July 2006 Archives by date

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Starting: Sat Jul 1 02:07:46 MST 2006
Ending: Mon Jul 31 11:10:19 MST 2006
Messages: 552

• [asterisk-dev] Exceptionally long queue length queuing - Asterisk1.2.9.1   Roy Sigurd Karlsbakk
• [asterisk-dev] reference count on ast_netsock objects   Frédérick Lefebvre
• [asterisk-dev] Discussion about Bugreport 0007382: [patch] calc_metric and max_penalty   Michael Konietzny
• [asterisk-dev] ringinuse   Urban
• [asterisk-dev] Zaptel channel lock bug?   Anton
• [asterisk-dev] Zaptel channel lock bug?   Paul Cadach
• [asterisk-dev] Zaptel prilocaldialplan troubles   Wolfgang Zweimueller
• [asterisk-dev] Something Wrong with User's List?   Steve Totaro
• [asterisk-dev] Re: Asterisk, rport and RFC 3581 problems - Is this a bug?   Marc
• [asterisk-dev] Solaris Zaptel Update   Joseph Benden
• [asterisk-dev] Solaris Zaptel Update   Josip Gracin
• [asterisk-dev] Something Wrong with User's List?   Olle E Johansson
• [asterisk-dev] Character set for AMI/HTTP: UTF8 ?   Olle E Johansson
• [asterisk-dev] CDR Accounting wrong   René Enskat [Teamware GmbH]
• [asterisk-dev] app_queue   Mailing Lists
• [asterisk-dev] capture and convert voice stream to a certain format   Jamel A
• [asterisk-dev] app_queue   Johansson Olle E
• [asterisk-dev] Patch to app_queue to add annoucements to the caller   Tristan
• [asterisk-dev] Patch to app_queue to add annoucements to the caller   Jared Smith
• [asterisk-dev] realtime architecture blocks iax network thread   Dawid Mielnik
• [asterisk-dev] Character set for AMI/HTTP: UTF8 ?   Jared Smith
• [asterisk-dev] realtime architecture blocks iax network thread   Joshua Colp
• [asterisk-dev] app_queue   Mailing Lists
• [asterisk-dev] Re: realtime architecture blocks iax network thread   Tony Mountifield
• [asterisk-dev] RES: asterisk-dev Digest, Vol 24, Issue 5   Roberto
• [asterisk-dev] FreeBSD Asterisk Segmentation Faults   J. Oquendo
• [asterisk-dev] RES: asterisk-dev Digest, Vol 24, Issue 5   Steven
• [asterisk-dev] FreeBSD Asterisk Segmentation Faults   Mike Fedyk
• [asterisk-dev] FreeBSD Asterisk Segmentation Faults   J. Oquendo
• [asterisk-dev] app_queue   Mailing Lists
• [asterisk-dev] capture and convert voice stream to a certain format   Kohler, Jeffrey
• [asterisk-dev] Help with outbound dialing problems   Daniel Corbe
• [asterisk-dev] Help with outbound dialing problems   Jared Smith
• [asterisk-dev] Config for echo cancel boards   Roberto
• [asterisk-dev] Config for echo cancel boards   Malcolm C. Davenport
• [asterisk-dev] Show codecs   John Martin
• [asterisk-dev] Show codecs   John Martin
• [asterisk-dev] Help! Zap Startup failure: why is libpri not defined ?   Julian Lyndon-Smith
• [asterisk-dev] Weird transcoding error (SIP, local channels): sip_write: Asked to transmit frame type 64, while native formats is 256 (read/write = 64/256)   Matt King
• [asterisk-dev] realtime architecture blocks iax network thread   Dawid Mielnik
• [asterisk-dev] AOC-D passthrough - patch on bugs.digium.com   Alexander Mayrhofer
• [asterisk-dev] Show codecs   Olle E Johansson
• [asterisk-dev] Asterisk User presence implementation   MELARGA COULIBALY
• [asterisk-dev] No AMI event sent on dialplan variable changes via AGI and setting CLID on dial   Tony Howat
• [asterisk-dev] Asterisk User presence implementation   Steve Langstaff
• [asterisk-dev] WG: CDR Accounting wrong   René Enskat [Teamware GmbH]
• [asterisk-dev] No AMI event sent on dialplan variable changes via AGI and setting CLID on dial   Olle E Johansson
• [asterisk-dev] The app_lock mutex in chan_agent.c   Kevin P. Fleming
• [asterisk-dev] SIP + T.38 == 'fax' extension?   Kevin P. Fleming
• [asterisk-dev] Proposal to seperate qualify & keep alive   Kevin P. Fleming
• [asterisk-dev] Proposal to seperate qualify & keep alive   Kevin P. Fleming
• [asterisk-dev] reference count on ast_netsock objects   Kevin P. Fleming
• [asterisk-dev] ringinuse   Kevin P. Fleming
• [asterisk-dev] app_queue   Kevin P. Fleming
• [asterisk-dev] Help! Zap Startup failure: why is libpri not defined ?   Kevin P. Fleming
• [asterisk-dev] Help! Zap Startup failure: why is libpri not defined ?   Julian Lyndon-Smith
• [asterisk-dev] The app_lock mutex in chan_agent.c   Stephen Davies
• [asterisk-dev] Proposal to seperate qualify & keep alive   Johansson Olle E
• [asterisk-dev] Character set for AMI/HTTP: UTF8 ?   Jeffrey C. Ollie
• [asterisk-dev] ringinuse   BJ Weschke
• [asterisk-dev] poll delay in ast_waitfor_nandfds   Jiang Jinke
• [asterisk-dev] realtime architecture blocks iax network thread   Joshua Colp
• [asterisk-dev] Multiple MySQL Database ‘Syncing’ Within an Asterisk Cluster   JR Richardson
• [asterisk-dev] Multiple MySQL Database ‘Syncing’ Within an Asterisk Cluster   Joseph Benden
• [asterisk-dev] Zaptel and hardware usages   Joseph Benden
• [asterisk-dev] Help with outbound dialing problems   Daniel Corbe
• [asterisk-dev] Zaptel and hardware usages   Tilghman Lesher
• [asterisk-dev] Proposal to seperate qualify & keep alive   Eric "ManxPower" Wieling
• [asterisk-dev] Voicemail NFS Cutting out   Bill Kunyiha
• [asterisk-dev] Voicemail NFS Cutting out   alex at pilosoft.com
• [asterisk-dev] Asterisk Bounty: Additional Dial | L options   webmaster at 5nf.net
• [asterisk-dev] Asterisk Bounty: Additional Dial | L options   Stephan A. Edelman
• [asterisk-dev] Strange interrupt issue with zaptel and chan_ss7   Kai Militzer
• Re: Re: [asterisk-dev] Multiple MySQL Database ‘Syncing’ Within an Asterisk Cluster   Fabe
• [asterisk-dev] Adding max_open_files option to asterisk.conf?   Roy Sigurd Karlsbakk
• [asterisk-dev] Adding max_open_files option to asterisk.conf?   Roy Sigurd Karlsbakk
• [asterisk-dev] [PATCH] CLID spoof option added to Dial application   Tony Howat
• [asterisk-dev] Re: [PATCH] CLID spoof option added to Dial application   Benny Amorsen
• [asterisk-dev] Adding max_open_files option to asterisk.conf?   Johansson Olle E
• [asterisk-dev] [PATCH] CLID spoof option added to Dial application   Johansson Olle E
• [asterisk-dev] Re: [PATCH] CLID spoof option added to Dial application   Roy Sigurd Karlsbakk
• [asterisk-dev] Adding max_open_files option to asterisk.conf?   Roy Sigurd Karlsbakk
• [asterisk-dev] feature "zaptel Disabled echo canceller because of tone (rx) on channel " work?   Vitaly Oborsky
• [asterisk-dev] realtime architecture blocks iax network thread   Dawid Mielnik
• [asterisk-dev] Re: [PATCH] CLID spoof option added to Dial application   Johansson Olle E
• [asterisk-dev] [PATCH] CLID spoof option added to Dial application   Alexander Lopez
• [asterisk-dev] Asterisk Bounty: Additional Dial | L options   Kaloyan Kovachev
• [asterisk-dev] Strange interrupt issue with zaptel and chan_ss7   Steve Underwood
• [asterisk-dev] configure script tests for mysql   Dan Austin
• [asterisk-dev] Out of tree applications   Dan Austin
• [asterisk-dev] Out of tree applications   Kevin P. Fleming
• [asterisk-dev] Out of tree applications   Dan Austin
• [asterisk-dev] Out of tree applications   Kevin P. Fleming
• [asterisk-dev] Out of tree applications   Dan Austin
• [asterisk-dev] RTP packetization (bugid 5162)   Dan Austin
• [asterisk-dev] Out of tree applications   Roy Sigurd Karlsbakk
• [asterisk-dev] Out of tree applications   Dan Austin
• [asterisk-dev] Hello everybody   Sameer Tamsekar
• [asterisk-dev] Adding max_open_files option to asterisk.conf?   Dinesh Nair
• [asterisk-dev] Re: [PATCH] CLID spoof option added to Dial application   Dinesh Nair
• [asterisk-dev] Manager Interface: Newchannel vs. Newstate   Brian Capouch
• [asterisk-dev] Strange interrupt issue with zaptel and chan_ss7   Kai Militzer
• [asterisk-dev] Manager Interface: Newchannel vs. Newstate   Russell Bryant
• [asterisk-dev] Help for new Asterisk developper   MELARGA COULIBALY
• [asterisk-dev] SOCKET   Sameer Tamsekar
• [asterisk-dev] SOCKET   Sameer Tamsekar
• [asterisk-dev] FastAGI behaviour in trunk changed recently?   Alistair Cunningham
• [asterisk-dev] segfaults in rtp.c   Julian Lyndon-Smith
• [asterisk-dev] segfaults in rtp.c   North Antara
• [asterisk-dev] Bug#7505 Introduce missing cause "Non-selected user clearing"   Armin Schindler
• [asterisk-dev] Bug#7505 Introduce missing cause "Non-selected user clearing"   Kevin P. Fleming
• [asterisk-dev] Changing Queue behaviour on agent hangup   Tristan
• [asterisk-dev] Bug#7505 Introduce missing cause "Non-selected user clearing"   Armin Schindler
• [asterisk-dev] Bug#7505 Introduce missing cause "Non-selected user clearing"   Klaus Darilion
• [asterisk-dev] Adding IE to PRI messages   Klaus Darilion
• [asterisk-dev] Help for Asterisk New developper   MELARGA COULIBALY
• Re: [asterisk-dev] Multiple MySQL Database ‘Syncing’ Within an Asterisk Cluster   Matt Riddell (IT)
• [asterisk-dev] chan_zap questions   Klaus Darilion
• [asterisk-dev] WG: CDR ist getting wrong status   René Enskat [Teamware GmbH]
• [asterisk-dev] chan_zap questions   Armin Schindler
• [asterisk-dev] Streamlining RTP code.   Constantine Filin
• [asterisk-dev] Incoming Jingle Calls   Landon McDowell
• [asterisk-dev] Incoming Jingle Calls   Eric "ManxPower" Wieling
• [asterisk-dev] Re: [svn-commits] trunk r1216 - /trunk/Makefile   Tzafrir Cohen
• [asterisk-dev] Incoming Jingle Calls   Landon McDowell
• [asterisk-dev] SIP To: header   sip
• [asterisk-dev] chan_zap questions   Kevin P. Fleming
• [asterisk-dev] Streamlining RTP code.   Kevin P. Fleming
• [asterisk-dev] Jingle Out Config   Landon McDowell
• [asterisk-dev] chan_zap questions   Armin Schindler
• [asterisk-dev] Hmm. "Virtual memory exhausted" building SVN trunk   Brian Capouch
• [asterisk-dev] Hmm. "Virtual memory exhausted" building SVN trunk   Luigi Rizzo
• [asterisk-dev] Hmm. "Virtual memory exhausted" building SVN trunk   Brian Capouch
• [asterisk-dev] Hmm. "Virtual memory exhausted" building SVN trunk   Kaloyan Kovachev
• [asterisk-dev] RE: asterisk-dev Digest, Vol 24, Issue 23   Constantine Filin
• [asterisk-dev] chan_zap questions   Klaus Darilion
• [asterisk-dev] Re: [asterisk-ss7] Strange interrupt issue with zaptel and chan_ss7   Kai Militzer
• [asterisk-dev] FastAGI behaviour in trunk changed recently?   Russell Bryant
• [asterisk-dev] SOCKET   Russell Bryant
• [asterisk-dev] SOCKET   Marc Haisenko
• [asterisk-dev] FastAGI behaviour in trunk changed recently?   Alistair Cunningham
• [asterisk-dev] chan_zap questions   Armin Schindler
• [asterisk-dev] chan_zap questions   Klaus Darilion
• [asterisk-dev] chan_zap questions   Armin Schindler
• [asterisk-dev] chan_zap questions   Klaus Darilion
• [asterisk-dev] SIP To: header   sip
• [asterisk-dev] chan_zap questions   Armin Schindler
• [asterisk-dev] chan_zap questions   Klaus Darilion
• [asterisk-dev] libpri ASN1_FIXUP question   Klaus Darilion
• [asterisk-dev] chan_zap questions   Armin Schindler
• [asterisk-dev] Bridge and queue question   Tristan
• [asterisk-dev] libpri ASN1_FIXUP question   Paul Cadach
• [asterisk-dev] can i send proper DNID to manager on hangup   Kamran Ahmad
• [asterisk-dev] FastAGI behaviour in trunk changed recently?   Russell Bryant
• [asterisk-dev] chan_sip.c request URI parsing   Ryan Mitchell
• [asterisk-dev] Multiple MySQL Database ‘Syncing’ Within an Asterisk Cluster   Joseph Benden
• [asterisk-dev] fix for domains with empty context   Andrey Rouskol
• [asterisk-dev] FastAGI behaviour in trunk changed recently?   Alistair Cunningham
• [asterisk-dev] chan_zap questions   Klaus Darilion
• [asterisk-dev] chan_zap questions   Armin Schindler
• [asterisk-dev] chan_zap questions   Klaus Darilion
• [asterisk-dev] chan_zap questions   Armin Schindler
• [asterisk-dev] fix for domains with empty context   John Todd
• [asterisk-dev] chan_zap questions   Klaus Darilion
• [asterisk-dev] chan_zap questions   Armin Schindler
• [asterisk-dev] chan_zap questions   Kevin P. Fleming
• [asterisk-dev] Bug in ast_rtp_read ???   Constantine Filin
• [asterisk-dev] chan_zap questions   Klaus Darilion
• [asterisk-dev] Hi   Ramsundar
• [asterisk-dev] chan_zap questions   Armin Schindler
• [asterisk-dev] Bug in ast_rtp_read ???   John Martin
• [asterisk-dev] Bug in ast_rtp_read ???   Russell Bryant
• AOC-support for PRI (was: [asterisk-dev] chan_zap questions)   Klaus Darilion
• [asterisk-dev] Some confusion about threads   Sameer
• [asterisk-dev] SIP INVITE retransmission, erroneous   Nikolaos Markolefas
• [asterisk-dev] SIP INVITE retransmission, erroneous   Kevin P. Fleming
• [asterisk-dev] Asterisk 1.2.10 and Zaptel 1.2.7 released!   Asterisk Development Team
• [asterisk-dev] SIP INVITE retransmission, erroneous   Jay Greasley
• [asterisk-dev] Maybe... Just maybe   J. Oquendo
• [asterisk-dev] re : maximum length of SIP VIA header   Dennis Yeung
• [asterisk-dev] Maybe... Just maybe   Matt Riddell (NZ)
• [asterisk-dev] asterisk operator script   HeverinM at cs.com
• [asterisk-dev] Using the Video Hardware as a DSP?   Haider Raza
• [asterisk-dev] Using the Video Hardware as a DSP?   Eric "ManxPower" Wieling
• [asterisk-dev] asterisk operator script   Peter Beckman
• [asterisk-dev] chan_phone.c:41:29: error: linux/compiler.h: No such file or directory   Axel Thimm
• [asterisk-dev] Blog Asterisk e VoIP   sergio sicari
• [asterisk-dev] Streamlining RTP code.   Roy Sigurd Karlsbakk
• [asterisk-dev] chan_phone.c:41:29: error: linux/compiler.h: No such file or directory   Tzafrir Cohen
• [asterisk-dev] chan_phone.c:41:29: error: linux/compiler.h: No such file or directory   Kevin P. Fleming
• [asterisk-dev] Closed bug ID 0007438 - I am able to reproduce   Tass Iliopoulos
• [asterisk-dev] Closed bug ID 0007438 - I am able to reproduce   Tilghman Lesher
• [asterisk-dev] channels up   voip at gmx-topmail.de
• [asterisk-dev] Question to a feature concerning call pickup   kibeki at gmx.net
• [asterisk-dev] Queries regarding Echo Cancellation in Zaptel-1.2.6   Chan Kwang Mien
• [asterisk-dev] Queries regarding Echo Cancellation in Zaptel-1.2.6   Paul Cadach
• [asterisk-dev] Queries regarding Echo Cancellation in Zaptel-1.2.6   Chan Kwang Mien
• [asterisk-dev] Queries regarding Echo Cancellation in Zaptel-1.2.6   Steve Davies
• [asterisk-dev] Queries regarding Echo Cancellation in Zaptel-1.2.6   Nicolas Bougues
• [asterisk-dev] asterisk sending connects when it shouldn't   Simone Cittadini
• [asterisk-dev] Queries regarding Echo Cancellation in Zaptel-1.2.6   Eric "ManxPower" Wieling
• [asterisk-dev] Channel problems   voip at gmx-topmail.de
• [asterisk-dev] asterisk sending connects when it shouldn't (is there a q931-INFORMATION equivalent in IAX2 ?)   Simone Cittadini
• [asterisk-dev] Queries regarding Echo Cancellation in Zaptel-1.2.6   Steven
• [asterisk-dev] Does anybody know how to do a producer/consumer based buffered writer in C?   Matt King
• [asterisk-dev] asterisk sending connects when it shouldn't   Paul Cadach
• [asterisk-dev] asterisk sending connects when it shouldn't   Eric "ManxPower" Wieling
• [asterisk-dev] Asterisk Configuration Outgoing Calls   André Figueiredo
• [asterisk-dev] Asterisk Configuration Outgoing Calls   Steven
• [asterisk-dev] Voice format when using another SIP server   Jamel A
• [asterisk-dev] asterisk sending connects when it shouldn't (is there a q931-INFORMATION equivalent in IAX2 ?)   Matthew Fredrickson
• [asterisk-dev] MeetMe without Zaptel or ztdummy   Anthony LaMantia
• [asterisk-dev] Channel problems   Anthony LaMantia
• [asterisk-dev] MeetMe without Zaptel or ztdummy   Matt Florell
• [asterisk-dev] H.263 Buffer size   John Martin
• [asterisk-dev] MeetMe without Zaptel or ztdummy   Matt Riddell (NZ)
• [asterisk-dev] H.263 Buffer size   Tilghman Lesher
• [asterisk-dev] asterisk sending connects when it shouldn't   Simone Cittadini
• [asterisk-dev] H.263 Buffer size   Sergio García Murillo
• [asterisk-dev] asterisk sending connects when it shouldn't   Paul Cadach
• [asterisk-dev] Queries regarding Echo Cancellation in Zaptel-1.2.6   Chan Kwang Mien
• [asterisk-dev] asterisk sending connects when it shouldn't (pre-answer OT)   Simone Cittadini
• [asterisk-dev] H.263 Buffer size   Tilghman Lesher
• [asterisk-dev] H.263 Buffer size   Sergio García Murillo
• [asterisk-dev] H.263 Buffer size   John Martin
• [asterisk-dev] Queries regarding Echo Cancellation in Zaptel-1.2.6   Paul Cadach
• [asterisk-dev] yacc error when compiling newest Asterisk 1.2 on FC3   Juan Carlos Castro y Castro
• [asterisk-dev] Channel problems   voip at gmx-topmail.de
• [asterisk-dev] call-limit and problem with freezy phones.   Vitaly Oborsky
• [asterisk-dev] call-limit and problem with freezy phones.   Martin Vít
• [asterisk-dev] Implementing Paging on the Linksys SPA9XX phones   John Lange
• [asterisk-dev] H.263 Buffer size   John Lange
• [asterisk-dev] call-limit and problem with freezy phones. also freezy zap channels with x101p card.   Vitaly Oborsky
• [asterisk-dev] H.263 Buffer size   John Martin
• [asterisk-dev] Does anybody know how to do a producer/consumer based buffered writer in C?   Kevin P. Fleming
• [asterisk-dev] fix for domains with empty context   Roy Sigurd Karlsbakk
• [asterisk-dev] Channel problems   Kevin P. Fleming
• [asterisk-dev] Method to present patches and feature changes   Bart Fisher
• [asterisk-dev] Method to present patches and feature changes   North Antara
• [asterisk-dev] Method to present patches and feature changes   Jared Smith
• [asterisk-dev] Re: [Asterisk-Security] Day early and a dollar short? (IAX2 and SIP problems)   John Todd
• [asterisk-dev] Re: [Asterisk-Security] Day early and a dollar short? (IAX2 and SIP problems)   Kevin P. Fleming
• [asterisk-dev] Re: [Asterisk-Security] Day early and a dollar short? (IAX2 and SIP problems)   John Lange
• [asterisk-dev] Web Application Communication   David R.
• [asterisk-dev] Web Application Communication   Stefan Reuter
• [asterisk-dev] Web Application Communication   Matthew Nicholson
• [asterisk-dev] Web Application Communication   Stefan Reuter
• [asterisk-dev] Web Application Communication   Matthew Nicholson
• [asterisk-dev] Web Application Communication   David R.
• [asterisk-dev] Web Application Communication   Matthew Nicholson
• [asterisk-dev] Web Application Communication   Jean-Denis Girard
• [asterisk-dev] Queries regarding Echo Cancellation in Zaptel-1.2.6   Chan Kwang Mien
• [asterisk-dev] How to write new application for Asterisk by using C   PhongDo
• [asterisk-dev] Queries regarding Echo Cancellation in Zaptel-1.2.6   Paul Cadach
• [asterisk-dev] How to send DNIS(B-party number) in IAX trunk   Sam Liang
• [asterisk-dev] How to write new application for Asterisk by using C   Russell Bryant
• [asterisk-dev] How is asterisk code tested ?   Pranav Peshwe
• [asterisk-dev] H.263 Buffer size   Sergio García Murillo
• [asterisk-dev] app_voicemail generates locking errors since 1.2.10   Kai Militzer
• [asterisk-dev] Web Application Communication   Tim Panton
• [asterisk-dev] Web Application Communication   Stefan Reuter
• [asterisk-dev] How is asterisk code tested ?   Maxim Vexler
• [asterisk-dev] How is asterisk code tested ?   Paul Cadach
• [asterisk-dev] asterisk sending connects when it shouldn't (and request for feature)   Simone Cittadini
• [asterisk-dev] asterisk sending connects when it shouldn't (andrequest for feature)   Paul Cadach
• [asterisk-dev] Does anybody know how to do a producer/consumerbased buffered writer in C?   Asterisk
• [asterisk-dev] Re: Does anybody know how to do a producer/consumer based buffered writer in C?   Matt King
• [asterisk-dev] H.263 Buffer size   Rich Adamson
• [asterisk-dev] Re: [Asterisk-Security] Day early and a dollar short? (IAX2 and SIP problems)   Rich Adamson
• [asterisk-dev] Re: Does anybody know how to do a producer/consumer based buffered writer in C?   Kevin P. Fleming
• [asterisk-dev] Does anybody know how to do a producer/consumerbased buffered writer in C?   Kevin P. Fleming
• [asterisk-dev] Re: [Asterisk-Security] Day early and a dollar short? (IAX2 and SIP problems)   Kevin P. Fleming
• [asterisk-dev] Voice Rec Integration   Jeremi Bergman
• [asterisk-dev] Voice Rec Integration   Yaakov Menken
• [asterisk-dev] Voice Rec Integration   Jeremi Bergman
• [asterisk-dev] Does anybody know how to do a producer/consumer based buffered writer in C?   C. Maj
• [asterisk-dev] Voice Rec Integration   C. Maj
• [asterisk-dev] Voice Rec Integration   Jeremi Bergman
• [asterisk-dev] Voice Rec Integration   Jared Smith
• [asterisk-dev] Does anybody know how to do a producer/consumer based buffered writer in C?   Kevin P. Fleming
• [asterisk-dev] Beta testers needed for new sound files   Kevin P. Fleming
• [asterisk-dev] Beta testers needed for new sound files   John Lange
• [asterisk-dev] Beta testers needed for new sound files   Kevin P. Fleming
• [asterisk-dev] Beta testers needed for new sound files   Strom Carlson
• [asterisk-dev] Beta testers needed for new sound files (now version 1.4.1)   Kevin P. Fleming
• [asterisk-dev] Beta testers needed for new sound files (now version 1.4.1)   John Lange
• [asterisk-dev] Beta testers needed for new sound files (now version 1.4.1)   Kevin P. Fleming
• [asterisk-dev] Beta testers needed for new sound files (now version 1.4.1)   Andrew Kohlsmith
• [asterisk-dev] Beta testers needed for new sound files   Alberto Sagredo
• [asterisk-dev] Beta testers needed for new sound files (now version 1.4.1)   John Lange
• [asterisk-dev] Beta testers needed for new sound files   Steve Kennedy
• [asterisk-dev] Beta testers needed for new sound files (now version 1.4.1)   Strom Carlson
• [asterisk-dev] Beta testers needed for new sound files   Kevin P. Fleming
• [asterisk-dev] Beta testers needed for new sound files   John Lange
• [asterisk-dev] Does anybody know how to do a producer/consumer based buffered writer in C?   Chris Howard
• [asterisk-dev] Beta testers needed for new sound files   Tzafrir Cohen
• [asterisk-dev] Beta testers needed for new sound files   Dan Austin
• [asterisk-dev] Beta testers needed for new sound files   Alberto Sagredo
• [asterisk-dev] Beta testers needed for new sound files   Kevin P. Fleming
• [asterisk-dev] Beta testers needed for new sound files   Rob Fugina
• [asterisk-dev] Beta testers needed for new sound files (now version 1.4.1)   Kaloyan Kovachev
• [asterisk-dev] Beta testers needed for new sound files   Kevin P. Fleming
• [asterisk-dev] Beta testers needed for new sound files (now version 1.4.1)   Kevin P. Fleming
• [asterisk-dev] Beta testers needed for new sound files (now version 1.4.1)   Kaloyan Kovachev
• [asterisk-dev] Beta testers needed for new sound files (now version 1.4.1)   Kevin P. Fleming
• [asterisk-dev] Beta testers needed for new sound files   Alberto Sagredo
• [asterisk-dev] Inherited veriables on blind transfer?   Alistair Cunningham
• [asterisk-dev] Re: Beta testers needed for new sound files (now version....   Paul Duffy
• [asterisk-dev] Beta testers needed for new sound files   Brian Capouch
• [asterisk-dev] How to send DNIS(B-party number) in IAX trunk   Sam Liang
• [asterisk-dev] [patch] IAX2 Native Bridge enhancements   Tim Davies
• [asterisk-dev] [RFC] optimisinf RTP traffic?   Roy Sigurd Karlsbakk
• [asterisk-dev] How is asterisk code tested ?   Rahul A. Jadhav
• [asterisk-dev] Queries regarding Echo Cancellation in Zaptel-1.2.6   Chan Kwang Mien
• [asterisk-dev] Queries regarding Echo Cancellation in Zaptel-1.2.6   Chan Kwang Mien
• [asterisk-dev] Queries regarding Echo Cancellation in Zaptel-1.2.6   Paul Cadach
• [asterisk-dev] Queries regarding Echo Cancellation in Zaptel-1.2.6   Chan Kwang Mien
• [asterisk-dev] Queries regarding Echo Cancellation in Zaptel-1.2.6   Paul Cadach
• [asterisk-dev] Queries regarding Echo Cancellation in Zaptel-1.2.6   Chan Kwang Mien
• [asterisk-dev] Beta testers needed for new sound files   Steve Kennedy
• [asterisk-dev] Making an application for asterisk   Kevin Khanye
• [asterisk-dev] [asterisk]: Scheduler and I/O Manager   PhongDo
• [asterisk-dev] [RFC] optimisinf RTP traffic? (Roy Sigurd Karlsbakk)   Constantine Filin
• [asterisk-dev] Queries regarding Echo Cancellation in Zaptel-1.2.6   Anton
• [asterisk-dev] Making an application for asterisk   Moises Silva
• [asterisk-dev] Making an application for asterisk   North Antara
• [asterisk-dev] Stream voice to Asterisk   Jamel A
• [asterisk-dev] [RFC] optimisinf RTP traffic? (Roy Sigurd Karlsbakk)   Roy Sigurd Karlsbakk
• [asterisk-dev] [patch] IAX2 Native Bridge enhancements   Jared Smith
• [asterisk-dev] chan_sip.c: sip_xmit   Lucas Alvarez
• [asterisk-dev] Beta testers needed for new sound files (now version 1.4.1)   John Lange
• [asterisk-dev] chan_sip.c: sip_xmit   Jared Smith
• [asterisk-dev] [RFC] optimisinf RTP traffic?   John Martin
• [asterisk-dev] [RFC] optimising RTP traffic?   SteveK
• [asterisk-dev] Re: [RFC] optimisinf RTP traffic? (Roy Sigurd Karlsbakk)   Benny Amorsen
• [asterisk-dev] Opinions requested: BUG ID 0007063: new app_voicemail feature   Mitch Sharp
• [asterisk-dev] Feedback / handoff request for AGI "BACKGROUND" command...   Josh McAllister
• [asterisk-dev] Re: [RFC] optimisinf RTP traffic? (Roy Sigurd Karlsbakk)   Roy Sigurd Karlsbakk
• [asterisk-dev] [RFC] optimising RTP traffic?   Roy Sigurd Karlsbakk
• [asterisk-dev] [RFC] optimising RTP traffic?   SteveK
• [asterisk-dev] Solaris zaptel   Joseph Benden
• [asterisk-dev] Re: [RFC] optimisinf RTP traffic? (Roy Sigurd Karlsbakk)   Dinesh Nair
• [asterisk-dev] Beta testers needed for new sound files (now version 1.4.1)   Brian Capouch
• [asterisk-dev] Beta testers needed for new sound files (now version 1.4.1)   Brian Capouch
• [asterisk-dev] Asterisk Core Dumps   Kohler, Jeffrey
• [asterisk-dev] Re: [RFC] optimisinf RTP traffic? (Roy Sigurd Karlsbakk)   Jeffrey C. Ollie
• [asterisk-dev] RE: Sphinx and asterisk   Stephan A. Edelman
• [asterisk-dev] About implementation of poll() in poll.c   Pranav Peshwe
• [asterisk-dev] Feedback / handoff request for AGI "BACKGROUND" command...   Jared Smith
• [asterisk-dev] About implementation of poll() in poll.c   SteveK
• [asterisk-dev] Voicemail - Failed to lock path....   Aaron Paxson
• [asterisk-dev] Asterisk Core Dumps   Kohler, Jeffrey
• [asterisk-dev] iax.conf and dtmfmode   Peter Beckman
• [asterisk-dev] Solaris zaptel   John Todd
• [asterisk-dev] Feedback / handoff request for AGI "BACKGROUND" command...   Tilghman Lesher
• [asterisk-dev] Solaris zaptel   Joseph Benden
• [asterisk-dev] iax.conf and dtmfmode   Russell Bryant
• [asterisk-dev] iax.conf and dtmfmode   Peter Beckman
• [asterisk-dev] [RFC] optimising RTP traffic?   Constantine Filin
• [asterisk-dev] Feedback / handoff request for AGI "BACKGROUND" command...   Tim Ringenbach
• [asterisk-dev] Asterisk Core Dumps   Eric "ManxPower" Wieling
• [asterisk-dev] Channel problems   voip at gmx-topmail.de
• [asterisk-dev] [RFC] optimising RTP traffic?   Roy Sigurd Karlsbakk
• [asterisk-dev] [RFC] optimising RTP traffic?   Tim Panton
• [asterisk-dev] [RFC] optimising RTP traffic?   Roy Sigurd Karlsbakk
• [asterisk-dev] cdr_xml ?   Tim Panton
• [asterisk-dev] after a bridge   Clod Patry
• [asterisk-dev] Jabber Components   Landon McDowell
• [asterisk-dev] t.38   David Hindmarsh
• [asterisk-dev] status of trunk   Geoff Karl
• [asterisk-dev] noob here ... development environment set up?   AK 4asterisk
• [asterisk-dev] Questions on concurrent Dial   Chan Kwang Mien
• [asterisk-dev] Questions on concurrent Dial   Rich Adamson
• [asterisk-dev] Questions on concurrent Dial   Chan Kwang Mien
• [asterisk-dev] noob here ... development environment set up?   Maxim Vexler
• [asterisk-dev] Questions on concurrent Dial   Eric "ManxPower" Wieling
• [asterisk-dev] Questions on concurrent Dial   Armin Schindler
• [asterisk-dev] noob here ... development environment set up?   Tilghman Lesher
• [asterisk-dev] Beta testers needed for new sound files (now version 1.4.1)   Gil Kloepfer
• [asterisk-dev] Beta testers needed for new sound files (now version 1.4.1)   Bob Atkins
• [asterisk-dev] noob here ... development environment set up?   Russell Bryant
• [asterisk-dev] noob here ... development environment set up?   Aaron Paxson
• [asterisk-dev] noob here ... development environment set up?   Jeffrey C. Ollie
• [asterisk-dev] [RFC] optimising RTP traffic?   Roy Sigurd Karlsbakk
• [asterisk-dev] Re: FastAGI behaviour in trunk changed recently?   Alistair Cunningham
• [asterisk-dev] Re: FastAGI behaviour in trunk changed recently?   Alistair Cunningham
• [asterisk-dev] Re: Does anybody know how to do aproducer/consumer based buffered writer in C?   Asterisk
• [asterisk-dev] app_page enhancements   Kristian Kielhofner
• [asterisk-dev] app_page enhancements   jeff oconnell
• [asterisk-dev] how to compile and install chan_jingle.so   Danish Samad
• [asterisk-dev] app_page enhancements   Kristian Kielhofner
• [asterisk-dev] app_page enhancements   jeff oconnell
• [asterisk-dev] app_page enhancements   Kristian Kielhofner
• [asterisk-dev] T.38 call with t38 in original SDP fails   David Hindmarsh
• [asterisk-dev] how to compile and install chan_jingle.so   Dome C.
• [asterisk-dev] about TCP support to SIP   vivek relan
• [asterisk-dev] app_page enhancements   Dinesh Nair
• [asterisk-dev] about TCP support to SIP   Jeffrey C. Ollie
• [asterisk-dev] about TCP support to SIP   Tilghman Lesher
• [asterisk-dev] app_page enhancements   jeff oconnell
• [asterisk-dev] IAX JB DTMF Issues   Greg Boehnlein
• [asterisk-dev] about TCP support to SIP   alamantia at midphase.com
• [asterisk-dev] Inherited veriables on blind transfer?   Kevin P. Fleming
• [asterisk-dev] after a bridge   Kevin P. Fleming
• [asterisk-dev] status of trunk   Kevin P. Fleming
• [asterisk-dev] app_queue::eventwhencalled   John Martin
• [asterisk-dev] app_queue::eventwhencalled   Tilghman Lesher
• [asterisk-dev] noob here ... development environment set up?   Kevin P. Fleming
• [asterisk-dev] noob here ... development environment set up?   Aaron Paxson
• [asterisk-dev] zaptel spans on sysfs   Tzafrir Cohen
• [asterisk-dev] Re: [asterisk-commits] russell: branch 1.2 r38288 - /branches/1.2/channels/chan_sip.c   Kevin P. Fleming
• [asterisk-dev] zttranscode - bad page state   Michael L. Young
• [asterisk-dev] Top Users in a MeetMe room???   Armando Alvarez
• [asterisk-dev] Opinions requested: BUG ID 0007063: new app_voicemail feature   Nic Bellamy
• [asterisk-dev] zttranscode - bad page state   Russell Bryant
• [asterisk-dev] Re: [asterisk-commits] russell: branch 1.2 r38288 - /branches/1.2/channels/chan_sip.c   Russell Bryant
• [asterisk-dev] Re: FastAGI behaviour in trunk changed recently?   Russell Bryant
• [asterisk-dev] Re: [asterisk-commits] russell: branch 1.2 r38288 - /branches/1.2/channels/chan_sip.c   Kaloyan Kovachev
• [asterisk-dev] Re: [asterisk-commits] russell: branch 1.2r38288 - /branches/1.2/channels/chan_sip.c   Paul Cadach
• [asterisk-dev] zttranscode - bad page state   Frank Sautter
• [asterisk-dev] Re: FastAGI behaviour in trunk changed recently?   Alistair Cunningham
• [asterisk-dev] Re: FastAGI behaviour in trunk changed recently?   Kaloyan Kovachev
• [asterisk-dev] Trasport parameter in sip.conf/chan_sip.c   vivek relan
• [asterisk-dev] Re: FastAGI behaviour in trunk changed recently?   Alistair Cunningham
• [asterisk-dev] Re: FastAGI behaviour in trunk changed recently?   Kaloyan Kovachev
• [asterisk-dev] T.38 call with t38 in original SDP fails   Kevin P. Fleming
• [asterisk-dev] Asterisk core dumps when using blind transfers and channel OOH323 from asterisk-addons   Dinesh Nair
• [asterisk-dev] Re: [asterisk-commits] russell: branch 1.2r38288 - /branches/1.2/channels/chan_sip.c   Russell Bryant
• [asterisk-dev] Asterisk core dumps when using blind transfers and channel OOH323 from asterisk-addons   Kevin P. Fleming
• [asterisk-dev] 'IAX2 call variable passing between servers '   Douglas Garstang
• [asterisk-dev] 'IAX2 call variable passing between servers '   Tilghman Lesher
• [asterisk-dev] 'IAX2 call variable passing between servers '   Douglas Garstang
• [asterisk-dev] 'IAX2 call variable passing between servers '   Domjan Attila
• [asterisk-dev] 'IAX2 call variable passing between servers '   Florian Overkamp
• [asterisk-dev] 'IAX2 call variable passing between servers '   Douglas Garstang
• [asterisk-dev] 'IAX2 call variable passing between servers '   Tim Panton
• [asterisk-dev] 'IAX2 call variable passing between servers '   Domjan Attila
• [asterisk-dev] 'IAX2 call variable passing between servers '   John Todd
• [asterisk-dev] 'IAX2 call variable passing between servers '   Douglas Garstang
• [asterisk-dev] 'IAX2 call variable passing between servers '   Watkins, Bradley
• [asterisk-dev] 'IAX2 call variable passing between servers '   Douglas Garstang
• [asterisk-dev] 'IAX2 call variable passing between servers '   Bruce Reeves
• [asterisk-dev] Extending call parking to display park extension on the handset display   Guillermo Roditi
• [asterisk-dev] Distribution Lists in Asterisk Voicemail   Aaron Paxson
• [asterisk-dev] Distribution Lists in Asterisk Voicemail   John Lange
• [asterisk-dev] 'IAX2 call variable passing between servers '   Matt Riddell (NZ)
• [asterisk-dev] Bounty: SIP conferencing fix   Brian Jones
• [asterisk-dev] Extending call parking to display park extension onthe handset display   Alexander Lopez
• [asterisk-dev] Extending call parking to display park extension onthe handset display   Guillermo Roditi
• [asterisk-dev] option for timestamps in log messages   Russell Bryant
• [asterisk-dev] option for timestamps in log messages   Jeffrey C. Ollie
• [asterisk-dev] option for timestamps in log messages   Jeffrey C. Ollie
• [asterisk-dev] jingle audio   Clod Patry
• [asterisk-dev] option for timestamps in log messages   Steven
• [asterisk-dev] Queries regarding Asterisk Manager API - ZapTransfer   Chan Kwang Mien
• [asterisk-dev] Extending call parking to display park extensiononthe handset display   Alexander Lopez
• [asterisk-dev] Distribution Lists in Asterisk Voicemail   Russell Bryant
• [asterisk-dev] Distribution Lists in Asterisk Voicemail   Douglas Garstang
• [asterisk-dev] Queries regarding Asterisk Manager API - ZapTransfer   Stefan Reuter
• [asterisk-dev] option for timestamps in log messages   Kaloyan Kovachev
• [asterisk-dev] 'IAX2 call variable passing between servers '   Stephen Davies
• [asterisk-dev] option for timestamps in log messages   Dinesh Nair
• [asterisk-dev] option for timestamps in log messages   Russell Bryant
• [asterisk-dev] Routing data modem calls   Brian Candler
• [asterisk-dev] 'IAX2 call variable passing between servers '   Roy Sigurd Karlsbakk
• [asterisk-dev] Re: FastAGI behaviour in trunk changed recently?   Alistair Cunningham
• [asterisk-dev] Routing data modem calls   Olivier Krief
• [asterisk-dev] Asterisk core dumps when using blind transfers and channel OOH323 from asterisk-addons   Dinesh Nair
• [asterisk-dev] asterisk dumps core on calls/transfers to Local/ext   Vahan Yerkanian
• [asterisk-dev] Routing data modem calls   Brian Candler
• [asterisk-dev] option for timestamps in log messages   Rich Adamson
• [asterisk-dev] Dialplan without AgentCallbackLogin   Asterisk
• [asterisk-dev] Dialplan without AgentCallbackLogin   Tristan
• [asterisk-dev] Local channel and SendText()   Dmytro Mishchenko
• [asterisk-dev] Routing data modem calls   Brian Candler
• [asterisk-dev] Routing data modem calls   Andrew Kohlsmith
• [asterisk-dev] Re: FastAGI behaviour in trunk changed recently?   Kaloyan Kovachev
• [asterisk-dev] Re: FastAGI behaviour in trunk changed recently?   Alistair Cunningham
• [asterisk-dev] Routing data modem calls   Olivier Krief
• [asterisk-dev] Re: FastAGI behaviour in trunk changed recently?   Kaloyan Kovachev
• [asterisk-dev] Routing data modem calls   Alexander Lopez
• [asterisk-dev] Re: FastAGI behaviour in trunk changed recently?   Alistair Cunningham
• [asterisk-dev] Routing data modem calls   John Lange
• [asterisk-dev] jingle audio   mogorman
• [asterisk-dev] Inherited veriables on blind transfer?   Alistair Cunningham
• [asterisk-dev] Routing data modem calls   Matthew Fredrickson
• [asterisk-dev] option for timestamps in log messages   Mr. James W. Laferriere
• [asterisk-dev] option for timestamps in log messages   Steven
• [asterisk-dev] Re: FastAGI behaviour in trunk changed recently?   Kaloyan Kovachev
• [asterisk-dev] Trasport parameter in sip.conf/chan_sip.c   Fabian Müller
• [asterisk-dev] 'IAX2 call variable passing between servers '   Tilghman Lesher
• [asterisk-dev] 'IAX2 call variable passing between servers '   Tilghman Lesher
• [asterisk-dev] Routing data modem calls   Steven
• [asterisk-dev] Routing data modem calls   Andrew Kohlsmith
• [asterisk-dev] jingle audio   Landon McDowell
• [asterisk-dev] Re: FastAGI behaviour in trunk changed recently?   Alistair Cunningham
• [asterisk-dev] Routing data modem calls   Stelios Koroneos
• [asterisk-dev] 'IAX2 call variable passing between servers '   Douglas Garstang
• [asterisk-dev] 'IAX2 call variable passing between servers '   Tilghman Lesher
• [asterisk-dev] option for timestamps in log messages   Steve Edwards
• [asterisk-dev] Asterisk core dumps when using blind transfers and channel OOH323 from asterisk-addons   Kevin P. Fleming
• [asterisk-dev] Speaking slots at AstriCon almost full   Steven Sokol
• [asterisk-dev] option for timestamps in log messages   Russell Bryant
• [asterisk-dev] Routing data modem calls   Brian Candler
• [asterisk-dev] Routing data modem calls   Brian Candler
• [asterisk-dev] Routing data modem calls   Brian Candler
• [asterisk-dev] Routing data modem calls   Steven Critchfield
• [asterisk-dev] 'IAX2 call variable passing between servers '   John Todd
• [asterisk-dev] DTMF Bug(s)?   Peter Beckman
• [asterisk-dev] DTMF Bug(s)?   Greg Boehnlein
• [asterisk-dev] Asterisk core dumps when using blind transfers and channel OOH323 from asterisk-addons   Dinesh Nair
• [asterisk-dev] option for timestamps in log messages   Roy Sigurd Karlsbakk
• [asterisk-dev] 'IAX2 call variable passing between servers '   Tim Panton
• [asterisk-dev] 'IAX2 call variable passing between servers '   Roy Sigurd Karlsbakk
• [asterisk-dev] 'IAX2 call variable passing between servers '   Tzafrir Cohen
• [asterisk-dev] option for timestamps in log messages   Tzafrir Cohen
• [asterisk-dev] 'IAX2 call variable passing between servers '   Tilghman Lesher
• [asterisk-dev] DTMF Bug(s)?   Peter Beckman
• [asterisk-dev] Queries regarding Hint for Group of Extensions   Chan Kwang Mien
• [asterisk-dev] 'IAX2 call variable passing between servers '   Tim Ringenbach
• [asterisk-dev] DTMF Bug(s)?   Greg Boehnlein
• [asterisk-dev] DTMF Bug(s)?   Tilghman Lesher
• [asterisk-dev] Re: FastAGI behaviour in trunk changed recently?   Kaloyan Kovachev
• [asterisk-dev] 'IAX2 call variable passing between servers '   Tim Panton
• [asterisk-dev] 'IAX2 call variable passing between servers '   Steven
• [asterisk-dev] Dial with H Option to Disconnect Call   Wilmar Campos
• [asterisk-dev] Problem with branch 1.2 r38420 - /branches/1.2/channels/chan_sip.c   asterisk at ntplx.net
• [asterisk-dev] 'IAX2 call variable passing between servers '   Tilghman Lesher
• [asterisk-dev] automatic "ztcfg" for analog cards   Tzafrir Cohen
• [asterisk-dev] Feature Request: Assigning number of calls accepted by Bandwidth   Mohamed A. Gombolaty
• [asterisk-dev] Asterisk functinality   Dovid B - Asterisk Dev.
• [asterisk-dev] automatic "ztcfg" for analog cards   Tzafrir Cohen
• [asterisk-dev] automatic "ztcfg" for analog cards   Steven
• [asterisk-dev] Asterisk functinality   Steven
• [asterisk-dev] Remote WMI notification   Jean-Yves Avenard
• [asterisk-dev] Re: asterisk-dev Digest, Vol 24, Issue 84   Dovid B - Asterisk Dev.
• [asterisk-dev] Re: asterisk-dev Digest, Vol 24, Issue 84   Aaron Paxson
• [asterisk-dev] Attended-transfer bug?   Guillermo Winkler
• [asterisk-dev] Re: FastAGI behaviour in trunk changed recently?   Alistair Cunningham
• [asterisk-dev] T.38 call with t38 in original SDP fails   David Hindmarsh
• [asterisk-dev] Re: FastAGI behaviour in trunk changed recently?   Tim Ringenbach
• [asterisk-dev] Routing data modem calls   Anton
• [asterisk-dev] Zap Subchannel naming and ISDN   Koopmann, Jan-Peter
• [asterisk-dev] automatic "ztcfg" for analog cards   Tzafrir Cohen
• [asterisk-dev] musiconhold   Juergen John
• [asterisk-dev] musiconhold   Aaron Paxson
• [asterisk-dev] musiconhold   Aaron Paxson
• [asterisk-dev] zaptel-1.2.7: vpm450m_fw.h missing?   Patrick
• [asterisk-dev] zaptel-1.2.7: vpm450m_fw.h missing?   Kevin P. Fleming
• [asterisk-dev] option for timestamps in log messages   Roy Sigurd Karlsbakk
• [asterisk-dev] 'IAX2 call variable passing between servers '   Roy Sigurd Karlsbakk
• [asterisk-dev] Queues.conf   franco

Last message date: Mon Jul 31 11:10:19 MST 2006
Archived on: Tue Sep 5 14:28:03 MST 2006

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