November 2003 Archives by thread

• Messages sorted by: [ subject ] [ author ] [ date ]
• More info on this list...

Starting: Sat Nov 1 16:58:07 MST 2003
Ending: Sun Nov 30 20:25:53 MST 2003
Messages: 225

• [Asterisk-Dev] cvs and msvc   lappin at buenavistawireless.com
• [Asterisk-Dev] app_disa2: Feedback requested   Eric Wieling
  • [Asterisk-Dev] app_disa2: Feedback requested   Steven Critchfield
    • [Asterisk-Dev] app_disa2: Feedback requested   Eric Wieling
  • [Asterisk-Dev] app_disa2: Feedback requested   John Todd
    • [Asterisk-Dev] app_disa2: Feedback requested   Eric Wieling
      • [Asterisk-Dev] app_disa2: Feedback requested   Tilghman Lesher
• [Asterisk-Dev] App DISA2: strdup memory leaks?   Steve Rodgers
  • [Asterisk-Dev] App DISA2: strdup memory leaks?   Steven Critchfield
• [Asterisk-Dev] Question about quad E1   Alastair Maw
  • [Asterisk-Dev] Question about quad E1   Micke Andersson
• [Asterisk-Dev] unsubscribe   Daniel Lee
• [Asterisk-Dev] (no subject)   Daniel Lee
  • [Asterisk-Dev] (no subject)   Steven Critchfield
• [Asterisk-Dev] Channel struct in pri_dchannel   Claude Klimos
  • [Asterisk-Dev] Channel struct in pri_dchannel   Martin Pycko
• [Asterisk-Dev] Extension of transfer functionality in app_dial.c   John Todd
• [Asterisk-Dev] PLAY MP3 IN EAGI IS DOWN !!!   Areski
  • [Asterisk-Dev] PLAY MP3 IN EAGI IS DOWN !!!   Tais M. Hansen
    • [Asterisk-Dev] PLAY MP3 IN EAGI IS DOWN !!!   Mark Spencer
  • [Asterisk-Dev] PLAY MP3 IN EAGI IS DOWN !!!   Steven Critchfield
    • [Asterisk-Dev] PLAY MP3 IN EAGI IS DOWN !!!   Areski
      • [Asterisk-Dev] PLAY MP3 IN EAGI IS DOWN !!!   Steven Critchfield
      • [Asterisk-Dev] PLAY MP3 IN EAGI IS DOWN !!!   Tilghman Lesher
• [Asterisk-Dev] EAGI Question   Eric Wieling
  • [Asterisk-Dev] EAGI Question   Steven Critchfield
• [Asterisk-Dev] [REPOST] [Asterisk-Users] ZapRAS docs needed...   Roy Sigurd Karlsbakk
  • [Asterisk-Dev] [REPOST] [Asterisk-Users] ZapRAS docs needed...   Steven Critchfield
• [Asterisk-Dev] cell phone locations?   Steven Critchfield
  • [Asterisk-Dev] cell phone locations?   John Todd
• [Asterisk-Dev] Distinctive ring inbound on FXO   William Waites
  • [Asterisk-Dev] Distinctive ring inbound on FXO   John Todd
• [Asterisk-Dev] test   Md. Mahabub Alam
• [Asterisk-Dev] Wifi IP phones...   Luis Mata
  • [Asterisk-Dev] Wifi IP phones...   Alexander Mayrhofer
    • [Asterisk-Dev] Wifi IP phones...   Luis Mata
      • [Asterisk-Dev] Wifi IP phones...   Michiel Betel
    • [Asterisk-Dev] Wifi IP phones...   Jared Mauch
      • [Asterisk-Dev] Wifi IP phones...   Andrew Kohlsmith
      • [Asterisk-Dev] Wifi IP phones...   John Todd
  • [Asterisk-Dev] Wifi IP phones...   Admin - Storm3
• [Asterisk-Dev] AGI/ Oh323 and Incoming IP address...   Luis Mata
  • [Asterisk-Dev] AGI/ Oh323 and Incoming IP address...   Michael Manousos
• [Asterisk-Dev] Wifi IP phones...   Bisker, Scott (7805)
  • [Asterisk-Dev] Wifi IP phones...   Michael Manousos
• [Asterisk-Dev] Wifi IP phones...   Bisker, Scott (7805)
  • [Asterisk-Dev] Wifi IP phones...   Michael Manousos
• [Asterisk-Dev] Need testers for new STUN build system   Chris Albertson
  • [Asterisk-Dev] DTFM Caller ID   Jorge Merlino
• [Asterisk-Dev] x100P   Todd Wallace
  • [Asterisk-Dev] x100P   Steven Critchfield
    • [Asterisk-Dev] x100P   Todd Wallace
      • [Asterisk-Dev] x100P   Luis Mata
  • [Asterisk-Dev] x100P   Eric Wieling
    • [Asterisk-Dev] x100P   Olle E. Johansson
• [Asterisk-Dev] DTMF detection on modem(ISDN)   Tomaz Izanc
• [Asterisk-Dev] Voice energy detection: coder wanted   John Todd
  • [Asterisk-Dev] Voice energy detection: coder wanted   David Davis
    • [Asterisk-Dev] Voice energy detection: coder wanted   John Todd
      • [Asterisk-Dev] Voice energy detection: coder wanted   Peter Nixon
      • [Asterisk-Dev] Mera Network Softswitch with OH323   Sergio Serrano Revuelto
  • [Asterisk-Dev] Voice energy detection: coder wanted   Stephen Davies
    • [Asterisk-Dev] Voice energy detection: coder wanted   John Todd
• [Asterisk-Dev] [Asterisk-Users] Help with Conference   Raj Kiran Talsani
• [Asterisk-Dev] VoiceXML: publicvoicexml integration ?   Florian Overkamp
  • [Asterisk-Dev] VoiceXML: publicvoicexml integration ?   Luis Mata
• [Asterisk-Dev] Voice energy detection: coder wanted   Freddi Hansen
  • [Asterisk-Dev] Voice energy detection: coder wanted   John Todd
• [Asterisk-Dev] Re: Voice energy detection: coder wanted   Freddi Hansen
  • [Asterisk-Dev] Re: Voice energy detection: coder wanted   Chris Albertson
• [Asterisk-Dev] IAX2 Bug: 0000456 callerid=   Linus Surguy
• [Asterisk-Dev] Oh323 and CDR   Luis Mata
  • [Asterisk-Dev] More to Oh323 and CDR   Luis Mata
    • Hmm...: [Asterisk-Dev] More to Oh323 and CDR   Luis Mata
    • [Asterisk-Dev] More to Oh323 and CDR   Michael Manousos
      • [Asterisk-Dev] More to Oh323 and CDR   Luis Mata
• [Asterisk-Dev] [PATCH] few fixes to the AGI code   Steven Critchfield
  • [Asterisk-Dev] [PATCH] few fixes to the AGI code   Steven Critchfield
• [Asterisk-Dev] IP v6 Supported?   Peter Brown
  • [Asterisk-Dev] IP v6 Supported?   Steven Critchfield
    • [Asterisk-Dev] IP v6 Supported?   John Brown (CV)
  • [Asterisk-Dev] Operator Console   David Hindmarsh
    • [Asterisk-Dev] Operator Console   Rich Adamson
      • [Asterisk-Dev] Operator Console   Gary
      • [Asterisk-Dev] Operator Console   asterisk at billheckel.com
      • [Asterisk-Dev] Operator Console   Paul Mahler
      • [Asterisk-Dev] Operator Console   David Hindmarsh
      • [Asterisk-Dev] Operator Console   Ken Walker
  • [Asterisk-Dev] IP v6 Supported?   Olle E. Johansson
• [Asterisk-Dev] question about looping of code.   Steven Critchfield
  • [Asterisk-Dev] question about looping of code.   Martin Pycko
    • [Asterisk-Dev] question about looping of code.   Steven Critchfield
• [Asterisk-Dev] Re: Voice energy detection: coder wanted   Francois Lambert
• [Asterisk-Dev] ISDN PRI TBCT   Gene Kochanowsky
• [Asterisk-Dev] IAX interface with asterisk - questions about using libiax   Grzegorz Nosek
• [Asterisk-Dev] asterisk cvs ebuilds for gentoo portage system   Dorian Gray
• [Asterisk-Dev] Open Source Linux PBX!   Quan Le Trung
  • [Asterisk-Dev] RE: [Asterisk-Users] Open Source Linux PBX!   Sergio Serrano Revuelto
  • [Asterisk-Dev] Open Source Linux PBX!   asterisk at billheckel.com
• [Asterisk-Dev] CVS repository changes   Thorsten Lockert
  • [Asterisk-Dev] SIP -> H.323   Sergio Serrano Revuelto
  • [Asterisk-Dev] Re: [Asterisk-Users] CVS repository changes   Rich Adamson
    • [Asterisk-Dev] Re: [Asterisk-Users] CVS repository changes   Thorsten Lockert
• [Asterisk-Dev] Text to speech on Asterisk   astdev at newww.com
  • [Asterisk-Dev] Text to speech on Asterisk   Chris Albertson
  • [Asterisk-Dev] Text to speech on Asterisk   Eric Wieling
    • [Asterisk-Dev] Text to speech on Asterisk   John Todd
      • [Asterisk-Dev] Text to speech on Asterisk   Peter Brown
      • [Asterisk-Dev] Text to speech on Asterisk   Ginés Gómez
      • [Asterisk-Dev] Text to speech on Asterisk   Tais M. Hansen
      • [Asterisk-Dev] Text to speech on Asterisk   Raj Kiran Talusani
      • [Asterisk-Dev] Text to speech on Asterisk   astdev at newww.com
      • [Asterisk-Dev] Text to speech on Asterisk   Peter Brown
      • [Asterisk-Dev] Text to speech on Asterisk   Mark Spencer
      • [Asterisk-Dev] Text to speech on Asterisk   John Todd
      • [Asterisk-Dev] Text to speech on Asterisk   astdev at newww.com
      • [Asterisk-Dev] Text to speech on Asterisk   Joseph Riss
      • [Asterisk-Dev] Text to speech on Asterisk   Dave
      • [Asterisk-Dev] Text to speech on Asterisk   Olle E. Johansson
      • [Asterisk-Dev] Text to speech on Asterisk   Tilghman Lesher
      • [Asterisk-Dev] Text to speech on Asterisk   Olle E. Johansson
      • [Asterisk-Dev] Text to speech on Asterisk   Mark Spencer
      • [Asterisk-Dev] Text to speech on Asterisk   Michael Bielicki
      • [Asterisk-Dev] Text to speech on Asterisk   Chris Albertson
• [Asterisk-Dev] Asterisk, MGCP, and the AUEP verb   Brady Kirby
  • [Asterisk-Dev] Asterisk, MGCP, and the AUEP verb   Pavel Litvinenko
• [Asterisk-Dev] Asterisk, MGCP, not processing RSIP   Brady Kirby
• [Asterisk-Dev] RE: [Asterisk-Users] Overhead Paging - app notes   John Todd
• [Asterisk-Dev] Consultant? List yourself on the Asterisk Wiki   Olle E. Johansson
• [Asterisk-Dev] Message lamp integration with legacy pbx during conversion   Josh J. Zuerner
• [Asterisk-Dev] Embedded Asterisk   Andy Powell
  • [Asterisk-Dev] Embedded Asterisk   Lars Boegild Thomsen
  • [Asterisk-Dev] Embedded Asterisk   Lars Boegild Thomsen
• [Asterisk-Dev] Voice XML APIs   steve.lane at covad.net
• [Asterisk-Dev] Asterisk Cdr   Kausik Chatterjee
• [Asterisk-Dev] MOH timing   Bob Knight
  • [Asterisk-Dev] MOH timing   Mark Spencer
• [Asterisk-Dev] Ethereal packet sniffer   Alastair Maw
  • [Asterisk-Dev] Ethereal packet sniffer   Tais M. Hansen
• [Asterisk-Dev] IAXtel problems   Jorge Merlino
• [Asterisk-Dev] Unregisterd SIP calls.   Luis Mata
  • [Asterisk-Dev] Unregisterd SIP calls.   Mark Spencer
    • [Asterisk-Dev] Dialing logged in Agent   James Coberly
• [Asterisk-Dev] Ethereal plugin for IAX2   Alastair Maw
  • [Asterisk-Dev] Ethereal plugin for IAX2   Steve Kann
  • [Asterisk-Dev] Re: Ethereal plugin for IAX2   Alastair Maw
• [Asterisk-Dev] Dialing logged in Agent   Alex Lopez
  • [Asterisk-Dev] Dialing logged in Agent   James Coberly
    • [Asterisk-Dev] Dialing logged in Agent   Richard Lyman
      • [Asterisk-Dev] Agentlogin - transfer issue   James Coberly
  • [Asterisk-Dev] Turnkey Systems   Michael at Matraex.com
    • [Asterisk-Dev] Turnkey Systems   Jeremy McNamara
• [Asterisk-Dev] Re: [Asterisk-cvs] asterisk/apps app_queue.c,1.39,1.40   James Sizemore
  • [Asterisk-Dev] Re: [Asterisk-cvs] asterisk/apps app_queue.c,1.39,1.40   James Sizemore
    • [Asterisk-Dev] Re: [Asterisk-cvs] asterisk/appsapp_queue.c,1.39,1.40   TC
      • [Asterisk-Dev] Re: [Asterisk-cvs] asterisk/appsapp_queue.c,1.39,1.40   James Sizemore
• [Asterisk-Dev] Cisco DTMF Issue   Josh J. Zuerner
• [Asterisk-Dev] IAX protocol - how do I transfer a call?   Grzegorz Nosek
  • [Asterisk-Dev] IAX protocol - how do I transfer a call?   Mark Spencer
    • [Asterisk-Dev] IAX protocol - how do I transfer a call?   Grzegorz Nosek
      • [Asterisk-Dev] IAX protocol - how do I transfer a call?   Matteo Brancaleoni
• [Asterisk-Dev] Should I hear the CAS signal on ADSI Phones?   Jonathan Biggs
• [Asterisk-Dev] Slow response on 7960 web services with php   Peter Grace
• [Asterisk-Dev] app_perl ??   Brian West
• [Asterisk-Dev] Asterisk server hangs   Francois Lambert
  • [Asterisk-Dev] Asterisk server hangs   Mark Spencer
  • [Asterisk-Dev] Asterisk server hangs   Chris Wilson
    • [Asterisk-Dev] Asterisk server hangs   Mark Spencer
      • [Asterisk-Dev] Asterisk server hangs   Chris Wilson
      • [Asterisk-Dev] Asterisk server hangs   Luis Mata
      • [Asterisk-Dev] Asterisk server hangs   Chris Wilson
      • [Asterisk-Dev] Asterisk server hangs   Rich Adamson
• [Asterisk-Dev] Small CVS problem   Matteo Brancaleoni
• [Asterisk-Dev] RE: Asterisk server hangs   Francois Lambert
  • [Asterisk-Dev] RE: Asterisk server hangs   Chris Wilson
• [Asterisk-Dev] Re: [Asterisk-cvs] asterisk/channels chan_sip.c,1.248,1.249   Olle E. Johansson
  • [Asterisk-Dev] Re: [Asterisk-cvs] asterisk/channels chan_sip.c,1.248,1.249   Thorsten Lockert
    • [Asterisk-Dev] Re: [Asterisk-cvs] asterisk/channels chan_sip.c,1.248,1.249   Olle E. Johansson
      • [Asterisk-Dev] Re: [Asterisk-cvs] asterisk/channels chan_sip.c,1.248,1.249   Thorsten Lockert
      • [Asterisk-Dev] Re: [Asterisk-cvs] asterisk/channels chan_sip.c,1.248,1.249   Olle E. Johansson
      • [Asterisk-Dev] Re: [Asterisk-cvs] asterisk/channels chan_sip.c,1.248,1.249   Brian West
      • [Asterisk-Dev] Re: [Asterisk-cvs] asterisk/channels chan_sip.c,1.248,1.249   Olle E. Johansson
      • [Asterisk-Dev] Re: [Asterisk-cvs] asterisk/channels chan_sip.c,1.248,1.249   Jeremy McNamara
      • [Asterisk-Dev] Re: [Asterisk-cvs] asterisk/channels chan_sip.c,1.248,1.249   Olle E. Johansson
      • [Asterisk-Dev] Re: [Asterisk-cvs] asterisk/channels chan_sip.c,1.248,1.249   asterisk at lists.styx.org
      • [Asterisk-Dev] Re: [Asterisk-cvs] asterisk/channels chan_sip.c,1.248,1.249   Olle E. Johansson
      • [Asterisk-Dev] Re: [Asterisk-cvs] asterisk/channels chan_sip.c,1.248,1.249   asterisk at lists.styx.org
      • [Asterisk-Dev] Re: [Asterisk-cvs] asterisk/channels chan_sip.c,1.248,1.249   asterisk at lists.styx.org
      • [Asterisk-Dev] Re: [Asterisk-cvs] asterisk/channels chan_sip.c,1.248,1.249   Conroy, Lawrence (SMTP)
      • [Asterisk-Dev] Re: [Asterisk-cvs] asterisk/channels chan_sip.c,1.248,1.249   asterisk at lists.styx.org
      • [Asterisk-Dev] Re: [Asterisk-cvs] asterisk/channels chan_sip.c,1.248,1.249   Olle E. Johansson
      • [Asterisk-Dev] Re: [Asterisk-cvs] asterisk/channels chan_sip.c,1.248,1.249   asterisk at lists.styx.org
      • [Asterisk-Dev] Re: [Asterisk-cvs] asterisk/channels chan_sip.c,1.248,1.249   Olle E. Johansson
    • [Asterisk-Dev] Re: [Asterisk-cvs] asterisk/channels chan_sip.c,1.248,1.249   Olle E. Johansson
    • [Asterisk-Dev] Re: [Asterisk-cvs] asterisk/channels chan_sip.c,1.248,1.249   asterisk at lists.styx.org
• [Asterisk-Dev] [Fwd: Re: [Sip-implementors] Call hold questions]   Olle E. Johansson
• [Asterisk-Dev] Athlon-XP and AMD kernels, still a problem?   Chris Albertson
• [Asterisk-Dev] Asterisk IP PBX and ENUM - tutorial slides   asterisk at lists.styx.org
  • [Asterisk-Dev] Asterisk IP PBX and ENUM - tutorial slides   John Todd
    • [Asterisk-Dev] Asterisk IP PBX and ENUM - tutorial slides   Jan Janak
• [Asterisk-Dev] ENUM Lookup -- X-IAX2   asterisk at lists.styx.org
• [Asterisk-Dev] ztdummy   Raj Kiran Talusani
  • [Asterisk-Dev] ztdummy   Steven Critchfield
• [Asterisk-Dev] [PATCH] unique channel identifiers for the LINK/UNLINK manager events   Klaus-Peter Junghanns
  • [Asterisk-Dev] [PATCH] unique channel identifiers for the LINK/UNLINK manager events   Olle E. Johansson
• [Asterisk-Dev] NTT FSK - Japanese Caller ID   Isamar Maia
• [Asterisk-Dev] bugs.digium.com and acknowledged status   Brian West
• [Asterisk-Dev] channel.c ast_call + ast_transfer OR instead of AND???   Michiel Betel
  • [Asterisk-Dev] channel.c ast_call + ast_transfer OR instead of AND???   Mark Spencer
    • [Asterisk-Dev] channel.c ast_call + ast_transfer OR instead of AND???   Michiel Betel
      • [Asterisk-Dev] channel.c ast_call + ast_transfer OR instead of AND???   Brian West
• [Asterisk-Dev] DEBUG_THREADS in * makefile DeadLocks   TC
  • [Asterisk-Dev] HowTo Debug a DeadLock in * was [DEBUG_THREADS in * makefile DeadLocks]   TC
• [Asterisk-Dev] SIP Registration Cancellation   Kang.ChenJi at c3smail.monmouth.army.mil
• [Asterisk-Dev] Mac OS X - Thanks   Conroy, Lawrence (SMTP)
  • [Asterisk-Dev] Mac OS X - Thanks   Olle E. Johansson
    • [Asterisk-Dev] Mac OS X - Thanks   Conroy, Lawrence (SMTP)
• [Asterisk-Dev] Adding caller ID playback to VM   Steve Rodgers
• [Asterisk-Dev] Proposed solution for exit code priority jumps   John Todd
• [Asterisk-Dev] help needed - SIP register response contact   Conroy, Lawrence (SMTP)
• [Asterisk-Dev] IAX phones and CPU usage problem   Claude Klimos
• [Asterisk-Dev] cdr_pgsql load problem   Dave Weis
  • [Asterisk-Dev] cdr_pgsql load problem   Vince Vielhaber
    • [Asterisk-Dev] cdr_pgsql load problem   Dave Weis
    • [Asterisk-Dev] cdr_pgsql load problem   Vince Vielhaber

Last message date: Sun Nov 30 20:25:53 MST 2003
Archived on: Tue Sep 5 14:26:47 MST 2006

• Messages sorted by: [ subject ] [ author ] [ date ]
• More info on this list...