December 2003 Archives by date

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Starting: Mon Dec 1 05:22:56 MST 2003
Ending: Wed Dec 31 21:59:41 MST 2003
Messages: 238

• [Asterisk-Dev] Recomended GSM channel banks for use with E1 WC   Peter Nixon
• [Asterisk-Dev] IAX phones and CPU usage problem   Grzegorz Nosek
• [Asterisk-Dev] help needed - SIP register response contact   Michael Procter
• [Asterisk-Dev] Asterisk with Voicetronix OpenLine4 card   Ahmad Faiz
• [Asterisk-Dev] help needed - SIP register response contact   Conroy, Lawrence (SMTP)
• [Asterisk-Dev] Compiling problem   Kang.ChenJi at c3smail.monmouth.army.mil
• [Asterisk-Dev] Compiling problem   Jared Mauch
• [Asterisk-Dev] Compiling problem   Vince Vielhaber
• [Asterisk-Dev] IAX2 call variable passing between servers   John Todd
• [Asterisk-Dev] Compiling problem   Michael Manousos
• [Asterisk-Dev] Compiling problem   Kang.ChenJi at c3smail.monmouth.army.mil
• [Asterisk-Dev] IAX2 call variable passing between servers   Michael Bielicki
• [Asterisk-Dev] Compiling problem   Jared Mauch
• [Asterisk-Dev] E1/PRA connection to existing PBX with Asterisk - SETUP ACK with Suggested Channel ID missing ?   Nicolas Dramais
• [Asterisk-Dev] Compiling problem   Alastair Maw
• [Asterisk-Dev] Compiling problem   Florian Overkamp
• [Asterisk-Dev] E1/PRA connection to existing PBX with Asterisk - SETUP ACK with Suggested Channel ID missing ?   Pertti Pikkarainen
• [Asterisk-Dev] IAX2 call variable passing between servers   Linus Surguy
• [Asterisk-Dev] Bug Num 0000456   Linus Surguy
• [Asterisk-Dev] include file?   Jared Mauch
• [Asterisk-Dev] asterisk -r comm issues   Jared Mauch
• [Asterisk-Dev] Bug Num 0000456   Jeremy McNamara
• [Asterisk-Dev] asterisk -r comm issues   Greg Varga
• [Asterisk-Dev] include file?   John Todd
• [Asterisk-Dev] asterisk -r comm issues   John Todd
• [Asterisk-Dev] asterisk -r comm issues   Jared Mauch
• [Asterisk-Dev] IAX2 call variable passing between servers   John Todd
• [Asterisk-Dev] IAX2 call variable passing between servers   Adam Hart
• [Asterisk-Dev] SMS over PRI/E1?   Roy Sigurd Karlsbakk
• [Asterisk-Dev] Bug Num 0000456   Linus Surguy
• [Asterisk-Dev] include file?   James Golovich
• [Asterisk-Dev] SMS over PRI/E1?   Mark Hagler
• [Asterisk-Dev] Bug Num 0000456   Jeremy McNamara
• [Asterisk-Dev] SMS over PRI/E1?   Emanuele Pucciarelli
• [Asterisk-Dev] SMS over PRI/E1?   Roy Sigurd Karlsbakk
• [Asterisk-Dev] SMS over PRI/E1?   Nicolas Bougues
• [Asterisk-Dev] IAX2 call variable passing between servers   Olle E. Johansson
• [Asterisk-Dev] Bug Num 0000456   Linus Surguy
• [Asterisk-Dev] IAX2 call variable passing between servers   Linus Surguy
• [Asterisk-Dev] VoiceXML Connectivity for Asterisk   Raj Kiran Talusani
• [Asterisk-Dev] SMS over PRI/E1?   Emanuele Pucciarelli
• [Asterisk-Dev] IAX2 call variable passing between servers   Grzegorz Nosek
• [Asterisk-Dev] Bug Num 0000456   Mark Spencer
• [Asterisk-Dev] IAX2 call variable passing between servers   Mark Spencer
• [Asterisk-Dev] Bug Num 0000456   Linus Surguy
• [Asterisk-Dev] IAX2 call variable passing between servers   John Todd
• [Asterisk-Dev] Implement missing features in Meetme application   Angel Carpintero
• [Asterisk-Dev] IAX2 call variable passing between servers   Alastair Maw
• [Asterisk-Dev] IAX2 call variable passing between servers   John Todd
• [Asterisk-Dev] Bug Num 0000456   Mark Spencer
• [Asterisk-Dev] IAX2 call variable passing between servers   Linus Surguy
• [Asterisk-Dev] Bug Num 0000456   Linus Surguy
• [Asterisk-Dev] IAX2 call variable passing between servers   Joshua M. Thompson
• [Asterisk-Dev] Bug Num 0000456   Mark Spencer
• [Asterisk-Dev] telephony applet anyone?   lappin at buenavistawireless.com
• [Asterisk-Dev] SMS over PRI/E1?   Brett Schwarz
• [Asterisk-Dev] asterisk -r comm issues   Michael Manousos
• [Asterisk-Dev] VoiceXML Connectivity for Asterisk   Steven Critchfield
• [Asterisk-Dev] command line editing?   Dave Weis
• [Asterisk-Dev] Re: E1/PRA connection to existing PBX with Asterisk - SETUP ACK with Suggested Channel ID missing ?   Nicolas Dramais
• [Asterisk-Dev] Re: VoiceXML Connectivity for Asterisk   Raj Kiran Talusani
• [Asterisk-Dev] IAX2 call variable passing between servers   William Waites
• [Asterisk-Dev] IAX2 call variable passing between servers   Olle E. Johansson
• [Asterisk-Dev] Implement missing features in Meetme application   Jeremy McNamara
• [Asterisk-Dev] IAX2 call variable passing between servers   Linus Surguy
• [Asterisk-Dev] VoiceXML Connectivity for Asterisk   TC
• [Asterisk-Dev] Re: VoiceXML Connectivity for Asterisk   Steven Critchfield
• [Asterisk-Dev] telephony applet anyone?   Peter Nixon
• [Asterisk-Dev] IAX2 call variable passing between servers   Dave Weis
• [Asterisk-Dev] VoiceXML Connectivity for Asterisk   Florian Overkamp
• [Asterisk-Dev] VoiceXML Connectivity for Asterisk   Steven Critchfield
• [Asterisk-Dev] Implement missing features in Meetme application   Olle E. Johansson
• [Asterisk-Dev] telephony applet anyone?   lappin at buenavistawireless.com
• [Asterisk-Dev] "Dial-out" Conference   Kang.ChenJi at c3smail.monmouth.army.mil
• [Asterisk-Dev] Asterisk Application Question   Adam
• [Asterisk-Dev] Asterisk Application Question   Steven Critchfield
• [Asterisk-Dev] Re: VoiceXML Connectivity for Asterisk   Raj Kiran Talusani
• [Asterisk-Dev] Application API   Jonathan Tew
• [Asterisk-Dev] lawful intercept   wasim at convergence.com.pk
• [Asterisk-Dev] VoiceXML Connectivity for Asterisk   Florian Overkamp
• [Asterisk-Dev] VoiceXML Connectivity for Asterisk   Olle E. Johansson
• [Asterisk-Dev] SMS over PRI/E1?   Roy Sigurd Karlsbakk
• [Asterisk-Dev] Implement missing features in Meetme application   Angel Carpintero
• [Asterisk-Dev] SMS over PRI/E1?   Grzegorz Nosek
• [Asterisk-Dev] "Dial-out" Conference   Luis Mata
• [Asterisk-Dev] Asterisk Application Question   Adam
• [Asterisk-Dev] "Dial-out" Conference   Kang.ChenJi at c3smail.monmouth.army.mil
• [Asterisk-Dev] "Dial-out" Conference   Walker Haddock
• [Asterisk-Dev] "Dial-out" Conference   Luis Mata
• [Asterisk-Dev] "Dial-out" Conference   Olle E. Johansson
• [Asterisk-Dev] SIP 407 error   jerk face
• [Asterisk-Dev] VoiceXML Connectivity for Asterisk   TC
• [Asterisk-Dev] linux 2.6?   Roy Sigurd Karlsbakk
• [Asterisk-Dev] linux 2.6?   Matteo Brancaleoni
• [Asterisk-Dev] AutoTools Build Enviroment [was: linux 2.6?]   Greg Varga
• [Asterisk-Dev] PBX-wilcalu on FreeBSD   Olle E. Johansson
• [Asterisk-Dev] Unknown RTP codec 72 received ??   Carlos Arnt
• [Asterisk-Dev] Autoconf support -- testers needed   William Waites
• [Asterisk-Dev] Autoconf support -- testers needed   Chris Albertson
• [Asterisk-Dev] Autoconf support -- testers needed   Greg Varga
• [Asterisk-Dev] Autoconf support -- testers needed   William Waites
• [Asterisk-Dev] linux 2.6?   Mark Spencer
• [Asterisk-Dev] tx_fax fscked up...   Roy Sigurd Karlsbakk
• [Asterisk-Dev] tx_fax fscked up...   Steven Critchfield
• [Asterisk-Dev] tx_fax fscked up...   TC
• [Asterisk-Dev] (no subject)   Nicholas Hart
• [Asterisk-Dev] SEGV in CVS version of Asterisk   Ron OHara
• [Asterisk-Dev] Architecture for CTI   Jonathan Tew
• [Asterisk-Dev] SEGV in CVS version of Asterisk   Steven Critchfield
• [Asterisk-Dev] VoiceMail file formats   Jonathan.Hopper at kaz-group.com
• [Asterisk-Dev] EAGI : Pipe saturation   Areski
• [Asterisk-Dev] SEGV in CVS version of Asterisk   Matteo Brancaleoni
• [Asterisk-Dev] SEGV in CVS version of Asterisk   Steven Critchfield
• [Asterisk-Dev] EAGI : Pipe saturation   Steven Critchfield
• [Asterisk-Dev] SEGV in CVS version of Asterisk   Jonathan Tew
• [Asterisk-Dev] SEGV in CVS version of Asterisk   Andrew Kohlsmith
• [Asterisk-Dev] SEGV in CVS version of Asterisk   Nicholas Hart
• [Asterisk-Dev] SEGV in CVS version of Asterisk   Chris Albertson
• [Asterisk-Dev] SEGV in CVS version of Asterisk   James Golovich
• [Asterisk-Dev] SEGV in CVS version of Asterisk   Steven Critchfield
• [Asterisk-Dev] Why use SIGABRT to kill mpg123 procs in res_musiconhold?   Tom Moertel
• [Asterisk-Dev] wiax   Rattana BIV
• [Asterisk-Dev] Why use SIGABRT to kill mpg123 procs in res_musiconhold?   Jared Smith
• [Asterisk-Dev] Why use SIGABRT to kill mpg123 procs in res_musiconhold?   Rich Adamson
• [Asterisk-Dev] SIP bug and Polycom phones   Christian Hecimovic
• [Asterisk-Dev] SIP bug and Polycom phones   Clif Jones
• [Asterisk-Dev] SIP bug and Polycom phones   Greg Varga
• [Asterisk-Dev] Voicetronix?   Keith Vogt
• [Asterisk-Dev] SIP bug and Polycom phones   Christian Hecimovic
• [Asterisk-Dev] Voicetronix?   Anthony Wood
• [Asterisk-Dev] SIP bug and Polycom phones   Clif Jones
• [Asterisk-Dev] Why use SIGABRT to kill mpg123 procs in res_musiconhold?   Sam Bingner
• [Asterisk-Dev] Fw: Job Opportunities   Steve Lynn
• [Asterisk-Dev] Why use SIGABRT to kill mpg123 procs in res_musiconhold?   Sam Bingner
• [Asterisk-Dev] CVS compile error??? Chan_zap.c   Luis Mata
• [Asterisk-Dev] CVS compile error??? Chan_zap.c   Tilghman Lesher
• [Asterisk-Dev] Just wanted to say hi   Leif Madsen
• [Asterisk-Dev] (no subject)   Ravi Cherukuri
• [Asterisk-Dev] Dynamic IAX connections   Steve Rodgers
• [Asterisk-Dev] Making functions obsolete   Olle E. Johansson
• [Asterisk-Dev] Dynamic IAX connections   Steven Critchfield
• [Asterisk-Dev] noises an Zap channel (TDM20B) while hdd activity   Thomas Haeger
• [Asterisk-Dev] Quintum CMS amd Chan_h323 Codec G729   Luis Mata
• [Asterisk-Dev] EAGI : Pipe saturation   Areski
• [Asterisk-Dev] Dynamic IAX connections   Steve Rodgers
• [Asterisk-Dev] EAGI : Pipe saturation   Steven Critchfield
• [Asterisk-Dev] Dynamic IAX connections   Steven Critchfield
• [Asterisk-Dev] Dynamic IAX connections   Steve Rodgers
• [Asterisk-Dev] VoiceTronix OpenSwitch12 Problem   Jacky Chen\(陳明君\)
• [Asterisk-Dev] Gnophone problems   Peter Nixon
• [Asterisk-Dev] EAGI : Pipe saturation   Areski
• [Asterisk-Dev] EAGI : Pipe saturation   Steven Critchfield
• [Asterisk-Dev] Dynamic IAX connections   Steven Critchfield
• [Asterisk-Dev] EAGI : Pipe saturation   Areski
• [Asterisk-Dev] EAGI : Pipe saturation   Steven Critchfield
• [Asterisk-Dev] VoiceTronix OpenSwitch12 Problem   Daniel Bichara
• [Asterisk-Dev] Dynamic IAX connections   Steve Rodgers
• [Asterisk-Dev] A Red Hat Linux 8.0 Question   Kang.ChenJi at c3smail.monmouth.army.mil
• [Asterisk-Dev] A Red Hat Linux 8.0 Question   Steven Critchfield
• [Asterisk-Dev] Core dump with latest CVS on RedHat-7.2   Darren Nickerson
• [Asterisk-Dev] Core dump with latest CVS on RedHat-7.2   Darren Nickerson
• [Asterisk-Dev] Re: VoiceTronix OpenSwitch12 Problem   Ahmad Faiz
• [Asterisk-Dev] Re: VoiceTronix OpenSwitch12 Problem   Ahmad Faiz
• [Asterisk-Dev] Re: [Asterisk-cvs] asterisk/channels chan_zap.c,1.154,1.155   James Sizemore
• [Asterisk-Dev] Core dump with latest CVS on RedHat-7.2   Chris Wilson
• [Asterisk-Dev] Re: [Asterisk-cvs] asterisk/appsapp_queue.c,1.39,1.40   James Sizemore
• [Asterisk-Dev] Re: VoiceTronix OpenSwitch12 Problem   Daniel Bichara
• [Asterisk-Dev] Re: [Dev] opteron anyone ?   Peter Nixon
• [Asterisk-Dev] opteron anyone ?   Reinhard Max
• [Asterisk-Dev] Re: [Asterisk-cvs] asterisk/channels chan_zap.c,1.154,1.155   Martin Pycko
• [Asterisk-Dev] RE: VoiceTronix OpenSwitch12 Problem   Ahmad Faiz
• [Asterisk-Dev] Core dump with latest CVS on RedHat-7.2   Darren Nickerson
• [Asterisk-Dev] RE: VoiceTronix OpenSwitch12 Problem   Jacky
• [Asterisk-Dev] RxFAX application   Sergio Serrano Revuelto
• [Asterisk-Dev] RxFAX application   Sergio Serrano Revuelto
• [Asterisk-Dev] call center - 1 phone ringing at a time   Mike Nugent
• [Asterisk-Dev] call center - 1 phone ringing at a time   Kevin Bockman
• [Asterisk-Dev] call center - 1 phone ringing at a time   Mike Nugent
• [Asterisk-Dev] MGCP patch to interoperate with Cisco IAD243X   Juan J. Sierralta P.
• [Asterisk-Dev] MGCP patch to interoperate with Cisco IAD243X   Jeremy McNamara
• [Asterisk-Dev] MGCP patch to interoperate with Cisco IAD243X   Karl Putland
• [Asterisk-Dev] MGCP patch to interoperate with Cisco IAD243X   Juan J. Sierralta P.
• [Asterisk-Dev] Re: [Asterisk-Users] RxFAX application   Masakazu Nakano
• [Asterisk-Dev] MGCP patch to interoperate with Cisco IAD243X   Juan J. Sierralta P.
• [Asterisk-Dev] MGCP patch to interoperate with Cisco IAD243X   Tilghman Lesher
• [Asterisk-Dev] MGCP patch to interoperate with Cisco IAD243X   Juan J. Sierralta P.
• [Asterisk-Dev] Autocreatepeer ?   Olle E. Johansson
• [Asterisk-Dev] RE: VoiceTronix OpenSwitch12 Problem   Paul Bagyenda
• [Asterisk-Dev] compile problem   Daniel ANDRE
• [Asterisk-Dev] compile problem   Rich Adamson
• [Asterisk-Dev] RE: [Asterisk-Users] RxFAX application   Sergio Serrano Revuelto
• [Asterisk-Dev] Autocreatepeer ?   John Bigelow
• [Asterisk-Dev] FWD: Cisco 12SP+ / VIP30 - Works with Asterisk (chan_skinny related)   Leif Madsen
• [Asterisk-Dev] H.323 Conferencing Server   Kang.ChenJi at c3smail.monmouth.army.mil
• [Asterisk-Dev] MGCP Problem   Sergi Gabunia
• [Asterisk-Dev] Dial() and non-BUSY failures   Stephen Davies
• [Asterisk-Dev] Dial() returns - +1 or +101 and what they mean   Stephen Davies
• [Asterisk-Dev] Dial() returns - +1 or +101 and what they mean   duncan
• [Asterisk-Dev] opteron anyone ?   Peter Nixon
• [Asterisk-Dev] Dial() returns - +1 or +101 and what they mean   Steven Critchfield
• [Asterisk-Dev] failover was: Dial() returns...   Mike Nugent
• [Asterisk-Dev] calling user hears 'pbx-transfer' after being parked   Barton Hodges
• [Asterisk-Dev] FWD: Cisco 12SP+ / VIP30 - Works with Asterisk (chan_skinny related)   Steven Critchfield
• [Asterisk-Dev] FWD: Cisco 12SP+ / VIP30 - Works with Asterisk (chan_skinny related)   Leif Madsen
• [Asterisk-Dev] FWD: Cisco 12SP+ / VIP30 - Works with Asterisk (chan_skinny related)   Steven Critchfield
• [Asterisk-Dev] FWD: Cisco 12SP+ / VIP30 - Works with Asterisk (chan_skinny related)   Jeremy McNamara
• [Asterisk-Dev] FWD: Cisco 12SP+ / VIP30 - Works with Asterisk (chan_skinny related)   Steven Critchfield
• [Asterisk-Dev] PRI error   Peter Nixon
• [Asterisk-Dev] FWD: Cisco 12SP+ / VIP30 - Works with Asterisk (chan_skinny related)   Steven Critchfield
• [Asterisk-Dev] UK phone sets that can be used with the Digium Cards   Hassan Raza
• [Asterisk-Dev] UK phone sets that can be used with the Digium Cards   Iain Stevenson
• [Asterisk-Dev] DTMF Error   Brent Franks
• [Asterisk-Dev] sip 30X   Martin Kihlgren
• [Asterisk-Dev] sip 30X   Olle E. Johansson
• [Asterisk-Dev] DTMF Error   Brian West
• [Asterisk-Dev] Strange bells in voicemail and voicemailmain ?   Florian Overkamp
• [Asterisk-Dev] Digium TE410P Card   Mike Nugent
• [Asterisk-Dev] Digium TE410P Card   Paul Mahler
• [Asterisk-Dev] DTMF Error   Olle E. Johansson
• [Asterisk-Dev] DTMF Error   Brent Franks
• [Asterisk-Dev] is ast_openstream works well?   Masakazu Nakano
• [Asterisk-Dev] Digium TE410P Card   Andrew Kohlsmith
• [Asterisk-Dev] Current database abstraction effort ?   Nicolas Bougues
• [Asterisk-Dev] Current database abstraction effort ?   CW_ASN
• [Asterisk-Dev] Digium TE410P Card   Andrew Kohlsmith
• [Asterisk-Dev] Digium TE410P Card   Ed Devine
• [Asterisk-Dev] Digium TE410P Card   James Sharp
• [Asterisk-Dev] Current database abstraction effort ?   Brian West
• [Asterisk-Dev] Current database abstraction effort ?   James Sharp
• [Asterisk-Dev] Current database abstraction effort ?   Brian West
• [Asterisk-Dev] Current database abstraction effort ?   James Sharp
• [Asterisk-Dev] Crappy New Year - AGI is b0rked   Jared Smith
• [Asterisk-Dev] Current database abstraction effort ?   Nicolas Bougues
• [Asterisk-Dev] Crappy New Year - AGI is b0rked   Steven Critchfield
• [Asterisk-Dev] Crappy New Year - AGI is b0rked   ml at neoninternet.com
• [Asterisk-Dev] Crappy New Year - AGI is b0rked   Paul Liew
• [Asterisk-Dev] Comedian Mail's ADSI implementation broken on phones needing unlock codes?   Darren Nickerson
• [Asterisk-Dev] Crappy New Year - AGI is b0rked   Steven Critchfield

Last message date: Wed Dec 31 21:59:41 MST 2003
Archived on: Tue Sep 5 14:26:48 MST 2006

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