August 2004 Archives by author

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Starting: Sun Aug 1 00:26:28 MST 2004
Ending: Tue Aug 31 18:41:59 MST 2004
Messages: 359

• [Asterisk-Dev] Can I play announcement to caller and callee when I answer the ca ll?   Tony Chan (ITApps)
• [Asterisk-Dev] how to support g.729 in ivr   3bs
• [Asterisk-Dev] How are outgoing calls handled?   Horacio J. Peña
• [Asterisk-Dev] how to get control of the file descriptor from pbx?   Horacio J. Peña
• [Asterisk-Dev] echo can in chan_modem_i4l   Niklas Ögren
• [Asterisk-Dev] DTMF and Broadvoice   Rich Adamson
• [Asterisk-Dev] VoIP SPAM, what's next ?   Rich Adamson
• [Asterisk-Dev] How do I use app_test?   Rich Adamson
• [Asterisk-Dev] Error   AdyaKamal
• [Asterisk-Dev] minumum linux hardware-less than 100 mb   Serkan Albayrak
• [Asterisk-Dev] min packets for small linux system   Serkan Albayrak
• [Asterisk-Dev] Adding extension information to queue log   Ilia Alshanetsky
• [Asterisk-Dev] UK Caller ID and cable company PSTN lines   Syed Amin
• [Asterisk-Dev] Good Explain about Virus in Voip.   Carlos Arnt
• [Asterisk-Dev] more on chan_capi   None Atall
• [Asterisk-Dev] RE: MYSQL application   Domjan Attila
• [Asterisk-Dev] RE: MYSQL application   Domjan Attila
• [Asterisk-Dev] Problem in compiling asterisk.c   Atif Azhar
• [Asterisk-Dev] Asterisk on the solaris/sparc platform?   Nicholas Bachmann
• [Asterisk-Dev] New Macro_Stdexten to fix a Sip device not login problem.   Virginia Batista
• [Asterisk-Dev] Asterisk on the solaris/sparc platform?   Brian Battle
• [Asterisk-Dev] Chan_VPB - Brazilian CallerID patch   Daniel Bichara
• [Asterisk-Dev] Chan_VPB - Brazilian CallerID patch   Daniel Bichara
• [Asterisk-Dev] Snom Programmable button Mini Howto and ring statepatch   Andre Bierwirth
• [Asterisk-Dev] minumum linux hardware-less than 100 mb   Greg Boehnlein
• [Asterisk-Dev] minumum linux hardware-less than 100 mb   Greg Boehnlein
• [Asterisk-Dev] Setting "Digest realm"   Greg Boehnlein
• [Asterisk-Dev] Setting "Digest realm"   Greg Boehnlein
• [Asterisk-Dev] Snom Programmable button Mini Howto and ring state patch   Greg Boehnlein
• [Asterisk-Dev] SIP bug and Polycom phones   Karl Brose
• [Asterisk-Dev] Network Crashed when we try to start TDMoE   Paul Cadach
• [Asterisk-Dev] How to use more PCI Cards with FXO/FXS in casethat TDMoE doesn't work   Paul Cadach
• [Asterisk-Dev] How to use more PCI Cards with FXO/FXS in case that TDMoE doesn't work   Paul Cadach
• [Asterisk-Dev] Agent problems with meetme::bug??   Navnit Chachan
• [Asterisk-Dev] park orbits application?   Dr. Michael J. Chudobiak
• [Asterisk-Dev] bounty: snom call parking   Dr. Michael J. Chudobiak
• [Asterisk-Dev] Snom Programmable button Mini Howto and ring state patch   Dr. Michael J. Chudobiak
• [Asterisk-Dev] bizarre registration behavior   Thomas B. Clark
• [Asterisk-Dev] UK Caller ID and cable company PSTN lines   Conroy, Lawrence (SMTP)
• [Asterisk-Dev] MacOS v10.3.5 Compile Problem   Conroy, Lawrence (SMTP)
• [Asterisk-Dev] properly replacing incominglimit/outgoinglimit   Ryan Courtnage
• [Asterisk-Dev] voicemail message number limits   Steven Critchfield
• [Asterisk-Dev] RE: MYSQL application   Steven Critchfield
• [Asterisk-Dev] RE: MYSQL application   Steven Critchfield
• [Asterisk-Dev] RE: MYSQL application   Steven Critchfield
• [Asterisk-Dev] query   Steven Critchfield
• [Asterisk-Dev] Adding extension information to queue log   Steven Critchfield
• [Asterisk-Dev] Error   Steven Critchfield
• [Asterisk-Dev] Asterisk channel variable access: a proposal   Brian Cuthie
• [Asterisk-Dev] sip hang when phone hangup :-(   Jan Czmok
• [Asterisk-Dev] Zaptel and Fedora Core 2 Works   Dan Davis
• [Asterisk-Dev] Pasive channels monitoring   Dan Davis
• [Asterisk-Dev] ast_bridge_call() warning   Eran Drory
• [Asterisk-Dev] VoIP SPAM, what's next ?   Duane
• [Asterisk-Dev] VoIP SPAM, what's next ?   Duane
• [Asterisk-Dev] internationalized DISA   Duane
• [Asterisk-Dev] mailbox number storage   Duane
• [Asterisk-Dev] PRI: Zap channel answers before get a CONNECT   Danilo Lotina F.
• [Asterisk-Dev] Channels variables from dial app.   Danilo Lotina F.
• [Asterisk-Dev] PRI: Zap channel answers before get a CONNECT   Danilo Lotina F.
• [Asterisk-Dev] Channels variables from dial app.   Danilo Lotina F.
• [Asterisk-Dev] RE: MYSQL application   Constantine Filin
• [Asterisk-Dev] MacOS v10.3.5 Runtime Error   Chris Foster
• [Asterisk-Dev] Contract Developers need   Timothy R. Fox
• [Asterisk-Dev] Please help: Problem with stuttering on TE410P...   Claus Futtrup
• [Asterisk-Dev] what's the difference between ast_waitstream and ast_waitstream_full   Rob Gagnon
• [Asterisk-Dev] problem with asterisk HEAD on RedHat 9.0   Rob Gagnon
• [Asterisk-Dev] Good Explain about Virus in Voip.   Rob Gagnon
• [Asterisk-Dev] query   Saurabh Garg
• [Asterisk-Dev] Query regading new system setup   Saurabh Garg
• [Asterisk-Dev] IAX >> FXO ?   Gary
• [Asterisk-Dev] MacOS v10.3.5 Compile Problem   Glenn
• [Asterisk-Dev] MacOS v10.3.5 Runtime Error   Glenn
• [Asterisk-Dev] MacOS v10.3.5 Compile Problem   Glenn
• [Asterisk-Dev] MacOS v10.3.5 Runtime Error   Glenn
• [Asterisk-Dev] Error   Bob Goddard
• [Asterisk-Dev] RE: [Asterisk-cvs] asterisk CREDITS,1.22,1.23   Adam Goryachev
• [Asterisk-Dev] Asterisk channel variable access: a proposal   Adam Goryachev
• [Asterisk-Dev] HELP with HFC Card, please   Sascha Growe
• [Asterisk-Dev] Missing in action: call timer stats?   Nicolas Gudino
• [Asterisk-Dev] Asterisk channel variable access: a proposal   Nicolas Gudino
• [Asterisk-Dev] UK Caller ID and cable company PSTN lines   David Gurr
• [Asterisk-Dev] UK Caller ID and cable company PSTN lines   David Gurr
• [Asterisk-Dev] Dial "S" option doesn't cancel timeout   David Gurr
• [Asterisk-Dev] Asterisk Receptionist Beta (Demo)   Kyle Hagan
• [Asterisk-Dev] DTMF and Broadvoice   Jeremy Hall
• [Asterisk-Dev] DTMF and Broadvoice   Jeremy Hall
• [Asterisk-Dev] DTMF and Broadvoice   Jeremy Hall
• [Asterisk-Dev] Compiling zaptel breaks on linux 2.4.27.   Adam E. Hampton
• [Asterisk-Dev] very simple ODBC application   Robert Hanzlik
• [Asterisk-Dev] [LONG] Call holding and native bridging with ast_h323   Kiel Hedjam
• [Asterisk-Dev] [LONG?] Impossible auto Dial with chan H323   Kiel Hedjam
• [Asterisk-Dev] [LONG?] Impossible auto Dial with chan H323   Kiel Hedjam
• [Asterisk-Dev] DTMF and Broadvoice   Greg Hill
• [Asterisk-Dev] bizarre registration behavior   Greg Hill
• [Asterisk-Dev] Snom Programmable button Mini Howto and ring statepatch   David Hindmarsh
• [Asterisk-Dev] Andre Bierwirth's ring state patches for SNOM 200 programmable buttons   David Hinkle
• [Asterisk-Dev] Snom Programmable button Mini Howto and ring state patch   David Hinkle
• [Asterisk-Dev] Snom Programmable button Mini Howto and ringstatepatch   David Hinkle
• [Asterisk-Dev] Snom Programmable button Mini Howto and ringstate patch   David Hinkle
• FW: [Asterisk-Dev] Snom Programmable button Mini Howto andringstate patch   David Hinkle
• [Asterisk-Dev] Snom Programmable button Mini Howto and ringstatepatch   David Hinkle
• [Asterisk-Dev] Asterisks   Huddleston, Robert
• [Asterisk-Dev] Error   Huddleston, Robert
• [Asterisk-Dev] IAX/SIP listener ???   Jan Hulala
• [Asterisk-Dev] International callback service with Asterisk   Jan Hulala
• [Asterisk-Dev] MYSQL application   Chris A. Icide
• [Asterisk-Dev] How to use more PCI Cards with FXO/FXS in case that TDMoE doesn't work   Chris A. Icide
• [Asterisk-Dev] Network Crashed when we try to start TDMoE   Chris A. Icide
• [Asterisk-Dev] res_data Issue   Robert Jackson
• [Asterisk-Dev] How to make RTP Packets NOT passing thru Asterisk?   Robert Jackson
• [Asterisk-Dev] DTMF and Broadvoice   Christopher Jacob
• [Asterisk-Dev] DTMF and Broadvoice   Christopher Jacob
• [Asterisk-Dev] DTMF and Broadvoice   Christopher Jacob
• [Asterisk-Dev] Transferring Calls   Christopher Jacob
• [Asterisk-Dev] Transferring Calls   Christopher Jacob
• [Asterisk-Dev] bugs.digium.com ...and what or who is "housekeeping" ????   Olle E. Johansson
• [Asterisk-Dev] Setting "Digest realm"   Olle E. Johansson
• [Asterisk-Dev] low level chan sip call control option   Olle E. Johansson
• [Asterisk-Dev] Patch to support an "outgoing proxy" on SIP register   Olle E. Johansson
• [Asterisk-Dev] DTMF and Broadvoice   James Jones
• [Asterisk-Dev] minumum linux hardware-less than 100 mb   Senad Jordanovic
• [Asterisk-Dev] minumum linux hardware-less than 100 mb   Senad Jordanovic
• [Asterisk-Dev] How to use more PCI Cards with FXO/FXS in case that TDMoE doesn't work   Klaus-Peter Junghanns
• [Asterisk-Dev] How to use more PCI Cards with FXO/FXS in case that TDMoE doesn't work   Klaus-Peter Junghanns
• [Asterisk-Dev] CVS HEAD (20040807) jitter buffer questions   Steve Kann
• [Asterisk-Dev] minumum linux hardware-less than 100 mb   Steve Kennedy
• [Asterisk-Dev] RE: MYSQL application   Andrew Kohlsmith
• [Asterisk-Dev] RE: MYSQL application   Andrew Kohlsmith
• [Asterisk-Dev] minumum linux hardware-less than 100 mb   Andrew Kohlsmith
• [Asterisk-Dev] minumum linux hardware-less than 100 mb   Andrew Kohlsmith
• [Asterisk-Dev] minumum linux hardware-less than 100 mb   Andrew Kohlsmith
• [Asterisk-Dev] DTMF and Broadvoice   Andrew Kohlsmith
• [Asterisk-Dev] CVS HEAD (20040807) jitter buffer questions   Andrew Kohlsmith
• [Asterisk-Dev] more iax2 weirdness (trunking)   Andrew Kohlsmith
• [Asterisk-Dev] CVS HEAD (20040807) jitter buffer questions   Andrew Kohlsmith
• [Asterisk-Dev] Asterisk channel variable access: a proposal   Andrew Kohlsmith
• [Asterisk-Dev] How to use more PCI Cards with FXO/FXS in case that TDMoE doesn't work   Andrew Kohlsmith
• [Asterisk-Dev] Asterisk channel variable access: a proposal   Andrew Kohlsmith
• [Asterisk-Dev] Asterisk channel variable access: a proposal   Andrew Kohlsmith
• [Asterisk-Dev] Pasive channels monitoring   Andrew Kohlsmith
• [Asterisk-Dev] __use_ast_pthread_create_instead__   Ben Kramer
• [Asterisk-Dev] __use_ast_pthread_create_instead__   Ben Kramer
• [Asterisk-Dev] __use_ast_pthread_create_instead__   Ben Kramer
• [Asterisk-Dev] __use_ast_pthread_create_instead__   Ben Kramer
• [Asterisk-Dev] problem with asterisk HEAD on RedHat 9.0   Ben Kramer
• [Asterisk-Dev] problem with asterisk HEAD on RedHat 9.0   Ben Kramer
• [Asterisk-Dev] Interesting fix on CVS   Ben Kramer
• [Asterisk-Dev] voicemail message number limits   Hemant Kumar
• [Asterisk-Dev] Problems with wcfxs.o   Tilghman Lesher
• [Asterisk-Dev] GET VARIABLE with AGI   Tilghman Lesher
• [Asterisk-Dev] Adding extension information to queue log   Tilghman Lesher
• [Asterisk-Dev] Using the return codes from extension.conf commands/applications   Tilghman Lesher
• [Asterisk-Dev] Transferring Calls   Tilghman Lesher
• [Asterisk-Dev] custom programming   Tilghman Lesher
• [Asterisk-Dev] International callback service with Asterisk   Tilghman Lesher
• [Asterisk-Dev] International callback service with Asterisk   Tilghman Lesher
• [Asterisk-Dev] International callback service with Asterisk   Tilghman Lesher
• [Asterisk-Dev] Asterisk channel variable access: a proposal   Tilghman Lesher
• [Asterisk-Dev] International callback service with Asterisk   Tilghman Lesher
• [Asterisk-Dev] Asterisk channel variable access: a proposal   Tilghman Lesher
• [Asterisk-Dev] Snom Programmable button Mini Howto and ring state patch   Tilghman Lesher
• [Asterisk-Dev] Asterisk on Solaris   Simon Lockhart
• [Asterisk-Dev] Asterisk channel variable access: a proposal   Alexander Lopez
• [Asterisk-Dev] Problem in compiling asterisk.c   Sunrise Ltd
• [Asterisk-Dev] My apologies for the broken threads   Sunrise Ltd
• [Asterisk-Dev] Assistance with localisation   Sunrise Ltd
• [Asterisk-Dev] Thanks to everybody who helped with translations   Sunrise Ltd
• [Asterisk-Dev] VoIP SPAM, what's next ?   Sunrise Ltd
• [Asterisk-Dev] MacOS v10.3.5 Compile Problem   Sunrise Ltd
• [Asterisk-Dev] __use_ast_pthread_create_instead__   Jolan Luff
• [Asterisk-Dev] __use_ast_pthread_create_instead__   Jolan Luff
• [Asterisk-Dev] __use_ast_pthread_create_instead__   Jolan Luff
• [Asterisk-Dev] voicemail message number limits   Richard Lyman
• [Asterisk-Dev] [PATCH] zaptel Makefile cleanup   Leif Madsen
• [Asterisk-Dev] Contract Developers need   Leif Madsen
• [Asterisk-Dev] DTMF and Broadvoice   Paul Mahler
• [Asterisk-Dev] Debug info over Manager API   C. Maj
• [Asterisk-Dev] Integration of AGI and Management API   C. Maj
• [Asterisk-Dev] International callback service with Asterisk   Paulo Mannheimer
• [Asterisk-Dev] International callback service with Asterisk   Paulo Mannheimer
• [Asterisk-Dev] Chan_VPB - Brazilian CallerID patch   Paulo Mannheimer
• [Asterisk-Dev] International callback service with Asterisk   Paulo H. Mannheimer
• [Asterisk-Dev] i->controllers   Maurizio Marini
• [Asterisk-Dev] more on chan_capi   Maurizio Marini
• [Asterisk-Dev] more on chan_capi   Maurizio Marini
• [Asterisk-Dev] more on chan_capi   Maurizio Marini
• [Asterisk-Dev] voicemail message number limits   Steve McMahon
• [Asterisk-Dev] End of public smear campaign by pimpsoft   Jeremy McNamara
• [Asterisk-Dev] What happened to #asterisk on irc.freenode.net   Jeremy McNamara
• [Asterisk-Dev] Commercial deployments of Asterix   Jeremy McNamara
• [Asterisk-Dev] Chan_VPB - Brazilian CallerID patch   Jorge Merlino
• [Asterisk-Dev] VoIP SPAM, what's next ?   Bill Michaelson
• [Asterisk-Dev] ztplay. How do I talk directly to zap channels?   Ben Miller
• [Asterisk-Dev] Some Queries   Mayank Mishra
• [Asterisk-Dev] Commercial deployments of Asterix   Mayank Mishra
• [Asterisk-Dev] MWI work around for Call Manager 4.0 (kinda)   Chris Modesitt
• [Asterisk-Dev] VoIP SPAM, what's next ?   Bill Moran
• [Asterisk-Dev] Re: Problem in compiling asterisk.c   Tony Mountifield
• [Asterisk-Dev] Re: what's the difference between ast_waitstream and ast_waitstream_full   Tony Mountifield
• [Asterisk-Dev] Re: SIP - asterisk does not handle re-INVITEs correctly   Tony Mountifield
• [Asterisk-Dev] Re: asterisk manager.c,1.67,1.68   Tony Mountifield
• [Asterisk-Dev] Debug info over Manager API   Ivan Myrvold
• [Asterisk-Dev] Debug info over Manager API   Ivan Myrvold
• [Asterisk-Dev] minumum linux hardware-less than 100 mb   Miroslav Nachev
• [Asterisk-Dev] Entry point & Minimal requirements for Linux Device Driver and connection with Asterisk IP PBX   Miroslav Nachev
• [Asterisk-Dev] Entry point & Minimal requirements for Linux Device Driver and connection with Asterisk IP PBX   Miroslav Nachev
• [Asterisk-Dev] Who knows where have open source SIP phone projects in windows platform   Miroslav Nachev
• [Asterisk-Dev] How to use more PCI Cards with FXO/FXS in case that TDMoE doesn't work   Miroslav Nachev
• [Asterisk-Dev] Network Crashed when we try to start TDMoE   Miroslav Nachev
• [Asterisk-Dev] How to use more PCI Cards with FXO/FXS in case that TDMoE doesn't work   Miroslav Nachev
• [Asterisk-Dev] DTMF and Broadvoice   Darren Nickerson
• [Asterisk-Dev] Banned from the channel?   Peter Nixon
• [Asterisk-Dev] RE: [Asterisk-cvs] asterisk CREDITS,1.22,1.23   Florian Overkamp
• [Asterisk-Dev] RE: [Asterisk-cvs] asterisk CREDITS,1.22,1.23   Florian Overkamp
• [Asterisk-Dev] SIP REFER transferring   Florian Overkamp
• [Asterisk-Dev] Asterisk channel variable access: a proposal   Florian Overkamp
• [Asterisk-Dev] Asterisk channel variable access: a proposal   Florian Overkamp
• [Asterisk-Dev] Support for Brazil's market   Marcelo Pacheco
• [Asterisk-Dev] How to use more PCI Cards with FXO/FXS in case that TDMoE doesn't work   Marcelo Pacheco
• [Asterisk-Dev] GET VARIABLE with AGI   Arnaud Pignard
• [Asterisk-Dev] Integration of AGI and Management API   David Pollak
• [Asterisk-Dev] Integration of AGI and Management API   David Pollak
• [Asterisk-Dev] Integration of AGI and Management API   David Pollak
• [Asterisk-Dev] Integration of AGI and Management API   David Pollak
• [Asterisk-Dev] IAX/SIP listener ???   David Pollak
• [Asterisk-Dev] Asterisk channel variable access: a proposal   David Pollak
• [Asterisk-Dev] Asterisk channel variable access: a proposal   David Pollak
• [Asterisk-Dev] [PATCH] zaptel Makefile cleanup   Steven Pritchard
• [Asterisk-Dev] Transferring Calls   Michael Procter
• [Asterisk-Dev] ATA 186 is not getting registered to Asterisk server   Rajeev
• [Asterisk-Dev] ATA 186 connected phone is not ringing   Rajeev
• [Asterisk-Dev] ATA 186 connected phone is not ringing (Rajeev) - Solved   Rajeev
• [Asterisk-Dev] VoIP SPAM, what's next ?   Soren Rathje
• [Asterisk-Dev] VoIP SPAM, what's next ?   Soren Rathje
• [Asterisk-Dev] VoIP SPAM, what's next ?   Soren Rathje
• [Asterisk-Dev] Support for Brazil's market   Soren Rathje
• [Asterisk-Dev] Change in #define ADSI_KEY_APPS   Seth Remington
• [Asterisk-Dev] HELP: BYE-request not sent to SIP-peer   Josh Roberson
• [Asterisk-Dev] Error   Josh Roberson
• [Asterisk-Dev] bizarre registration behavior   Wolfgang S. Rupprecht
• [Asterisk-Dev] Asterisk - sip communicator problem behind NAT   Partha Sarathi
• [Asterisk-Dev] Try new patch (ZAPTEL/PRI)   Partha Sarathi
• [Asterisk-Dev] How to make RTP Packets NOT passing thru Asterisk?   Partha Sarathi
• [Asterisk-Dev] Some Queries   Schaefer, Mark
• [Asterisk-Dev] MYSQL application   Gunnar Schaller
• [Asterisk-Dev] start answer end in sql cdrs   Holger Schurig
• [Asterisk-Dev] MYSQL application   Holger Schurig
• [Asterisk-Dev] Problem in compiling asterisk.c   Holger Schurig
• [Asterisk-Dev] Problem in compiling asterisk.c   Holger Schurig
• [Asterisk-Dev] RE: MYSQL application   Holger Schurig
• [Asterisk-Dev] query   Holger Schurig
• [Asterisk-Dev] min packets for small linux system   Holger Schurig
• [Asterisk-Dev] Asterisk - sip communicator problem behind NAT   Holger Schurig
• [Asterisk-Dev] Asterisk channel variable access: a proposal   Holger Schurig
• [Asterisk-Dev] Asterisk channel variable access: a proposal   Holger Schurig
• [Asterisk-Dev] Asterisk channel variable access: a proposal   Holger Schurig
• [Asterisk-Dev] Commercial deployments of Asterix   Holger Schurig
• [Asterisk-Dev] Asterisk channel variable access: a proposal   Holger Schurig
• [Asterisk-Dev] How do I use app_test?   Richard Scobie
• [Asterisk-Dev] SIP Response 403   Mitul Sen
• [Asterisk-Dev] Using the return codes from extension.conf commands/applications   Andreas Sikkema
• [Asterisk-Dev] Preventing Asterisk from codec conversion when reinviting   Andreas Sikkema
• [Asterisk-Dev] SQL extensions   Matthew Simpson
• [Asterisk-Dev] End of public smear campaign by pimpsoft   Derek Smithies
• [Asterisk-Dev] __use_ast_pthread_create_instead__   Derek Smithies
• [Asterisk-Dev] AstriCon Reminder: Please register today   Steven Sokol
• [Asterisk-Dev] New List Server   Mark Spencer
• [Asterisk-Dev] UK Caller ID and cable company PSTN lines   Chris Stenton
• [Asterisk-Dev] VoIP SPAM, what's next ?   Scott Stingel
• [Asterisk-Dev] Asterisks   Scott Stingel
• [Asterisk-Dev] PRI: Zap channel answers before get a CONNECT   Peter Svensson
• [Asterisk-Dev] How to use more PCI Cards with FXO/FXS in case that TDMoE doesn't work   Peter Svensson
• [Asterisk-Dev] i4l (isdn) question / remarks ?   The Sysop
• [Asterisk-Dev] voicemail message number limits   Steve Szmidt
• [Asterisk-Dev] voicemail message number limits   Steve Szmidt
• [Asterisk-Dev] SIP Response 403   Steve Szmidt
• [Asterisk-Dev] VoIP SPAM, what's next ?   Steve Szmidt
• [Asterisk-Dev] International callback service with Asterisk   Steve Szmidt
• [Asterisk-Dev] Error   Steve Szmidt
• [Asterisk-Dev] Error   Steve Szmidt
• [Asterisk-Dev] Debug info over Manager API   TC
• [Asterisk-Dev] Pasive channels monitoring   TC
• [Asterisk-Dev] Pasive channels monitoring   TC
• [Asterisk-Dev] minumum linux hardware-less than 100 mb   James H. Thompson
• [Asterisk-Dev] using round-robin dns for sip registrations   Mark Thompson
• [Asterisk-Dev] X100P and polarity reversal   Alexander Timoshenko
• [Asterisk-Dev] minumum linux hardware-less than 100 mb   John Todd
• [Asterisk-Dev] Asterisk channel variable access: a proposal   John Todd
• [Asterisk-Dev] Asterisk channel variable access: a proposal   John Todd
• [Asterisk-Dev] Asterisk channel variable access: a proposal   John Todd
• [Asterisk-Dev] Asterisk channel variable access: a proposal   John Todd
• [Asterisk-Dev] Asterisk channel variable access: a proposal   John Todd
• [Asterisk-Dev] RE: MYSQL application   John Turner
• [Asterisk-Dev] Problems with wcfxs.o   Warren Volz
• [Asterisk-Dev] Problems with wcfxs.o   Warren Volz
• [Asterisk-Dev] asterisk.c,1.110,1.111   Kevin Walsh
• [Asterisk-Dev] Bogus (d) channels   Kevin Walsh
• [Asterisk-Dev] RE: [Asterisk-cvs] asterisk CREDITS,1.22,1.23   Kevin Walsh
• [Asterisk-Dev] RE: MYSQL application   Kevin Walsh
• [Asterisk-Dev] SIP Response 403   Kevin Walsh
• [Asterisk-Dev] VoIP SPAM, what's next ?   Kevin Walsh
• [Asterisk-Dev] X100P and polarity reversal   Kevin Walsh
• [Asterisk-Dev] SIP or IAX provider for Madison WI?   John Warren
• [Asterisk-Dev] Asterisk GUIs at Astricon   John Warren
• [Asterisk-Dev] custom programming   John Warren
• [Asterisk-Dev] custom programming   John Warren
• [Asterisk-Dev] Error   Eric Wieling
• [Asterisk-Dev] Pasive channels monitoring   Guillermo Winkler
• [Asterisk-Dev] Pasive channels monitoring   Guillermo Winkler
• [Asterisk-Dev] what's the difference between ast_waitstream and ast_waitstream_full   YULONG ZHOU
• [Asterisk-Dev] HELP: BYE-request not sent to SIP-peer   Roland Zagler
• [Asterisk-Dev] Error when streaming file in AGI   Roland Zagler
• [Asterisk-Dev] sdp: asterisk responds with codecs not in INVITE   Alex Zeffertt
• [Asterisk-Dev] SIP - asterisk does not handle re-INVITEs correctly   Alex Zeffertt
• [Asterisk-Dev] SIP - asterisk does not handle re-INVITEs correctly   Alex Zeffertt
• [Asterisk-Dev] SIP - asterisk does not handle re-INVITEs correctly   Alex Zeffertt
• [Asterisk-Dev] Patch to support an "outgoing proxy" on SIP register   Dietmar Zlabinger
• [Asterisk-Dev] Patch to support an "outgoing proxy" on SIP register   Dietmar Zlabinger
• [Asterisk-Dev] bizarre registration behavior   alex at pilosoft.com
• [Asterisk-Dev] How to use more PCI Cards with FXO/FXS in case that TDMoE doesn't work   alex at pilosoft.com
• [Asterisk-Dev] Asterisk channel variable access: a proposal   alex at pilosoft.com
• [Asterisk-Dev] new hardware integration   boris.vincent at mindspeed.com
• [Asterisk-Dev] Asterisks   t d
• [Asterisk-Dev] Error   dbruce at bananatel.com
• [Asterisk-Dev] voicemail message number limits   dking at pimpsoft.com
• [Asterisk-Dev] voicemail message number limits   dking at pimpsoft.com
• [Asterisk-Dev] End of public smear campaign by pimpsoft   dking at pimpsoft.com
• [Asterisk-Dev] Contract Developers need   olcay ergun
• [Asterisk-Dev] Astricon - call for help   joachim
• [Asterisk-Dev] how to support g.729 in ivr   joachim
• [Asterisk-Dev] VoIP SPAM, what's next ?   matt.riddell at sineapps.com
• [Asterisk-Dev] res_data Issue   matt.riddell at sineapps.com
• [Asterisk-Dev] Interesting fix on CVS   matt.riddell at sineapps.com
• [Asterisk-Dev] Integration of AGI and Management API   mattf
• [Asterisk-Dev] Integration of AGI and Management API   mattf
• [Asterisk-Dev] Integration of AGI and Management API   mattf
• [Asterisk-Dev] Integration of AGI and Management API   mattf
• [Asterisk-Dev] International callback service with Asterisk   mirza sahib
• [Asterisk-Dev] Commercial deployments of Asterix   mirza sahib
• [Asterisk-Dev] sip call setup   mark spowage
• [Asterisk-Dev] low level chan sip call control option   mark spowage
• [Asterisk-Dev] IAXy frame timestamps   steve at daviesfam.org
• [Asterisk-Dev] Forking the dialplan via Dial(LOCAL/...&LOCAL/...) (was Re: [Asterisk-Users] Dialplan question)   steve at daviesfam.org
• [Asterisk-Dev] UK Caller ID and cable company PSTN lines   steve at daviesfam.org
• [Asterisk-Dev] CVS HEAD (20040807) jitter buffer questions   steve at daviesfam.org
• [Asterisk-Dev] CVS HEAD (20040807) jitter buffer questions   steve at daviesfam.org
• [Asterisk-Dev] more iax2 weirdness (trunking)   steve at daviesfam.org
• [Asterisk-Dev] CVS HEAD (20040807) jitter buffer questions   steve at daviesfam.org
• [Asterisk-Dev] Asterisk Developers Wanted - NY   voipjobs at optonline.net
• [Asterisk-Dev] Who knows where have open source SIP phone projects in windows platform   chaye wala
• [Asterisk-Dev] What happened to #asterisk on irc.freenode.net   chaye wala
• [Asterisk-Dev] International callback service with Asterisk   chaye wala
• [Asterisk-Dev] International callback service with Asterisk   chaye wala
• [Asterisk-Dev] Try new patch (ZAPTEL/PRI)   moritz winterberg
• [Asterisk-Dev] Try new patch (ZAPTEL/PRI)   moritz winterberg
• [Asterisk-Dev] Try new patch (ZAPTEL/PRI)   moritz winterberg
• [Asterisk-Dev] Test new patch (ZAPTEL/PRI)   moritz winterberg
• [Asterisk-Dev] Who knows where have open source SIP phone projects in windows platform   xin yi

Last message date: Tue Aug 31 18:41:59 MST 2004
Archived on: Tue Sep 5 14:27:01 MST 2006

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