490 messages

com.digium.lists.asterisk-dev [All Lists]

2005 December [All Months]

Page 1 (Messages 1 to 100): 1 2 3 4 5

[Asterisk-Dev] TDM400 answering POTS voicemail polarity reversal - Rich Adamson
[Asterisk-Dev] Help Debugging Dropped Call Audio - Possibly Fixed - Florian Overkamp
[Asterisk-Dev] Help Debugging Dropped Call Audio - Possibly Fixed - Kevin P. Fleming
[Asterisk-Dev] Help Debugging Dropped Call Audio - Possibly Fixed - SteveK
[Asterisk-Dev] Help Debugging Dropped Call Audio - Possibly Fixed - Kevin P. Fleming
[Asterisk-Dev] calloc vs malloc ? - Tilghman Lesher
[Asterisk-Dev] Help Debugging Dropped Call Audio - Possibly Fixed - Florian Overkamp
[Asterisk-Dev] Help Debugging Dropped Call Audio - Possibly Fixed - Kevin P. Fleming
[Asterisk-Dev] Help Debugging Dropped Call Audio - Possibly Fixed - Matt Roth
[Asterisk-Dev] calloc vs malloc ? - Russell Bryant
[Asterisk-Dev] Help Debugging Dropped Call Audio - Possibly Fixed - Matthew Roth
[Asterisk-Dev] Help Debugging Dropped Call Audio - Possibly Fixed - Matthew Roth
[Asterisk-Dev] Help Debugging Dropped Call Audio - Possibly Fixed - Kevin P. Fleming
[Asterisk-Dev] Help Debugging Dropped Call Audio - Mike Benoit
[Asterisk-Dev] ast_channel behaviour - Atif Rasheed
[Asterisk-Dev] SQLite Realtime Driver - Steven Sokol
[Asterisk-Dev] Help Debugging Dropped Call Audio - Matthew Roth
[Asterisk-Dev] SQLite Realtime Driver - Steven Critchfield
[Asterisk-Dev] Help Debugging Dropped Call Audio - Possibly Fixed - Matthew Roth
[Asterisk-Dev] Help Debugging Dropped Call Audio - Possibly Fixed - Matthew Roth
[Asterisk-Dev] Help Debugging Dropped Call Audio - Matthew Roth
[Asterisk-Dev] Help Debugging Dropped Call Audio - Possibly Fixed - Kevin P. Fleming
[Asterisk-Dev] Help Debugging Dropped Call Audio - Possibly Fixed - Kevin P. Fleming
[Asterisk-Dev] ChanSpy() records files with funky permissions - Juan Carlos Castro y Castro
[Asterisk-Dev] channel monitoring - use MixMonitor instead of Monitor - Wolfgang Pichler
[Asterisk-Dev] channel monitoring - use MixMonitor instead of Monitor - BJ Weschke
[Asterisk-Dev] channel monitoring - use MixMonitor instead of Monitor - Wolfgang Pichler
[Asterisk-Dev] ChanSpy() records files with funky permissions - Alexander Lopez
[Asterisk-Dev] Re: [Asterisk-Users] Asterisk does not handle call from a Cisco IAD correctly - James Sizemore
[Asterisk-Dev] Re: [Asterisk-Users] Asterisk does not handle call from a Cisco IAD correctly - Tilghman Lesher
[Asterisk-Dev] Re: [Asterisk-Users] Asterisk does not handle call from a Cisco IAD correctly - Tilghman Lesher
[Asterisk-Dev] ChanSpy() records files with funky permissions - Alexander Lopez
[Asterisk-Dev] Re: [Asterisk-Users] Asterisk does not handle call from a Cisco IAD correctly - Kevin P. Fleming
[Asterisk-Dev] Asterisk extra logging to file - ast guy
[Asterisk-Dev] Asterisk extra logging to file - BJ Weschke
[Asterisk-Dev] Asterisk extra logging to file - ast guy
[Asterisk-Dev] Asterisk extra logging to file - BJ Weschke
[Asterisk-Dev] Re: [Asterisk-Users] Asterisk does not handle call from a Cisco IAD correctly - James Sizemore
[Asterisk-Dev] asterisk 1.2 g729 compile errors - hemant surjuse
[Asterisk-Dev] asterisk 1.2 g729 compile errors - Steven Critchfield
[Asterisk-Dev] Help Debugging Dropped Call Audio - Steven Critchfield
[Asterisk-Dev] ztdummy? is it necessary? - Jason DiCioccio
[Asterisk-Dev] ztdummy? is it necessary? - BJ Weschke
[Asterisk-Dev] Help Debugging Dropped Call Audio - Possibly Fixed - Matt Roth
[Asterisk-Dev] Re: ztdummy? is it necessary? - Jason DiCioccio
[Asterisk-Dev] Re: ztdummy? is it necessary? - Dan Austin
[Asterisk-Dev] Re: ztdummy? is it necessary? - Andrew Latham
[Asterisk-Dev] Re: ztdummy? is it necessary? - Jason DiCioccio
[Asterisk-Dev] any reason for #define FREE in the code ? - Luigi Rizzo
[Asterisk-Dev] Re: ztdummy? is it necessary? - Steven Critchfield
[Asterisk-Dev] Re: ztdummy? is it necessary? - Tzafrir Cohen
[Asterisk-Dev] any reason for #define FREE in the code ? - Kevin P. Fleming
[Asterisk-Dev] any reason for #define FREE in the code ? - Luigi Rizzo
[Asterisk-Dev] any reason for #define FREE in the code ? - Kevin P. Fleming
[Asterisk-Dev] Re: ztdummy? is it necessary? - Eric "ManxPower" Wieling
[Asterisk-Dev] Re: ztdummy? is it necessary? - North Antara
[Asterisk-Dev] Re: ztdummy? is it necessary? - Olle E Johansson
[Asterisk-Dev] Re: ztdummy? is it necessary? - Christian Richter
[Asterisk-Dev] Re: ztdummy? is it necessary? - Kevin P. Fleming
[Asterisk-Dev] ast_callerid_parse - Luigi Rizzo
[Asterisk-Dev] Realtime call controll - Kaloyan Kovachev
[Asterisk-Dev] Anybody experienced infinite loops in pbx_substitute_variables_helper_full? - Juan Carlos Castro y Castro
SV: [Asterisk-Dev] Realtime call controll - Kaloyan Kovachev
[Asterisk-Dev] I AM A DUMBASS (was: Anybody experienced infinite loops in...) - Juan Carlos Castro y Castro
[Asterisk-Dev] I AM A DUMBASS (was: Anybody experienced infiniteloops in...) - Alexander Lopez
SV: [Asterisk-Dev] Realtime call controll - Kaloyan Kovachev
[Asterisk-Dev] C++ AGI debuggin - Matthew A. Nicholson
[Asterisk-Dev] ast_callerid_parse - Brian Capouch
[Asterisk-Dev] ast_callerid_parse - BJ Weschke
[Asterisk-Dev] ast_callerid_parse - Kevin P. Fleming
[Asterisk-Dev] ast_callerid_parse - Olle E Johansson
[Asterisk-Dev] Packetization discussion - Dan Austin
[Asterisk-Dev] Asterisk does not handle call from a Cisco IAD correctly - James Sizemore
[Asterisk-Dev] Asterisk does not handle call from a Cisco IAD correctly - Kevin P. Fleming
[Asterisk-Dev] ast_callerid_parse - Luigi Rizzo
[Asterisk-Dev] RPID Issue - Ray Van Dolson
[Asterisk-Dev] ast_callerid_parse - Olle E Johansson
[Asterisk-Dev] Race issue in channel.c involving uniqueint on Asterisk 1.2.1 - Werner Johansson
[Asterisk-Dev] Race issue in channel.c involving uniqueint on Asterisk 1.2.1 - Olle E Johansson
[Asterisk-Dev] Race issue in channel.c involving uniqueint on Asterisk 1.2.1 [bugid 6091] - Werner Johansson
[Asterisk-Dev] Race issue in channel.c involving uniqueint on Asterisk 1.2.1 - Luigi Rizzo
[Asterisk-Dev] Problem on ZAP channel - rbra...@adiance.com
[Asterisk-Dev] Problem on ZAP channel - Steve Totaro
[Asterisk-Dev] Problem on ZAP channel - Steven
[Asterisk-Dev] Race issue in channel.c involving uniqueint on Asterisk 1.2.1 - Tilghman Lesher
[Asterisk-Dev] Race issue in channel.c involving uniqueint on Asterisk 1.2.1 - Luigi Rizzo
[Asterisk-Dev] Race issue in channel.c involving uniqueint onAsterisk 1.2.1 - Luigi Rizzo
[Asterisk-Dev] Question on using system(find args -exec rm {} \;) - bd...@prosodiemail.com
[Asterisk-Dev] Problem on ZAP channel - Steve Totaro
[Asterisk-Dev] Problem on ZAP channel - Steve Totaro
[Asterisk-Dev] Coding Standard for Asterisk? - Steve Murphy
[Asterisk-Dev] Race issue in channel.c involving uniqueint onAsterisk 1.2.1 - Luigi Rizzo
[Asterisk-Dev] chan_sip.c : ignoring domain part for incoming INVITE's causes conflicts between domains? - Bruno Rocha
[Asterisk-Dev] Coding Standard for Asterisk? - Steven Critchfield
[Asterisk-Dev] Fax Support - rbra...@adiance.com
[Asterisk-Dev] chan_sip.c : ignoring domain part for incoming INVITE's causes conflicts between domains? - Olle E Johansson
[Asterisk-Dev] LOCAL_USER_ADD / LOCAL_USER_REMOVE semantics ? - Luigi Rizzo
[Asterisk-Dev] LOCAL_USER_ADD / LOCAL_USER_REMOVE semantics ? - Russell Bryant
[Asterisk-Dev] chan_sip.c : ignoring domain part for incomingINVITE's causes conflicts between domains? - Enzo Michelangeli
[Asterisk-Dev] Voicemail through outlook - S.Ammad Jami

Page 1 (Messages 1 to 100): 1 2 3 4 5