June 2004 Archives by date

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Starting: Tue Jun 1 00:19:37 MST 2004
Ending: Wed Jun 30 21:46:32 MST 2004
Messages: 460

• [Asterisk-Dev] HEAD - Advanced voicemail behaviour change   John Todd
• [Asterisk-Dev] HEAD - Advanced voicemail behaviour change   Rob Gagnon
• [Asterisk-Dev] HEAD - Advanced voicemail behaviour change   Paul Crick
• [Asterisk-Dev] HEAD - Advanced voicemail behaviour change   steve at daviesfam.org
• [Asterisk-Dev] quadBRI installation problem   Michael Sandee
• [Asterisk-Dev] quadBRI installation problem   GIBERT Frédéric
• [Asterisk-Dev] quadBRI installation problem   GIBERT Frédéric
• [Asterisk-Dev] quadBRI installation problem   GIBERT Frédéric
• [Asterisk-Dev] HEAD - Advanced voicemail behaviour change   Brian Cuthie
• [Asterisk-Dev] HEAD - Advanced voicemail behaviour change   denon
• [Asterisk-Dev] MGCP Clients   J C
• [Asterisk-Dev] HEAD - Advanced voicemail behaviour change   Jesse Janzer
• [Asterisk-Dev] Some (lack of) answers regarding the wakeup call application...   Rob Fugina
• [Asterisk-Dev] Some (lack of) answers regarding the wakeup ca ll application...   Luckcuck Nick-LCKN001
• [Asterisk-Dev] HEAD - Advanced voicemail behaviour change   Andrew Kohlsmith
• [Asterisk-Dev] MWI indications and AGI Voicemail   Brian Cuthie
• [Asterisk-Dev] Some (lack of) answers regarding the wakeup call application...   Eric Wieling
• [Asterisk-Dev] Record Application Problem   Pedro Bessa Goncalves
• [Asterisk-Dev] Creating An Asterisk Data Model   Robert Withrow
• [Asterisk-Dev] MWI indications and AGI Voicemail   Paul Crick
• [Asterisk-Dev] Re: [Core Asterisk 0001760]: Macros do not clear argument variables when calling other macros   Eric Wieling
• [Asterisk-Dev] Re: [Applications 0001758]: VoicemailMain2 does not respond in anticipated manner   steve at daviesfam.org
• [Asterisk-Dev] Re: [Applications 0001758]: VoicemailMain2 does not respond in anticipated manner   Eric Wieling
• [Asterisk-Dev] Re: [Applications 0001758]: VoicemailMain2 does not respond in anticipated manner   Fran Boon
• [Asterisk-Dev] ast_data code first public release ready for testing   Rob Gagnon
• [Asterisk-Dev] ast_data code first public release ready for testing   Joshua M. Thompson
• [Asterisk-Dev] Re: [Applications 0001758]: VoicemailMain2 doesnot respond in anticipated manner   Kevin Walsh
• [Asterisk-Dev] Re: [Applications 0001758]: VoicemailMain2 doesnot respond in anticipated manner   brian
• [Asterisk-Dev] ast_data code first public release ready fortesting   brian
• [Asterisk-Dev] Re: [Applications 0001758]: VoicemailMain2 does not respond in anticipated manner   steve at daviesfam.org
• [Asterisk-Dev] Problems with TE405P and generic hdlc   Patrick J. Conroy
• [Asterisk-Dev] zap call pickup problem - more details   Andrew Kohlsmith
• [Asterisk-Dev] ast_data code first public release ready fortesting   Greg Varga
• [Asterisk-Dev] Newbie Question   Murray Hooper
• [Asterisk-Dev] Newbie Question   Steven Critchfield
• [Asterisk-Dev] Newbie Question   Eric Wieling
• [Asterisk-Dev] Basic Conferencing from IAX2, other VoIP Channels   Steven Sokol
• [Asterisk-Dev] Problem: Sip/Zap (ONE WAY AUDIO) CVS 02-06-2004 10:00AM PST   Steve McMahon
• [Asterisk-Dev] MSI support   Ronaldo S Pereira
• [Asterisk-Dev] VoIP x PSTN Gateway   Ronaldo S Pereira
• [Asterisk-Dev] Basic Conferencing from IAX2, other VoIP Channels   Shannon Mitchell
• [Asterisk-Dev] VoIP x PSTN Gateway   Steven Sokol
• [Asterisk-Dev] VoIP x PSTN Gateway   Greg Varga
• [Asterisk-Dev] buglet with * and libpri????   Steven Critchfield
• [Asterisk-Dev] MSI support   Eric Wieling
• [Asterisk-Dev] Basic Conferencing from IAX2, other VoIP Channels   Nicolas Gudino
• [Asterisk-Dev] app_meetme crash   Jared Mauch
• [Asterisk-Dev] app_meetme crash   Rob Gagnon
• [Asterisk-Dev] app_meetme crash   Fabian Stelzer
• [Asterisk-Dev] app_meetme crash   Rob Gagnon
• [Asterisk-Dev] Intercom and Paging   James H. Thompson
• [Asterisk-Dev] app_meetme crash   Rob Gagnon
• [Asterisk-Dev] app_meetme crash   Rob Gagnon
• [Asterisk-Dev] Re: [Iaxclient-devel] Basic Conferencing from IAX2, other VoIP Channels   Steve Kann
• [Asterisk-Dev] RE: [Asterisk-cvs] asterisk/channels chan_zap.c,1.240,1.241   Kevin Walsh
• [Asterisk-Dev] Re: [Iaxclient-devel] Basic Conferencing from IAX2, other VoIP Channels   Adam Hart
• [Asterisk-Dev] Variable manipulation in extensions.conf   oi geli
• [Asterisk-Dev] Problems with TE405P and generic hdlc   Schaefer, Mark
• [Asterisk-Dev] XTunnels: freeware for SIP tunnelling   John Todd
• [Asterisk-Dev] Call authentication in Asterisk   Aram Ter-Martirosyan
• [Asterisk-Dev] HEAD - Advanced voicemail behaviour change   Mark Elkins
• [Asterisk-Dev] Call authentication in Asterisk   Apollon Koutlides
• [Asterisk-Dev] Call authentication in Asterisk   Fran Boon
• [Asterisk-Dev] HEAD - Advanced voicemail behaviour change   Chris Lee
• [Asterisk-Dev] HEAD - Advanced voicemail behaviour change   Steve Totaro
• [Asterisk-Dev] HEAD - Advanced voicemail behaviour change   Steven Kokinos
• [Asterisk-Dev] Call authentication in Asterisk   Aram Ter-Martirosyan
• [Asterisk-Dev] HEAD - Advanced voicemail behaviour change   asterisk at jdennis.net
• [Asterisk-Dev] Basic Conferencing from IAX2, other VoIP Channels   Steven Sokol
• [Asterisk-Dev] Re: Call authentication in Asterisk   Peter Nixon
• [Asterisk-Dev] zaptel wcusb will not load   Wichert Akkerman
• [Asterisk-Dev] Call authentication in Asterisk   Fran Boon
• [Asterisk-Dev] Re: zaptel wcusb will not load   Peter Nixon
• [Asterisk-Dev] New Channel - Serial + Sound Card -> GSM Mobile   Adam Goryachev
• [Asterisk-Dev] PATCH - Adds a set'able text string for each channel   Adam Goryachev
• [Asterisk-Dev] PATCH - Adds a set'able text string for each channel   James Golovich
• [Asterisk-Dev] Patch: Voicemail interruptions   Kevin Walsh
• [Asterisk-Dev] New Channel - Serial + Sound Card -> GSM Mobile   Dan
• [Asterisk-Dev] Re: zaptel wcusb will not load   Wichert Akkerman
• [Asterisk-Dev] Hardware question (wrong list?) LG GDK Digital phones   Steve Hanselman
• [Asterisk-Dev] Redirecting a channel to an app without the dial plan?   Steven Sokol
• [Asterisk-Dev] chan_sip2 outboundproxy   David Beckemeyer
• [Asterisk-Dev] Continuing dial plan after caller hangs up   John James
• [Asterisk-Dev] Continuing dial plan after caller hangs up   Eric Wieling
• [Asterisk-Dev] Continuing dial plan after caller hangs up   Steven Critchfield
• [Asterisk-Dev] Continuing dial plan after caller hangs up   Alric
• [Asterisk-Dev] Continuing dial plan after caller hangs up   John Todd
• [Asterisk-Dev] IAX Conferencing (3-way calling) Patch   Steven Sokol
• [Asterisk-Dev] chan_sip2 outboundproxy   Olle E. Johansson
• [Asterisk-Dev] Re: Creating An Asterisk Data Model   Peter Nixon
• [Asterisk-Dev] Re: Creating An Asterisk Data Model   Peter Nixon
• [Asterisk-Dev] Re: zaptel wcusb will not load   Peter Nixon
• [Asterisk-Dev] Continuing dial plan after caller hangs up   John James
• [Asterisk-Dev] Adding a dial option to update the DB on bridge   Steve Rodgers
• [Asterisk-Dev] Re: Creating An Asterisk Data Model   Paul Mahler
• [Asterisk-Dev] Re: Creating An Asterisk Data Model   Miroslav Nachev
• [Asterisk-Dev] Re: Creating An Asterisk Data Model   Miroslav Nachev
• [Asterisk-Dev] Re: Creating An Asterisk Data Model   Paul Mahler
• [Asterisk-Dev] Re: Re[2]: Re: Creating An Asterisk Data Model   Peter Nixon
• [Asterisk-Dev] Re: Creating An Asterisk Data Model   James H. Thompson
• [Asterisk-Dev] Re: Creating An Asterisk Data Model   Paul Mahler
• [Asterisk-Dev] Adding a dial option to update the DB on bridge   Steve Rodgers
• [Asterisk-Dev] Adding a dial option to update the DB on bridge   Tilghman Lesher
• [Asterisk-Dev] (Asterisk and A400) too much jitter with chan_h323   ooseghale at telecomlabs.net
• [Asterisk-Dev] load_module error with chan_oh323   ooseghale at telecomlabs.net
• [Asterisk-Dev] zaptel/wcusb fixes   Wichert Akkerman
• [Asterisk-Dev] Re: New Channel - Serial + Sound Card -> GSM Mobile   jbarre
• [Asterisk-Dev] Re: New Channel - Serial + Sound Card -> GSM Mobile   Dan
• [Asterisk-Dev] Adding a dial option to update the DB on bridge   hwstar at rodgers.sdcoxmail.com
• [Asterisk-Dev] (Asterisk and A400) too much jitter with chan_h323   Jeremy McNamara
• AW: [Asterisk-Dev] Re: New Channel - Serial + Sound Card -> GSM Mobile   Markku Korpi
• [Asterisk-Dev] zaptel/wcusb fixes   creslin at digium.com
• [Asterisk-Dev] zaptel/wcusb fixes   Wichert Akkerman
• [Asterisk-Dev] asterisk SIP registration   Kristopher Lalletti
• [Asterisk-Dev] Adding a dial option to update the DB on bridge   Steve Rodgers
• [Asterisk-Dev] asterisk SIP registration   Olle E. Johansson
• [Asterisk-Dev] STUN support   David Beckemeyer
• [Asterisk-Dev] Sub delimiter syntax options in extensions.conf   Nicolas Bougues
• [Asterisk-Dev] STUN support   George Petrov Georgiev-Rusiichev
• [Asterisk-Dev] STUN support   Fran Boon
• [Asterisk-Dev] Error on Chan_zap and prid_dchannel   george bush
• [Asterisk-Dev] enum app is rfc2916 compliant?   Gustavo García Bernardo
• [Asterisk-Dev] Presence, Barge-in, and auto-conferencing   John Todd
• [Asterisk-Dev] enum app is rfc2916 compliant?   Tony Hoyle
• [Asterisk-Dev] STUN support   Jeremy McNamara
• [Asterisk-Dev] STUN support   Olle E. Johansson
• [Asterisk-Dev] STUN support   Jeremy McNamara
• [Asterisk-Dev] STUN support   Karl Brose
• [Asterisk-Dev] STUN support   Chris Albertson
• [Asterisk-Dev] Re: STUN support   David Beckemeyer
• [Asterisk-Dev] Asterisk and DECT   steve at daviesfam.org
• [Asterisk-Dev] Asterisk SIP Problem   Pedro Bessa Goncalves
• [Asterisk-Dev] Asterisk SIP Problem   Adam Hart
• [Asterisk-Dev] Asterisk SIP Problem   Luis Mata
• [Asterisk-Dev] Support for SIP message 300 multiple choices   Umar Sear
• [Asterisk-Dev] Asterisk SIP Problem   Pedro Bessa Goncalves
• [Asterisk-Dev] Support for SIP message 300 multiple choices   Olle E. Johansson
• [Asterisk-Dev] Quicknet LineJack   Gustavo García Bernardo
• [Asterisk-Dev] Passing overlap digits from one pri-E1 to another   Peter Svensson
• [Asterisk-Dev] freebsd rtp.c resource not available errors   James H. Cloos Jr.
• [Asterisk-Dev] Support for SIP message 300 multiple choices (SONUS)   John Todd
• [Asterisk-Dev] Error installing Prepaid App   oi geli
• [Asterisk-Dev] OpenBSD-Patch for current CVS   Folke Ashberg
• [Asterisk-Dev] freebsd rtp.c resource not available errors   Dr. Rich Murphey
• [Asterisk-Dev] Reading channel variables in other channels   Steve Rodgers
• [Asterisk-Dev] OpenBSD-Patch for current CVS   Olle E. Johansson
• [Asterisk-Dev] freebsd rtp.c resource not available errors   Olle E. Johansson
• [Asterisk-Dev] Error installing Prepaid App   steve at daviesfam.org
• [Asterisk-Dev] Passing overlap digits from one pri-E1 to another   ePyron Felix Deierlein
• [Asterisk-Dev] Passing overlap digits from one pri-E1 to another   Peter Svensson
• AW: [Asterisk-Dev] Passing overlap digits from one pri-E1 to another   Thomas Haeger
• [Asterisk-Dev] Error installing Prepaid App   reseaux
• [Asterisk-Dev] Error installing Prepaid App   steve at daviesfam.org
• [Asterisk-Dev] Error installing Prepaid App   reseaux
• [Asterisk-Dev] Error installing Prepaid App   Soren Rathje
• [Asterisk-Dev] Error installing Prepaid App   reseaux
• [Asterisk-Dev] Error installing Prepaid App   reseaux
• [Asterisk-Dev] Error installing Prepaid App   reseaux
• AW: [Asterisk-Dev] Passing overlap digits from one pri-E1 to another   Peter Svensson
• [Asterisk-Dev] SIP question   GIBERT Frédéric
• [Asterisk-Dev] Error installing Prepaid App   Soren Rathje
• [Asterisk-Dev] SIP question   Olle E. Johansson
• [Asterisk-Dev] Passing overlap digits from one pri-E1 to another   Peter Svensson
• [Asterisk-Dev] Problem with E1   David Morillo
• [Asterisk-Dev] SIP question   Steve McMahon
• [Asterisk-Dev] Problem with X100P & Local Telco   Steve McMahon
• [Asterisk-Dev] Problem with X100P & Local Telco   Steven Critchfield
• [Asterisk-Dev] SIP External IP Address Issue   Steve McMahon
• [Asterisk-Dev] Problems with ring debounce on TDM400P FXO channel   Ron Frederick
• [Asterisk-Dev] Problem with E1   Scott Stingel
• [Asterisk-Dev] Reaching variables from the other side of the call   Olle E. Johansson
• [Asterisk-Dev] Bluetooth handsfree channel   Andreas Bayer
• [Asterisk-Dev] Reaching variables from the other side of the call   Michael Manousos
• [Asterisk-Dev] Problems with ring debounce on TDM400P FXO channel   Ryan Courtnage
• [Asterisk-Dev] Problems with ring debounce on TDM400P FXO channel   Kristopher Lalletti
• [Asterisk-Dev] Problems with ring debounce on TDM400P FXO channel   Rich Adamson
• [Asterisk-Dev] Problems with ring debounce on TDM400P FXO channel   Gerald Short
• [Asterisk-Dev] Centralized voicemail   Soren Rathje
• [Asterisk-Dev] Centralized voicemail   Rob Gagnon
• [Asterisk-Dev] Problems with ring debounce on TDM400P FXO channel   Rich Adamson
• [Asterisk-Dev] Problems with ring debounce on TDM400P FXO channel   Rich Adamson
• [Asterisk-Dev] E option in meetme.conf   Ryan Courtnage
• [Asterisk-Dev] E option in meetme.conf   Daniel Daley
• [Asterisk-Dev] E option in meetme.conf   Ryan Courtnage
• [Asterisk-Dev] Overlap digits on pri-E1 etc in Sweden   Peter Svensson
• [Asterisk-Dev] Overlap digits on pri-E1 etc in Sweden   brian k. west
• [Asterisk-Dev] AGI Scripting encountering "Ouch ... error while writing audio data: : Broken pipe " with Music On Hold disabled.   Kraig Beahn
• [Asterisk-Dev] Bluetooth handsfree channel   jbarre
• [Asterisk-Dev] Bluetooth handsfree channel   Dan
• IAX bluetooth [GSM mobile] gateway - was Re: [Asterisk-Dev] Bluetooth handsfree channel   jbarre
• IAX bluetooth [GSM mobile] gateway - was Re: [Asterisk-Dev] Bluetooth handsfree channel   Dan
• IAX bluetooth [GSM mobile] gateway - was Re: [Asterisk-Dev] Bluetooth handsfree channel   jbarre
• IAX bluetooth [GSM mobile] gateway - was Re: [Asterisk-Dev] Bluetooth handsfree channel   Rainer Jochem
• IAX bluetooth [GSM mobile] gateway - was Re: [Asterisk-Dev] Bluetooth handsfree channel   Dan
• IAX bluetooth [GSM mobile] gateway - was Re: [Asterisk-Dev] Bluetooth handsfree channel   Dan
• [Asterisk-Dev] Bluetooth handsfree channel   Andreas Bayer
• [Asterisk-Dev] Overlap digits on pri-E1 etc in Sweden   Peter Svensson
• IAX bluetooth [GSM mobile] gateway - was Re: [Asterisk-Dev] Bluetooth handsfree channel   Jon Radon
• [Asterisk-Dev] AGI Scripting encountering "Ouch ... error while writing audio data: : Broken pipe " with Music On Hold disabled.   Steven Critchfield
• IAX bluetooth [GSM mobile] gateway - was Re: [Asterisk-Dev] Bluetooth handsfree channel   Dan
• [Asterisk-Dev] Centralized voicemail   Soren Rathje
• IAX bluetooth [GSM mobile] gateway - was Re: [Asterisk-Dev] Bluetooth handsfree channel   Andreas Bayer
• [Asterisk-Dev] Centralized voicemail   Rob Gagnon
• [Asterisk-Dev] IRQ misses and crackles during disk activity   John Matthews
• [Asterisk-Dev] IRQ misses and crackles during disk activity   Steven Critchfield
• [Asterisk-Dev] handsfree application and asterisk   Andreas Bayer
• [Asterisk-Dev] Centralized voicemail   Rob Gagnon
• [Asterisk-Dev] IRQ misses and crackles during disk activity   Peter Svensson
• [Asterisk-Dev] Remote-Party-ID patch   Niklas Ögren
• [Asterisk-Dev] Making libiax2 speak TCP (through udp tunnelin g)   Whisker, Peter
• [Asterisk-Dev] GR-303 support?   Kevin P. Fleming
• [Asterisk-Dev] Centralized voicemail   Rob Gagnon
• [Asterisk-Dev] GR-303 support?   Dave Weis
• [Asterisk-Dev] GR-303 support?   Ray Burkholder
• [Asterisk-Dev] GR-303 support?   Dave Weis
• [Asterisk-Dev] GR-303 support?   Kevin P. Fleming
• [Asterisk-Dev] Centralized voicemail   Soren Rathje
• [Asterisk-Dev] GR-303 support?   Dave Weis
• [Asterisk-Dev] DTMF digits relay with Quintum to Oh323 version 0.5.10   Luis Mata
• [Asterisk-Dev] Centralized voicemail   Rob Gagnon
• [Asterisk-Dev] GR-303 support?   Kevin P. Fleming
• [Asterisk-Dev] Re: IRQ misses and crackles during disk activity   John Matthews
• [Asterisk-Dev] Centralized voicemail   Soren Rathje
• [Asterisk-Dev] Instrumenting IAX?   steve at daviesfam.org
• [Asterisk-Dev] Instrumenting IAX?   John Todd
• [Asterisk-Dev] Which one is the better Prepaid app   oi geli
• [Asterisk-Dev] ast_data, mysql, md5secret   Gunnar Schaller
• [Asterisk-Dev] ast_data, mysql, md5secret   Gunnar Schaller
• [Asterisk-Dev] Building Zaptel on MIPS   Christian Hecimovic
• [Asterisk-Dev] Bug or feature !   usedcanon
• [Asterisk-Dev] ast_data, mysql, md5secret   Rob Gagnon
• [Asterisk-Dev] Help needed regarding E1 card for back-to-back connectivity   sthiti
• [Asterisk-Dev] ast_data, mysql, md5secret   Gunnar Schaller
• [Asterisk-Dev] ast_data, mysql, md5secret   Gunnar Schaller
• [Asterisk-Dev] RTSP channel   Neutel Rodrigues
• [Asterisk-Dev] app_dial - Feature or bug ?   Umar Sear
• [Asterisk-Dev] ast_data, mysql, md5secret   Rob Gagnon
• [Asterisk-Dev] Eicon diva server 2M problem   Scannachiappolo
• [Asterisk-Dev] RE: [Asterisk-Users] Simplified Voicemail app / keeping peace withcohabitants   Brad Bergman
• [Asterisk-Dev] cdr_odbc and SQLite   David Creemer
• [Asterisk-Dev] cdr_odbc and SQLite   brian
• [Asterisk-Dev] appl Dial variables behavior   Danilo Lotina F.
• [Asterisk-Dev] Modified Prepaid App Error   oi geli
• [Asterisk-Dev] Channel Status   Ryan Courtnage
• [Asterisk-Dev] IAX2 no compatible codecs   Jason Penton
• [Asterisk-Dev] Eicon diva server 2M problem   Scannachiappolo
• [Asterisk-Dev] Re: cdr_odbc and SQLite   David Creemer
• [Asterisk-Dev] Help Compiling astman on Redhat FC1   Dorian Logan
• [Asterisk-Dev] Eicon diva server 2M problem   Helmut Adams
• [Asterisk-Dev] app_dial - Feature or bug ?   Umar Sear
• [Asterisk-Dev] problems with dtmf detection on X100P   Thomas Haeger
• [Asterisk-Dev] cvs [server aborted]: could not find desired version ...   Dr. Rich Murphey
• [Asterisk-Dev] PTHREAD_MUTEX_RECURSIVE   Jim Gottlieb
• [Asterisk-Dev] Re: PTHREAD_MUTEX_RECURSIVE   Jim Gottlieb
• [Asterisk-Dev] PTHREAD_MUTEX_RECURSIVE   Brian K. West
• [Asterisk-Dev] PTHREAD_MUTEX_RECURSIVE   Sam Bingner
• [Asterisk-Dev] Re: cvs [server aborted]: could not find desired version ...   Tony Mountifield
• [Asterisk-Dev] Accept patch to make iax2's calc_timestamp suspicious about the ast_frame.delivery value?   steve at daviesfam.org
• [Asterisk-Dev] Simple kde frontend for astersisk   Andreas Bayer
• [Asterisk-Dev] Accept patch to make iax2's calc_timestamp suspicious about the ast_frame.delivery value?   Andrew Kohlsmith
• [Asterisk-Dev] Bluetooth channel   Andreas Bayer
• [Asterisk-Dev] How are IAX2 timestamps supposed to work?   steve at daviesfam.org
• [Asterisk-Dev] Bluetooth channel   Dan
• [Asterisk-Dev] Accept patch to make iax2's calc_timestamp suspicious about the ast_frame.delivery value?   Steven Critchfield
• [Asterisk-Dev] Simple kde frontend for astersisk   Steven Critchfield
• [Asterisk-Dev] Simple kde frontend for astersisk   Andreas Bayer
• [Asterisk-Dev] Streaming File in Conversation   asterisk-dev at aconectarse.com
• [Asterisk-Dev] Simple kde frontend for astersisk   Steven Critchfield
• [Asterisk-Dev] Simple kde frontend for astersisk   Andreas Bayer
• [Asterisk-Dev] WCFXS module "robust" parameter   Richard Scobie
• [Asterisk-Dev] PTHREAD_MUTEX_RECURSIVE   Dr. Rich Murphey
• [Asterisk-Dev] #asterisk is +r now, meaning register your nick with nickserv   Brian K. West
• [Asterisk-Dev] PTHREAD_MUTEX_RECURSIVE   Alfred R. Nurnberger
• [Asterisk-Dev] PTHREAD_MUTEX_RECURSIVE   Dr. Rich Murphey
• [Asterisk-Dev] PTHREAD_MUTEX_RECURSIVE   Dr. Rich Murphey
• [Asterisk-Dev] Status of autoconf integration?   Robert Bedell
• [Asterisk-Dev] Status of autoconf integration?   Jeremy McNamara
• [Asterisk-Dev] PTHREAD_MUTEX_RECURSIVE   steve at daviesfam.org
• [Asterisk-Dev] SendURl -> getting the url in iax client   Navnit Chachan
• [Asterisk-Dev] PTHREAD_MUTEX_RECURSIVE   Dr. Rich Murphey
• [Asterisk-Dev] PTHREAD_MUTEX_RECURSIVE   steve at daviesfam.org
• [Asterisk-Dev] PTHREAD_MUTEX_RECURSIVE   Sam Bingner
• [Asterisk-Dev] Support for Novatel Merlin G201   Andreas Bayer
• [Asterisk-Dev] Support for Novatel Merlin G201   Brancaleoni Matteo
• [Asterisk-Dev] Support for Novatel Merlin G201   Bruce Ferrell
• [Asterisk-Dev] Support for Novatel Merlin G201   Andreas Bayer
• [Asterisk-Dev] PTHREAD_MUTEX_RECURSIVE   Dr. Rich Murphey
• [Asterisk-Dev] PTHREAD_MUTEX_RECURSIVE   Jeremy McNamara
• [Asterisk-Dev] SIP Peer handling for outbound calls   Kristopher Lalletti
• [Asterisk-Dev] Support for Novatel Merlin G201   Bruce Ferrell
• [Asterisk-Dev] PTHREAD_MUTEX_RECURSIVE   Alfred R. Nurnberger
• [Asterisk-Dev] PTHREAD_MUTEX_RECURSIVE   Jeremy McNamara
• [Asterisk-Dev] Status of autoconf integration?   Robert Bedell
• [Asterisk-Dev] Status of autoconf integration?   Jon Radon
• [Asterisk-Dev] Status of autoconf integration?   Dr. Rich Murphey
• [Asterisk-Dev] PTHREAD_MUTEX_RECURSIVE   Dr. Rich Murphey
• [Asterisk-Dev] Grandstream Autoanswer   James H. Thompson
• [Asterisk-Dev] Status of autoconf integration?   asterisk at jdennis.net
• [Asterisk-Dev] Harald Baron/EBAROH/CH/Ascom ist nicht anwesend.   Harald Baron
• [Asterisk-Dev] How are IAX2 timestamps supposed to work?   Derek Smithies
• [Asterisk-Dev] Status of autoconf integration?   Steven Critchfield
• [Asterisk-Dev] Re: [Asterisk-cvs] asterisk/apps app_dial.c,1.72,1.73   Tais M. Hansen
• [Asterisk-Dev] Re: [Asterisk-cvs] asterisk/apps app_dial.c,1.72,1.73   Michael Manousos
• [Asterisk-Dev] Re: [Asterisk-cvs] asterisk/apps app_dial.c,1.72,1.73   Tais M. Hansen
• [Asterisk-Dev] IAX2 timestamping issues identified from trace analysis   steve at daviesfam.org
• [Asterisk-Dev] Skype support   Andreas Bayer
• [Asterisk-Dev] Skype support   steve at daviesfam.org
• [Asterisk-Dev] Harald Baron/EBAROH/CH/Ascom ist nicht anwesend.   Rob Gagnon
• [Asterisk-Dev] Skype support   Rob Gagnon
• [Asterisk-Dev] Skype support   Brian K. West
• [Asterisk-Dev] chan_h323 dtmf   Kelvin Chua
• [Asterisk-Dev] using 2 single pri cards on 1 server   jan
• [Asterisk-Dev] Skype support   Soren Rathje
• [Asterisk-Dev] Asterisk SIP outboundproxy   Kristopher Lalletti
• [Asterisk-Dev] Problems with inuse counter.   Claus Futtrup
• [Asterisk-Dev] Asterisk SIP outboundproxy   Robert Bedell
• [Asterisk-Dev] Little patch so chan_capi 0.3.4 will compile with latest CVS HEAD   steve at daviesfam.org
• [Asterisk-Dev] RE:reading conference numbers from database -newbie   Tarun Shroff
• [Asterisk-Dev] Re: [Asterisk-cvs] asterisk/channels chan_modem.c,1.22,1.23 chan_modem_i4l.c,1.15,1.16   Trevor Peirce
• [Asterisk-Dev] Re: [Asterisk-cvs] asterisk/channels chan_modem.c,1.22,1.23 chan_modem_i4l.c,1.15,1.16   Niklas Ögren
• [Asterisk-Dev] Re: [Asterisk-cvs] asterisk/channels chan_modem.c,1.22,1.23 chan_modem_i4l.c,1.15,1.16   James Golovich
• [Asterisk-Dev] Re: [Asterisk-cvs] asterisk/channels chan_modem.c,1.22,1.23 chan_modem_i4l.c,1.15,1.16   Trevor Peirce
• [Asterisk-Dev] Modified Prepaid App Database error   oi geli
• [Asterisk-Dev] Modified Prepaid App Database error   Dmitri Pavlenkov
• [Asterisk-Dev] DTMF digits relay with Quintum to Oh323 version 0.5.10   Kelvin Chua
• [Asterisk-Dev] DTMF digits relay with Quintum to Oh323 version 0.5.10   Dmitry Mishchenko
• [Asterisk-Dev] Codecs G729 and G723.1   Sudhir Kumar
• [Asterisk-Dev] Digium/Asterisk in Paris   Mark Spencer
• [Asterisk-Dev] Codecs G729 and G723.1   Brian K. West
• [Asterisk-Dev] Patch for postgres enabled app_voicemail.c   Matt Davies | MattDavies.Net
• [Asterisk-Dev] UPDATE Patch for postgres enabled app_voicemail.c   Matt Davies | MattDavies.Net
• [Asterisk-Dev] Getting URL to IAX Client Agent   Navnit Chachan
• [Asterisk-Dev] Zaptel compile errors in latest CVS   Chris Foster
• [Asterisk-Dev] (no subject)   Florian Overkamp
• [Asterisk-Dev] Re: [Asterisk-Users] using 2 single pri cards on 1 server   Mike Sturdee
• [Asterisk-Dev] Digium/Asterisk in Paris   Mouhamed Mahi SY
• [Asterisk-Dev] Dynamic Configuration from MS SQL Server   James Dutton
• [Asterisk-Dev] Dynamic Configuration from MS SQL Server   Michael Devenijn
• [Asterisk-Dev] Dynamic Configuration from MS SQL Server   James Dutton
• [Asterisk-Dev] Help with building on Mac OS X   Mike Mitchell
• [Asterisk-Dev] Status of autoconf integration?   Steve
• [Asterisk-Dev] Re: Codecs G729 and G723.1   Sudhir Kumar
• [Asterisk-Dev] UPDATE Patch for postgres enabled app_voicemail.c   Rob Gagnon
• [Asterisk-Dev] Dynamic Configuration from MS SQL Server   Rob Gagnon
• [Asterisk-Dev] UPDATE Patch for postgres enabled app_voicemail.c   Matt Davies | MattDavies.Net
• [Asterisk-Dev] Help with building on Mac OS X   Dr. Rich Murphey
• [Asterisk-Dev] Channel Status   Ryan Courtnage
• [Asterisk-Dev] Help with building on Mac OS X   Mike Mitchell
• [Asterisk-Dev] Getting URL to IAX Client Agent   Jean-Denis Girard
• [Asterisk-Dev] Re: Codecs G729 and G723.1   Kevin P. Fleming
• [Asterisk-Dev] Getting URL to IAX Client Agent   Navnit Chachan
• [Asterisk-Dev] Advanced ADSI scripts   alex at pilosoft.com
• [Asterisk-Dev] current error with today cvs   Richard Neese
• [Asterisk-Dev] current error with today cvs   Rob Gagnon
• [Asterisk-Dev] Stable branch usable? Development branch better?   Paul Mahler
• [Asterisk-Dev] Getting URL to IAX Client Agent   Jean-Denis Girard
• [Asterisk-Dev] rfc3581 - implemented correctly in * ?   Ryan Courtnage
• [Asterisk-Dev] Getting URL to IAX Client Agent   Steven Sokol
• [Asterisk-Dev] Getting URL to IAX Client Agent   Steve Kann
• [Asterisk-Dev] rfc3581 - implemented correctly in * ?   Ryan Courtnage
• [Asterisk-Dev] Getting URL to IAX Client Agent   Jean-Denis Girard
• [Asterisk-Dev] Getting URL to IAX Client Agent   Tilghman Lesher
• [Asterisk-Dev] Getting URL to IAX Client Agent   Jean-Denis Girard
• [Asterisk-Dev] rfc3581 - implemented correctly in * ?   John Todd
• [Asterisk-Dev] rfc3581 - implemented correctly in * ?   Ryan Courtnage
• [Asterisk-Dev] Advanced ADSI scripts   TC
• [Asterisk-Dev] Stable branch usable? Development branch better?   Brian
• [Asterisk-Dev] Advanced ADSI scripts   alex at pilosoft.com
• [Asterisk-Dev] Advanced ADSI scripts   TC
• [Asterisk-Dev] Advanced ADSI scripts   Andrew Kohlsmith
• [Asterisk-Dev] Advanced ADSI scripts   TC
• [Asterisk-Dev] HylaFAX and spandsp   Steve Underwood
• [Asterisk-Dev] HylaFAX and spandsp   Kevin P. Fleming
• [Asterisk-Dev] HylaFAX and spandsp   Steve Underwood
• [Asterisk-Dev] HylaFAX and spandsp   Kevin P. Fleming
• [Asterisk-Dev] Small Zaptel patch   Kevin Walsh
• [Asterisk-Dev] DTMF digits relay with Quintum to Oh323 version 0.5.10   Luis Mata
• [Asterisk-Dev] Advanced ADSI scripts   TC
• [Asterisk-Dev] HylaFAX and spandsp   Andrew Kohlsmith
• [Asterisk-Dev] Advanced ADSI scripts   Andrew Kohlsmith
• [Asterisk-Dev] Manager Command Reference   Rooster
• [Asterisk-Dev] Manager Command Reference   Brancaleoni Matteo
• [Asterisk-Dev] SIP users in MySQL   Filipe Murias
• [Asterisk-Dev] HylaFAX and spandsp   Christian Villa Real Lopes
• [Asterisk-Dev] HylaFAX and spandsp   Florian Overkamp
• [Asterisk-Dev] HylaFAX and spandsp   Bruce Ferrell
• [Asterisk-Dev] Advanced ADSI scripts   TC
• [Asterisk-Dev] Manager Command Reference   Rooster
• [Asterisk-Dev] sms sending and receiving   Andreas Bayer
• [Asterisk-Dev] HylaFAX and spandsp   Bruce Ferrell
• [Asterisk-Dev] Re: sms sending and receiving   Hans Mueller
• [Asterisk-Dev] voip online status   Andreas Bayer
• [Asterisk-Dev] voip online status   Rooster
• [Asterisk-Dev] DTMF digits relay with Quintum to Oh323 version 0.5.10   Kelvin Chua
• [Asterisk-Dev] E option in meetme.conf   Tilghman Lesher
• [Asterisk-Dev] Getting URL to IAX Client Agent   Navnit Chachan
• [Asterisk-Dev] Getting URL to IAX Client Agent   Navnit Chachan
• [Asterisk-Dev] Getting URL to IAX Client Agent   Adam Hart
• [Asterisk-Dev] Manager Command Reference   Jason Penton
• [Asterisk-Dev] Getting URL to IAX Client Agent   Navnit Chachan
• [Asterisk-Dev] HylaFAX and spandsp   Terry Goodwin
• [Asterisk-Dev] HylaFAX and spandsp   Thomas
• [Asterisk-Dev] voip online status   Nicolas Gudino
• [Asterisk-Dev] Manager Command Reference   Shannon Mitchell
• [Asterisk-Dev] E option in meetme.conf   Ryan Courtnage
• [Asterisk-Dev] Test POP's   Kurtz
• [Asterisk-Dev] Test POP's   Filipe Murias
• [Asterisk-Dev] Test POP's   Senad Jordanovic
• [Asterisk-Dev] OS X 10.3 Patch   John Turner
• [Asterisk-Dev] Test POP's   Tais M. Hansen
• [Asterisk-Dev] Re: 'a' and 'o' extensions do not work with app_voicemail.c (was: Newbie needs help)   Chad Scott
• [Asterisk-Dev] OS X 10.3 Patch   Tilghman Lesher
• [Asterisk-Dev] OS X 10.3 Patch   Jeremy McNamara
• [Asterisk-Dev] PRI U2U display messages   Alfred R. Nurnberger
• [Asterisk-Dev] E option in meetme.conf   Tilghman Lesher
• [Asterisk-Dev] Getting URL to IAX Client Agent   Adam Hart
• [Asterisk-Dev] Getting URL to IAX Client Agent   Navnit Chachan
• [Asterisk-Dev] Manager Command Reference   James Golovich
• [Asterisk-Dev] PRI U2U display messages   Klaus-Peter Junghanns
• [Asterisk-Dev] Customized Call Parking   Adnan Shah
• [Asterisk-Dev] PRI U2U display messages   Paul Cadach
• [Asterisk-Dev] German localization for dates   Andreas Granig
• [Asterisk-Dev] German localization for dates   Tilghman Lesher
• [Asterisk-Dev] Re: [Asterisk-Users] Customized Call Parking   Greg Boehnlein
• [Asterisk-Dev] Re: sms sending and receiving   Stefan Tichy
• [Asterisk-Dev] PRI U2U display messages   Alfred R. Nurnberger
• [Asterisk-Dev] channel.c:1508 ast_set_read_format: Unable to find a path from ULAW ????   Matt Davies | MattDavies.Net
• [Asterisk-Dev] enumLookup   Andreas Bayer
• [Asterisk-Dev] enumLookup   Duane
• [Asterisk-Dev] HylaFAX and spandsp   Terry Wilson
• [Asterisk-Dev] HylaFAX and spandsp   Greg Boehnlein
• [Asterisk-Dev] HylaFAX and spandsp   Ken Wiesner
• [Asterisk-Dev] Bugfix for CVS-HEAD-06/26/04-21:56:45   programmer_ted
• [Asterisk-Dev] Bugfix for CVS-HEAD-06/26/04-21:56:45   programmer_ted
• [Asterisk-Dev] Bugfix for CVS-HEAD-06/26/04-21:56:45   programmer_ted
• [Asterisk-Dev] How to get the SIP Response in AGI Application.   santosh bettad
• [Asterisk-Dev] channel development   mark spowage
• [Asterisk-Dev] Problems in chan_zap.c with libr2 support.   Danilo Lotina F.
• [Asterisk-Dev] RE: Problems in chan_zap.c with libr2 support.   Danilo Lotina F.
• [Asterisk-Dev] Problems in chan_zap.c with libr2 support.   CW_ASN
• [Asterisk-Dev] Calling Reload Remotely   James Dutton
• [Asterisk-Dev] RE: [Asterisk-cvs] zaptel README.Linux26,1.2,1.3 zaptel.h,1.33,1.34   Kevin Walsh
• [Asterisk-Dev] RE: [Asterisk-cvs] zaptel README.Linux26,1.2,1.3 zaptel.h,1.33,1.34   James Golovich
• [Asterisk-Dev] Calling Reload Remotely   Dmitri Pavlenkov
• [Asterisk-Dev] Calling Reload Remotely   Michael Sandee
• [Asterisk-Dev] RE: [Asterisk-cvs] zaptel README.Linux26,1.2,1.3zaptel.h,1.33,1.34   Kevin Walsh
• [Asterisk-Dev] RE: [Asterisk-cvs] zaptel README.Linux26,1.2,1.3zaptel.h,1.33,1.34   James Golovich
• [Asterisk-Dev] Bugfix for CVS-HEAD-06/26/04-21:56:45   programmer_ted
• [Asterisk-Dev] Bugfix for CVS-HEAD-06/26/04-21:56:45   programmer_ted
• [Asterisk-Dev] HylaFAX and spandsp   Ken Wiesner
• [Asterisk-Dev] Calling Reload Remotely   brian
• [Asterisk-Dev] Calling Reload Remotely   Tilghman Lesher
• [Asterisk-Dev] Calling Reload Remotely   Jeremy McNamara
• [Asterisk-Dev] sip call routing   mark spowage
• [Asterisk-Dev] channel driver sample   mark spowage
• [Asterisk-Dev] sip call routing   CW_ASN
• [Asterisk-Dev] enumLookup   Aaron J. Angel
• [Asterisk-Dev] Calling Reload Remotely   Brian K. West
• [Asterisk-Dev] CVsup up   Brian K. West
• [Asterisk-Dev] Calling Reload Remotely   Michael Sandee

Last message date: Wed Jun 30 21:46:32 MST 2004
Archived on: Tue Sep 5 14:26:57 MST 2006

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