948 messages

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

2006 September [All Months]

Page 1 (Messages 1 to 100): 1 2 3 4 5 6 7 8 9 10

[asterisk-dev] Re: [asterisk-announce] Digium G.729 codec binaries updated - Roy Sigurd Karlsbakk
[asterisk-dev] Re: [asterisk-announce] Digium G.729 codec binaries updated - Kevin P. Fleming
[asterisk-dev] Re: [asterisk-announce] Digium G.729 codec binaries updated - Roy Sigurd Karlsbakk
[asterisk-dev] Re: [asterisk-announce] Digium G.729 codec binaries updated - Kevin P. Fleming
[asterisk-dev] SIP Interoperability in Asterisk? - Olle E Johansson
[asterisk-dev] Asterisk Presence - Alexandr Olekhnovich
[asterisk-dev] A rewrite of Asterisk::AGI - Jean-Michel Hiver
[asterisk-dev] Asterisk, Asterisk-Addons, Zaptel and Libpri 1.4 betas released! - Matt Riddell (IT)
[asterisk-dev] Asterisk Presence - Johansson Olle E
[asterisk-dev] SIP to SIP unhold and no voice - Aragon Gouveia
[asterisk-dev] (no subject) - Yoann Aubineau
[asterisk-dev] To bweschke regarding app FollowMe - BJ Weschke
[asterisk-dev] core dump asterisk 1.2.12.1 - ChanSpy ??? - Dov Bigio
[asterisk-dev] Asterisk 1.4 and spandsp3/txfax/rxfax - aste...@ntplx.net
[asterisk-dev] 1.4.0-beta2 and g729 - Dan Austin
[asterisk-dev] 1.4.0-beta2 and g729 - Julian Lyndon-Smith
[asterisk-dev] core dump asterisk 1.2.12.1 - ChanSpy ??? - Sebastian Auriol
[asterisk-dev] A rewrite of Asterisk::AGI - Peter Beckman
[asterisk-dev] Developer Summit at Astricon USA 2006 - Kevin P. Fleming
[asterisk-dev] 1.4.0-beta2 and g729 - Dan Austin
[asterisk-dev] Asterisk behind NAT1 and SIP Phone behind NAT2 - Jay Ray
[asterisk-dev] OT But So Ungodly Important - Rushowr
[asterisk-dev] Re: [asterisk-announce] Digium G.729 codec binaries updated for Asterisk 1.4 beta - Roy Sigurd Karlsbakk
[asterisk-dev] Some 1.4.0-beta2 and Solaris 10 issues... - Jason Parker
[asterisk-dev] Re: [asterisk-announce] Digium G.729 codec binaries updated for Asterisk 1.4 beta - Kevin P. Fleming
[asterisk-dev] Asterisk 1.4 and spandsp3/txfax/rxfax - aste...@ntplx.net
[asterisk-dev] Asterisk 1.4 and spandsp3/txfax/rxfax - Kevin P. Fleming
[asterisk-dev] Some 1.4.0-beta2 and Solaris 10 issues... - Jason Parker
[asterisk-dev] Question about hup_handler & deadlocks - Jay Hoover
[SPAM] Re: [asterisk-dev] Problems compiling chan_h323 of 1.4beta2 version - Vlasis Hatzistavrou Mailing Lists Account
[asterisk-dev] Question about hup_handler & deadlocks - Tilghman Lesher
[asterisk-dev] Asterisk 1.4 and spandsp3/txfax/rxfax - Kevin P. Fleming
[asterisk-dev] Question about hup_handler & deadlocks - Jay Hoover
[asterisk-dev] SIP Interoperability in Asterisk? - Dinesh Nair
[asterisk-dev] Dual core - Raphael Jacquot
[asterisk-dev] Question about hup_handler & deadlocks - Tim Panton
[asterisk-dev] Solution allowing Asterisk send and receiving serial data from/to pbx - Paulo Garcia
[asterisk-dev] Codec not changed when making an Attended xfer (REFER) - Chan Kwang Mien
[asterisk-dev] Advice of charge - Klaus Darilion
[asterisk-dev] Advice of charge - Christian Richter
[asterisk-dev] ZAPTEL in trunk - bug/hardcoded d-chan in zaptel? E1-T1 issue. - Anton
[asterisk-dev] ZAPTEL in trunk - bug/hardcoded d-chan in zaptel? E1-T1 issue. - Anton
[asterisk-dev] CDR-csv records - Bro
[asterisk-dev] ZAPTEL in trunk - bug/hardcoded d-chan in zaptel?E1-T1 issue. - Anton
[asterisk-dev] Problem with 1.4b2 and native sounds - Matt O'Gorman
[asterisk-dev] Re: Dual core - Tomislav Parčina
[asterisk-dev] Problem with 1.4b2 and native sounds - Dan Austin
[asterisk-dev] New codec support in chan_skinny - Dan Austin
[asterisk-dev] New codec support in chan_skinny - Dan Austin
[asterisk-dev] Voicemail imap storage. - Kevin P. Fleming
[asterisk-dev] Rate limiting traffic to address potential DoS issues? - Steven
[asterisk-dev] Rate limiting traffic to address potential DoS issues? - Kristian Kielhofner
[asterisk-dev] Re: Rate limiting and firewall failures on DoS - J. Oquendo
[asterisk-dev] Extension exact match or alphabetical still? - Anton
[asterisk-dev] Rate limiting traffic to address potential DoS issues? - Tim Panton
[asterisk-dev] How to send SIP replies to another Asterisk? - Mikael Magnusson
[asterisk-dev] chan_sccp rtp patch for 1.4? - Andreas Anderson
[asterisk-dev] chan_sccp rtp patch for 1.4? - Pavel Jezek
[asterisk-dev] Passing DTMF through MeetMe - Matt Florell
[asterisk-dev] Re: Advice of charge - Tomislav Parčina
[asterisk-dev] G.729 codec problem - Rosa De Santis
[asterisk-dev] Rate limiting traffic to address potential DoS issues? - Jared Smith
[asterisk-dev] Re: Passing DTMF through MeetMe - Tony Mountifield
[asterisk-dev] Rate limiting traffic to address potential DoS issues? - Kristian Kielhofner
Load Testing (was Re: [asterisk-dev] Rate limiting traffic to address potential DoS issues?) - Jeremy McNamara
[asterisk-dev] Re: AGI SayNumber gender option - Tony Mountifield
[asterisk-dev] AGI SayNumber gender option - Andrew Latham
Load Testing (was Re: [asterisk-dev] Rate limiting traffic to address potential DoS issues?) - Greg Boehnlein
[asterisk-dev] Bug 5811 - Aaron Daniel
[asterisk-dev] displaysystemname in manager.conf - Kevin P. Fleming
[asterisk-dev] unable to call AT&T audio conference bridge - asterisk-user
[asterisk-dev] List of Zaptel/Libpri 1.2->1.4 changes - Matthew Fredrickson
[asterisk-dev] Fwd: Configuring Asterisk 1.4-beta2 to work with jingle - Raffaele Porzio
[asterisk-dev] how to compile and install chan_jingle.so - Raffaele Porzio
[asterisk-dev] Checking h323.h presence... no??? - Vlasis Hatzistavrou
[asterisk-dev] Re: AGI SayNumber gender option - Andrei Koulik
[asterisk-dev] Checking h323.h presence... no??? - Paul Cadach
[asterisk-dev] Crash with latest 1.2 svn r43186 - Martin Vít
[asterisk-dev] Re: Advice of charge - Tomislav Parčina
[asterisk-dev] how to compile and install chan_jingle.so - Raffaele Porzio
[asterisk-dev] Checking h323.h presence... no??? - Vlasis Hatzistavrou
[asterisk-dev] Re: Advice of charge - Klaus Darilion
[asterisk-dev] shutting down zaptel spans - Kevin P. Fleming
[asterisk-dev] Re: Crash with latest 1.2 svn r43186 - Martin Vít
[asterisk-dev] shutting down zaptel spans - Paul Cadach
[asterisk-dev] MOH not working on custom channel driver - Earle Clubb
[asterisk-dev] MOH not working on custom channel driver - Kevin P. Fleming
[asterisk-dev] shutting down zaptel spans - Klaus Darilion
[asterisk-dev] shutting down zaptel spans - C. Maj
[asterisk-dev] shutting down zaptel spans - Tzafrir Cohen
[asterisk-dev] MOH not working on custom channel driver - Earle Clubb
[asterisk-dev] MOH not working on custom channel driver - Kevin P. Fleming
[asterisk-dev] Re: kpfleming: branch 1.2 r43895 - /branches/1.2/cli.c - Kevin P. Fleming
[asterisk-dev] MOH not working on custom channel driver - Paul Cadach
[asterisk-dev] Fwd: How can I unistall Asterisk? - Raphaël Jacquot
[asterisk-dev] Fwd: How can I unistall Asterisk? - Kevin P. Fleming
[asterisk-dev] several compile problems after SVN 44056 changes - Luigi Rizzo
[asterisk-dev] several compile problems after SVN 44056 changes - Luigi Rizzo
[asterisk-dev] developers help - Michael Rozov
[asterisk-dev] Re: [asterisk-commits] pcadach: branch 1.4 r44090 - /branches/1.4/main/rtp.c - Kevin P. Fleming

Page 1 (Messages 1 to 100): 1 2 3 4 5 6 7 8 9 10