September 2005 Archives by author

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Starting: Thu Sep 1 01:41:47 MST 2005
Ending: Fri Sep 30 21:02:18 MST 2005
Messages: 822

• [Asterisk-Dev] - Call progress event?   Daniel Montejo Biosca (hotmail)
• [Asterisk-Dev] Compiling Asterisk for dual architectue   Aarno Syvänen
• [Asterisk-Dev] AgentCallbackLogin, 1.2-beta1 issues   José Pablo Ezequiel Fernández
• [Asterisk-Dev] asterisk 'stable'?   José Pablo Ezequiel Fernández
• [Asterisk-Dev] FW: [Asterisk-Users] RxFax/TxFax - Compile Problem   José Pablo Ezequiel Fernández
• [Asterisk-Dev] Request Netiquite   José Pablo Ezequiel Fernández
• [Asterisk-Dev] removing depreciated code?   José Pablo Ezequiel Fernández
• [Asterisk-Dev] ast_channel_tech   José Pablo Ezequiel Fernández
• [Asterisk-Dev] delayed sip answer   Jens Kübler
• [Asterisk-Dev] Database backed Asterisk   Jens Kübler
• [Asterisk-Dev] Doxygen and comments   José Pablo Ezequiel Fernández
• [Asterisk-Dev] Realtime agents.   José Pablo Ezequiel Fernández
• [Asterisk-Dev] Realtime agents.   José Pablo Ezequiel Fernández
• [Asterisk-Dev] Agents in realtime   José Pablo Ezequiel Fernández
• [Asterisk-Dev] FW: [Asterisk-Users] RxFax/TxFax - Compile Problem   José Pablo Ezequiel Fernández
• [Asterisk-Dev] Re: #3986 bounty   José Pablo Ezequiel Fernández
• [Asterisk-Dev] Questions   Rich Adamson
• [Asterisk-Dev] Stable change log suggestion?   Rich Adamson
• [Asterisk-Dev] Stable change log suggestion?   Rich Adamson
• [Asterisk-Dev] Stable change log suggestion?   Rich Adamson
• [Asterisk-Dev] SIP Jitter Buffer Testing   Mark Aiken
• [Asterisk-Dev] SIPit17 in Stockholm :: Asterisk under pressure!!   Mark Aiken
• [Asterisk-Dev] what do I need to implement odbc   Mark Aiken
• Malfunctioning SendDTMF() and Playtones() (was: Re: [Asterisk-Dev] need help)   Mark Aiken
• [Asterisk-Dev] Database backed Asterisk   Mark Aiken
• [Asterisk-Dev] Database backed Asterisk   Mark Aiken
• [Asterisk-Dev] CSTA interface comments   Mark Aiken
• [Asterisk-Dev] SIP failover using Asterisk and openais   Mark Aiken
• [Asterisk-Dev] SIP failover using Asterisk and openais   Mark Aiken
• [Asterisk-Dev] [OT] [Very OT] socket connection from perl to visual basic   Mark Aiken
• [Asterisk-Dev] ASTCC-adding more than one trunk to one route   Faramarz Amidafshar
• [Asterisk-Dev] [patch] IM Manager Event   Andreas Anderson
• [Asterisk-Dev] cable modems?   Colin Anderson
• [Asterisk-Dev] TDMoE and tap device   Colin Anderson
• [Asterisk-Dev] TDMoE and tap device   Colin Anderson
• [Asterisk-Dev] Asterisk and AMD64 - Crashes   Woody Anderson
• [Asterisk-Dev] Asterisk and AMD64 - Crashes   Woody Anderson
• [Asterisk-Dev] Asterisk and AMD64 - Crashes   Woody Anderson
• [Asterisk-Dev] asterisk 'stable'?   Woody Anderson
• [Asterisk-Dev] config mgmt plan?   Woody Anderson
• [Asterisk-Dev] Real-time Linux claims single-digit microsecond responsiveness   Carlos Antunes
• [Asterisk-Dev] Is the ChanIsAvail command thread safe?   Carlos Antunes
• [Asterisk-Dev] Is the ChanIsAvail command thread safe?   Carlos Antunes
• [Asterisk-Dev] Re: Answering Machine detection   Carlos Antunes
• [Asterisk-Dev] callerid spill   Gurminder Arora
• [Asterisk-Dev] Doesn't finishes callerid spill   Gurminder Arora
• [Asterisk-Dev] PLEASE HELP!! CALLERID FAILS!!   Gurminder Arora
• [Asterisk-Dev] Re: [Asterisk-Users] MusicOnHold not working   Gurminder Arora
• [Asterisk-Dev] App directed pickup (bug #4865)   Joshua Colp - Asterlink
• [Asterisk-Dev] SpanDSP memory leak?   Domjan Attila
• [Asterisk-Dev] The CallerIDPres not working?   Domjan Attila
• [Asterisk-Dev] Input and testing.   Dan Austin
• [Asterisk-Dev] oh323 inband-info AST_CONTROL_PROGRESS   Igor - BZ
• [Asterisk-Dev] spelling error in iax.conf.sample   Michiel van Baak
• [Asterisk-Dev] Questions   Michiel van Baak
• [Asterisk-Dev] Want to add a VM delete event - where in app_voicemail.c?   Jordan Bean
• [Asterisk-Dev] Want to add a VM delete event - where inapp_voicemail.c?   Jordan Bean
• [Asterisk-Dev] Want to add a VM delete event - wherein app_voicemail.c?   Jordan Bean
• Advantage / Disadvantage to install asterisk from CVS was: [Asterisk-Dev] Asterisk run as non root   Joseph Benden
• [Asterisk-Dev] Bug with toll free between 2 asterisk servers (special toll free signaling?)   Joseph Benden
• [Asterisk-Dev] codec issues   Dov Bigio
• [Asterisk-Dev] H323/SIP<--->PSTN with asterisk. Is it Possible?   Reynold Blur
• [Asterisk-Dev] Re: [Asterisk-cvs] asterisk logger.c,1.81,1.82   Matthew Boehm
• [Asterisk-Dev] Can't Add SDP Madness   Matthew Boehm
• [Asterisk-Dev] Can't Add SDP Madness   Matthew Boehm
• [Asterisk-Dev] Properly Destroying Variables   Matthew Boehm
• [Asterisk-Dev] Properly Destroying Variables   Matthew Boehm
• [Asterisk-Dev] Asterisk and AMD64 - Crashes   Greg Boehnlein
• [Asterisk-Dev] asterisk 'stable'?   Greg Boehnlein
• [Asterisk-Dev] asterisk 'stable'?   Greg Boehnlein
• [Asterisk-Dev] asterisk 'stable'?   Greg Boehnlein
• [Asterisk-Dev] asterisk 'stable'?   Greg Boehnlein
• [Asterisk-Dev] Questions   Greg Boehnlein
• [Asterisk-Dev] Questions   Greg Boehnlein
• [Asterisk-Dev] Questions   Greg Boehnlein
• [Asterisk-Dev] Re: #3986 bounty   Greg Boehnlein
• [Asterisk-Dev] Suspected memory leak in 1.0.9   Greg Boehnlein
• [Asterisk-Dev] chan_h323 1.128 compile fix   Brad Borgald
• [Asterisk-Dev] "register" statement: why can't it reference an account?   Brad Borgald
• [Asterisk-Dev] Real-time Linux claims single-digit microsecond responsiveness   Rus V. Brushkoff
• [Asterisk-Dev] RSA auth broken in IAX2?   Russell Bryant
• [Asterisk-Dev] CVS Servers   Russell Bryant
• [Asterisk-Dev] removing depreciated code?   Russell Bryant
• [Asterisk-Dev] Stable change log suggestion?   Russell Bryant
• [Asterisk-Dev] Stable change log suggestion?   Russell Bryant
• [Asterisk-Dev] Open source time card application for Asterisk   Chuck Bunn
• [Asterisk-Dev] Open source time card application for Asterisk   Chuck Bunn
• [Asterisk-Dev] Open source time card application for Asterisk   Chuck Bunn
• [Asterisk-Dev] Re: Asterisk Time Card   Chuck Bunn
• [Asterisk-Dev] Open source time card application for Asterisk   Chuck Bunn
• [Asterisk-Dev] Asterisk AGI php, and Asterisk PHP are written as functions instead of classes and objects.   Chuck Bunn
• [Asterisk-Dev] Open source time card application for Asterisk   Chuck Bunn
• [Asterisk-Dev] Asterisk AGI php, and Asterisk PHP are written as functions instead of classes and objects.   Chuck Bunn
• [Asterisk-Dev] Open source time card application for Asterisk   Chuck Bunn
• [Asterisk-Dev] Asterisk Bounty VoiceMail-n-Email Synchronization =$1125   Matthew Butt
• [Asterisk-Dev] startup   Nir Simionovich - CTO
• [Asterisk-Dev] [Fwd: 802.1p support via SO_Priority& 8021q modulein RTP.c]   Paul Cadach
• [Asterisk-Dev] Redirection reason   Paul Cadach
• [Asterisk-Dev] How complete is Q.SIG support?   Paul Cadach
• [Asterisk-Dev] ISDN SETUP on incoming SMS?   Paul Cadach
• [Asterisk-Dev] ISDN SETUP on incoming SMS?   Paul Cadach
• [Asterisk-Dev] ISDN SETUP on incoming SMS?   Paul Cadach
• [Asterisk-Dev] ISDN SETUP on incoming SMS?   Paul Cadach
• [Asterisk-Dev] Is CVS-HEAD the next 1.2 Beta?   Brian Capouch
• [Asterisk-Dev] Is CVS-HEAD the next 1.2 Beta?   Brian Capouch
• [Asterisk-Dev] I need help to make a simple iax switch/proxy   Brian Capouch
• [Asterisk-Dev] asterisk 'stable'?   Brian Capouch
• [Asterisk-Dev] NoOp not returning anything?   Brian Capouch
• [Asterisk-Dev] removing depreciated code?   Brian Capouch
• [Asterisk-Dev] TDMoE and tap device   Dome Charoenyost
• [Asterisk-Dev] app_dial.c ANSWEREDTIME value is not return when caller cancels the call   Raymond Chen
• [Asterisk-Dev] question about ast_async_goto   Sergio Chersovani
• [Asterisk-Dev] #3986 bounty   Sergio Chersovani
• [Asterisk-Dev] #3986 bounty   Sergio Chersovani
• [Asterisk-Dev] Open source time card application for Asterisk   Mike Clark
• [Asterisk-Dev] Open G.729 / G.723.1 update, fixed memory leak   Tzafrir Cohen
• [Asterisk-Dev] Open G.729 / G.723.1 update, fixed memory leak   Tzafrir Cohen
• [Asterisk-Dev] Asterisk and AMD64 - Crashes   Tzafrir Cohen
• Advantage / Disadvantage to install asterisk from CVS was: [Asterisk-Dev] Asterisk run as non root   Tzafrir Cohen
• Advantage / Disadvantage to install asterisk from CVS was: [Asterisk-Dev] Asterisk run as non root   Tzafrir Cohen
• [Asterisk-Dev] asterisk 'stable'?   Tzafrir Cohen
• [Asterisk-Dev] need help   Tzafrir Cohen
• [Asterisk-Dev] maximum concurrent ZAP channels .... max conf ports ...   Tzafrir Cohen
• [Asterisk-Dev] Questions   Tzafrir Cohen
• [Asterisk-Dev] Asterisk and GlobalCall   Tzafrir Cohen
• [Asterisk-Dev] removing depreciated code?   Tzafrir Cohen
• [Asterisk-Dev] Stable change log suggestion?   Tzafrir Cohen
• [Asterisk-Dev] Voicemail password change + Ast Realtime   Tzafrir Cohen
• [Asterisk-Dev] Mac Mini on Debian - Asterisk   Tzafrir Cohen
• [Asterisk-Dev] Maximum wait for I/O in bridged channels   Juan Jose Comellas
• [Asterisk-Dev] Zaptel compile error after kernel switch to SMP   David Cook
• [Asterisk-Dev] Asterisk Compiled on Arm with static libraries   Jawhny Cooke
• [Asterisk-Dev] Asterisk and AMD64 - Crashes   Dave Cotton
• [Asterisk-Dev] Asterisk and AMD64 - Crashes   Dave Cotton
• Advantage / Disadvantage to install asterisk from CVS was: [Asterisk-Dev] Asterisk run as non root   Dave Cotton
• [Asterisk-Dev] getting started   Jonathan k. Creasy
• [Asterisk-Dev] getting started   Jonathan k. Creasy
• [Asterisk-Dev] getting started   Jonathan k. Creasy
• [Asterisk-Dev] getting started   Jonathan k. Creasy
• [Asterisk-Dev] asterisk 'stable'?   Jonathan k. Creasy
• [Asterisk-Dev] need help   Jonathan k. Creasy
• [Asterisk-Dev] I need help to make a simple iax switch/proxy   Steven Critchfield
• [Asterisk-Dev] Asterisk and AMD64 - Crashes   Steven Critchfield
• [Asterisk-Dev] asterisk 'stable'?   Steven Critchfield
• [Asterisk-Dev] asterisk 'stable'?   Steven Critchfield
• [Asterisk-Dev] asterisk 'stable'?   Steven Critchfield
• [Asterisk-Dev] Newbie needs advice   Steven Critchfield
• [Asterisk-Dev] High Availability again   Steven Critchfield
• [Asterisk-Dev] Questions   Steven Critchfield
• [Asterisk-Dev] need help   Steven Critchfield
• [Asterisk-Dev] need help   Steven Critchfield
• [Asterisk-Dev] Open source time card application for Asterisk   Steven Critchfield
• [Asterisk-Dev] removing depreciated code?   Steven Critchfield
• [Asterisk-Dev] Open source time card application for Asterisk   Steven Critchfield
• [Asterisk-Dev] GUI's   Steven Critchfield
• [Asterisk-Dev] GUI's   Steven Critchfield
• [Asterisk-Dev] ISDN SETUP on incoming SMS?   Steven Critchfield
• [Asterisk-Dev] ast_channel_tech   Steven Critchfield
• [Asterisk-Dev] SIP failover using Asterisk and openais   Steven Dake
• [Asterisk-Dev] SIP failover using Asterisk and openais   Steven Dake
• [Asterisk-Dev] SIP failover using Asterisk and openais   Steven Dake
• [Asterisk-Dev] Licensing of Application over Asterisk   Amit Dang
• [Asterisk-Dev] startup   Klaus Darilion
• [Asterisk-Dev] Problem redirecting to voicemail through a SIPproxy (Looks like a bug)   Chris St Denis
• [Asterisk-Dev] ast_xxx function..?   Chris St Denis
• [Asterisk-Dev] Asterisk and AMD64 - Crashes   Mark Dickie
• [Asterisk-Dev] Asterisk and AMD64 - Crashes   Mark Dickie
• [Asterisk-Dev] removing depreciated code?   Rod Dorman
• [Asterisk-Dev] Stable change log suggestion?   Rod Dorman
• [Asterisk-Dev] Open G.729 / G.723.1 update, fixed memory leak   Paradise Dove
• [Asterisk-Dev] Open G.729 / G.723.1 update, fixed memory leak   Paradise Dove
• [Asterisk-Dev] hints not working on CVS HEAD   Paradise Dove
• [Asterisk-Dev] hints not working on CVS HEAD   Paradise Dove
• [Asterisk-Dev] Device State   Mark Edwards
• [Asterisk-Dev] hangupcause - tracethrough from PRI   Mark Edwards
• [Asterisk-Dev] what do I need to implement odbc   Mark Edwards
• [Asterisk-Dev] Re: How to measure delay in meetme?   Steve Edwards
• [Asterisk-Dev] Open source time card application for Asterisk   Steve Edwards
• [Asterisk-Dev] Authorisation field for voicemail.conf   Mark Elkins
• [Asterisk-Dev] [Fwd: 802.1p support via SO_Priority& 8021q module in RTP.c]   Alex Y Fadeyev
• [Asterisk-Dev] Asterisk and AMD64 - Crashes   Brian C. Fertig
• [Asterisk-Dev] Asterisk and AMD64 - Crashes   Brian C. Fertig
• [Asterisk-Dev] need help   Brian C. Fertig
• [Asterisk-Dev] Interesting Core Dump   Brian C. Fertig
• [Asterisk-Dev] Question about RTP port usage   Bryan Field-Elliot
• [Asterisk-Dev] Question about RTP port usage   Bryan Field-Elliot
• [Asterisk-Dev] undefined symbol: ast_config_load error with latest cvs install   Kevin P. Fleming
• [Asterisk-Dev] undefined symbol: ast_config_load error with latest cvs install   Kevin P. Fleming
• [Asterisk-Dev] Account Code?   Kevin P. Fleming
• [Asterisk-Dev] Doxygen and comments   Kevin P. Fleming
• [Asterisk-Dev] Realtime agents.   Kevin P. Fleming
• [Asterisk-Dev] spelling error in iax.conf.sample   Kevin P. Fleming
• [Asterisk-Dev] Call and video bandwidths   Kevin P. Fleming
• [Asterisk-Dev] Input and testing.   Kevin P. Fleming
• [Asterisk-Dev] Call and video bandwidths   Kevin P. Fleming
• [Asterisk-Dev] Input and testing.   Kevin P. Fleming
• [Asterisk-Dev] Authorisation field for voicemail.conf   Kevin P. Fleming
• [Asterisk-Dev] undefined symbol: ast_config_load error with latest cvs install   Kevin P. Fleming
• [Asterisk-Dev] Authorisation field for voicemail.conf   Kevin P. Fleming
• [Asterisk-Dev] rfc3389 status?   Kevin P. Fleming
• [Asterisk-Dev] rfc3389 status?   Kevin P. Fleming
• [Asterisk-Dev] rfc3389 status?   Kevin P. Fleming
• [Asterisk-Dev] zaptel "rest value" of 0xff is not appropriate when ALAW is in use   Kevin P. Fleming
• [Asterisk-Dev] rfc3389 status?   Kevin P. Fleming
• [Asterisk-Dev] getting started   Kevin P. Fleming
• [Asterisk-Dev] AgentCallbackLogin, 1.2-beta1 issues   Kevin P. Fleming
• [Asterisk-Dev] Re: AgentCallbackLogin, 1.2-beta1 issues   Kevin P. Fleming
• [Asterisk-Dev] Is the ChanIsAvail command thread safe?   Kevin P. Fleming
• [Asterisk-Dev] Is the ChanIsAvail command thread safe?   Kevin P. Fleming
• [Asterisk-Dev] Is the ChanIsAvail command thread safe?   Kevin P. Fleming
• [Asterisk-Dev] Is the ChanIsAvail command thread safe?   Kevin P. Fleming
• [Asterisk-Dev] asterisk 'stable'?   Kevin P. Fleming
• [Asterisk-Dev] asterisk 'stable'?   Kevin P. Fleming
• [Asterisk-Dev] asterisk 'stable'?   Kevin P. Fleming
• [Asterisk-Dev] R1.502 of chan_zap.c kills callerid on a x101p   Kevin P. Fleming
• [Asterisk-Dev] SIP INVITE vs TO URI   Kevin P. Fleming
• [Asterisk-Dev] SIP INVITE vs TO URI   Kevin P. Fleming
• [Asterisk-Dev] SIP INVITE vs TO URI   Kevin P. Fleming
• [Asterisk-Dev] SIP INVITE vs TO URI   Kevin P. Fleming
• [Asterisk-Dev] asterisk 'stable'?   Kevin P. Fleming
• [Asterisk-Dev] config mgmt plan?   Kevin P. Fleming
• [Asterisk-Dev] Can't Add SDP Madness   Kevin P. Fleming
• [Asterisk-Dev] Can't Add SDP Madness   Kevin P. Fleming
• Re [Asterisk-Dev] SIP Jitter Buffer Testing   Kevin P. Fleming
• [Asterisk-Dev] Questions   Kevin P. Fleming
• [Asterisk-Dev] Questions   Kevin P. Fleming
• [Asterisk-Dev] BOUNTY: 100$ Name before Voicemail Playback   Kevin P. Fleming
• [Asterisk-Dev] Asterisk and GlobalCall   Kevin P. Fleming
• [Asterisk-Dev] Asterisk and GlobalCall   Kevin P. Fleming
• [Asterisk-Dev] Asterisk Dialogic channel   Kevin P. Fleming
• [Asterisk-Dev] BOUNTY: 100$ Name before Voicemail Playback   Kevin P. Fleming
• [Asterisk-Dev] BOUNTY: 100$ Name before Voicemail Playback   Kevin P. Fleming
• [Asterisk-Dev] Asterisk Dialogic channel   Kevin P. Fleming
• [Asterisk-Dev] Asterisk Dialogic channel   Kevin P. Fleming
• [Asterisk-Dev] BOUNTY: 100$ Name before Voicemail Playback   Kevin P. Fleming
• [Asterisk-Dev] removing depreciated code?   Kevin P. Fleming
• [Asterisk-Dev] removing depreciated code?   Kevin P. Fleming
• [Asterisk-Dev] removing depreciated code?   Kevin P. Fleming
• [Asterisk-Dev] Echo cancellation on GPU   Kevin P. Fleming
• [Asterisk-Dev] BOUNTY: 100$ Name before Voicemail Playback   Kevin P. Fleming
• [Asterisk-Dev] Redirection reason   Kevin P. Fleming
• [Asterisk-Dev] Command "sip show objets"   Kevin P. Fleming
• [Asterisk-Dev] Redirection reason   Kevin P. Fleming
• [Asterisk-Dev] SIP failover using Asterisk and openais   Kevin P. Fleming
• [Asterisk-Dev] "register" statement: why can't it reference an account?   Kevin P. Fleming
• [Asterisk-Dev] "register" statement: why can't it reference an account?   Kevin P. Fleming
• [Asterisk-Dev] ast_xxx function..?   Kevin P. Fleming
• [Asterisk-Dev] ast_xxx function..?   Kevin P. Fleming
• [Asterisk-Dev] The CallerIDPres not working?   Kevin P. Fleming
• [Asterisk-Dev] ast_xxx function..?   Kevin P. Fleming
• [Asterisk-Dev] ast_xxx function..?   Kevin P. Fleming
• [Asterisk-Dev] Answering Machine detection   Matt Florell
• [Asterisk-Dev] Asterisk Dialogic channel   Matt Florell
• [Asterisk-Dev] Asterisk Dialogic channel   Matt Florell
• [Asterisk-Dev] Channels capactiy of asterisk   Dario Flores
• [Asterisk-Dev] ZyXel 2000W   El Flynn
• [Asterisk-Dev] * Crashed   Chee Foong
• [Asterisk-Dev] Bug# 4824: CLI prompt always shown underneath cli command output   Corey Frang
• [Asterisk-Dev] Interesting AEL / Variable issue   Corey Frang
• [Asterisk-Dev] Interesting AEL / Variable issue   Corey Frang
• [Asterisk-Dev] Ring requested on channel already in use   Matt Fredrickson
• [Asterisk-Dev] Ring requested on channel already in use   Matt Fredrickson
• [Asterisk-Dev] Ring requested on channel already in use   Matt Fredrickson
• [Asterisk-Dev] asterisk 'stable'?   Matt Fredrickson
• [Asterisk-Dev] asterisk 'stable'?   Matt Fredrickson
• [Asterisk-Dev] fxotune.c broken by design?   Matt Fredrickson
• [Asterisk-Dev] maximum concurrent ZAP channels .... max conf ports ...   Matthew Fredrickson
• [Asterisk-Dev] How complete is Q.SIG support?   Matthew Fredrickson
• [Asterisk-Dev] ISDN SETUP on incoming SMS?   Matthew Fredrickson
• [Asterisk-Dev] Hardware HDLC in zaptel   Matthew Fredrickson
• [Asterisk-Dev] - Call progress event?   Matthew Fredrickson
• [Asterisk-Dev] Compiling Asterisk for dual architectue   Jeff Gehlbach
• [Asterisk-Dev] Properly Destroying Variables   Michael Giagnocavo
• [Asterisk-Dev] Questions   Michael Giagnocavo
• [Asterisk-Dev] BOUNTY: 100$ Name before Voicemail Playback   Matt Gibson
• [Asterisk-Dev] BOUNTY: 100$ Name before Voicemail Playback   Matt Gibson
• [Asterisk-Dev] BOUNTY: 100$ Name before Voicemail Playback   Matthew Gibson
• [Asterisk-Dev] BOUNTY: 100$ Name before Voicemail Playback   Matthew Gibson
• [Asterisk-Dev] BOUNTY: 100$ Name before Voicemail Playback   Matthew Gibson
• [Asterisk-Dev] BOUNTY: 100$ Name before Voicemail Playback   Matthew Gibson
• [Asterisk-Dev] Re: Answering Machine detection   Gilmore, Gerry
• [Asterisk-Dev] Re: Answering Machine detection   Gilmore, Gerry
• [Asterisk-Dev] Re: Asterisk Dialogic channel   Gilmore, Gerry
• [Asterisk-Dev] Open source time card application for Asterisk   Gilmore, Gerry
• [Asterisk-Dev] Open source time card application for Asterisk   Gilmore, Gerry
• [Asterisk-Dev] smsq   Bob Goddard
• [Asterisk-Dev] smsq   Bob Goddard
• [Asterisk-Dev] I need help to make a simple iax switch/proxy   Goldenear
• [Asterisk-Dev] I need help to make a simple iax switch/proxy   Goldenear
• [Asterisk-Dev] asterisk 'stable'?   Robert Goodyear
• [Asterisk-Dev] Open G.729 / G.723.1 update, fixed memory leak   Adam Goryachev
• [Asterisk-Dev] Problem with Parking announcement and RedirectAction   Josip Gracin
• [Asterisk-Dev] How not to record music on hold?   Josip Gracin
• [Asterisk-Dev] Problem with 'cvs diff', getting empty diffs   Josip Gracin
• [Asterisk-Dev] Problem with 'cvs diff', getting empty diffs   Josip Gracin
• [Asterisk-Dev] Bug 2905 - patch for macro arguments in M() modifier   Ed Greenberg
• [Asterisk-Dev] Open G.729 / G.723.1 update, fixed memory leak   Edwin Groothuis
• [Asterisk-Dev] getting started   Edwin Groothuis
• [Asterisk-Dev] Re: Asterisk-Dev Digest, Vol 14, Issue 31   Adam Gundy
• [Asterisk-Dev] Suspected memory leak in 1.0.9   Craig Guy
• [Asterisk-Dev] Suspected memory leak in 1.0.9   Craig Guy
• [Asterisk-Dev] G.722 patch uploaded   Chris Halls
• [Asterisk-Dev] FW: [Asterisk-Users] RxFax/TxFax - Compile Problem   Steve Hanselman
• [Asterisk-Dev] Is CVS-HEAD the next 1.2 Beta?   Kevin Hanson
• [Asterisk-Dev] Is CVS-HEAD the next 1.2 Beta?   Kevin Hanson
• [Asterisk-Dev] ast_xxx function..?   Matt Hess
• [Asterisk-Dev] ast_xxx function..?   Matt Hess
• [Asterisk-Dev] ast_xxx function..?   Matt Hess
• [Asterisk-Dev] ast_xxx function..?   Matt Hess
• [Asterisk-Dev] ast_xxx function..?   Matt Hess
• [Asterisk-Dev] ast_xxx function..?   Matt Hess
• [Asterisk-Dev] problem receiveing fax whit capi   Frank van den Heuvel
• [Asterisk-Dev] PLEASE HELP!! CALLERID FAILS!!   John Hill
• [Asterisk-Dev] PLEASE HELP!! CALLERID FAILS!!   John Hill
• [Asterisk-Dev] Re: PLEASE HELP!! CALLERID FAILS!!   John Hill
• [Asterisk-Dev] R1.502 of chan_zap.c kills callerid on a x101p   John Hill
• [Asterisk-Dev] IAX busy indication but not invalidating the call   Lee Howard
• [Asterisk-Dev] IAX timing of VOICE frames   Lee Howard
• [Asterisk-Dev] IAX timing of VOICE frames   Lee Howard
• [Asterisk-Dev] RSA auth broken in IAX2?   Jerris, Michael MI
• [Asterisk-Dev] rfc3389 status?   Jerris, Michael MI
• [Asterisk-Dev] Question about RTP port usage   Jerris, Michael MI
• <SPAM>RE: [Asterisk-Dev] Asterisk and AMD64 - Crashes   Jerris, Michael MI
• [Asterisk-Dev] Asterisk and AMD64 - Crashes   Jerris, Michael MI
• [Asterisk-Dev] asterisk 'stable'?   Jerris, Michael MI
• [Asterisk-Dev] SIP INVITE vs TO URI   Jerris, Michael MI
• [Asterisk-Dev] Call dropped 100% of time when incoming IAX routedto outgoing CAPI   Jerris, Michael MI
• [Asterisk-Dev] removing depreciated code?   Jerris, Michael MI
• [Asterisk-Dev] ISDN SETUP on incoming SMS?   Jerris, Michael MI
• [Asterisk-Dev] Interesting AEL / Variable issue   Jerris, Michael MI
• [Asterisk-Dev] ast_channel_tech   Jerris, Michael MI
• [Asterisk-Dev] ast_xxx function..?   Jerris, Michael MI
• [Asterisk-Dev] Doxygen and comments   Olle E. Johansson
• [Asterisk-Dev] [patch] IM Manager Event   Olle E. Johansson
• [Asterisk-Dev] advance debugging?   Olle E. Johansson
• [Asterisk-Dev] Ring requested on channel already in use   Olle E. Johansson
• [Asterisk-Dev] Authorisation field for voicemail.conf   Olle E. Johansson
• [Asterisk-Dev] RSA auth broken in IAX2?   Olle E. Johansson
• [Asterisk-Dev] rfc3389 status?   Olle E. Johansson
• [Asterisk-Dev] rfc3389 status?   Olle E. Johansson
• [Asterisk-Dev] Subscribe/Notify support   Olle E. Johansson
• [Asterisk-Dev] SIPit17 in Stockholm :: Asterisk under pressure!!   Olle E. Johansson
• [Asterisk-Dev] SIPit17 in Stockholm :: Asterisk under pressure!!   Olle E. Johansson
• [Asterisk-Dev] "register" statement: why can't it reference an account?   Olle E. Johansson
• [Asterisk-Dev] SIP failover using Asterisk and openais   Olle E. Johansson
• [Asterisk-Dev] Problem redirecting to voicemail through a SIPproxy (Looks like a bug)   Olle E. Johansson
• [Asterisk-dev] sip calleridnum   Olle E. Johansson
• [Asterisk-Dev] Current status on _outgoing_ Swedish/Dutch DTMFCLIP for TDM400 FXS interfaces?   Werner Johansson
• [Asterisk-Dev] Asterisk and AMD64 - Crashes   Joseph
• [Asterisk-Dev] Asterisk and AMD64 - Crashes   Joseph
• [Asterisk-Dev] Asterisk and AMD64 - Crashes   Joseph
• [Asterisk-Dev] Asterisk and AMD64 - Crashes   Joseph
• [Asterisk-Dev] Asterisk and AMD64 - Crashes   Joseph
• <SPAM>RE: [Asterisk-Dev] Asterisk and AMD64 - Crashes   Joseph
• <SPAM>RE: [Asterisk-Dev] Asterisk and AMD64 - Crashes   Joseph
• [Asterisk-Dev] Asterisk and AMD64 - Crashes   Joseph
• [Asterisk-Dev] Asterisk and AMD64 - Crashes   Joseph
• [Asterisk-Dev] Asterisk and AMD64 - Crashes   Joseph
• [Asterisk-Dev] Asterisk and AMD64 - Crashes   Joseph
• <SPAM>RE: [Asterisk-Dev] Asterisk and AMD64 - Crashes   Joseph
• [Asterisk-Dev] Asterisk and AMD64 - Crashes   Joseph
• [Asterisk-Dev] Asterisk and AMD64 - Crashes   Joseph
• [Asterisk-Dev] Asterisk run as non root   Joseph
• Advantage / Disadvantage to install asterisk from CVS was: [Asterisk-Dev] Asterisk run as non root   Joseph
• Advantage / Disadvantage to install asterisk from CVS was: [Asterisk-Dev] Asterisk run as non root   Joseph
• [Asterisk-Dev] Open source time card application for Asterisk   Joseph
• [Asterisk-Dev] Open source time card application for Asterisk   Joseph
• [Asterisk-Dev] Answering Machine detection   Nitin Joshi
• [Asterisk-Dev] Re: Answering Machine detection   Nitin Joshi
• [Asterisk-Dev] H.263 format Video   Areski K
• [Asterisk-Dev] asterisk 'stable'?   Steve Kann
• [Asterisk-Dev] rfc3389 status?   Roy Sigurd Karlsbakk
• [Asterisk-Dev] Question about RTP port usage   Roy Sigurd Karlsbakk
• [Asterisk-Dev] asterisk 'stable'?   Roy Sigurd Karlsbakk
• [Asterisk-Dev] Not for 1.2, was SIP RTP JitterBuffer in Asterisk - now aimed for 1.3 dev   Roy Sigurd Karlsbakk
• [Asterisk-Dev] asterisk 'stable'?   Roy Sigurd Karlsbakk
• [Asterisk-Dev] asterisk 'stable'?   Roy Sigurd Karlsbakk
• [Asterisk-Dev] #3986 bounty   Roy Sigurd Karlsbakk
• [Asterisk-Dev] Suspected memory leak in 1.0.9   Roy Sigurd Karlsbakk
• [Asterisk-Dev] #3986 bounty   Roy Sigurd Karlsbakk
• [Asterisk-Dev] #3986 bounty   Roy Sigurd Karlsbakk
• [Asterisk-Dev] removing depreciated code?   Roy Sigurd Karlsbakk
• [Asterisk-Dev] removing depreciated code?   Roy Sigurd Karlsbakk
• [Asterisk-Dev] removing depreciated code?   Roy Sigurd Karlsbakk
• [Asterisk-Dev] removing depreciated code?   Roy Sigurd Karlsbakk
• [Asterisk-Dev] removing depreciated code?   Roy Sigurd Karlsbakk
• [Asterisk-Dev] removing depreciated code?   Roy Sigurd Karlsbakk
• [Asterisk-Dev] removing depreciated code?   Roy Sigurd Karlsbakk
• [Asterisk-Dev] removing depreciated code?   Roy Sigurd Karlsbakk
• [Asterisk-Dev] removing depreciated code?   Roy Sigurd Karlsbakk
• [Asterisk-Dev] MATH() vs expressions?   Roy Sigurd Karlsbakk
• [Asterisk-Dev] ISDN SETUP on incoming SMS?   Roy Sigurd Karlsbakk
• [Asterisk-Dev] ISDN SETUP on incoming SMS?   Roy Sigurd Karlsbakk
• [Asterisk-Dev] ISDN SETUP on incoming SMS?   Roy Sigurd Karlsbakk
• [Asterisk-Dev] ISDN SETUP on incoming SMS?   Roy Sigurd Karlsbakk
• [Asterisk-Dev] Fraud Possibility in CDR   Roy Sigurd Karlsbakk
• [Asterisk-Dev] ast_xxx function..?   Roy Sigurd Karlsbakk
• [Asterisk-Dev] Digium Dual T1 Spans 2xxp   Ricky Keele
• [Asterisk-Dev] Dual Span T1   Ricky Keele
• [Asterisk-Dev] Conferences under asterisk   Alex Kobalto
• [Asterisk-Dev] Open G.729 / G.723.1 update, fixed memory leak   Andrew Kohlsmith
• [Asterisk-Dev] Open G.729 / G.723.1 update, fixed memory leak   Andrew Kohlsmith
• [Asterisk-Dev] Open G.729 / G.723.1 update, fixed memory leak   Andrew Kohlsmith
• [Asterisk-Dev] I need help to make a simple iax switch/proxy   Andrew Kohlsmith
• [Asterisk-Dev] I need help to make a simple iax switch/proxy   Andrew Kohlsmith
• [Asterisk-Dev] Digium Dual T1 Spans 2xxp   Andrew Kohlsmith
• [Asterisk-Dev] asterisk 'stable'?   Andrew Kohlsmith
• Re [Asterisk-Dev] SIP Jitter Buffer Testing   Andrew Kohlsmith
• [Asterisk-Dev] SIP Jitter Buffer Testing   Andrew Kohlsmith
• [Asterisk-Dev] need help   Andrew Kohlsmith
• [Asterisk-Dev] Questions   Andrew Kohlsmith
• [Asterisk-Dev] need help   Andrew Kohlsmith
• [Asterisk-Dev] need help   Andrew Kohlsmith
• [Asterisk-Dev] Questions   Andrew Kohlsmith
• [Asterisk-Dev] Re: #3986 bounty   Andrew Kohlsmith
• [Asterisk-Dev] need help   Andrew Kohlsmith
• [Asterisk-Dev] goiax expanded with free us domestic calling   Andrew Kohlsmith
• [Asterisk-Dev] Echo cancellation on GPU   Andrew Kohlsmith
• [Asterisk-Dev] Hardware HDLC in zaptel   Andrew Kohlsmith
• [Asterisk-Dev] Suspected memory leak in 1.0.9   Thralling Penguin LLC
• [Asterisk-Dev] need help   Lakmal
• [Asterisk-Dev] Bug with toll free between 2 asterisk servers (special toll free signaling?)   John Lange
• [Asterisk-Dev] Bug with toll free between 2 asterisk servers (special toll free signaling?)   John Lange
• [Asterisk-Dev] Bug with toll free between 2 asterisk servers(special toll free signaling?)   John Lange
• [Asterisk-Dev] Asterisk 1.0.9 long term stability   Sig Lange
• reason values (was: [Asterisk-Dev] Re: Originate Call and Unique ID)   Joerg Lauer
• reason values (was: [Asterisk-Dev] Re: Originate Call and Unique ID)   Tilghman Lesher
• [Asterisk-Dev] Impossible for an extension to send Flash to Bell CO?   Tilghman Lesher
• [Asterisk-Dev] Impossible for an extension to send Flash to Bell CO?   Tilghman Lesher
• [Asterisk-Dev] NoOp not returning anything?   Tilghman Lesher
• [Asterisk-Dev] NoOp not returning anything?   Tilghman Lesher
• [Asterisk-Dev] Suspected memory leak in 1.0.9   Tilghman Lesher
• [Asterisk-Dev] Suspected memory leak in 1.0.9   Tilghman Lesher
• [Asterisk-Dev] Re: #3986 bounty   Tilghman Lesher
• [Asterisk-Dev] Asterisk Dialogic channel   Tilghman Lesher
• [Asterisk-Dev] removing depreciated code?   Tilghman Lesher
• [Asterisk-Dev] removing depreciated code?   Tilghman Lesher
• [Asterisk-Dev] Open source time card application for Asterisk   Tilghman Lesher
• [Asterisk-Dev] Open source time card application for Asterisk   Tilghman Lesher
• [Asterisk-Dev] Open source time card application for Asterisk   Tilghman Lesher
• [Asterisk-Dev] Want to add a VM delete event - where in app_voicemail.c?   Tilghman Lesher
• [Asterisk-Dev] Want to add a VM delete event - where inapp_voicemail.c?   Tilghman Lesher
• [Asterisk-Dev] play tone for auto monitoring   Keiron Liddle
• [Asterisk-Dev] asterisk doesn't send '200 result..." after a VERBOSE command   Arnaud Ligot
• [Asterisk-Dev] Subscribe/Notify support   David (Dalei) Liu
• [Asterisk-Dev] SIP INVITE vs TO URI   William Lloyd
• [Asterisk-Dev] SIP INVITE vs TO URI   William Lloyd
• [Asterisk-Dev] SIP INVITE vs TO URI   William Lloyd
• [Asterisk-Dev] INVITE URI from contact?   William Lloyd
• [Asterisk-Dev] Database backed Asterisk   William Lloyd
• [Asterisk-Dev] Mac Mini on Debian - Asterisk   Tom Lo
• [Asterisk-Dev] CVS Servers   Alexander Lopez
• [Asterisk-Dev] CVS Servers   Alexander Lopez
• [Asterisk-Dev] FW: [Asterisk-Users] RxFax/TxFax - Compile Problem   Alexander Lopez
• [Asterisk-Dev] Caller Name: Asterisk reading too fast   Alexander Lopez
• [Asterisk-Dev] Request Netiquite   Alexander Lopez
• [Asterisk-Dev] Request Netiquite   Alexander Lopez
• [Asterisk-Dev] odbc problems (newbie odbc developer)   Julian Lyndon-Smith
• [Asterisk-Dev] what do I need to implement odbc   Julian Lyndon-Smith
• [Asterisk-Dev] what do I need to implement odbc   Julian Lyndon-Smith
• [Asterisk-Dev] ACD: Limiting time spent with Agent   Begumisa Gerald M
• [Asterisk-Dev] IAX PBX responds to IAX registration with expires time=0   Maciek
• [Asterisk-Dev] Database backed Asterisk   Leif Madsen
• [Asterisk-Dev] Database backed Asterisk   Leif Madsen
• [Asterisk-Dev] Want to add a VM delete event - wherein app_voicemail.c?   Leif Madsen
• [Asterisk-Dev] asterisk-oh323: New versions 0.6.7 and 0.7.3   Michael Manousos
• [Asterisk-Dev] Call and video bandwidths   John Martin
• [Asterisk-Dev] Call and video bandwidths   John Martin
• [Asterisk-Dev] Account Code?   Matt
• [Asterisk-Dev] ***TESTING NEEDED*** JITTER BUFFER FOR SIP   Matt
• [Asterisk-Dev] ***TESTING NEEDED*** JITTER BUFFER FOR SIP   Matt
• [Asterisk-Dev] asterisk 'stable'?   Matt
• [Asterisk-Dev] * debug problem   Matt
• [Asterisk-Dev] ZyXel 2000W   John Matthews
• [Asterisk-Dev] Fwd: CallerID and CDR   Sherwood McGowan
• [Asterisk-Dev] PrivacyManager on a per user basis?   Sherwood McGowan
• [Asterisk-Dev] PrivacyManager on a per user basis?   Sherwood McGowan
• [Asterisk-Dev] getting started   Sherwood McGowan
• [Asterisk-Dev] asterisk 'stable'?   Sherwood McGowan
• [Asterisk-Dev] asterisk 'stable'?   Sherwood McGowan
• [Asterisk-Dev] asterisk 'stable'?   Sherwood McGowan
• [Asterisk-Dev] asterisk 'stable'?   Sherwood McGowan
• [Asterisk-Dev] SIP INVITE vs TO URI   Sherwood McGowan
• [Asterisk-Dev] SIP INVITE vs TO URI   Sherwood McGowan
• [Asterisk-Dev] SIP INVITE vs TO URI   Sherwood McGowan
• [Asterisk-Dev] NoOp not returning anything?   Sherwood McGowan
• [Asterisk-Dev] NoOp not returning anything?   Sherwood McGowan
• [Asterisk-Dev] NoOp not returning anything?   Sherwood McGowan
• [Asterisk-Dev] NoOp not returning anything?   Sherwood McGowan
• [Asterisk-Dev] NoOp not returning anything?   Sherwood McGowan
• [Asterisk-Dev] SIPit17 in Stockholm :: Asterisk under pressure!!   Sherwood McGowan
• [Asterisk-Dev] need help   Sherwood McGowan
• [Asterisk-Dev] A couple of bugs?   Sherwood McGowan
• [Asterisk-Dev] Re: #3986 bounty   Sherwood McGowan
• [Asterisk-Dev] Fraud Possibility in CDR   Sherwood McGowan
• [Asterisk-Dev] Fraud Possibility in CDR   Sherwood McGowan
• [Asterisk-Dev] The CallerIDPres not working?   Sherwood McGowan
• [Asterisk-Dev] The CallerIDPres not working?   Sherwood McGowan
• [Asterisk-Dev] The CallerIDPres not working?   Sherwood McGowan
• [Asterisk-Dev] The CallerIDPres not working?   Sherwood McGowan
• [Asterisk-Dev] Open Source time card application for Asterisk   Rob McKrill
• [Asterisk-Dev] ZyXel 2000W   Steve McMahon
• [Asterisk-Dev] T.38 source code   Jeremy McNamara
• [Asterisk-Dev] Open G.729 / G.723.1 update, fixed memory leak   Jeremy McNamara
• [Asterisk-Dev] Ring requested on channel already in use   Jeremy McNamara
• [Asterisk-Dev] Ring requested on channel already in use   Jeremy McNamara
• [Asterisk-Dev] asterisk 'stable'?   Jeremy McNamara
• [Asterisk-Dev] need help   Jeremy McNamara
• [Asterisk-Dev] #3986 bounty   Jeremy McNamara
• [Asterisk-Dev] #3986 bounty   Jeremy McNamara
• [Asterisk-Dev] #3986 bounty   Jeremy McNamara
• [Asterisk-Dev] removing depreciated code?   Jeremy McNamara
• [Asterisk-Dev] removing depreciated code?   Jeremy McNamara
• [Asterisk-Dev] need help   Enzo Michelangeli
• Malfunctioning SendDTMF() and Playtones() (was: Re: [Asterisk-Dev] need help)   Enzo Michelangeli
• Malfunctioning SendDTMF() and Playtones() (was: Re:[Asterisk-Dev] need help)   Enzo Michelangeli
• [Asterisk-Dev] Bug with toll free between 2 asterisk servers(special toll free signaling?)   Enzo Michelangeli
• [Asterisk-Dev] "register" statement: why can't it reference an account?   Enzo Michelangeli
• [Asterisk-Dev] Re: undefined symbol: ast_config_load error with latest cvs install   Tony Mountifield
• [Asterisk-Dev] Re: Is the ChanIsAvail command thread safe?   Tony Mountifield
• [Asterisk-Dev] Re: PLEASE HELP!! CALLERID FAILS!!   Tony Mountifield
• [Asterisk-Dev] Re: NoOp not returning anything?   Tony Mountifield
• [Asterisk-Dev] Re: need help   Tony Mountifield
• [Asterisk-Dev] Re: need help   Tony Mountifield
• [Asterisk-Dev] Re: #3986 bounty   Tony Mountifield
• [Asterisk-Dev] Re: asterisk 'stable'?   Tony Mountifield
• [Asterisk-Dev] Re: asterisk 'stable'?   Tony Mountifield
• [Asterisk-Dev] Re: asterisk 'stable'?   Tony Mountifield
• [Asterisk-Dev] Re: #3986 bounty   Tony Mountifield
• [Asterisk-Dev] Re: Asterisk Dialogic channel   Tony Mountifield
• [Asterisk-Dev] How complete is Q.SIG support?   Tony Mountifield
• [Asterisk-Dev] Re: How complete is Q.SIG support?   Tony Mountifield
• [Asterisk-Dev] TDMoE and tap device   Andreas Mueller
• [Asterisk-Dev] Call dropped 100% of time when incoming IAX routed to outgoing CAPI   Christopher Mylonas
• [Asterisk-Dev] Open source time card application for Asterisk & then hotel software   Christopher Mylonas
• [Asterisk-Dev] removing depreciated code?   Dinesh Nair
• [Asterisk-Dev] removing depreciated code?   Dinesh Nair
• [Asterisk-Dev] How to write an asterisk application   Thang Nguyen
• [Asterisk-Dev] Re: How to measure delay in meetme?   Clive Nicolson
• [Asterisk-Dev] zaptel API description?   Kristian Nielsen
• [Asterisk-Dev] Asterisk and GlobalCall   Kristian Nielsen
• [Asterisk-Dev] Asterisk and GlobalCall   Kristian Nielsen
• [Asterisk-Dev] Asterisk and GlobalCall   Kristian Nielsen
• [Asterisk-Dev] Problem with 'cvs diff', getting empty diffs   Kristian Nielsen
• [Asterisk-Dev] Redirection reason   Kristian Nielsen
• [Asterisk-Dev] Redirection reason   Kristian Nielsen
• [Asterisk-Dev] Redirection reason   Kristian Nielsen
• [Asterisk-Dev] SIP failover using Asterisk and openais   Kristian Nielsen
• [Asterisk-Dev] ast_channel_tech   Kristian Nielsen
• [Asterisk-dev] sip calleridnum   Michal Olejnik
• [Asterisk-Dev] T.38 source code   Florian Overkamp
• [Asterisk-Dev] T.38 source code   Florian Overkamp
• [Asterisk-Dev] T.38 source code   Florian Overkamp
• [Asterisk-Dev] SIP Jitter Buffer Testing   George Pajari
• Re [Asterisk-Dev] SIP Jitter Buffer Testing   George Pajari
• [Asterisk-Dev] Re: #3986 bounty   Chris Parker
• [Asterisk-Dev] ***TESTING NEEDED*** JITTER BUFFER FOR SIP   Patrick
• Advantage / Disadvantage to install asterisk from CVS was: [Asterisk-Dev] Asterisk run as non root   Patrick
• [Asterisk-Dev] Asterisk and AMD64 - Crashes   Paul
• [Asterisk-Dev] Asterisk and AMD64 - Crashes   Paul
• [Asterisk-Dev] Asterisk and AMD64 - Crashes   Paul
• [Asterisk-Dev] asterisk 'stable'?   Paul
• [Asterisk-Dev] Asterisk run as non root   Paul
• Advantage / Disadvantage to install asterisk from CVS was: [Asterisk-Dev] Asterisk run as non root   Paul
• [Asterisk-Dev] asterisk 'stable'?   Paul
• [Asterisk-Dev] need help   Paul
• [Asterisk-Dev] need help   Paul
• [Asterisk-Dev] what do I need to implement odbc   Paul
• [Asterisk-Dev] need help   Paul
• [Asterisk-Dev] Questions   Paul
• [Asterisk-Dev] need help   Paul
• [Asterisk-Dev] need help   Paul
• [Asterisk-Dev] Questions   Paul
• [Asterisk-Dev] need help   Paul
• [Asterisk-Dev] T.38 source code   Rosario Pingaro
• [Asterisk-Dev] Open G.729 / G.723.1 update, fixed memory leak   Daniel Pocock
• [Asterisk-Dev] Open G.729 / G.723.1 update, fixed memory leak   Daniel Pocock
• [Asterisk-Dev] Open G.729 / G.723.1 update, fixed memory leak   Daniel Pocock
• [Asterisk-Dev] Questions   Shawn Porter
• [Asterisk-Dev] maximum concurrent ZAP channels .... max conf ports ...   Vamsi Pottangi
• [Asterisk-Dev] zaptel "rest value" of 0xff is not appropriatewhen ALAW is in use   Michael Procter
• [Asterisk-Dev] zaptel "rest value" of 0xff is not appropriatewhenALAW is in use   Michael Procter
• [Asterisk-Dev] ISDN SETUP on incoming SMS?   Michael Procter
• [Asterisk-Dev] spandsp crashing asterisk   Atif Rasheed
• [Asterisk-Dev] asterisk 'stable'?   Matt Riddell
• [Asterisk-Dev] asterisk 'stable'?   Matt Riddell
• [Asterisk-Dev] BOUNTY: 100$ Name before Voicemail Playback   Matt Riddell
• [Asterisk-Dev] BOUNTY: 100$ Name before Voicemail Playback   Matt Riddell
• [Asterisk-Dev] BOUNTY: 100$ Name before Voicemail Playback   Matt Riddell
• [Asterisk-Dev] BOUNTY: 100$ Name before Voicemail Playback   Matt Riddell
• [Asterisk-Dev] BOUNTY: 100$ Name before Voicemail Playback   Matt Riddell
• [Asterisk-Dev] The CallerIDPres not working?   Josh Roberson
• [Asterisk-Dev] ast_channel_tech   Steve Rodgers
• [Asterisk-Dev] Re: [Asterisk-cvs] asterisk logger.c,1.81,1.82   Frank Sautter
• [Asterisk-Dev] problem receiveing fax whit capi   Armin Schindler
• [Asterisk-Dev] problem receiveing fax whit capi   Armin Schindler
• [Asterisk-Dev] Echo cancellation on GPU   Richard Scobie
• [Asterisk-Dev] Command "sip show objets"   Ezequiel A. Sculli
• [Asterisk-Dev] zaptel "rest value" of 0xff is not appropriate when ALAW is in use   Hans Petter Selasky
• [Asterisk-Dev] ISDN SETUP on incoming SMS?   Hans Petter Selasky
• [Asterisk-Dev] High Availability again   Sergio Serrano
• [Asterisk-Dev] SpanDSP memory leak?   Darren Sessions
• [Asterisk-Dev] Re: #3986 bounty   Andreas Sikkema
• [Asterisk-Dev] Asterisk AGI php, and Asterisk PHP are written as functions instead of classes and objects.   Moises Silva
• [Asterisk-Dev] goiax expanded with free us domestic calling   Matthew Simpson
• [Asterisk-Dev] Ring requested on channel already in use   James Sizemore
• [Asterisk-Dev] zaptel API description?   Bob Smith
• [Asterisk-Dev] getting started   Jared Smith
• [Asterisk-Dev] asterisk 'stable'?   Nathan C. Smith
• [Asterisk-Dev] asterisk 'stable'?   Nathan C. Smith
• [Asterisk-Dev] I need help to make a simple iax switch/proxy   Derek Smithies
• [Asterisk-Dev] I need help to make a simple iax switch/proxy   Derek Smithies
• [Asterisk-Dev] I need help to make a simple iax switch/proxy   Derek Smithies
• [Asterisk-Dev] I need help to make a simple iax switch/proxy   Derek Smithies
• [Asterisk-Dev] IAX timing of VOICE frames   Derek Smithies
• [Asterisk-Dev] IAX timing of VOICE frames   Derek Smithies
• [Asterisk-Dev] #3986 bounty   Derek Smithies
• [Asterisk-Dev] T.38 source code   Craig Southeren
• [Asterisk-Dev] T.38 source code   Craig Southeren
• [Asterisk-Dev] Database backed Asterisk   Serge Sozonoff
• [Asterisk-Dev] Database backed Asterisk   Serge Sozonoff
• [Asterisk-Dev] Database backed Asterisk   Serge Sozonoff
• [Asterisk-Dev] Database backed Asterisk   Serge Sozonoff
• [Asterisk-Dev] GUI's   Serge Sozonoff
• [Asterisk-Dev] GUI's   Serge Sozonoff
• [Asterisk-Dev] I need help to make a simple iax switch/proxy   Steve
• [Asterisk-Dev] I need help to make a simple iax switch/proxy   Steve
• [Asterisk-Dev] asterisk 'stable'?   Steve
• [Asterisk-Dev] SIP INVITE vs TO URI   Steve
• [Asterisk-Dev] Newbie needs advice   Steve
• [Asterisk-Dev] SIP Jitter Buffer Testing   SteveK
• [Asterisk-Dev] removing depreciated code?   SteveK
• [Asterisk-Dev] Echo cancellation on GPU   SteveK
• [Asterisk-Dev] I need help to make a simple iax switch/proxy   Steven
• [Asterisk-Dev] undefined symbol: ast_config_load error with latest cvs install   Steven
• [Asterisk-Dev] I need help to make a simple iax switch/proxy   Steven
• [Asterisk-Dev] need help to compile asterisk under ubuntu   Steven
• [Asterisk-Dev] Digium Dual T1 Spans 2xxp   Steven
• [Asterisk-Dev] Re: [Asterisk-Users] MusicOnHold not working   Steven
• [Asterisk-Dev] removing depreciated code?   Steven
• [Asterisk-Dev] removing depreciated code?   Steven
• [Asterisk-Dev] Licensing of Application over Asterisk   Steven
• [Asterisk-Dev] Open source time card application for Asterisk   Steven
• [Asterisk-Dev] ast_channel_tech   Steven
• [Asterisk-Dev] Open source time card application for Asterisk   Joe Stewart
• [Asterisk-Dev] Re: Asterisk Time Card   Joe Stewart
• [Asterisk-Dev] Asterisk AGI php, and Asterisk PHP are written as functions instead of classes and objects.   Joe Stewart
• [Asterisk-Dev] startup   Daniel Swarbrick
• [Asterisk-Dev] How to submit code - res_config_pgsql ?   Daniel Swarbrick
• [Asterisk-Dev] How to submit code - res_config_pgsql ?   Daniel Swarbrick
• [Asterisk-Dev] ast realtime and ast_mutex_lock   Daniel Swarbrick
• [Asterisk-Dev] Voicemail password change + Ast Realtime   Daniel Swarbrick
• [Asterisk-Dev] Voicemail password change + Ast Realtime   Daniel Swarbrick
• [Asterisk-Dev] rfc3389 status?   Mike Taht
• [Asterisk-Dev] ast_channel_tech   Mike Taht
• [Asterisk-Dev] res_monitor improvement   Tamas
• [Asterisk-Dev] Caller Name: Asterisk reading too fast   J Thomas
• [Asterisk-Dev] fxotune.c broken by design?   Rob Thomas
• [Asterisk-Dev] hints not working on CVS HEAD   Rob Thomas
• [Asterisk-Dev] Re: smsq   Stefan Tichy
• [Asterisk-Dev] Re: ZyXel 2000W ( --> asterisk-users )   Stefan Tichy
• [Asterisk-Dev] Input and testing.   John Todd
• [Asterisk-Dev] Echo cancellation on GPU   John Todd
• [Asterisk-Dev] CSTA interface comments   John Todd
• [Asterisk-Dev] SIP failover using Asterisk and openais   John Todd
• [Asterisk-Dev] SIP failover using Asterisk and openais   John Todd
• [Asterisk-Dev] Re: #3986 bounty   Steve Totaro
• [Asterisk-Dev] removing depreciated code?   Steve Totaro
• [Asterisk-Dev] T.38 source code   Steve Underwood
• [Asterisk-Dev] G.722 patch uploaded   Steve Underwood
• [Asterisk-Dev] T.38 source code   Steve Underwood
• [Asterisk-Dev] T.38 source code   Steve Underwood
• [Asterisk-Dev] ISDN SETUP on incoming SMS?   Steve Underwood
• [Asterisk-Dev] ISDN SETUP on incoming SMS?   Steve Underwood
• [Asterisk-Dev] Does Asterisk support Meetme realtime now?   VOIP
• [Asterisk-Dev] Want to add a VM delete event - where in app_voicemail.c?   Joel Vandal
• [Asterisk-Dev] need help   Martin Vit
• [Asterisk-Dev] TDMoE and tap device   Martin Vit
• [Asterisk-Dev] Re: #3986 bounty   Chris Wade
• [Asterisk-Dev] [OT] [Very OT] socket connection from perl to visual basic   Chris Wade
• [Asterisk-Dev] [OT] [Very OT] socket connection from perl to visual basic   Chris Wade
• [Asterisk-Dev] Open source time card application for Asterisk & then hotel software   Dave Weis
• [Asterisk-Dev] Mac Mini on Debian - Asterisk   Dave Weis
• [Asterisk-Dev] smsq   Drew Wells
• [Asterisk-Dev] smsq   Drew Wells
• [Asterisk-Dev] Re: smsq   Drew Wells
• [Asterisk-Dev] Request Netiquite   BJ Weschke
• [Asterisk-Dev] Open source time card application for Asterisk   BJ Weschke
• [Asterisk-Dev] TDMoE and tap device   BJ Weschke
• [Asterisk-Dev] TDMoE and tap device   BJ Weschke
• [Asterisk-Dev] GUI's   BJ Weschke
• [Asterisk-Dev] Is CVS-HEAD the next 1.2 Beta?   Brian West
• [Asterisk-Dev] Input and testing.   Brian West
• [Asterisk-Dev] Input and testing.   Brian West
• [Asterisk-Dev] Input and testing.   Brian West
• [Asterisk-Dev] I need help to make a simple iax switch/proxy   Brian West
• [Asterisk-Dev] asterisk 'stable'?   Brian West
• [Asterisk-Dev] Caller Name: Asterisk reading too fast   Brian West
• [Asterisk-Dev] SIP Jitter Buffer Testing   Brian West
• [Asterisk-Dev] Asterisk Dialogic channel   Brian West
• [Asterisk-Dev] Asterisk Dialogic channel   Brian West
• [Asterisk-Dev] Open source time card application for Asterisk   Brian West
• [Asterisk-Dev] TDMoE and tap device   Brian West
• [Asterisk-Dev] TDMoE and tap device   Brian West
• [Asterisk-Dev] SpanDSP memory leak?   Brian K. West
• [Asterisk-Dev] Input and testing.   Brian K. West
• [Asterisk-Dev] undefined symbol: ast_config_load error with latest cvs install   Brian K. West
• [Asterisk-Dev] Ring requested on channel already in use   Brian K. West
• [Asterisk-Dev] PrivacyManager on a per user basis?   Brian K. West
• [Asterisk-Dev] rfc3389 status?   Brian K. West
• [Asterisk-Dev] rfc3389 status?   Brian K. West
• [Asterisk-Dev] rfc3389 status?   Brian K. West
• [Asterisk-Dev] rfc3389 status?   Brian K. West
• [Asterisk-Dev] rfc3389 status?   Brian K. West
• [Asterisk-Dev] Asterisk and AMD64 - Crashes   Brian K. West
• [Asterisk-Dev] Asterisk and AMD64 - Crashes   Brian K. West
• [Asterisk-Dev] Asterisk and AMD64 - Crashes   Brian K. West
• <SPAM>RE: [Asterisk-Dev] Asterisk and AMD64 - Crashes   Brian K. West
• [Asterisk-Dev] Asterisk and AMD64 - Crashes   Brian K. West
• [Asterisk-Dev] Asterisk and AMD64 - Crashes   Brian K. West
• [Asterisk-Dev] Asterisk and AMD64 - Crashes   Brian K. West
• [Asterisk-Dev] Asterisk and AMD64 - Crashes   Brian K. West
• [Asterisk-Dev] asterisk 'stable'?   Brian K. West
• [Asterisk-Dev] asterisk 'stable'?   Brian K. West
• [Asterisk-Dev] SIP INVITE vs TO URI   Brian K. West
• [Asterisk-Dev] SIP INVITE vs TO URI   Brian K. West
• [Asterisk-Dev] SIP INVITE vs TO URI   Brian K. West
• [Asterisk-Dev] NoOp not returning anything?   Brian K. West
• [Asterisk-Dev] NoOp not returning anything?   Brian K. West
• [Asterisk-Dev] Re: [Asterisk-cvs] asterisk logger.c,1.81,1.82   Brian K. West
• [Asterisk-Dev] FW: [Asterisk-Users] RxFax/TxFax - Compile Problem   Brian K. West
• [Asterisk-Dev] SIP Jitter Buffer Testing   Brian K. West
• [Asterisk-Dev] need help   Brian K. West
• [Asterisk-Dev] need help   Brian K. West
• [Asterisk-Dev] need help   Brian K. West
• [Asterisk-Dev] what do I need to implement odbc   Brian K. West
• [Asterisk-Dev] what do I need to implement odbc   Brian K. West
• [Asterisk-Dev] what do I need to implement odbc   Brian K. West
• [Asterisk-Dev] Questions   Brian K. West
• [Asterisk-Dev] Questions   Brian K. West
• [Asterisk-Dev] need help   Brian K. West
• [Asterisk-Dev] Questions   Brian K. West
• [Asterisk-Dev] Questions   Brian K. West
• [Asterisk-Dev] Bug in SIP message Handling   Gene Willingham
• [Asterisk-Dev] delayed sip answer   Mark Willis
• [Asterisk-Dev] delayed sip answer   Mark Willis
• [Asterisk-Dev] T.38 source code   Terry Wilson
• [Asterisk-Dev] asterisk 'stable'?   Terry Wilson
• [Asterisk-Dev] Open G.729 / G.723.1 update, fixed memory leak   David Woodhouse
• [Asterisk-Dev] Mac Mini on Debian - Asterisk   David Woodhouse
• [Asterisk-Dev] pass through g.723/g.729 as a b2b ua   Ma Zhiyong
• [Asterisk-Dev] SIP Jitter Buffer Testing   Zoa
• [Asterisk-Dev] How to submit code - res_config_pgsql ?   Zoa
• [Asterisk-Dev] Ring requested on channel already in use   alan
• [Asterisk-Dev] Ring requested on channel already in use   alan
• [Asterisk-Dev] Ring requested on channel already in use   alan
• [Asterisk-Dev] Re: Asterisk-Dev Digest, Vol 14, Issue 19   alan
• [Asterisk-Dev] Ring requested on channel already in use   alan
• [Asterisk-Dev] AgentCallbackLogin, 1.2-beta1 issues   alan
• [Asterisk-Dev] Re: AgentCallbackLogin, 1.2-beta1 issues   alan
• [Asterisk-Dev] Re: AgentCallbackLogin, 1.2-beta1 issues   alan
• [Asterisk-Dev] Re: Ring requested on channel already in use   alan
• [Asterisk-Dev] cable modems?   alex at pilosoft.com
• [Asterisk-Dev] Fwd: CallerID and CDR   alex at pilosoft.com
• [Asterisk-Dev] asterisk 'stable'?   alex at pilosoft.com
• [Asterisk-Dev] asterisk 'stable'?   alex at pilosoft.com
• [Asterisk-Dev] asterisk 'stable'?   alex at pilosoft.com
• [Asterisk-Dev] SIP Jitter Buffer Testing   alex at pilosoft.com
• [Asterisk-Dev] SIP Jitter Buffer Testing   alex at pilosoft.com
• [Asterisk-Dev] advance debugging?   david.j as
• [Asterisk-Dev] T.38 source code   asterisk at ntplx.net
• [Asterisk-Dev] G.722 patch uploaded   asterisk at ntplx.net
• [Asterisk-Dev] App directed pickup (bug #4865)   asterisk at ntplx.net
• [Asterisk-Dev] need help   julien bossart
• [Asterisk-Dev] Newbie needs advice   jose luis campos
• [Asterisk-Dev] undefined symbol: ast_config_load error with latest cvs install   ddiffa
• [Asterisk-Dev] undefined symbol: ast_config_load error with latest cvs install   ddiffa
• [Asterisk-Dev] undefined symbol: ast_config_load error with latest cvs install   ddiffa
• [Asterisk-Dev] ast_channel_tech   flavio
• [Asterisk-Dev] ast_channel_tech   flavio
• [Asterisk-Dev] (no subject)   flavio
• [Asterisk-Dev] ast_channel_tech   flavio
• [Asterisk-Dev] [patch] IM Manager Event   harry gaillac
• [Asterisk-Dev] RAGI - Ruby Asterisk Gateway Interface   joe heitzeberg
• [Asterisk-Dev] cable modems?   julian howard
• [Asterisk-Dev] Is the ChanIsAvail command thread safe?   hugolivude
• [Asterisk-Dev] Is the ChanIsAvail command thread safe?   hugolivude
• [Asterisk-Dev] Impossible for an extension to send Flash to Bell CO?   hugolivude
• [Asterisk-Dev] Impossible for an extension to send Flash to Bell CO?   hugolivude
• [Asterisk-Dev] need help   hugolivude
• [Asterisk-Dev] Impossible for an extension to send Flash to Bell CO?   hugolivude
• [Asterisk-Dev] Bug in the blindxfer and atxfer commands?   hugolivude
• [Asterisk-Dev] Error while starting Asterisk   jibumathewemail-ast at yahoo.com
• [Asterisk-Dev] Re: Answering Machine detection (steve@daviesfam.org)   manish kumar
• [Asterisk-Dev] NoOp not returning anything?   lconroy
• [Asterisk-Dev] delayed sip answer   lconroy
• [Asterisk-Dev] Echo cancellation on GPU   lconroy
• [Asterisk-Dev] how to fix this bug?E400P port2 and port 4 can not work   oncemore
• [Asterisk-Dev] Asterisk and GlobalCall   tim panton
• [Asterisk-Dev] problem receiveing fax whit capi   roberto.ziano
• [Asterisk-Dev] problem receiveing fax whit capi   roberto.ziano
• [Asterisk-Dev] startup   himanshoo kumar saxena
• [Asterisk-Dev] G.722 patch uploaded   steve at daviesfam.org
• [Asterisk-Dev] Strange code in chan_zap.c   steve at daviesfam.org
• [Asterisk-Dev] Answering Machine detection   steve at daviesfam.org
• [Asterisk-Dev] zaptel "rest value" of 0xff is not appropriate when ALAW is in use   steve at daviesfam.org
• [Asterisk-Dev] zaptel "rest value" of 0xff is not appropriate when ALAW is in use   steve at daviesfam.org
• [Asterisk-Dev] chan_zap bug - pri spans end up stuck DOWN   steve at daviesfam.org
• [Asterisk-Dev] zaptel "rest value" of 0xff is not appropriate when ALAW is in use   steve at daviesfam.org
• [Asterisk-Dev] zaptel "rest value" of 0xff is not appropriatewhen ALAW is in use   steve at daviesfam.org
• [Asterisk-Dev] zaptel "rest value" of 0xff is not appropriatewhen ALAW is in use   steve at daviesfam.org
• [Asterisk-Dev] Dual Span T1   steve at daviesfam.org
• [Asterisk-Dev] zaptel "rest value" of 0xff is not appropriate when ALAW is in use   steve at daviesfam.org
• [Asterisk-Dev] SIP Jitter Buffer Testing   steve at daviesfam.org
• [Asterisk-Dev] SIPit17 in Stockholm :: Asterisk under pressure!!   steve at daviesfam.org
• [Asterisk-Dev] VoicemailDetect   steve at daviesfam.org
• [Asterisk-Dev] High Availability again   steve at daviesfam.org
• [Asterisk-Dev] Re: Ring requested on channel already in use   steve at daviesfam.org
• [Asterisk-Dev] Echo cancellation on GPU   steve at daviesfam.org
• [Asterisk-Dev] Echo cancellation on GPU   steve at daviesfam.org
• [Asterisk-Dev] need help to compile asterisk under ubuntu   julienasterisk t
• [Asterisk-Dev] Mac Mini on Debian - Asterisk   tech

Last message date: Fri Sep 30 21:02:18 MST 2005
Archived on: Tue Sep 5 14:27:38 MST 2006

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