [asterisk-dev] [asterisk-commits] tilghman: trunk r114773 - in /trunk: apps/ include/asterisk/ main/   Kevin P. Fleming
• [asterisk-dev] Overlapped dialling on SIP trunks for outgoing calls   Raj Jain
• [asterisk-dev] Overlapped dialling on SIP trunks for outgoing calls   Tilghman Lesher
• [asterisk-dev] Overlapped dialling on SIP trunks for outgoing calls   Klaus Darilion
• [asterisk-dev] 1.2.28 patch - a minor point   Paul Hewlett
• [asterisk-dev] 1.2.28 patch - a minor point   Kevin P. Fleming
• [asterisk-dev] Overlapped dialling on SIP trunks for outgoing calls   Tilghman Lesher
• [asterisk-dev] Overlapped dialling on SIP trunks for outgoing calls   Raj Jain
• [asterisk-dev] adding SIP Headers a SIP dialog   Marco Happenhofer
• [asterisk-dev] Overlapped dialling on SIP trunks for outgoing calls   Benny Amorsen
• [asterisk-dev] Overlapped dialling on SIP trunks for outgoing calls   Tilghman Lesher
• [asterisk-dev] Overlapped dialling on SIP trunks for outgoing calls   Tilghman Lesher
• [asterisk-dev] adding SIP Headers a SIP dialog   Klaus Darilion
• [asterisk-dev] zap not coming online on fedora 8   Sean Bright
• [asterisk-dev] API changes 1.4.0 -> 1.6.0, review requested on specific files   Jeff Peeler
• [asterisk-dev] Portability in Asterisk   Aadilkhan Maniyar
• [asterisk-dev] iax2_sched_replace: Unable to cancel schedule   Timo Teräs
• [asterisk-dev] [asterisk-addons-commits] tilghman: branch 1.4 r578 - /branches/1.4/Makefile.rules   Kevin P. Fleming
• [asterisk-dev] Invalid SPF record in lists.digium.com domain   Boris Bakchiev
• [asterisk-dev] Invalid SPF record in lists.digium.com domain   Alexander Lopez
• [asterisk-dev] How to add new app/res to configure script   Gabriel Ortiz Lour
• [asterisk-dev] Question   Venefax
• [asterisk-dev] API changes 1.4.0 -> 1.6.0, review requested on specific files   Russell Bryant
• [asterisk-dev] Invalid SPF record in lists.digium.com domain   Russell Bryant
• [asterisk-dev] Asterisk 1.6 peer matching working differently   Johansson Olle E
• [asterisk-dev] Asterisk 1.6 peer matching working differently   Simon Perreault

Last message date: Wed Apr 30 15:27:33 CDT 2008
Archived on: Wed Apr 30 15:27:37 CDT 2008

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