November 2005 Archives by author

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Starting: Tue Nov 1 04:12:38 MST 2005
Ending: Wed Nov 30 23:49:00 MST 2005
Messages: 736

• [Asterisk-Dev] How are SIP calls connected/bridged ?   Arnaud
• [Asterisk-Dev] How are SIP calls connected/bridged ?   Arnaud
• [Asterisk-Dev] How are SIP calls connected/bridged ?   Arnaud
• [Asterisk-Dev] has somebody ever tested faxdetection ?   Thomas Häger
• [Asterisk-Dev] has somebody ever tested faxdetection ?   Thomas Häger
• [Asterisk-Dev] E1 channel allocation   Denis Galvão
• [Asterisk-Dev] E1 channel allocation   Denis Galvão
• [Asterisk-Dev] Can anoncvs make 'cvs commit' in CVS asterisk ?   Juan Manuel Guerrero de los Ríos
• [Asterisk-Dev] Realtime fails using unixODBC and FreeTDS against MSSQL   Trælnes AS
• [Asterisk-Dev] ast_load_realtime   José Pablo Ezequiel Fernández
• [Asterisk-Dev] reload causing SIP phone to not be registered   Rich Adamson
• [Asterisk-Dev] reload causing SIP phone to not be registered   Rich Adamson
• [Asterisk-Dev] reload causing SIP phone to not be registered   Rich Adamson
• [Asterisk-Dev] Re: [Asterisk-Users] Warning -- chan_iax2.c: ast_sched_runq tasks   Rich Adamson
• [Asterisk-Dev] Security issue mumblings   Rich Adamson
• [Asterisk-Dev] CVS server problem?   Rich Adamson
• [Asterisk-Dev] asterisk 1.2 g729 compile errors   Diyanat Ali
• [Asterisk-Dev] asterisk 1.2 g729 compile errors   Diyanat Ali
• [Asterisk-Dev] app_conference errors   Diyanat Ali
• [Asterisk-Dev] extensions.conf limitations, and how to overcome them   Michael Anderson
• [Asterisk-Dev] app_page   Michael Anderson
• [Asterisk-Dev] Asterisk project converts to Subversion versioncontrol system   North Antara
• [Asterisk-Dev] Subversion checkouts   North Antara
• [Asterisk-Dev] Handling AST_FORMAT_SLINEAR at 48000Hz   Carlos Antunes
• [Asterisk-Dev] chan_modem_i4l (only noise is received, reason for the problem)   David Arendt
• [Asterisk-Dev] Develop SoftPhone with IAXclient!!   Alberto Lavariega Arista
• [Asterisk-Dev] reload causing SIP phone to not be registered   James Armstrong
• [Asterisk-Dev] reload causing SIP phone to not be registered   James Armstrong
• [Asterisk-Dev] Asterisk make clean and make looping?? (LONG)   James Armstrong
• [Asterisk-Dev] Voicemail password changes not sticking   James Armstrong
• [Asterisk-Dev] DID number not saved when incoming fax detected on Zap   James Armstrong
• [Asterisk-Dev] trying to generate bt, but info threads not working...   Ron Arts
• [Asterisk-Dev] chan_exosip2   Joshua Colp - Asterlink
• [Asterisk-Dev] chan_exosip2   Joshua Colp - Asterlink
• [Asterisk-Dev] 1.2.0-beta1 MeetMe bug?   Dan Austin
• [Asterisk-Dev] Re: 1.2.0-beta1 MeetMe bug?   Dan Austin
• [Asterisk-Dev] MeetMe participant time tracking   Dan Austin
• [Asterisk-Dev] MeetMe participant time tracking   Dan Austin
• [Asterisk-Dev] MeetMe participant time tracking   Dan Austin
• [Asterisk-Dev] Version 1.2.0 chan_h323 (ooh323c)   Dan Austin
• [Asterisk-Dev] Version 1.2.0 chan_h323 (ooh323c)   Dan Austin
• [Asterisk-Dev] Version 1.2.0 chan_h323 (ooh323c)   Dan Austin
• [Asterisk-Dev] MeetMe assumes 20ms...   Dan Austin
• [Asterisk-Dev] MeetMe assumes 20ms...   Dan Austin
• [Asterisk-Dev] Missing audio from Zaptel channels   Rod Bacon
• [Asterisk-Dev] Re: [Asterisk-Users] Missing audio from Zaptel channels - SOLVED!?   Rod Bacon
• [Asterisk-Dev] echo, delay and dropped packets in channel driver   Boris Bakchiev
• [Asterisk-Dev] echo, delay and dropped packets in channel driver   Boris Bakchiev
• [Asterisk-Dev] Asterisk on Cygiwn   Time Bandit
• [Asterisk-Dev] Delphi ActiveX component   Time Bandit
• [Asterisk-Dev] AGIphp Installation   Umair Bari
• [Asterisk-Dev] Harald Baron/EBAROH/CH/Ascom ist nicht anwesend.   Harald Baron
• [Asterisk-Dev] asterisk.spec and the redhat dir   Joseph Benden
• [Asterisk-Dev] asterisk.spec and the redhat dir   Joseph Benden
• [Asterisk-Dev] Asterisk Locking - queue application   Dov Bigio
• [Asterisk-Dev] queue_log for agent stats   Dov Bigio
• [Asterisk-Dev] Queue - Transfer & server stability   Dov Bigio
• [Asterisk-Dev] Queue - Transfer & server stability   Dov Bigio
• [Asterisk-Dev] Asterisk 1.2.0-beta2 ParkAndAnnounce question   Steve Blair
• [Asterisk-Dev] How to get Referred-By header   Steve Blair
• [Asterisk-Dev] Re: bug 4252 - increasing delay over time channels in MeetMe   Kevin Bockman
• [Asterisk-Dev] chan_bluetooth   Sven Boeckelmann
• [Asterisk-Dev] AST_FRAME_DATA   Greg Boehnlein
• [Asterisk-Dev] asterisk.spec and the redhat dir   Greg Boehnlein
• [Asterisk-Dev] asterisk.spec and the redhat dir   Greg Boehnlein
• [Asterisk-Dev] asterisk.spec and the redhat dir   Greg Boehnlein
• [Asterisk-Dev] Action item :: Check if your patch is up to date!   Greg Boehnlein
• [Asterisk-Dev] MeetMe assumes 20ms...   Greg Boehnlein
• [Asterisk-Dev] echo, delay and dropped packets in channel driver   Kris Boutilier
• [Asterisk-Dev] echo, delay and dropped packets in channel driver   Kris Boutilier
• [Asterisk-Dev] Monitor Function broken since Beta-2 ?   Max Bressel
• [Asterisk-Dev] sip gateway provider recommendations?   John Brookes
• [Asterisk-Dev] ODBC Voicemail Storage   Russell Bryant
• [Asterisk-Dev] Additions to function DUNDiLookup   Russell Bryant
• [Asterisk-Dev] app_datetime unused ? and sayunixtime cleanup...   Russell Bryant
• [Asterisk-Dev] Module that works in Asterisk 1.0.x and 1.2.x   Russell Bryant
• [Asterisk-Dev] gmake error for asterisk-addons-1.2.0   Russell Bryant
• [Asterisk-Dev] Compiler debug switch in the main Asterisk Makefile   Russell Bryant
• [Asterisk-Dev] chan_bluetooth   Russell Bryant
• [Asterisk-Dev] lock.h change on 10-28 (rev 1.39) means you can never use pthread_mutex_t?   Brian Capouch
• [Asterisk-Dev] banned from the *-users list?   Brian Capouch
• [Asterisk-Dev] CVS server problem?   Brian Capouch
• [Asterisk-Dev] CVS server problem?   Brian Capouch
• [Asterisk-Dev] asterisk 1.2 g729 compile errors   Mik Cheez
• [Asterisk-Dev] jb-rtp-sip.scx.new.patch compiled error in Fedora 3   Raymond Chen
• [Asterisk-Dev] Module that works in Asterisk 1.0.x and 1.2.x   Sergio Chersovani
• [Asterisk-Dev] reload and hints problem   Sergio Chersovani
• [Asterisk-Dev] dialplan hint case sensitive   Sergio Chersovani
• [Asterisk-Dev] ChanIsAvail() works for SIP?   Guo-Wei Chiuan
• [Asterisk-Dev] Asterisk project converts to Subversion version control system   James Cloos
• [Asterisk-Dev] ztcfg and wctdm/wcfxo   Tzafrir Cohen
• [Asterisk-Dev] Asterisk make clean and make looping??   Tzafrir Cohen
• [Asterisk-Dev] Asterisk make clean and make looping??   Tzafrir Cohen
• [Asterisk-Dev] New manager API command   Tzafrir Cohen
• [Asterisk-Dev] Asterisk on Cygiwn   Tzafrir Cohen
• [Asterisk-Dev] chan_exosip2 for asterisk   Tzafrir Cohen
• [Asterisk-Dev] Version 1.2.0 chan_h323 (ooh323c)   Tzafrir Cohen
• [Asterisk-Dev] Install destination on Solaris   Tzafrir Cohen
• [Asterisk-Dev] Install destination on Solaris   Tzafrir Cohen
• [Asterisk-Dev] AGIphp Installation   Tzafrir Cohen
• [Asterisk-Dev] Bounty - Connecting Asterisk to SER   Joshua Colp
• [Asterisk-Dev] Module that works in Asterisk 1.0.x and 1.2.x   Juan Jose Comellas
• [Asterisk-Dev] Module that works in Asterisk 1.0.x and 1.2.x   Juan Jose Comellas
• [Asterisk-Dev] Module that works in Asterisk 1.0.x and 1.2.x   Juan Jose Comellas
• [Asterisk-Dev] chan_bluetooth   Dave Cotton
• [Asterisk-Dev] RFC2833 Event Duration   Ryan Courtnage
• [Asterisk-Dev] Re: multiple registrations of same credentials   Jonathan k. Creasy
• [Asterisk-Dev] Re: multiple registrations of same credentials   Jonathan k. Creasy
• [Asterisk-Dev] [OTAnn] Feedback   Jonathan k. Creasy
• [Asterisk-Dev] New manager API command   Jonathan k. Creasy
• [Asterisk-Dev] Asterisk on Cygiwn   Jonathan k. Creasy
• [Asterisk-Dev] Asterisk make clean and make looping??   Steven Critchfield
• [Asterisk-Dev] extensions.conf limitations, and how to overcome them   Steven Critchfield
• [Asterisk-Dev] New manager API command   Steven Critchfield
• [Asterisk-Dev] Develop SoftPhone with IAXclient!!   Steven Critchfield
• [Asterisk-Dev] Asterisk project converts to Subversion version control system   Steven Critchfield
• [Asterisk-Dev] Asterisk project converts to Subversion version control system   Steven Critchfield
• [Asterisk-Dev] SIP media negotiation problem   Steven Critchfield
• [Asterisk-Dev] New SVN email subject   Steven Critchfield
• [Asterisk-Dev] New SVN email subject   Steven Critchfield
• [Asterisk-Dev] AST_FLAG_DEFER_DTMF (would like dialogic-r4 like semantics)   Steven Critchfield
• [Asterisk-Dev] res_config_mysql.c connection problems => bug   Loic DIDELOT
• [Asterisk-Dev] res_config_mysql.c connection problems => bug   Loic DIDELOT
• [Asterisk-Dev] res_config_mysql.c connection problems => bug   Loic DIDELOT
• [Asterisk-Dev] res_config_mysql.c connection problems => bug   Loic DIDELOT
• [Asterisk-Dev] res_config_mysql.c connection problems => bug   Loic DIDELOT
• [Asterisk-Dev] A problem of MixMonitor   Gerald Dachs
• [Asterisk-Dev] chan_bluetooth and asterisk-1.2   Dan
• [Asterisk-Dev] chan_bluetooth and asterisk-1.2   Dan
• [Asterisk-Dev] chan_bluetooth and asterisk-1.2   Dan
• [Asterisk-Dev] chan_bluetooth and asterisk-1.2   Dan
• [Asterisk-Dev] chan_bluetooth and asterisk-1.2   Dan
• [Asterisk-Dev] Disposition failed in Asterisk-1.2.0-stable   Aaron Daniel
• [Asterisk-Dev] Echo cancel on PRI   Klaus Darilion
• [Asterisk-Dev] Bounty - Connecting Asterisk to SER   Klaus Darilion
• [Asterisk-Dev] Bug 5590 and Release Candidates   Paul Davidson
• [Asterisk-Dev] sip gateway provider recommendations?   Rusty Dekema
• [Asterisk-Dev] Asterisk make clean and make looping??   Saul Diaz
• [Asterisk-Dev] Asterisk make clean and make looping??   Saul Diaz
• [Asterisk-Dev] Asterisk make clean and make looping??   Saul Diaz
• [Asterisk-Dev] Compatility in the manager   Saul Diaz
• [Asterisk-Dev] Asterisk make clean and make looping?? (LONG)   Saul Diaz
• [Asterisk-Dev] Security issue mumblings   Saul Diaz
• [Asterisk-Dev] New manager API command   Saul Diaz
• [Asterisk-Dev] Asterisk make clean and make looping?? (Possible solution)   Saul Diaz
• [Asterisk-Dev] New manager API command   Saul Diaz
• [Asterisk-Dev] New manager API command   Saul Diaz
• [Asterisk-Dev] chan_misdn in asterisk beta 2 ?   Ray Van Dolson
• [Asterisk-Dev] Asterisk or Polycom Bug? - Message waiting   Rod Dorman
• [Asterisk-Dev] app_datetime unused ? and sayunixtime cleanup...   Rod Dorman
• [Asterisk-Dev] cvs revision tags   Rod Dorman
• [Asterisk-Dev] cvs revision tags   Rod Dorman
• [Asterisk-Dev] CVS server problem?   Rod Dorman
• [Asterisk-Dev] Re: cvs revision tags   Rod Dorman
• [Asterisk-Dev] Re: cvs revision tags   Rod Dorman
• [Asterisk-Dev] Re: cvs revision tags   Rod Dorman
• [Asterisk-Dev] FYI: Internet-Draft for IAX: Inter-Asterisk eXchange Version 2   Rod Dorman
• [Asterisk-Dev] HELP! on disconnecting stale calls.   Paradise Dove
• [Asterisk-Dev] HELP! on disconnecting stale calls.   Paradise Dove
• [Asterisk-Dev] Asterisk project converts to Subversion version control system   Paradise Dove
• [Asterisk-Dev] Re: TDMoE + kernel badness   Fabio Ferrari
• [Asterisk-Dev] reload causing SIP phone to not be registered   Brian C. Fertig
• [Asterisk-Dev] lock.h change on 10-28 (rev 1.39) means you can never use pthread_mutex_t?   Kevin P. Fleming
• [Asterisk-Dev] chan_misdn in asterisk beta 2 ?   Kevin P. Fleming
• [Asterisk-Dev] Re: chan_misdn in asterisk beta 2 ?   Kevin P. Fleming
• [Asterisk-Dev] chan_misdn in asterisk beta 2 ?   Kevin P. Fleming
• [Asterisk-Dev] Asterisk make clean and make looping??   Kevin P. Fleming
• [Asterisk-Dev] reload causing SIP phone to not be registered   Kevin P. Fleming
• [Asterisk-Dev] chan_misdn in asterisk beta 2 ?   Kevin P. Fleming
• [Asterisk-Dev] New ChangeLog in CVS HEAD Asterisk tree   Kevin P. Fleming
• [Asterisk-Dev] Re: multiple registrations of same credentials   Kevin P. Fleming
• [Asterisk-Dev] New ChangeLog in CVS HEAD Asterisk tree   Kevin P. Fleming
• [Asterisk-Dev] New ChangeLog in CVS HEAD Asterisk tree   Kevin P. Fleming
• [Asterisk-Dev] Send tone to caller on answer   Kevin P. Fleming
• [Asterisk-Dev] since 2005-10-28, pthread_mutex_t can never be used in apps that include asterisk's lock.h ??   Kevin P. Fleming
• [Asterisk-Dev] lock.h change on 10-28 (rev 1.39) means you can never use pthread_mutex_t?   Kevin P. Fleming
• [Asterisk-Dev] Local channel optimisation   Kevin P. Fleming
• [Asterisk-Dev] lock.h change on 10-28 (rev 1.39) means you cannever use pthread_mutex_t?   Kevin P. Fleming
• [Asterisk-Dev] Asterisk make clean and make looping?? (Possible solution)   Kevin P. Fleming
• [Asterisk-Dev] Asterisk make clean and make looping?? (Possible solution)   Kevin P. Fleming
• [Asterisk-Dev] Asterisk make clean and make looping?? (Possible solution)   Kevin P. Fleming
• [Asterisk-Dev] Asterisk make clean and make looping?? (Possible solution)   Kevin P. Fleming
• [Asterisk-Dev] ZT_DIAL Operation   Kevin P. Fleming
• [Asterisk-Dev] always_inline change...   Kevin P. Fleming
• [Asterisk-Dev] other -Werror issues - time_t printf format...   Kevin P. Fleming
• [Asterisk-Dev] RFC2833 Event Duration   Kevin P. Fleming
• [Asterisk-Dev] Asterisk or Polycom Bug?   Kevin P. Fleming
• [Asterisk-Dev] RFC2833 Event Duration   Kevin P. Fleming
• [Asterisk-Dev] ZT_DIAL Operation   Kevin P. Fleming
• [Asterisk-Dev] Security issue mumblings   Kevin P. Fleming
• [Asterisk-Dev] Security issue mumblings   Kevin P. Fleming
• [Asterisk-Dev] Security issue mumblings - SIP   Kevin P. Fleming
• [Asterisk-Dev] other -Werror issues - time_t printf format...   Kevin P. Fleming
• [Asterisk-Dev] Security issue mumblings - SIP   Kevin P. Fleming
• [Asterisk-Dev] queue_log for agent stats   Kevin P. Fleming
• [Asterisk-Dev] Security issue mumblings - SIP   Kevin P. Fleming
• [Asterisk-Dev] asterisk.spec and the redhat dir   Kevin P. Fleming
• [Asterisk-Dev] trivial patch requesting comments before filing   Kevin P. Fleming
• [Asterisk-Dev] trivial patch requesting comments before filing   Kevin P. Fleming
• [Asterisk-Dev] Asterisk 1.2.0-rc1 Released!   Kevin P. Fleming
• [Asterisk-Dev] CVS HEAD tree frozen until release   Kevin P. Fleming
• [Asterisk-Dev] Bug 5590 and Release Candidates   Kevin P. Fleming
• [Asterisk-Dev] E1 channel allocation   Kevin P. Fleming
• [Asterisk-Dev] Calling apps and parsing return values   Kevin P. Fleming
• [Asterisk-Dev] Asterisk 1.2.0-rc2 Released!   Kevin P. Fleming
• [Asterisk-Dev] app_datetime unused ? and sayunixtime cleanup...   Kevin P. Fleming
• [Asterisk-Dev] Handling AST_FORMAT_SLINEAR at 48000Hz   Kevin P. Fleming
• [Asterisk-Dev] Install destination on Solaris   Kevin P. Fleming
• [Asterisk-Dev] Install destination on Solaris   Kevin P. Fleming
• [Asterisk-Dev] cvs revision tags   Kevin P. Fleming
• [Asterisk-Dev] Asterisk make clean and make looping?? (Possiblesolution)   Kevin P. Fleming
• [Asterisk-Dev] generation of config files from source code ?   Kevin P. Fleming
• [Asterisk-Dev] Compiler debug switch in the main Asterisk Makefile   Kevin P. Fleming
• [Asterisk-Dev] Conditional Goto and GoSub use 'n'?   Kevin P. Fleming
• [Asterisk-Dev] Asterisk project converts to Subversion version control system   Kevin P. Fleming
• [Asterisk-Dev] Asterisk project converts to Subversion version control system   Kevin P. Fleming
• [Asterisk-Dev] Asterisk project converts to Subversion version control system   Kevin P. Fleming
• [Asterisk-Dev] Subversion file permissions   Kevin P. Fleming
• [Asterisk-Dev] zaptel-1.2.0 compilation fails on 2.5.15-rc2-git6   Kevin P. Fleming
• [Asterisk-Dev] zaptel-1.2.0 compilation fails on 2.5.15-rc2-git6   Kevin P. Fleming
• [Asterisk-Dev] New SVN email subject   Kevin P. Fleming
• [Asterisk-Dev] New SVN email subject   Kevin P. Fleming
• [Asterisk-Dev] New SVN email subject   Kevin P. Fleming
• [Asterisk-Dev] AST_FLAG_DEFER_DTMF (would like dialogic-r4 like semantics)   Kevin P. Fleming
• [Asterisk-Dev] AST_FLAG_DEFER_DTMF (would like dialogic-r4 like semantics)   Kevin P. Fleming
• [Asterisk-Dev] ZT_DIAL Operation   Matthew Fredrickson
• [Asterisk-Dev] E1 channel allocation   Matthew Fredrickson
• [Asterisk-Dev] mtp-2   Matthew Fredrickson
• [Asterisk-Dev] mtp-2   Matthew Fredrickson
• [Asterisk-Dev] mtp-2   Matthew Fredrickson
• [Asterisk-Dev] INSTALL_PREFIX   Hans Fugal
• [Asterisk-Dev] Install destination on Solaris   Hans Fugal
• [Asterisk-Dev] zaptel-1.2.0 compilation fails on 2.5.15-rc2-git6   Jean-Denis Girard
• [Asterisk-Dev] Install destination on Solaris   Bob Goddard
• [Asterisk-Dev] Call Admission control   Mohamed A. Gombolaty
• [Asterisk-Dev] CAC in Asterisk   Mohamed A. Gombolaty
• [Asterisk-Dev] Get number function   Tristram Graham
• [Asterisk-Dev] Re: Get number function   Tristram Graham
• [Asterisk-Dev] Send tone to caller on answer   Ed Greenberg
• [Asterisk-Dev] Changes to the way DISA and CDR work together   Ed Greenberg
• [Asterisk-Dev] Calling apps and parsing return values   Ed Greenberg
• [Asterisk-Dev] More work on DISA   Ed Greenberg
• [Asterisk-Dev] gmake issues (solved, #5885 on mantis)   Edwin Groothuis
• [Asterisk-Dev] gmake issues (solved, #5885 on mantis)   Edwin Groothuis
• [Asterisk-Dev] LDAP RealTime driver   Manuel Guesdon
• [Asterisk-Dev] festival voice   Manuel Guesdon
• [Asterisk-Dev] festival voice   Manuel Guesdon
• [Asterisk-Dev] queue_log for agent stats   Kyle Hagan
• [Asterisk-Dev] Problem with hanging up a SIP channel   Marc Haisenko
• [Asterisk-Dev] Problem with hanging up a SIP channel   Marc Haisenko
• [Asterisk-Dev] Chan_sip version 1, 2 and NG: 3   Marc Haisenko
• [Asterisk-Dev] Problem with hanging up a SIP channel   Marc Haisenko
• [Asterisk-Dev] Problem with hanging up a SIP channel   Marc Haisenko
• [Asterisk-Dev] Problem with hanging up a SIP channel   Marc Haisenko
• [Asterisk-Dev] Re: multiple registrations of same credentials   Kevin Hanson
• [Asterisk-Dev] Re: multiple registrations of same credentials   Kevin Hanson
• [Asterisk-Dev] A problem of MixMonitor   Kevin Hanson
• [Asterisk-Dev] Asterisk 1.2.0-rc1 Released!   Kevin Hanson
• [Asterisk-Dev] Asterisk 1.2.0-rc1 Released!   Kevin Hanson
• [Asterisk-Dev] chan_exosip2   Kevin Hanson
• [Asterisk-Dev] chan_exosip2   Kevin Hanson
• [Asterisk-Dev] chan_exosip2   Kevin Hanson
• [Asterisk-Dev] chan_exosip2   Kevin Hanson
• [Asterisk-Dev] reload and hints problem   Kevin Hanson
• [Asterisk-Dev] CVS server problem?   Kevin Hanson
• [Asterisk-Dev] Re: cvs revision tags   Kevin Hanson
• [Asterisk-Dev] Call-Limit   Kevin Hanson
• [Asterisk-Dev] Disposition failed in Asterisk-1.2.0-stable   Kevin Hanson
• [Asterisk-Dev] Module that works in Asterisk 1.0.x and 1.2.x   Martin Harriss
• [Asterisk-Dev] musiconhold -vs- musicclass problems setting the differnt class of music   Ronald Hartmann
• [Asterisk-Dev] 1.2.0 bug or mis-config? chan_sip.c   Bryan Heitman
• [Asterisk-Dev] New app: app_page comments   Edwin Horton
• [Asterisk-Dev] ODBC Voicemail Storage   Edwin Horton
• [Asterisk-Dev] 482 Loop Detected problem   Charles Huang
• [Asterisk-Dev] New Jersey ATT Vocie T1 Asterisk Toll free number not working   Charles Huang
• [Asterisk-Dev] Multiple Pickup   Chih-Wei Huang
• [Asterisk-Dev] Asterisk make clean and make looping??   Chih-Wei Huang
• [Asterisk-Dev] A problem of MixMonitor   Chih-Wei Huang
• [Asterisk-Dev] say.c cleanup   Chih-Wei Huang
• [Asterisk-Dev] sip show peers doesn't work?   Chih-Wei Huang
• [Asterisk-Dev] bug 4252 - increasing delay over time channels in MeetMe   Chih-Wei Huang
• [Asterisk-Dev] Re: bug 4252 - increasing delay over time channels in MeetMe   Chih-Wei Huang
• [Asterisk-Dev] Multiple Pickup   Chih-Wei Huang
• [Asterisk-Dev] Re: bug 4252 - increasing delay over time channels in MeetMe   Chih-Wei Huang
• [Asterisk-Dev] A question about G option of Dial   Chih-Wei Huang
• [Asterisk-Dev] Re: A question about G option of Dial   Chih-Wei Huang
• [Asterisk-Dev] Re: [Asterisk-Users] Warning -- chan_iax2.c: ast_sched_runq tasks   Mark Hulber
• [Asterisk-Dev] DID number not saved when incoming fax detected on Zap   Mark Hulber
• [Asterisk-Dev] Asterisk project converts to Subversion version control system   Mark Hulber
• [Asterisk-Dev] Subversion checkouts   Mark Hulber
• [Asterisk-Dev] New SVN email subject   Mark Hulber
• [Asterisk-Dev] New SVN email subject   Mark Hulber
• [Asterisk-Dev] Collect Calls   IT
• [Asterisk-Dev] Collect Calls   IT
• [Asterisk-Dev] Collect Calls   IT
• [Asterisk-Dev] Collect Calls   IT
• [Asterisk-Dev] E1 connection problem   Hugh Jackman
• [Asterisk-Dev] An Ast-user question for Ast-dev experts   Hugh Jackman
• [Asterisk-Dev] An Ast-user question for Ast-dev experts   Hugh Jackman
• [Asterisk-Dev] Re: mtp-2 (janvb@caselaboratories.com)   Hadi Jadallah
• [Asterisk-Dev] lock.h change on 10-28 (rev 1.39) means you cannever use pthread_mutex_t?   Jerris, Michael MI
• [Asterisk-Dev] [repost] cli output issue   Jerris, Michael MI
• [Asterisk-Dev] Asterisk on Cygiwn   Jerris, Michael MI
• [Asterisk-Dev] Another weird dialplan question   Jerris, Michael MI
• [Asterisk-Dev] chan_exosip2 for asterisk   Jerris, Michael MI
• [Asterisk-Dev] Asterisk on Cygiwn   Jerris, Michael MI
• [Asterisk-Dev] Hardware Manual   Jerris, Michael MI
• [Asterisk-Dev] Delphi ActiveX component   Jerris, Michael MI
• [Asterisk-Dev] AST_FRAME_DATA   Olle E. Johansson
• [Asterisk-Dev] Security issue mumblings - SIP   Olle E. Johansson
• [Asterisk-Dev] Security issue mumblings - SIP   Olle E. Johansson
• [Asterisk-Dev] Security issue mumblings - SIP   Olle E. Johansson
• [Asterisk-Dev] say.c cleanup   Olle E. Johansson
• [Asterisk-Dev] chan_exosip2   Olle E. Johansson
• [Asterisk-Dev] chan_exosip2   Olle E. Johansson
• [Asterisk-Dev] chan_exosip2   Olle E. Johansson
• [Asterisk-Dev] chan_exosip2   Olle E. Johansson
• [Asterisk-Dev] chan_exosip2   Olle E. Johansson
• [Asterisk-Dev] chan_exosip2   Olle E. Johansson
• [Asterisk-Dev] app_page   Olle E. Johansson
• [Asterisk-Dev] generation of config files from source code ?   Olle E. Johansson
• [Asterisk-Dev] Re: cvs revision tags   Olle E. Johansson
• [Asterisk-Dev] Re: [Asterisk-Users] Asterisk crash: "using deprecated BYE/Also transfer method"   Olle E. Johansson
• [Asterisk-Dev] Queue - Transfer & server stability   Olle E. Johansson
• [Asterisk-Dev] 482 Loop Detected problem   Olle E. Johansson
• [Asterisk-Dev] Re: 482 Loop Detected problem   Olle E. Johansson
• [Asterisk-Dev] Re: 482 Loop Detected problem   Olle E. Johansson
• [Asterisk-Dev] Action item :: Check if your patch is up to date!   Olle E. Johansson
• [Asterisk-Dev] get register info from peer section in sip.conf ?   Olle E. Johansson
• [Asterisk-Dev] Chan_sip version 1, 2 and NG: 3   Olle E. Johansson
• [Asterisk-Dev] Chan_sip version 1, 2 and NG: 3   Olle E. Johansson
• [Asterisk-Dev] G.729 SDP bug in sip_chan.c (asterisk v1.2.0)   Olle E. Johansson
• [Asterisk-Dev] How are SIP calls connected/bridged ?   Olle E. Johansson
• [Asterisk-Dev] chan_misdn in asterisk beta 2 ?   Senad Jordanovic
• [Asterisk-Dev] New ChangeLog in CVS HEAD Asterisk tree   Steve Kann
• [Asterisk-Dev] echo, delay and dropped packets in channel driver   Steve Kann
• [Asterisk-Dev] Re: Handling AST_FORMAT_SLINEAR at 48000Hz   Steve Kann
• [Asterisk-Dev] Re: cvs revision tags   Steve Kann
• [Asterisk-Dev] Sox Mixing Command With Monitor?   Nate Kapi
• [Asterisk-Dev] Sox Mixing Command With Monitor?   Nate Kapi
• [Asterisk-Dev] another little try   Roy Sigurd Karlsbakk
• [Asterisk-Dev] banned from the *-users list?   Roy Sigurd Karlsbakk
• [Asterisk-Dev] Asterisk Fax support using T.38   Lilantha Karunaratne
• [Asterisk-Dev] sco connection error: connection reset by peer   Khaled
• [Asterisk-Dev] Changing Port 5060 with 81   Abdul Lateef Khan
• [Asterisk-Dev] RE: Changing Port 5060 with 81   Abdul Lateef Khan
• [Asterisk-Dev] call forwarding   Abdul Lateef Khan
• [Asterisk-Dev] AGIphp Installation   Abdul Lateef Khan
• [Asterisk-Dev] AGIphp Installation   Abdul Lateef Khan
• [Asterisk-Dev] AGIphp Installation   Abdul Lateef Khan
• [Asterisk-Dev] festival voice   Abdul Lateef Khan
• [Asterisk-Dev] festival voice   Abdul Lateef Khan
• [Asterisk-Dev] Compiler debug switch in the main Asterisk Makefile   Kristian Kielhofner
• [Asterisk-Dev] Re: Require Help in Detection of Human Voice or Answering machine on Called telephone number   Claude Klimos
• [Asterisk-Dev] RFC2833 Event Duration   Andrew Kohlsmith
• [Asterisk-Dev] Echo cancel on PRI   Andrew Kohlsmith
• [Asterisk-Dev] Send tone to caller on answer   Kaloyan Kovachev
• [Asterisk-Dev] RE: Changing Port 5060 with 81   Kaloyan Kovachev
• [Asterisk-Dev] Re: Handling AST_FORMAT_SLINEAR at 48000Hz   Kaloyan Kovachev
• [Asterisk-Dev] generation of config files from source code ?   Kaloyan Kovachev
• [Asterisk-Dev] generation of config files from source code ?   Kaloyan Kovachev
• [Asterisk-Dev] FYI: Internet-Draft for IAX: Inter-Asterisk eXchange Version 2   Kaloyan Kovachev
• [Asterisk-Dev] echo, delay and dropped packets in channel driver   Ben Kramer
• [Asterisk-Dev] echo, delay and dropped packets in channel driver   Ben Kramer
• [Asterisk-Dev] echo, delay and dropped packets in channel driver   Ben Kramer
• [Asterisk-Dev] echo, delay and dropped packets in channel driver   Ben Kramer
• [Asterisk-Dev] Patch to allow Dial() to ignore call forward   John Lange
• [Asterisk-Dev] Security issue mumblings   Andrew Latham
• [Asterisk-Dev] Asterisk make clean and make looping??   Tilghman Lesher
• [Asterisk-Dev] musiconhold -vs- musicclass problems setting the differnt class of music   Tilghman Lesher
• [Asterisk-Dev] Asterisk or Polycom Bug?   Tilghman Lesher
• [Asterisk-Dev] Security issue mumblings   Tilghman Lesher
• [Asterisk-Dev] Patch to allow Dial() to ignore call forward   Tilghman Lesher
• [Asterisk-Dev] E1 channel allocation   Tilghman Lesher
• [Asterisk-Dev] Re: bug 4252 - increasing delay over time channels in MeetMe   Tilghman Lesher
• [Asterisk-Dev] Re: bug 4252 - increasing delay over time channels in MeetMe   Tilghman Lesher
• [Asterisk-Dev] Re: bug 4252 - increasing delay over time channels in MeetMe   Tilghman Lesher
• [Asterisk-Dev] MeetMe participant time tracking   Tilghman Lesher
• [Asterisk-Dev] app_datetime unused ? and sayunixtime cleanup...   Tilghman Lesher
• [Asterisk-Dev] app_datetime unused ? and sayunixtime cleanup...   Tilghman Lesher
• [Asterisk-Dev] Handling AST_FORMAT_SLINEAR at 48000Hz   Tilghman Lesher
• [Asterisk-Dev] cvs revision tags   Tilghman Lesher
• [Asterisk-Dev] generation of config files from source code ?   Tilghman Lesher
• [Asterisk-Dev] regular expression behavior   Tilghman Lesher
• [Asterisk-Dev] Problem with hanging up a SIP channel   Tilghman Lesher
• [Asterisk-Dev] Problem with hanging up a SIP channel   Tilghman Lesher
• [Asterisk-Dev] res_config_mysql.c connection problems => bug   Tilghman Lesher
• [Asterisk-Dev] Asterisk project converts to Subversion version control system   Tilghman Lesher
• [Asterisk-Dev] New SVN email subject   Tilghman Lesher
• [Asterisk-Dev] AST_FLAG_DEFER_DTMF (would like dialogic-r4 like semantics)   Tilghman Lesher
• 回复： Re: [Asterisk-Dev] queue_log for agent stats   Gary Li
• [Asterisk-Dev] AST_FLAG_DEFER_DTMF (would like dialogic-r4 like semantics)   Greg Lim
• [Asterisk-Dev] New ChangeLog in CVS HEAD Asterisk tree   Simon Lockhart
• [Asterisk-Dev] Install destination on Solaris   Simon Lockhart
• [Asterisk-Dev] Install destination on Solaris   Simon Lockhart
• [Asterisk-Dev] Install destination on Solaris   Simon Lockhart
• [Asterisk-Dev] $10 Bounty for Bug #5407   Alexander Lopez
• [Asterisk-Dev] How are SIP calls connected/bridged ?   Alexander Lopez
• [Asterisk-Dev] 1.2.0 Manager Action: Agents bug?   Alexander Lopez
• [Asterisk-Dev] Security issue mumblings   Alexander O. Lopez
• [Asterisk-Dev] Source Code for PalmTool 1.47 available anywhere?   Ralf P. Loserth
• [Asterisk-Dev] Require Help in Detection of Human Voice or Answering machine on Called telephone number   Richard Lyman
• [Asterisk-Dev] Help X101p kewlstart   MZ
• [Asterisk-Dev] Subversion file permissions   Mikael Magnusson
• [Asterisk-Dev] Chan_sip: video capabilities, call bandwidth and RTCP   John Martin
• [Asterisk-Dev] Chan_sip: video capabilities, call bandwidthand RTCP   John Martin
• [Asterisk-Dev] Asterisk or Polycom Bug?   Martin Mateev
• [Asterisk-Dev] Asterisk or Polycom Bug?   Will McCown
• [Asterisk-Dev] Asterisk or Polycom Bug?   Will McCown
• [Asterisk-Dev] Asterisk or Polycom Bug? - Message waiting   Will McCown
• [Asterisk-Dev] Problems with zap lines in 1.2 Beta 2 or 1.2 RC1, but ok in Beta1   Will McCown
• [Asterisk-Dev] reload causing SIP phone to not be registered   Sherwood McGowan
• [Asterisk-Dev] extensions.conf limitations, and how to overcome them   Steve McMahon
• [Asterisk-Dev] 482 Loop Detected problem   Jeremy McNamara
• [Asterisk-Dev] Bounty - Connecting Asterisk to SER   Jeremy McNamara
• [Asterisk-Dev] Re: chan_misdn in asterisk beta 2 ?   Doug Meredith
• [Asterisk-Dev] Re: 482 Loop Detected problem   Doug Meredith
• [Asterisk-Dev] Re: 482 Loop Detected problem   Doug Meredith
• [Asterisk-Dev] Re: Echo cancel on PRI   Doug Meredith
• [Asterisk-Dev] Sox Mixing Command With Monitor?   Jorge Merlino
• [Asterisk-Dev] Asterisk app_ices   Daniel Mikusa
• [Asterisk-Dev] error while building zaptal   Daniel Mikusa
• [Asterisk-Dev] res_config_mysql.c connection problems => bug   Ariel Monaco
• [Asterisk-Dev] Sip Invite in application   Markus Monka
• [Asterisk-Dev] Local channel optimisation   Tony Mountifield
• [Asterisk-Dev] Re: other -Werror issues - time_t printf format...   Tony Mountifield
• [Asterisk-Dev] Re: extensions.conf limitations, and how to overcome them   Tony Mountifield
• [Asterisk-Dev] Re: sip show peers doesn't work?   Tony Mountifield
• [Asterisk-Dev] Re: bug 4252 - increasing delay over time channels in MeetMe   Tony Mountifield
• [Asterisk-Dev] Inconsistencies between .call files and Manager API   Tony Mountifield
• [Asterisk-Dev] Re: bug 4252 - increasing delay over time channels in MeetMe   Tony Mountifield
• [Asterisk-Dev] Re: 1.2.0-beta1 MeetMe bug?   Tony Mountifield
• [Asterisk-Dev] Re: Handling AST_FORMAT_SLINEAR at 48000Hz   Tony Mountifield
• [Asterisk-Dev] Re: A question about G option of Dial   Tony Mountifield
• [Asterisk-Dev] Re: Handling AST_FORMAT_SLINEAR at 48000Hz   Tony Mountifield
• [Asterisk-Dev] Re: cvs revision tags   Tony Mountifield
• [Asterisk-Dev] Re: cvs revision tags   Tony Mountifield
• [Asterisk-Dev] Re: cvs revision tags   Tony Mountifield
• [Asterisk-Dev] lock.h change on 10-28 (rev 1.39) means you can never use pthread_mutex_t?   Dinesh Nair
• [Asterisk-Dev] Security issue mumblings - SIP   Dinesh Nair
• [Asterisk-Dev] mtp-2   Dinesh Nair
• [Asterisk-Dev] get register info from peer section in sip.conf ?   Dinesh Nair
• [Asterisk-Dev] Require Help in Detection of Human Voice or Answering machine on Called telephone number   Justin Newman
• [Asterisk-Dev] Asterisk Fax support using T.38   Florian Overkamp
• [Asterisk-Dev] Echo cancel on PRI   Florian Overkamp
• [Asterisk-Dev] Install destination on Solaris   Chris Parker
• [Asterisk-Dev] Install destination on Solaris   Chris Parker
• [Asterisk-Dev] Current HEAD: res_config_odbc.c compilation failure   Patrick
• [Asterisk-Dev] Current HEAD: res_config_odbc.c compilation failure   Patrick
• [Asterisk-Dev] asterisk.spec and the redhat dir   Patrick
• [Asterisk-Dev] asterisk.spec and the redhat dir   Patrick
• [Asterisk-Dev] zaptel-1.2.0 compilation fails on 2.5.15-rc2-git6   Patrick
• [Asterisk-Dev] zaptel-1.2.0 compilation fails on 2.5.15-rc2-git6   Patrick
• [Asterisk-Dev] Asterisk on Cygiwn   Paul
• [Asterisk-Dev] Asterisk on Cygiwn   Paul
• [Asterisk-Dev] Meetme: Sending DTMF to other users in a conference   Vamsi Pottangi
• [Asterisk-Dev] chan_bluetooth   Brent Priddy
• [Asterisk-Dev] chan_bluetooth   Brent Priddy
• [Asterisk-Dev] chan_bluetooth   Brent Priddy
• [Asterisk-Dev] chan_bluetooth and asterisk-1.2   Brent Priddy
• [Asterisk-Dev] Re: multiple registrations of same credentials   Michael Procter
• [Asterisk-Dev] Re: multiple registrations of same credentials   Michael Procter
• [Asterisk-Dev] asterisk.spec and the redhat dir   Jason Pyeron
• [Asterisk-Dev] asterisk.spec and the redhat dir   Jason Pyeron
• [Asterisk-Dev] trivial patch requesting comments before filing   Jason Pyeron
• [Asterisk-Dev] asterisk.spec and the redhat dir   Jason Pyeron
• [Asterisk-Dev] asterisk.spec and the redhat dir   Jason Pyeron
• [Asterisk-Dev] asterisk.spec and the redhat dir   Jason Pyeron
• [Asterisk-Dev] asterisk.spec and the redhat dir   Jason Pyeron
• [Asterisk-Dev] trivial patch requesting comments before filing   Jason Pyeron
• [Asterisk-Dev] asterisk.spec and the redhat dir   Jason Pyeron
• [Asterisk-Dev] Asterisk on Cygiwn   Jason Pyeron
• [Asterisk-Dev] New manager API command   Stefan Reuter
• [Asterisk-Dev] chan_misdn in asterisk beta 2 ?   Christian Richter
• [Asterisk-Dev] Asterisk make clean and make looping?? (LONG)   Christian Richter
• [Asterisk-Dev] Re: chan_misdn in asterisk beta 2 ?   Christian Richter
• [Asterisk-Dev] AST_FRAME_DATA   Christian Richter
• [Asterisk-Dev] AST_FRAME_DATA   Christian Richter
• [Asterisk-Dev] AST_FRAME_DATA   Christian Richter
• [Asterisk-Dev] Asterisk or Polycom Bug?   Matt Riddell
• [Asterisk-Dev] Asterisk or Polycom Bug?   Matt Riddell
• [Asterisk-Dev] extensions.conf limitations, and how to overcome them   Matt Riddell
• [Asterisk-Dev] [OTAnn] Feedback   Matt Riddell
• [Asterisk-Dev] New manager API command   Matt Riddell
• [Asterisk-Dev] Asterisk Fax support using T.38   Matt Riddell
• [Asterisk-Dev] Can anoncvs make 'cvs commit' in CVS asterisk ?   Matt Riddell
• [Asterisk-Dev] Develop SoftPhone with IAXclient!!   Matt Riddell
• [Asterisk-Dev] Asterisk on Cygiwn   Matt Riddell
• [Asterisk-Dev] chan_exosip2 for asterisk   Matt Riddell
• [Asterisk-Dev] Asterisk on Cygiwn   Matt Riddell
• [Asterisk-Dev] Re: bug 4252 - increasing delay over time channels in MeetMe   Matt Riddell
• [Asterisk-Dev] chan_exosip2 for asterisk   Matt Riddell
• [Asterisk-Dev] Re: bug 4252 - increasing delay over time channels in MeetMe   Matt Riddell
• [Asterisk-Dev] banned from the *-users list?   Matt Riddell
• [Asterisk-Dev] LDAP RealTime driver   Matt Riddell
• [Asterisk-Dev] Need help for developing application like "musiconhold" . . .   Matt Riddell
• [Asterisk-Dev] Source Code for PalmTool 1.47 available anywhere?   Matt Riddell
• [Asterisk-Dev] hello   Matt Riddell
• [Asterisk-Dev] Chan_sip: video capabilities, call bandwidth and RTCP   Matt Riddell
• [Asterisk-Dev] Chan_sip: video capabilities, call bandwidthand RTCP   Matt Riddell
• [Asterisk-Dev] Require Help in Detection of Human Voice or Answering machine on Called telephone number   Matt Riddell
• [Asterisk-Dev] res_config_mysql.c connection problems => bug   Matt Riddell
• [Asterisk-Dev] res_config_mysql.c connection problems => bug   Matt Riddell
• [Asterisk-Dev] always_inline change...   Luigi Rizzo
• [Asterisk-Dev] other -Werror issues - time_t printf format...   Luigi Rizzo
• [Asterisk-Dev] other -Werror issues - time_t printf format...   Luigi Rizzo
• [Asterisk-Dev] say.c cleanup   Luigi Rizzo
• [Asterisk-Dev] say.c cleanup, request for clarification   Luigi Rizzo
• [Asterisk-Dev] say.c cleanup   Luigi Rizzo
• [Asterisk-Dev] say.c cleanup -feedback wanted   Luigi Rizzo
• [Asterisk-Dev] app_datetime unused ? and sayunixtime cleanup...   Luigi Rizzo
• [Asterisk-Dev] app_datetime unused ? and sayunixtime cleanup...   Luigi Rizzo
• [Asterisk-Dev] app_datetime unused ? and sayunixtime cleanup...   Luigi Rizzo
• [Asterisk-Dev] Re: Handling AST_FORMAT_SLINEAR at 48000Hz   Luigi Rizzo
• [Asterisk-Dev] generation of config files from source code ?   Luigi Rizzo
• [Asterisk-Dev] generation of config files from source code ?   Luigi Rizzo
• [Asterisk-Dev] generation of config files from source code ?   Luigi Rizzo
• [Asterisk-Dev] Action item :: Check if your patch is up to date!   Luigi Rizzo
• [Asterisk-Dev] get register info from peer section in sip.conf ?   Luigi Rizzo
• [Asterisk-Dev] get register info from peer section in sip.conf ?   Luigi Rizzo
• [Asterisk-Dev] no DNID copied when bridging channels ?   Luigi Rizzo
• [Asterisk-Dev] no DNID copied when bridging channels ?   Luigi Rizzo
• [Asterisk-Dev] possible bug in app_dial.c ?   Luigi Rizzo
• [Asterisk-Dev] gmake issues   Luigi Rizzo
• [Asterisk-Dev] gmake issues   Luigi Rizzo
• [Asterisk-Dev] gmake issues (solved, #5885 on mantis)   Luigi Rizzo
• [Asterisk-Dev] gmake issues (solved, #5885 on mantis)   Luigi Rizzo
• [Asterisk-Dev] Compiler debug switch in the main Asterisk Makefile   Steve Rodgers
• [Asterisk-Dev] Compiler debug switch in the main Asterisk Makefile   Steve Rodgers
• [Asterisk-Dev] Line status detection for Zap ifaces ( fxo )   Gabriel Rojas
• [Asterisk-Dev] Line status detection for Zap ifaces ( fxo ) ( cont'd )   Gabriel Rojas
• [Asterisk-Dev] Call-Limit   Rushowr
• [Asterisk-Dev] Call-Limit   Rushowr
• [Asterisk-Dev] 482 Loop Detected problem   Rushowr
• [Asterisk-Dev] Conditional Goto and GoSub use 'n'?   Rushowr
• [Asterisk-Dev] Small question about CDR's   Jan Saell
• [Asterisk-Dev] Re: [Asterisk-Users] New asterisk management tool   Jan Saell
• [Asterisk-Dev] sip gateway provider recommendations?   Jan Saell
• [Asterisk-Dev] Missing RTP header in iLBC   Jan Saell
• [Asterisk-Dev] Require Help in Detection of Human Voice or Answering machine on Called telephone number   Muhammad Asim Sajjad
• [Asterisk-Dev] say.c cleanup   Frank Sautter
• [Asterisk-Dev] Subversion checkouts   Richard Scobie
• [Asterisk-Dev] ZT_DIAL Operation   Sergio Serrano
• [Asterisk-Dev] ZT_DIAL Operation   Sergio Serrano
• [Asterisk-Dev] Bounty - Connecting Asterisk to SER   Shidan
• [Asterisk-Dev] Send tone to caller on answer   Andreas Sikkema
• [Asterisk-Dev] app_festival normal behaviour??   Moises Silva
• [Asterisk-Dev] Zap gain changing patch   Denis Smirnov
• [Asterisk-Dev] Patch to allow Dial() to ignore call forward   Jared Smith
• [Asterisk-Dev] Voicemail password changes not sticking   Jared Smith
• [Asterisk-Dev] Hardware Manual   Jared Smith
• [Asterisk-Dev] chan_exosip2   Jared Smith
• [Asterisk-Dev] Interrupted playback of messages (ast_waitstream)   Jared Smith
• [Asterisk-Dev] mtp-2   Jared Smith
• [Asterisk-Dev] regular expression behavior   Jared Smith
• [Asterisk-Dev] G.729 SDP bug in sip_chan.c (asterisk v1.2.0)   Jared Smith
• [Asterisk-Dev] HELP! on disconnecting stale calls.   Jared Smith
• [Asterisk-Dev] app_festival normal behaviour??   Jared Smith
• [Asterisk-Dev] Re: cvs revision tags   SteveK
• [Asterisk-Dev] Collect Calls   Steven
• [Asterisk-Dev] Asterisk on Cygiwn   Steven
• [Asterisk-Dev] chan_exosip2   Steven
• [Asterisk-Dev] Asterisk on Cygiwn   Steven
• [Asterisk-Dev] Interrupted playback of messages (ast_waitstream)   Lars Sundqvist
• [Asterisk-Dev] Asterisk project converts to Subversion version control system   Mike Taht
• [Asterisk-Dev] Asterisk make clean and make looping?? (Possiblesolution)   Soner Tari
• [Asterisk-Dev] Asterisk make clean and make looping?? (Possible solution)   David Taylor
• [Asterisk-Dev] Re: Asterisk make clean and make looping?? (Possible solution)   David Taylor
• [Asterisk-Dev] Asterisk 1.2 Released!   Asterisk Development Team
• [Asterisk-Dev] Asterisk project converts to Subversion version control system   Asterisk Development Team
• [Asterisk-Dev] Need help for developing application like "musiconhold" . . .   Pratik Thakkar
• [Asterisk-Dev] mtp-2   Jacob Tinning
• [Asterisk-Dev] Getting caller id   Jacob Tinning
• [Asterisk-Dev] Handling AST_FORMAT_SLINEAR at 48000Hz   Jacob Tinning
• [Asterisk-Dev] Security issue mumblings   John Todd
• [Asterisk-Dev] Recommendations for SIP PBX/SP interoperability draft   John Todd
• [Asterisk-Dev] Re: Require Help in Detection of Human Voice or Answering machine on Called telephone number   John Todd
• [Asterisk-Dev] Install destination on Solaris   Stephen Uhler
• [Asterisk-Dev] has somebody ever tested faxdetection ?   Steve Underwood
• [Asterisk-Dev] has somebody ever tested faxdetection ?   Steve Underwood
• [Asterisk-Dev] Re: bug 4252 - increasing delay over time channels in MeetMe   Steve Underwood
• [Asterisk-Dev] Require Help in Detection of Human Voice or Answering machine on Called telephone number   Steve Underwood
• [Asterisk-Dev] reload causing SIP phone to not be registered   Asterisk User
• [Asterisk-Dev] Problem in call conference   Vinay
• [Asterisk-Dev] Re: Fw: Portace JUDR. Jan Grosam   Martin Vit
• [Asterisk-Dev] Re: TDMoE + kernel badness   Martin Vit
• [Asterisk-Dev] bug or expected behaviour : vars in template contexts (1.2.0/HEAD)   Bruno Voigt
• [Asterisk-Dev] Multiple Pickup   Chris Wade
• [Asterisk-Dev] Get number function   Chris Wade
• [Asterisk-Dev] Module that works in Asterisk 1.0.x and 1.2.x   Chris Wade
• [Asterisk-Dev] [repost] cli output issue   Christopher L. Wade
• [Asterisk-Dev] How to start?   Isack Waserman
• [Asterisk-Dev] UA   Isack Waserman
• [Asterisk-Dev] error while building zaptal   Isack Waserman
• [Asterisk-Dev] voicemail   Isack Waserman
• [Asterisk-Dev] Additions to function DUNDiLookup   Watkins, Bradley
• [Asterisk-Dev] extensions.conf limitations, and how to overcome them   Quinn Weaver
• [Asterisk-Dev] Asterisk make clean and make looping??   Robert Webb
• [Asterisk-Dev] Asterisk make clean and make looping??   Robert Webb
• [Asterisk-Dev] Asterisk make clean and make looping??   Robert Webb
• [Asterisk-Dev] Asterisk make clean and make looping?? (LONG)   Robert Webb
• [Asterisk-Dev] Asterisk make clean and make looping?? (Possible solution)   Robert Webb
• [Asterisk-Dev] Asterisk make clean and make looping?? (Possible solution)   Robert Webb
• [Asterisk-Dev] Asterisk make clean and make looping?? (Possible solution)   Robert Webb
• [Asterisk-Dev] Asterisk make clean and make looping?? (Possible solution)   Robert Webb
• [Asterisk-Dev] Asterisk make clean and make looping?? (Possiblesolution)   Robert Webb
• [Asterisk-Dev] A problem of MixMonitor   BJ Weschke
• [Asterisk-Dev] Asterisk Locking - queue application   BJ Weschke
• [Asterisk-Dev] musiconhold -vs- musicclass problems setting the differnt class of music   BJ Weschke
• [Asterisk-Dev] Current HEAD: res_config_odbc.c compilation failure   BJ Weschke
• [Asterisk-Dev] How to start?   BJ Weschke
• [Asterisk-Dev] Collect Calls   BJ Weschke
• [Asterisk-Dev] Collect Calls   BJ Weschke
• [Asterisk-Dev] Monitor Function broken since Beta-2 ?   BJ Weschke
• [Asterisk-Dev] Asterisk 1.2.0-beta2 ParkAndAnnounce question   BJ Weschke
• [Asterisk-Dev] Delphi ActiveX component   BJ Weschke
• [Asterisk-Dev] app_page   BJ Weschke
• [Asterisk-Dev] chan_misdn in asterisk beta 2 ?   Eric "ManxPower" Wieling
• [Asterisk-Dev] $10 Bounty for Bug #5407   Eric "ManxPower" Wieling
• [Asterisk-Dev] Echo cancel on PRI   Eric "ManxPower" Wieling
• [Asterisk-Dev] Re: chan_misdn in asterisk beta 2 ?   David Woodhouse
• [Asterisk-Dev] AST_FRAME_DATA   David Woodhouse
• [Asterisk-Dev] chan_bluetooth   David Woodhouse
• [Asterisk-Dev] chan_bluetooth   David Woodhouse
• [Asterisk-Dev] chan_bluetooth   David Woodhouse
• [Asterisk-Dev] chan_bluetooth   David Woodhouse
• [Asterisk-Dev] chan_bluetooth and asterisk-1.2   David Woodhouse
• [Asterisk-Dev] chan_bluetooth and asterisk-1.2   David Woodhouse
• [Asterisk-Dev] chan_bluetooth and asterisk-1.2   David Woodhouse
• [Asterisk-Dev] chan_bluetooth and asterisk-1.2   David Woodhouse
• [Asterisk-Dev] zaptel-1.2.0 compilation fails on 2.5.15-rc2-git6   David Woodhouse
• [Asterisk-Dev] zaptel-1.2.0 compilation fails on 2.5.15-rc2-git6   David Woodhouse
• [Asterisk-Dev] gmake error for asterisk-addons-1.2.0   Vahan Yerkanian
• [Asterisk-Dev] Asterisk project converts to Subversion version control system   Vahan Yerkanian
• [Asterisk-Dev] queue_log for agent stats   Darren Younger
• [Asterisk-Dev] Billing and consultative transfer   Sebastian Zaprzalski
• [Asterisk-Dev] ooh323 crash   Ma Zhiyong
• [Asterisk-Dev] ooh323 cause asterisk reload broken   Ma Zhiyong
• [Asterisk-Dev] chan_misdn in asterisk beta 2 ?   Zoa
• [Asterisk-Dev] chan_misdn in asterisk beta 2 ? - my mistake   Zoa
• [Asterisk-Dev] Re: chan_misdn in asterisk beta 2 ?   Zoa
• [Asterisk-Dev] chan_misdn in asterisk beta 2 ?   Zoa
• [Asterisk-Dev] chan_misdn in asterisk beta 2 ?   Zoa
• [Asterisk-Dev] chan_bluetooth   Theo Zourzouvillys
• [Asterisk-Dev] chan_bluetooth   Theo Zourzouvillys
• [Asterisk-Dev] chan_bluetooth   Theo Zourzouvillys
• [Asterisk-Dev] chan_bluetooth   Theo Zourzouvillys
• [Asterisk-Dev] chan_bluetooth   Theo Zourzouvillys
• [Asterisk-Dev] Security issue mumblings - SIP   imran ahmed
• [Asterisk-Dev] Security issue mumblings - SIP   imran ahmed
• [Asterisk-Dev] MeetMe participant time tracking   imran ahmed
• [Asterisk-Dev] cvs revision tags   imran ahmed
• [Asterisk-Dev] Echo cancel on PRI   imran ahmed
• [Asterisk-Dev] Re: Echo cancel on PRI   imran ahmed
• [Asterisk-Dev] Sox Mixing Command With Monitor?   imran ahmed
• [Asterisk-Dev] Problem with hanging up a SIP channel   imran ahmed
• [Asterisk-Dev] Problem with hanging up a SIP channel   imran ahmed
• [Asterisk-Dev] Problem with hanging up a SIP channel   imran ahmed
• [Asterisk-Dev] Problem with hanging up a SIP channel   imran ahmed
• [Asterisk-Dev] Re: Queue - Transfer & server stability   alan
• [Asterisk-Dev] 1.2.0 Manager Action: Agents bug?   alan
• [Asterisk-Dev] Re: 1.2.0 Manager Action: Agents bug?   alan
• [Asterisk-Dev] Re: bug 4252 - increasing delay over time channels in MeetMe   alex at pilosoft.com
• [Asterisk-Dev] Echo cancel on PRI   alex at pilosoft.com
• [Asterisk-Dev] An Ast-user question for Ast-dev experts   alex at pilosoft.com
• [Asterisk-Dev] New Jersey ATT Vocie T1 Asterisk Toll free number not working   alex at pilosoft.com
• [Asterisk-Dev] extensions.conf limitations, and how to overcome them   asterisk at funkspiel.org
• [Asterisk-Dev] extensions.conf limitations, and how to overcome them   asterisk at funkspiel.org
• [Asterisk-Dev] Another weird dialplan question   asterisk at funkspiel.org
• [Asterisk-Dev] Asterisk project converts to Subversion version control system   asterisk at ntplx.net
• [Asterisk-Dev] Asterisk make clean and make looping??   asterisk at ropeguru.com
• [Asterisk-Dev] Asterisk make clean and make looping??   asterisk at ropeguru.com
• [Asterisk-Dev] Asterisk make clean and make looping?? (LONG)   asterisk at ropeguru.com
• [Asterisk-Dev] Asterisk make clean and make looping?? (Possible solution)   asterisk at ropeguru.com
• [Asterisk-Dev] chan_sip/sip.conf   bosinclar at laposte.net
• [Asterisk-Dev] Subversion file permissions   brett at websmyths.com
• [Asterisk-Dev] Delphi ActiveX component   contato at tomasfrota.eti.br
• [Asterisk-Dev] extensions.conf limitations, and how to overcome them   critch at basesys.com
• [Asterisk-Dev] Help Connecting Cisco Router to Asterisk   ddiffa
• [Asterisk-Dev] Asterisk 1.2 will have support for SIMPLE notifications of device presence   harry gaillac
• [Asterisk-Dev] chan_exosip2 for asterisk   harry gaillac
• [Asterisk-Dev] chan_exosip2 for asterisk   harry gaillac
• [Asterisk-Dev] chan_exosip2 for asterisk   harry gaillac
• [Asterisk-Dev] chan_exosip2   harry gaillac
• [Asterisk-Dev] chan_exosip2   harry gaillac
• [Asterisk-Dev] chan_exosip2   harry gaillac
• [Asterisk-Dev] chan_exosip2   harry gaillac
• [Asterisk-Dev] Tr: Re: [Asterisk-Users] open letter   harry gaillac
• [Asterisk-Dev] Nat   harry gaillac
• [Asterisk-Dev] nat   harry gaillac
• [Asterisk-Dev] hello   harry gaillac
• [Asterisk-Dev] Bounty - Connecting Asterisk to SER   harry gaillac
• [Asterisk-Dev] Bounty - Connecting Asterisk to SER   harry gaillac
• [Asterisk-Dev] Bounty - Connecting Asterisk to SER   harry gaillac
• [Asterisk-Dev] Bounty - Connecting Asterisk to SER   harry gaillac
• [Asterisk-Dev] Tr: RE : [Asterisk-Users] Asterisk doesn't start   harry gaillac
• [Asterisk-Dev] Changing Port 5060 with 81   ast guy
• [Asterisk-Dev] Getting caller id   ast guy
• [Asterisk-Dev] Getting caller id   ast guy
• [Asterisk-Dev] G.729 SDP bug in sip_chan.c (asterisk v1.2.0)   hcb+asterisk-dev at unco.de
• [Asterisk-Dev] asterisk 1.2 g729 compile errors   hcb+asterisk-dev at unco.de
• [Asterisk-Dev] Handling AST_FORMAT_SLINEAR at 48000Hz   voipwala hindustani
• [Asterisk-Dev] Handling AST_FORMAT_SLINEAR at 48000Hz   voipwala hindustani
• [Asterisk-Dev] Re: Handling AST_FORMAT_SLINEAR at 48000Hz   voipwala hindustani
• [Asterisk-Dev] Compiler debug switch in the main Asterisk Makefile   hwstar at rodgers.sdcoxmail.com
• [Asterisk-Dev] Compiler debug switch in the main Asterisk Makefile   hwstar at rodgers.sdcoxmail.com
• [Asterisk-Dev] Hardware Manual   janvb at caselaboratories.com
• [Asterisk-Dev] Hardware Manual - MTP2   janvb at caselaboratories.com
• [Asterisk-Dev] another little try   janvb at caselaboratories.com
• [Asterisk-Dev] mtp-2   janvb at caselaboratories.com
• [Asterisk-Dev] mtp-2   janvb at caselaboratories.com
• [Asterisk-Dev] mtp-2   janvb at caselaboratories.com
• [Asterisk-Dev] mtp-2   janvb at caselaboratories.com
• [Asterisk-Dev] mtp-2   janvb at caselaboratories.com
• [Asterisk-Dev] mtp-2   janvb at caselaboratories.com
• [Asterisk-Dev] mtp-2   janvb at caselaboratories.com
• [Asterisk-Dev] Bounty - Connecting Asterisk to SER   janvb at caselaboratories.com
• [Asterisk-Dev] Asterisk make clean and make looping??   tim panton
• [Asterisk-Dev] Asterisk make clean and make looping??   tim panton
• [Asterisk-Dev] Install destination on Solaris   tim panton
• [Asterisk-Dev] [OTAnn] Feedback   shenanigans
• [Asterisk-Dev] New manager API command   snacktime
• [Asterisk-Dev] New asterisk management tool   snacktime
• [Asterisk-Dev] Re: New asterisk management tool   snacktime
• [Asterisk-Dev] regular expression behavior   snacktime
• [Asterisk-Dev] regular expression behavior   snacktime
• [Asterisk-Dev] regular expression behavior   snacktime
• [Asterisk-Dev] lock.h change on 10-28 (rev 1.39) means you can never use pthread_mutex_t?   steve at daviesfam.org
• [Asterisk-Dev] since 2005-10-28, pthread_mutex_t can never be used in apps that include asterisk's lock.h ??   steve at daviesfam.org
• [Asterisk-Dev] lock.h change on 10-28 (rev 1.39) means you can never use pthread_mutex_t?   steve at daviesfam.org
• [Asterisk-Dev] lock.h change on 10-28 (rev 1.39) means you can never use pthread_mutex_t?   steve at daviesfam.org
• [Asterisk-Dev] Re: [Asterisk-Users] Warning -- chan_iax2.c: ast_sched_runq tasks   steve at daviesfam.org
• [Asterisk-Dev] echo, delay and dropped packets in channel driver   steve at daviesfam.org
• [Asterisk-Dev] Silence Detection in G729 payload   suresh suresh
• [Asterisk-Dev] SIP media negotiation problem   anthony thomas
• [Asterisk-Dev] Asterisk on Cygiwn   don vanfossen

Last message date: Wed Nov 30 23:49:00 MST 2005
Archived on: Tue Sep 5 14:27:44 MST 2006

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