January 2006 Archives by author

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Starting: Sun Jan 1 12:29:38 MST 2006
Ending: Tue Jan 31 21:20:41 MST 2006
Messages: 663

• [Asterisk-Dev] Re: Voicemail to email volume change patch   Matt Riddell (IT)
• [Asterisk-Dev] moving sounds out of asterisk repository   Matt Riddell (IT)
• [Asterisk-Dev] Re: click-to-call cleint   Matt Riddell (IT)
• [Asterisk-Dev] Re: click-to-call cleint   Matt Riddell (IT)
• [Asterisk-Dev] Re: click-to-call cleint   Matt Riddell (IT)
• [Asterisk-Dev] Asterisk Port to Windows   Matt Riddell (IT)
• [asterisk-dev] rtp.c: Unable to allocate socket: Too many open files   Matt Riddell (IT)
• [asterisk-dev] (nessun oggetto)   Matt Riddell (IT)
• [asterisk-dev] Asterisk 1.2.3 Released - Critical Update... Thanks for the stability!   Matt Riddell (IT)
• [asterisk-dev] Asterisk 1.2.3 Released - Critical Update... Thanksfor the stability!   Matt Riddell (IT)
• [asterisk-dev] Detection of Answering Machine   Matt Riddell (IT)
• [asterisk-dev] Detection of Answering Machine   Matt Riddell (IT)
• [asterisk-dev] Garbled Audio   Matt Riddell (IT)
• [asterisk-dev] php agi problem   ijb 007
• [asterisk-dev] php agi problem   ijb 007
• [asterisk-dev] svn 1.2   Martin Vít
• [asterisk-dev] Zaptel Makefile -O4   Martin Vít
• [asterisk-dev] Zaptel Makefile -O4   Martin Vít
• [Asterisk-Dev] How to recover caller and called numbers with Asterisk Manager   Amaury BOSSÉ
• SV: [Asterisk-Dev] asterisk-dev censorship   Håkon Nessjøen
• [asterisk-dev] Dail application, timeout not working...   Håkon Nessjøen
• [Asterisk-Dev] REFER / NOTIFY methods according to RFC-3515   Adrian A
• [Asterisk-Dev] ztcfg-dude?   Mark Abene
• [Asterisk-Dev] ztcfg-dude?   Mark Abene
• [Asterisk-Dev] zaptel echo preload patch   Rich Adamson
• [asterisk-dev] Asterisk 1.2.3 Released - Critical Update... Thanks for the stability!   Rich Adamson
• [asterisk-dev] Asterisk 1.2.3 Released - Critical Update... Thanks for the stability!   Rich Adamson
• [asterisk-dev] gr303   Rich Adamson
• [asterisk-dev] gr303   MountaiNEt Admin
• [Asterisk-Dev] Asterisk Port to Windows   Chris Albertson
• [Asterisk-Dev] Re: click-to-call cleint   Colin Anderson
• [Asterisk-Dev] Re: click-to-call cleint   Colin Anderson
• [Asterisk-Dev] Re: click-to-call cleint   Colin Anderson
• [Asterisk-Dev] WAS: click-to-call cleint NOW: XML Manager I/F str aw poll   Colin Anderson
• [Asterisk-Dev] application modules   Andrew
• [asterisk-dev] MIPSEL and Asterisk   Shireesh Annam
• [Asterisk-Dev] Asterisk Hard Phone + Soft Video Client Integration   North Antara
• [Asterisk-Dev] what next Asterisk ???   North Antara
• [asterisk-dev] bounty update $5000.00 - Asterisk bounty PRI 2B channel transfer for NI2 PRI line   North Antara
• [asterisk-dev] Re: asterisk-dev Digest, Vol 18, Issue 62   North Antara
• [asterisk-dev] Mantis bug site evolving?   North Antara
• [asterisk-dev] The s extention doesn't work   North Antara
• [asterisk-dev] "Issue #6439 - the "timebomb" bug. Patch by Markster over GPRS"   Domjan Attila
• [Asterisk-Dev] Setting conference room timeout   Dan Austin
• [Asterisk-Dev] Re: Setting conference room timeout   Dan Austin
• [Asterisk-Dev] Setting conference room timeout   Dan Austin
• [Asterisk-Dev] Clear MeetMe monitor only mode?   Dan Austin
• [asterisk-dev] Re: Bugs that Need Your Input!   Dan Austin
• [asterisk-dev] Re: Bugs that Need Your Input!   Dan Austin
• [asterisk-dev] frame size   Dan Austin
• [Asterisk-Dev] How to recover caller and called numbers with Asterisk Manager   amaury BOSSE
• [Asterisk-Dev] Asterisk Port to Windows   Michiel van Baak
• [Asterisk-Dev] Bugs that Need Your Input!   Boris Bakchiev
• [asterisk-dev] Re: asterisk-dev Digest, Vol 18, Issue 62   Brian Bell
• [asterisk-dev] Re: asterisk-dev Digest, Vol 18, Issue 63   Brian Bell
• [Asterisk-Dev] Asterisk 1.2.2 Released!   Ibs Bis
• [Asterisk-Dev] How to get the transfer number   Steve Blair
• [Asterisk-Dev] Re: click-to-call cleint   Kevin Bockman
• [Asterisk-Dev] Bugs that Need Your Input!   Greg Boehnlein
• [Asterisk-Dev] Asterisk Port to Windows   Greg Boehnlein
• [Asterisk-Dev] Asterisk Port to Windows   Greg Boehnlein
• [asterisk-dev] Dialstatus Oddity in 1.2   Greg Boehnlein
• [asterisk-dev] Asterisk 1.2.3 Released - Critical Update... Thanks for the stability!   Peter Braidwood
• [Asterisk-Dev] Asterisk Port to Windows   Chad Brown
• [Asterisk-Dev] Asterisk Port to Windows   Chad Brown
• [Asterisk-Dev] Minor typo and a few files left open   Russell Bryant
• [Asterisk-Dev] moving sounds out of asterisk repository   Russell Bryant
• [Asterisk-Dev] moving sounds out of asterisk repository   Russell Bryant
• [Asterisk-Dev] moving sounds out of asterisk repository   Russell Bryant
• [Asterisk-Dev] Asterisk 1.2.2 Released!   Russell Bryant
• [asterisk-dev] No audio? Update your Asterisk   Russell Bryant
• [asterisk-dev] Locking in astmm.c   Russell Bryant
• [asterisk-dev] Locking in astmm.c   Russell Bryant
• [Asterisk-Dev] Is it possible to use username and password from SugarCRm for voicemail?   Chuck Bunn
• [asterisk-dev] SayUnixTime() Bug   Butler, Larry
• [asterisk-dev] Asterisk NAT Prolem   Dome C.
• [asterisk-dev] Asterisk NAT Prolem   Dome C.
• [asterisk-dev] chan_h323.so   Paul Cadach
• Antw: Re: [asterisk-dev] chan_h323.so   Paul Cadach
• [asterisk-dev] Passing AOC information across channels   Paul Cadach
• [Asterisk-Dev] Latest SVN update causes CallerID problem on Polycom phones   Brian Capouch
• [Asterisk-Dev] moving sounds out of asterisk repository   Brian Capouch
• [Asterisk-Dev] Audio format woes (ChanSpy)   Juan Carlos Castro y Castro
• [Asterisk-Dev] Having to hack old (pre-1.0) version of Asterisk   Juan Carlos Castro y Castro
• [Asterisk-Dev] PING users of Manuel Guesdon's LDAP extensions   Juan Carlos Castro y Castro
• Antw: Re: [asterisk-dev] chan_h323.so   Juan Carlos Castro y Castro
• [Asterisk-Dev] app_queue enable call progress to callee   Raymond Chen
• [asterisk-dev] how to enable app_queue inband call progress to caller   Raymond Chen
• [Asterisk-Dev] Asterisk Hard Phone + Soft Video Client Integration   Sergio Chersovani
• [Asterisk-Dev] ringback after PlayBack()   Guo-Wei Chiuan
• [Asterisk-Dev] ringback after PlayBack()   Guo-Wei Chiuan
• [Asterisk-Dev] ringback after PlayBack()   Guo-Wei Chiuan
• [Asterisk-Dev] ringback after PlayBack()   Guo-Wei Chiuan
• [Asterisk-Dev] double ringing tone on 1.2.1 bug or misconfiguration ?   Simone Cittadini
• SV: [Asterisk-Dev] asterisk-dev censorship   Tzafrir Cohen
• [Asterisk-Dev] zaptel path   Tzafrir Cohen
• [Asterisk-Dev] zaptel path   Tzafrir Cohen
• [Asterisk-Dev] A few driver ideas...   Tzafrir Cohen
• [Asterisk-Dev] building debs from asterisk+debian svn   Tzafrir Cohen
• [Asterisk-Dev] building debs from asterisk+debian svn   Tzafrir Cohen
• [Asterisk-Dev] Race issue in channel.c involving uniqueint on Asterisk 1.2.1   Tzafrir Cohen
• [Asterisk-Dev] moving sounds out of asterisk repository   Tzafrir Cohen
• [Asterisk-Dev] moving sounds out of asterisk repository   Tzafrir Cohen
• [Asterisk-Dev] WAS: click-to-call cleint NOW: XML Manager I/F str aw poll   Tzafrir Cohen
• [Asterisk-Dev] WAS: click-to-call cleint NOW: XML Manager I/F str aw poll   Tzafrir Cohen
• [Asterisk-Dev] Asterisk 1.2.2 Released!   Tzafrir Cohen
• [asterisk-dev] Re: Bugs that Need Your Input!   Tzafrir Cohen
• [asterisk-dev] Re: ztdummy question   Tzafrir Cohen
• [asterisk-dev] Re: ztdummy question   Tzafrir Cohen
• [asterisk-dev] Re: ztdummy question   Tzafrir Cohen
• Antw: Re: [asterisk-dev] chan_h323.so   Tzafrir Cohen
• [asterisk-dev] ztdummy question   Sean Cook
• [asterisk-dev] Re: ztdummy question   Sean Cook
• [Asterisk-Dev] (nessun oggetto)   Giancarlo Corso
• [asterisk-dev] (nessun oggetto)   Giancarlo Corso
• [asterisk-dev] Asterisk 1.2.3 Released - Critical Update... Thanks for the stability!   James Courtier-Dutton
• [Asterisk-Dev] When do we change manager version?   Steven Critchfield
• [Asterisk-Dev] Having to hack old (pre-1.0) version of Asterisk   Steven Critchfield
• [Asterisk-Dev] Voicemail to email volume change patch   Steven Critchfield
• [Asterisk-Dev] Voicemail to email volume change patch   Steven Critchfield
• [Asterisk-Dev] Re: Voicemail to email volume change patch   Steven Critchfield
• [Asterisk-Dev] Asterisk Port to Windows   Steven Critchfield
• [asterisk-dev] bounty update $5000.00 - Asterisk bounty PRI 2B channel transfer for NI2 PRI line   Steven Critchfield
• [asterisk-dev] app_background and app_cepstral   Steven Critchfield
• [asterisk-dev] MIPSEL and Asterisk   Steven Critchfield
• [asterisk-dev] Passing AOC information across channels   Steven Critchfield
• [asterisk-dev] gr303   Steven Critchfield
• [asterisk-dev] Passing AOC information across channels   Steven Critchfield
• [asterisk-dev] Help   Chuck Crosby
• [Asterisk-Dev] Echo cancellation   Aaron Daniel
• [Asterisk-Dev] Echo cancellation   Aaron Daniel
• [Asterisk-Dev] Voicemail to email volume change patch   Aaron Daniel
• [Asterisk-Dev] Voicemail to email volume change patch   Aaron Daniel
• [Asterisk-Dev] Voicemail to email volume change patch   Aaron Daniel
• [asterisk-dev] "Issue #6439 - the "timebomb" bug. Patch by Markster over GPRS"   Aaron Daniel
• [asterisk-dev] volume boosting ?   Matt Darnell
• [asterisk-dev] Help   Malcolm Davenport
• [asterisk-dev] disclaimer   Malcolm Davenport
• [Asterisk-Dev] Re: click-to-call cleint   Paul Davidson
• [Asterisk-Dev] Question on using system(find args -exec rm {} \; )   Rusty Dekema
• [Asterisk-Dev] Re: click-to-call cleint   Saul Diaz
• [Asterisk-Dev] test   Rod Dorman
• [asterisk-dev] Garbled Audio   Rod Dorman
• [Asterisk-Dev] res/res_agi.c set accountcode   Cristian Draghici
• [Asterisk-Dev] res/res_agi.c set accountcode   Cristian Draghici
• [Asterisk-Dev] how to run Asterisk in gdb or other debugger?   Cristian Draghici
• [Asterisk-Dev] res/res_agi.c set accountcode   Cristian Draghici
• [Asterisk-Dev] sip/iax video   Cristian Draghici
• [asterisk-dev] php agi problem   Cristian Draghici
• [asterisk-dev] php agi problem   Cristian Draghici
• [asterisk-dev] OK, IAX2 - something is major wack - 1.2.3   Edward Eastman
• [asterisk-dev] custom mysql cdrs   Edward Eastman
• [asterisk-dev] app_queue: feature request and possible patch   Boris Erdmann
• [asterisk-dev] app_queue: feature request and possible patch   Boris Erdmann
• [asterisk-dev] Asterisk 1.2.3 Released - Critical Update...Thanks forthe stability!   Mario Evangelista-Silva
• [Asterisk-Dev] AGI variables   Innocent Evil
• [asterisk-dev] Detection of Answering Machine   Innocent Evil
• [asterisk-dev] Question on SIP Domains and registration   Barry Flanagan
• [asterisk-dev] SIP domain support for authentication and virtual hosting   Barry Flanagan
• [Asterisk-Dev] zaptel path   Kevin P. Fleming
• [Asterisk-Dev] zaptel path   Kevin P. Fleming
• [Asterisk-Dev] Echo cancellation   Kevin P. Fleming
• [Asterisk-Dev] Echo cancellation   Kevin P. Fleming
• [Asterisk-Dev] res/res_agi.c set accountcode   Kevin P. Fleming
• [Asterisk-Dev] locate parking lot in res_features.c ?   Kevin P. Fleming
• [Asterisk-Dev] memory-pool based string field management for structures   Kevin P. Fleming
• [Asterisk-Dev] bug in greek support for voicemail   Kevin P. Fleming
• [Asterisk-Dev] Remotely reboot SIP Phones ?   Kevin P. Fleming
• [Asterisk-Dev] Agents, members - who's taking the call in the queue?   Kevin P. Fleming
• [Asterisk-Dev] When do we change manager version?   Kevin P. Fleming
• [Asterisk-Dev] Latest SVN update causes CallerID problem on Polycom phones   Kevin P. Fleming
• [Asterisk-Dev] building debs from asterisk+debian svn   Kevin P. Fleming
• [Asterisk-Dev] potentially nasty bugs in string pools usage   Kevin P. Fleming
• [Asterisk-Dev] potentially nasty bugs in string pools usage   Kevin P. Fleming
• [Asterisk-Dev] building debs from asterisk+debian svn   Kevin P. Fleming
• [Asterisk-Dev] Race issue in channel.c involving uniqueint on Asterisk 1.2.1   Kevin P. Fleming
• [Asterisk-Dev] Race issue in channel.c involving uniqueint on Asterisk 1.2.1   Kevin P. Fleming
• [Asterisk-Dev] potentially nasty bugs in string pools usage   Kevin P. Fleming
• [Asterisk-Dev] potentially nasty bugs in string pools usage   Kevin P. Fleming
• [Asterisk-Dev] Race issue in channel.c involving uniqueint on Asterisk 1.2.1   Kevin P. Fleming
• [Asterisk-Dev] Re: trunk - r7863 /trunk/apps/app_voicemail.c   Kevin P. Fleming
• [Asterisk-Dev] Questions Regarding the ast_copy_string() Janitor Project   Kevin P. Fleming
• [Asterisk-Dev] Questions Regarding the ast_copy_string() Janitor Project   Kevin P. Fleming
• [Asterisk-Dev] G.729 or G.729A codec   Kevin P. Fleming
• [Asterisk-Dev] problem with the public svn server ?   Kevin P. Fleming
• [Asterisk-Dev] LOCAL_USER cleanup update   Kevin P. Fleming
• [Asterisk-Dev] LOCAL_USER cleanup update   Kevin P. Fleming
• [Asterisk-Dev] Authenticate App with Rights Check.   Kevin P. Fleming
• [Asterisk-Dev] LOCAL_USER cleanup update   Kevin P. Fleming
• [Asterisk-Dev] chan_sip: handling "Multiple Choice"   Kevin P. Fleming
• [Asterisk-Dev] moving sounds out of asterisk repository   Kevin P. Fleming
• [Asterisk-Dev] Re: [asterisk-commits] trunk - r8063 in /trunk: channels/ configs/ doc/   Kevin P. Fleming
• [Asterisk-Dev] moving sounds out of asterisk repository   Kevin P. Fleming
• [Asterisk-Dev] ztcfg-dude?   Kevin P. Fleming
• [Asterisk-Dev] svnmerge script for developer branches updated and automerging available   Kevin P. Fleming
• [Asterisk-Dev] [patch] Function to check passcode against vm passcode   Kevin P. Fleming
• [Asterisk-Dev] Subversion 'group' projects now available   Kevin P. Fleming
• [Asterisk-Dev] WAS: click-to-call cleint NOW: XML Manager I/F str aw poll   Kevin P. Fleming
• [Asterisk-Dev] [patch] Function to check passcode against vmpasscode   Kevin P. Fleming
• [Asterisk-Dev] "true" is -1 or 1 in boolean functions ?   Kevin P. Fleming
• [Asterisk-Dev] ringback after PlayBack()   Kevin P. Fleming
• [Asterisk-Dev] Subversion 'group' projects now available   Kevin P. Fleming
• [asterisk-dev] Asterisk NAT Prolem   Kevin P. Fleming
• [asterisk-dev] Re: [asterisk-commits] trunk - r8300 in /trunk: channels/chan_zap.c configs/zapata.conf.sample   Kevin P. Fleming
• [Asterisk-Dev] Bugs that Need Your Input!   Kevin P. Fleming
• [asterisk-dev] ztdummy question   Kevin P. Fleming
• [Asterisk-Dev] Asterisk 1.2.2 Released!   Kevin P. Fleming
• [asterisk-dev] Need to implement whisper mode.   Kevin P. Fleming
• [asterisk-dev] extension ordering   Kevin P. Fleming
• [asterisk-dev] No audio? Update your Asterisk   Kevin P. Fleming
• [asterisk-dev] language-specific sound directories   Kevin P. Fleming
• [asterisk-dev] language-specific sound directories   Kevin P. Fleming
• [asterisk-dev] language-specific sound directories   Kevin P. Fleming
• [asterisk-dev] Why asterisk _binary_ links with ssl?   Kevin P. Fleming
• [asterisk-dev] Why asterisk _binary_ links with ssl?   Kevin P. Fleming
• [asterisk-dev] Locking in astmm.c   Kevin P. Fleming
• [asterisk-dev] PLC and G.729A?   Kevin P. Fleming
• [Asterisk-Dev] app_bridge testers wanted   Matt Florell
• [Asterisk-Dev] More Zaptel/BRI questions   Matthew Fredrickson
• [Asterisk-Dev] More Zaptel/BRI questions   Matthew Fredrickson
• [Asterisk-Dev] zaptel echo preload   Matthew Fredrickson
• [Asterisk-Dev] zaptel echo preload   Matthew Fredrickson
• [asterisk-dev] IM/Presence with asterisk. asterisk NewBie   Matthew Fredrickson
• [Asterisk-Dev] Remotely reboot SIP Phones ?   Jian Hong GUAN
• [Asterisk-Dev] Handling SIP response 480   Papadopoulos Georgios
• [Asterisk-Dev] bug in greek support for voicemail   Papadopoulos Georgios
• [Asterisk-Dev] Subversion 'group' projects now available   Bob Goddard
• [asterisk-dev] rtp.c: Unable to allocate socket: Too many open files   Bob Goddard
• [Asterisk-Dev] what might corrupt...? (fixed, but have a question)   Goldfinger, Todd A
• [asterisk-dev] Passing AOC information across channels   Adam Goryachev
• [Asterisk-Dev] WAS: click-to-call cleint NOW: XML Manager I/F str aw poll   Daniel Grace
• [asterisk-dev] RE: asterisk-dev Digest, Vol 18, Issue 64   Dail Granholm
• [asterisk-dev] ael loading problem - not all output is printed   Ed Greenberg
• [asterisk-dev] No audio? Update your Asterisk   Ed Greenberg
• [Asterisk-Dev] Quadra software - Changing to Opensource   Kyle Hagan
• [Asterisk-Dev] [PATCH] Fix bug in handle_request_info   Marc Haisenko
• [Asterisk-Dev] [PATCH] Fix bug in handle_request_info   Marc Haisenko
• [asterisk-dev] Setting the context in a SIP channel   Marc Haisenko
• [asterisk-dev] Setting the context in a SIP channel   Marc Haisenko
• [asterisk-dev] Setting the context in a SIP channel   Marc Haisenko
• [asterisk-dev] Setting the context in a SIP channel   Marc Haisenko
• [asterisk-dev] Setting the context in a SIP channel   Marc Haisenko
• [Asterisk-Dev] Latest SVN update causes CallerID problem on Polycom phones   Trevor G. Hammonds
• [Asterisk-Dev] Latest SVN update causes CallerID problem onPolycom phones   Trevor G. Hammonds
• [Asterisk-Dev] CallerID Number Not Properly Displaying on Polycom / Progress on bug 6150   Trevor G. Hammonds
• [asterisk-dev] No application 'SIPAddHeader'   Trevor G. Hammonds
• [asterisk-dev] No application 'SIPAddHeader'   Trevor G. Hammonds
• [asterisk-dev] No application 'SIPAddHeader'   Trevor G. Hammonds
• [Asterisk-Dev] Minor typo and a few files left open   Steve Hanselman
• [Asterisk-Dev] moderation on asterisk-dev   Kevin Hanson
• [asterisk-dev] memory leaks   Beau Hargis
• [asterisk-dev] memory leaks   Beau Hargis
• [asterisk-dev] memory leaks   Beau Hargis
• [asterisk-dev] memory leaks   Beau Hargis
• [Asterisk-Dev] A few driver ideas...   James Harper
• [Asterisk-Dev] A few driver ideas...   James Harper
• [Asterisk-Dev] A few driver ideas...   James Harper
• [Asterisk-Dev] A few driver ideas...   James Harper
• [Asterisk-Dev] More Zaptel/BRI questions   James Harper
• [Asterisk-Dev] More Zaptel/BRI questions   James Harper
• [Asterisk-Dev] More Zaptel/BRI questions   James Harper
• [Asterisk-Dev] zaptel echo preload   James Harper
• [Asterisk-Dev] zaptel echo preload patch   James Harper
• [Asterisk-Dev] zaptel echo preload   James Harper
• [asterisk-dev] fsk stuff   James Harper
• [OT] Re: [asterisk-dev] Garbled Audio   James Harper
• [Asterisk-Dev] new developer question   Eric Hartley
• [asterisk-dev] Problems submitting a patch   Eric Hartley
• [asterisk-dev] RTP Payload size definition   Henri Herscher
• [Asterisk-Dev] How to get the transfer number   Chih-Wei Huang
• [Asterisk-Dev] Problems about applicationmap   Chih-Wei Huang
• [Asterisk-Dev] app_bridge testers wanted   Chih-Wei Huang
• [Asterisk-Dev] double ringing tone on 1.2.1 bug or misconfiguration ?   Mark Hulber
• [Asterisk-Dev] Asterisk Port to Windows   Jeremy Huntwork
• [Asterisk-Dev] chan_sip.c : ignoring domain part for incomingINVITE's causes conflicts between domains?   Olle E Johansson
• [Asterisk-Dev] snom programmable buttons   Olle E Johansson
• [Asterisk-Dev] locate parking lot in res_features.c ?   Olle E Johansson
• [Asterisk-Dev] RED ALERT! Bug marshals need your help! NOW!   Olle E Johansson
• [Asterisk-Dev] Agents, members - who's taking the call in the queue?   Olle E Johansson
• [Asterisk-Dev] When do we change manager version?   Olle E Johansson
• [Asterisk-Dev] REFER / NOTIFY methods according to RFC-3515   Olle E Johansson
• [Asterisk-Dev] Re-invite Issue   Olle E Johansson
• [Asterisk-Dev] SIP transfer using the REFER method according to RFC3515   Olle E Johansson
• [Asterisk-Dev] REFER / NOTIFY methods according to RFC-3515   Olle E Johansson
• [Asterisk-Dev] WAS: click-to-call cleint NOW: XML Manager I/F str aw poll   Olle E Johansson
• [Asterisk-Dev] Subversion 'group' projects now available   Olle E Johansson
• [asterisk-dev] app_queue: feature request and possible patch   Olle E Johansson
• [asterisk-dev] CSTA support for asterisk... or begining this module/project   Olle E Johansson
• [asterisk-dev] No audio? Update your Asterisk   Olle E Johansson
• [asterisk-dev] Asterisk 1.2.3 Released - Critical Update... Thanks for the stability!   Olle E Johansson
• [Asterisk-Dev] Dial returning DIALSTATUS of NOANSWER and not BUSY   Olle E Johansson
• [Asterisk-Dev] Dial returning DIALSTATUS of NOANSWER and not BUSY   Olle E Johansson
• [asterisk-dev] Passing AOC information across channels   Olle E Johansson
• [asterisk-dev] Passing AOC information across channels   Olle E Johansson
• [asterisk-dev] Passing AOC information across channels   Olle E Johansson
• [asterisk-dev] Passing AOC information across channels   Olle E Johansson
• [asterisk-dev] Passing AOC information across channels   Olle E Johansson
• [asterisk-dev] hints on an application?   Olle E Johansson
• [asterisk-dev] Why asterisk _binary_ links with ssl?   Olle E Johansson
• [asterisk-dev] Question on SIP Domains and registration   Olle E Johansson
• [asterisk-dev] SIP domain support for authentication and virtual hosting   Olle E Johansson
• [asterisk-dev] Passing AOC information across channels   Olle E Johansson
• [asterisk-dev] Passing AOC information across channels   Olle E Johansson
• [asterisk-dev] Help on chan_bluetooth   Benedicto Junior
• [asterisk-dev] The s extention doesn't work   Amine KECHAOU
• [Asterisk-Dev] test   Roy Sigurd Karlsbakk
• [Asterisk-Dev] moderation on asterisk-dev   Roy Sigurd Karlsbakk
• [Asterisk-Dev] asterisk-dev censorship   Roy Sigurd Karlsbakk
• [asterisk-dev] PLC and G.729A?   Roy Sigurd Karlsbakk
• [Asterisk-Dev] Attended TRANSFER with Asterisk Manager   Lilantha Karunaratne
• [Asterisk-Dev] ruby-agi 1.1.0 released   Mohammad Khan
• [Asterisk-Dev] [patch] Function to check passcode against vm passcode   Gil Kloepfer
• [Asterisk-Dev] [patch] Function to check passcode against vm passcode   Gil Kloepfer
• [Asterisk-Dev] [patch] Function to check passcode against vmpasscode   Gil Kloepfer
• [Asterisk-Dev] [patch] Function to check passcode against vmpasscode   Gil Kloepfer
• [asterisk-dev] SIP REGISTER problems with Adit 3104   Gil Kloepfer
• [Asterisk-Dev] potentially nasty bugs in string pools usage   Andrew Kohlsmith
• [Asterisk-Dev] ztcfg-dude?   Andrew Kohlsmith
• [Asterisk-Dev] app_dial: Privacy option 2 returns dial-status ANSWER / option_priority_jumping not respected   Koopmann, Jan-Peter
• [asterisk-dev] Re: Bugs that Need Your Input!   Koopmann, Jan-Peter
• [asterisk-dev] Re: Bugs that Need Your Input!   Koopmann, Jan-Peter
• [asterisk-dev] Passing AOC information across channels   Koopmann, Jan-Peter
• [asterisk-dev] Passing AOC information across channels   Koopmann, Jan-Peter
• [asterisk-dev] Passing AOC information across channels   Koopmann, Jan-Peter
• [asterisk-dev] Passing AOC information across channels   Koopmann, Jan-Peter
• [asterisk-dev] Passing AOC information across channels   Koopmann, Jan-Peter
• [asterisk-dev] Passing AOC information across channels   Koopmann, Jan-Peter
• [Asterisk-Dev] How to get the transfer number   Kaloyan Kovachev
• [Asterisk-Dev] res/res_agi.c set accountcode   Kaloyan Kovachev
• [Asterisk-Dev] How to get the transfer number   Kaloyan Kovachev
• [Asterisk-Dev] How to recover caller and called numbers withAsterisk Manager   Kaloyan Kovachev
• [Asterisk-Dev] application modules   Kaloyan Kovachev
• [Asterisk-Dev] moving sounds out of asterisk repository   Kaloyan Kovachev
• [asterisk-dev] Dail application, timeout not working...   Kaloyan Kovachev
• [Asterisk-Dev] silencesupp header in SDP   Steve Langstaff
• [Asterisk-Dev] REFER / NOTIFY methods according to RFC-3515   Lea
• [Asterisk-Dev] SIP transfer using the REFER method according to RFC3515   Lea
• [Asterisk-Dev] REFER / NOTIFY methods according to RFC-3515   Lea
• [Asterisk-Dev] REFER / NOTIFY methods according to RFC-3515   Lea
• [Asterisk-Dev] REFER / NOTIFY methods according to RFC-3515   Lea
• [Asterisk-Dev] REFER / NOTIFY methods according to RFC-3515   Lea
• [asterisk-dev] ztdummy question   Ben Lear
• [Asterisk-Dev] chan_sip.c - BUG REPORT: Request-URI not in compliance with RFC3261   Daniel Leeds
• [Asterisk-Dev] Race issue in channel.c involving uniqueint on Asterisk 1.2.1   Tilghman Lesher
• [Asterisk-Dev] Race issue in channel.c involving uniqueint on Asterisk 1.2.1   Tilghman Lesher
• [Asterisk-Dev] Race issue in channel.c involving uniqueint on Asterisk 1.2.1   Tilghman Lesher
• [Asterisk-Dev] Question on using system(find args -exec rm {} \; )   Tilghman Lesher
• [Asterisk-Dev] Agents, members - who's taking the call in the queue?   Tilghman Lesher
• [Asterisk-Dev] possible bug in pbx.c::substring() ?   Tilghman Lesher
• [Asterisk-Dev] "true" is -1 or 1 in boolean functions ?   Tilghman Lesher
• [Asterisk-Dev] [patch] Function to check passcode against vmpasscode   Tilghman Lesher
• [asterisk-dev] Re: [asterisk-commits] trunk - r8300 in /trunk: channels/chan_zap.c configs/zapata.conf.sample   Tilghman Lesher
• [asterisk-dev] extension ordering   Tilghman Lesher
• [asterisk-dev] Problems submitting a patch   Tilghman Lesher
• [asterisk-dev] Zaptel Makefile -O4   Tilghman Lesher
• [asterisk-dev] Asterisk 1.2.3 Released - Critical Update... Thanks for the stability!   Tilghman Lesher
• [asterisk-dev] Setting the context in a SIP channel   Tilghman Lesher
• [asterisk-dev] memory leaks   Tilghman Lesher
• [asterisk-dev] Setting the context in a SIP channel   Tilghman Lesher
• [asterisk-dev] Setting the context in a SIP channel   Tilghman Lesher
• [asterisk-dev] memory leaks   Tilghman Lesher
• [asterisk-dev] Passing AOC information across channels   Tilghman Lesher
• [Asterisk-Dev] how to run Asterisk in gdb or other debugger?   Peter Loeppky
• [asterisk-dev] frame size   Fernando Lombardo
• [asterisk-dev] frame size   Fernando Lombardo
• [Asterisk-Dev] RE: [Asterisk-Users] ACD with polycom ip phones   Alexander Lopez
• [Asterisk-Dev] How to recover caller and called numbers withAsterisk Manager   Alexander Lopez
• [Asterisk-Dev] Agents, members - who's taking the call in the queue?   Alexander Lopez
• [Asterisk-Dev] When do we change manager version?   Alexander Lopez
• [Asterisk-Dev] Authenticate App with Rights Check.   Alexander Lopez
• [Asterisk-Dev] [patch] Function to check passcode against vmpasscode   Alexander Lopez
• [Asterisk-Dev] [patch] Function to check passcode againstvmpasscode   Alexander Lopez
• [Asterisk-Dev] Asterisk Port to Windows   Alexander Lopez
• [asterisk-dev] Need to implement whisper mode.   Alexander Lopez
• [asterisk-dev] Need to implement whisper mode.   Alexander Lopez
• [asterisk-dev] Re: Need to implement whisper mode.   Alexander Lopez
• [asterisk-dev] No application 'SIPAddHeader'   Alexander Lopez
• [asterisk-dev] Need to implement whisper mode.   Alexander Lopez
• [asterisk-dev] Modification to ChanisAvail to save BRIDGEDCHAN   Alexander Lopez
• [asterisk-dev] Modification to ChanisAvail to save BRIDGEDCHAN   Alexander Lopez
• [Asterisk-Dev] Problem on ZAP channel   Richard Lyman
• [Asterisk-Dev] zaptel echo preload   Richard Lyman
• [Asterisk-Dev] zaptel echo preload   Richard Lyman
• [Asterisk-Dev] When do we change manager version?   Julian Lyndon-Smith
• [Asterisk-Dev] Handling SIP response 480   Mikael Magnusson
• [Asterisk-Dev] REFER / NOTIFY methods according to RFC-3515   Mikael Magnusson
• [Asterisk-Dev] transfer() is not compliant with SIP RFC-3515 and RFC-3261   Mikael Magnusson
• [asterisk-dev] Escaping of # in URI?   Mikael Magnusson
• [Asterisk-Dev] Facility Name requested on channel 0/2 not in use on span 1   Henry Margies
• [Asterisk-Dev] Facility Name requested on channel 0/2 not in use on span 1   Henry Margies
• [asterisk-dev] disclaimer   Ivan F. Martinez
• [Asterisk-Dev] sip/iax video   Matt
• [Asterisk-Dev] click-to-call cleint   Matt
• [Asterisk-Dev] Problem on ZAP channel   Steve McMahon
• [Asterisk-Dev] Having to hack old (pre-1.0) version of Asterisk   Jeremy McNamara
• [Asterisk-Dev] Asterisk Port to Windows   Jeremy McNamara
• [Asterisk-Dev] Asterisk Port to Windows   Jeremy McNamara
• [asterisk-dev] sip register question   Zahid Mehmood
• [Asterisk-Dev] Re: click-to-call cleint   Phil Menico
• [Asterisk-Dev] Re: click-to-call cleint   Yaakov Menken
• [Asterisk-Dev] Re: Setting conference room timeout   Tony Mountifield
• [Asterisk-Dev] Re: trunk - r7863 /trunk/apps/app_voicemail.c   Tony Mountifield
• [asterisk-dev] Re: Need to implement whisper mode.   Tony Mountifield
• [asterisk-dev] Re: Bugs that Need Your Input!   Tony Mountifield
• [asterisk-dev] Re: ztdummy question   Tony Mountifield
• [asterisk-dev] Re: Asterisk 1.2.2 Released!   Tony Mountifield
• [asterisk-dev] Re: ztdummy question   Tony Mountifield
• [asterisk-dev] Re: ztdummy question   Tony Mountifield
• [Asterisk-Dev] potentially nasty bugs in string pools usage   Steve Murphy
• [Asterisk-Dev] AEL2 -- call to arms   Steve Murphy
• [asterisk-dev] Mantis bug site evolving?   Steve Murphy
• [Asterisk-Dev] Race issue in channel.c involving uniqueint on Asterisk 1.2.1   Dinesh Nair
• [asterisk-dev] Passing AOC information across channels   Dinesh Nair
• [Asterisk-Dev] zaptel path   Igor Neves
• [Asterisk-Dev] zaptel path   Igor Neves
• [Asterisk-Dev] zaptel path   Igor Neves
• [Asterisk-Dev] zaptel path   Igor Neves
• [Asterisk-Dev] Asterisk Port to Windows   Igor Neves
• [asterisk-dev] chan_h323.so   Hoai-Anh Ngo-Vi
• Antw: Re: [asterisk-dev] chan_h323.so   Hoai-Anh Ngo-Vi
• Antw: Re: [asterisk-dev] chan_h323.so   Hoai-Anh Ngo-Vi
• Antw: Re: [asterisk-dev] chan_h323.so   Hoai-Anh Ngo-Vi
• Antw: Re: [asterisk-dev] chan_h323.so   Hoai-Anh Ngo-Vi
• Antw: Re: [asterisk-dev] chan_h323.so   Hoai-Anh Ngo-Vi
• [Asterisk-Dev] How to adapt call termination keys in Asterisk.   Obelix
• [asterisk-dev] No application 'SIPAddHeader'   Sergey Okhapkin
• [asterisk-dev] extension ordering   Sergey Okhapkin
• [Asterisk-Dev] what next Asterisk ???   Taiwo Oluyemi
• [Asterisk-Dev] Festival problems with asterisk   David Otero
• [Asterisk-Dev] Using AGI script for festival   David Otero
• [Asterisk-Dev] Festival problems with asterisk   David Otero
• [Asterisk-Dev] When do we change manager version?   Florian Overkamp
• [Asterisk-Dev] Disconnecting call 'SIP/X.X.X.X-085340d0' for lack of RTP activity in 11 seconds   Javier Oviedo
• [Asterisk-Dev] Re: [Asterisk-Users] Disconnecting call 'SIP/X.X.X.X-085340d0' for lack of RTP activity in 11 seconds   Javier Oviedo
• [Asterisk-Dev] Authenticate App with Rights Check.   Alvaro Parres
• [Asterisk-Dev] G.729 or G.729A codec   Avin Patel
• [asterisk-dev] rtp.c: Unable to allocate socket: Too many open files   Quang Minh Phan
• [Asterisk-Dev] 482 Loop Detected problem   Matteo Piazza
• [Asterisk-Dev] Dialplan Loop Detection   Wolfgang Pichler
• [asterisk-dev] Escaping of # in URI?   Daniel Pocock
• [Asterisk-Dev] why doesn't asterisk support TCP   Michael Prochaska
• [Asterisk-Dev] res_config_mysql with ldap authentication   Michael Prochaska
• [Asterisk-Dev] Asterisk 1.2.2 Released!   Aryanto Rachmad
• [Asterisk-Dev] Bugs that Need Your Input!   Matt Ranney
• [Asterisk-Dev] Setting conference room timeout   Atif Rasheed
• [Asterisk-Dev] silencesupp header in SDP   Atif Rasheed
• [Asterisk-Dev] silencesupp header in SDP   Atif Rasheed
• [Asterisk-Dev] silencesupp header in SDP   Atif Rasheed
• [Asterisk-Dev] When do we change manager version?   Stefan Reuter
• [Asterisk-Dev] A few driver ideas...   Christian Richter
• [Asterisk-Dev] A few driver ideas...   Christian Richter
• [Asterisk-Dev] A few driver ideas...   Christian Richter
• [Asterisk-Dev] bug in greek support for voicemail   Christian Richter
• [Asterisk-Dev] Problems about applicationmap   Christian Richter
• [Asterisk-Dev] Having to hack old (pre-1.0) version of Asterisk   Christian Richter
• [Asterisk-Dev] utils.c: too large or negative timestamps   Christian Richter
• [asterisk-dev] Internationalization Issues   Christian Richter
• [asterisk-dev] Passing AOC information across channels   Christian Richter
• [Asterisk-Dev] Race issue in channel.c involving uniqueint on Asterisk 1.2.1   Luigi Rizzo
• [Asterisk-Dev] Race issue in channel.c involving uniqueint on Asterisk 1.2.1   Luigi Rizzo
• [Asterisk-Dev] Race issue in channel.c involving uniqueint onAsterisk 1.2.1   Luigi Rizzo
• [Asterisk-Dev] locate parking lot in res_features.c ?   Luigi Rizzo
• [Asterisk-Dev] locate parking lot in res_features.c ?   Luigi Rizzo
• [Asterisk-Dev] potentially nasty bugs in string pools usage   Luigi Rizzo
• [Asterisk-Dev] potentially nasty bugs in string pools usage   Luigi Rizzo
• [Asterisk-Dev] potentially nasty bugs in string pools usage   Luigi Rizzo
• [Asterisk-Dev] potentially nasty bugs in string pools usage   Luigi Rizzo
• [Asterisk-Dev] potentially nasty bugs in string pools usage   Luigi Rizzo
• [Asterisk-Dev] Race issue in channel.c involving uniqueint on Asterisk 1.2.1   Luigi Rizzo
• [Asterisk-Dev] Race issue in channel.c involving uniqueint on Asterisk 1.2.1   Luigi Rizzo
• [Asterisk-Dev] Race issue in channel.c involving uniqueint on Asterisk 1.2.1   Luigi Rizzo
• [Asterisk-Dev] Questions Regarding the ast_copy_string() Janitor Project   Luigi Rizzo
• [Asterisk-Dev] problem with the public svn server ?   Luigi Rizzo
• [Asterisk-Dev] LOCAL_USER cleanup update   Luigi Rizzo
• [Asterisk-Dev] LOCAL_USER cleanup update   Luigi Rizzo
• [Asterisk-Dev] LOCAL_USER cleanup update   Luigi Rizzo
• [Asterisk-Dev] LOCAL_USER cleanup update   Luigi Rizzo
• [Asterisk-Dev] LOCAL_USER cleanup update   Luigi Rizzo
• [Asterisk-Dev] "true" is -1 or 1 in boolean functions ?   Luigi Rizzo
• [Asterisk-Dev] possible bug in pbx.c::substring() ?   Luigi Rizzo
• [Asterisk-Dev] intended behaviour of ast_extension_close() ?   Luigi Rizzo
• [Asterisk-Dev] possible bug in pbx.c::substring() ?   Luigi Rizzo
• [Asterisk-Dev] "true" is -1 or 1 in boolean functions ?   Luigi Rizzo
• [asterisk-dev] extension ordering   Luigi Rizzo
• [asterisk-dev] extension ordering   Luigi Rizzo
• [asterisk-dev] extension ordering   Luigi Rizzo
• [asterisk-dev] No audio? Update your Asterisk   Luigi Rizzo
• [asterisk-dev] "Issue #6439 - the "timebomb" bug. Patch by Markster over GPRS"   Luigi Rizzo
• [asterisk-dev] volume boosting ?   Luigi Rizzo
• [asterisk-dev] volume boosting ?   Luigi Rizzo
• [asterisk-dev] language-specific sound directories   Luigi Rizzo
• [asterisk-dev] language-specific sound directories   Luigi Rizzo
• [asterisk-dev] language-specific sound directories   Luigi Rizzo
• [asterisk-dev] dubious code in channel.c   Luigi Rizzo
• [asterisk-dev] branch with my changes available on svn.digium.com   Luigi Rizzo
• [Asterisk-Dev] Asterisk Port to Windows   Josh Roberson
• [Asterisk-Dev] chan_sip: handling "Multiple Choice"   Bruno Rocha
• [asterisk-dev] CSTA support for asterisk... or begining this module/project   Fernando Romo
• [asterisk-dev] Asterisk 1.2.3 Released - Critical Update... Thanks for the stability!   Fernando Romo
• [asterisk-dev] Asterisk 1.2.3 Released - Critical Update... Thanks for the stability!   Fernando Romo
• [asterisk-dev] Asterisk 1.2.3 Released - Critical Update... Thanks for the stability!   Fernando Romo
• [Asterisk-Dev] "true" is -1 or 1 in boolean functions ?   Elazar Rosenthal
• [Asterisk-Dev] Questions Regarding the ast_copy_string() Janitor Project   Matt Roth
• [Asterisk-Dev] Questions Regarding the ast_copy_string() Janitor Project   Matt Roth
• [asterisk-dev] No audio? Update your Asterisk   Matt Roth
• [asterisk-dev] Locking in astmm.c   Matt Roth
• [Asterisk-Dev] Reading sound and recognizing DTMF sounds at the same time in eagi script ?   Robert Rozman
• [asterisk-dev] dial out and message playback   Danish Samad
• [Asterisk-Dev] why doesn't asterisk support TCP   SanYeiChen
• [asterisk-dev] Passing AOC information across channels   Frank Sautter
• [asterisk-dev] No audio? Update your Asterisk   Matt Schulte
• [Asterisk-Dev] app_bridge testers wanted   Heath Schultz
• [Asterisk-Dev] A few driver ideas...   Hans Petter Selasky
• [Asterisk-Dev] Question on using system(find args -exec rm {} \; )   Noel Sharpe
• [Asterisk-Dev] Googles VoIP protocol   Shidan
• [Asterisk-Dev] Re: click-to-call cleint   Shidan
• [Asterisk-Dev] WAS: click-to-call cleint NOW: XML Manager I/F str aw poll   Shidan
• [asterisk-dev] Fwd: [Asterisk-Users] Transfer (SIP REFER) - AccountCode not available?   Shidan
• [Asterisk-Dev] Asterisk Port to Windows   Adrian Sietsma
• [Asterisk-Dev] Dialplan Loop Detection   Andreas Sikkema
• [Asterisk-Dev] res/res_agi.c set accountcode   Moises Silva
• [Asterisk-Dev] How to recover caller and called numbers withAsterisk Manager   Moises Silva
• [Asterisk-Dev] new developer question   Moises Silva
• [Asterisk-Dev] AGI variables   Moises Silva
• [Asterisk-Dev] silencesupp header in SDP   Moises Silva
• [Asterisk-Dev] silencesupp header in SDP   Moises Silva
• [asterisk-dev] attended transfer TRANSFER_CONTEXT, why from transferee first?   Moises Silva
• [asterisk-dev] php agi problem   Moises Silva
• [Asterisk-Dev] Asterisk Clusters   Goran Skular
• [Asterisk-Dev] Asterisk Clusters   Goran Skular
• [Asterisk-Dev] queue_log parser   Goran Skular
• [asterisk-dev] Why asterisk _binary_ links with ssl?   Denis Smirnov
• [Asterisk-Dev] Voicemail to email volume change patch   Jared Smith
• [asterisk-dev] app_queue: feature request and possible patch   Jared Smith
• [asterisk-dev] SIP REGISTER problems with Adit 3104   Jared Smith
• [Asterisk-Dev] "true" is -1 or 1 in boolean functions ?   Derek Smithies
• [Asterisk-Dev] More Zaptel/BRI questions   Steven
• [Asterisk-Dev] moving sounds out of asterisk repository   Steven
• [Asterisk-Dev] new developer question   Steven
• [Asterisk-Dev] moving sounds out of asterisk repository   Steven
• [Asterisk-Dev] click-to-call cleint   Steven
• [Asterisk-Dev] WAS: click-to-call cleint NOW: XML Manager I/F str aw poll   Steven
• [Asterisk-Dev] Asterisk 1.2.2 Released!   Steven
• [Asterisk-Dev] Asterisk Port to Windows   Steven
• [OT] Re: [asterisk-dev] Garbled Audio   Steven
• [OT] Re: [asterisk-dev] Garbled Audio   Steven
• [Asterisk-Dev] how to run Asterisk in gdb or other debugger?   TTulinsky at netscape.net
• [Asterisk-Dev] Re: Voicemail to email volume change patch   Mike Taht
• [asterisk-dev] "Issue #6439 - the "timebomb" bug. Patch by Markster over GPRS"   Tamas
• [asterisk-dev] "Issue #6439 - the "timebomb" bug. Patch by Markster over GPRS"   Tamas
• [Asterisk-Dev] Asterisk 1.2.2 Released!   Asterisk Development Team
• [asterisk-dev] Asterisk Development and Release Cycle   Asterisk Development Team
• [asterisk-dev] Asterisk 1.2.3 Released - Critical Update   Asterisk Development Team
• [asterisk-dev] Asterisk 1.2.4 and Zaptel 1.2.3   The Asterisk Development Team
• [Asterisk-Dev] Re: Voicemail to email volume change patch   Robert A. Thompson
• [Asterisk-Dev] Re: Voicemail to email volume change patch   Robert A. Thompson
• [Asterisk-Dev] Re: Voicemail to email volume change patch   Robert A. Thompson
• [Asterisk-Dev] Re: Re: Voicemail to email volume change patch   Robert A. Thompson
• [asterisk-dev] extension ordering   John Todd
• [asterisk-dev] Dialstatus Oddity in 1.2   John Todd
• [asterisk-dev] CSTA support for asterisk... or begining this module/project   John Todd
• [Asterisk-Dev] Mute a given channel.   Jan du Toit
• [Asterisk-Dev] Channel muting support in the Asterisk Manager Interface.   Jan du Toit
• [asterisk-dev] Place were Meetme recordings is stored.   Jan du Toit
• [Asterisk-Dev] Race issue in channel.c involving uniqueint onAsterisk 1.2.1   Steve Totaro
• [Asterisk-Dev] asterisk-dev censorship   Steve Totaro
• [Asterisk-Dev] Asterisk Hard Phone + Soft Video Client Integration   Steve Totaro
• [asterisk-dev] bounty update $5000.00 - Asterisk bounty PRI 2Bchannel transfer for NI2 PRI line   Steve Totaro
• [Asterisk-Dev] Asterisk Port to Windows   Steve Totaro
• [Asterisk-Dev] zaptel echo preload   Steve Underwood
• [Asterisk-Dev] memory-pool based string field management for structures   Martin Vit
• [asterisk-dev] Can't get WaitForSilence to work   John Vogel
• [Asterisk-Dev] asterisk 1.2 g729 compile errors   Charles Wang
• [Asterisk-Dev] Asterisk Clusters   Watkins, Bradley
• [asterisk-dev] RTP Payload size definition   Robert Webb
• [Asterisk-Dev] moving sounds out of asterisk repository   BJ Weschke
• [Asterisk-Dev] [PATCH] Fix bug in handle_request_info   BJ Weschke
• [Asterisk-Dev] Re: [asterisk-commits] trunk - r8063 in /trunk: channels/ configs/ doc/   BJ Weschke
• [Asterisk-Dev] time stamp in queue_log   BJ Weschke
• [Asterisk-Dev] Re: click-to-call cleint   BJ Weschke
• [Asterisk-Dev] Asterisk Port to Windows   BJ Weschke
• [Asterisk-Dev] Asterisk Port to Windows   BJ Weschke
• [asterisk-dev] No audio? Update your Asterisk   BJ Weschke
• [asterisk-dev] Asterisk 1.2.3 Released - Critical Update... Thanks for the stability!   BJ Weschke
• [asterisk-dev] Modification to ChanisAvail to save BRIDGEDCHAN   BJ Weschke
• [Asterisk-Dev] ringback after PlayBack()   Eric "ManxPower" Wieling
• [Asterisk-Dev] Dialplan Loop Detection   Eric "ManxPower" Wieling
• [Asterisk-Dev] ringback after PlayBack()   Eric "ManxPower" Wieling
• [Asterisk-Dev] Re-invite Issue   Gene Willingham
• [Asterisk-Dev] RE: Re-invite Issue Revisited   Gene Willingham
• [asterisk-dev] app_background and app_cepstral   Jason Wolfe
• [Asterisk-Dev] Race issue in channel.c involving uniqueint on Asterisk 1.2.1   David Woodhouse
• [Asterisk-Dev] Asterisk Hard Phone + Soft Video Client Integration   Harry Yeh
• [Asterisk-Dev] what's mean when ackcall in agent.conf is set to "always"?   Peng Yong
• [Asterisk-Dev] Re: [asterisk-commits] trunk - r8063 in /trunk: channels/ configs/ doc/   Peng Yong
• [Asterisk-Dev] time stamp in queue_log   Peng Yong
• [Asterisk-Dev] time stamp in queue_log   Peng Yong
• [Asterisk-Dev] automated response   Michael Young
• [Asterisk-Dev] (no subject)   Ahmed Zaky
• [Asterisk-Dev] moderation on asterisk-dev   Zoa
• [Asterisk-Dev] When do we change manager version?   Zoa
• [asterisk-dev] No audio? Update your Asterisk   Zoa
• [Asterisk-Dev] G.729 or G.729A codec   Vlasis Hatzistavrou - asterisk mailing list account
• [asterisk-dev] IM/Presence with asterisk. asterisk NewBie   roswel ajf
• [Asterisk-Dev] ztcfg-dude?   alex at pilosoft.com
• [asterisk-dev] bounty update $5000.00 - Asterisk bounty PRI 2B channel transfer for NI2 PRI line   alex at pilosoft.com
• [asterisk-dev] frame size   asterisk at ropeguru.com
• [asterisk-dev] Garbled Audio   burak balasaygun
• [Asterisk-Dev] Question on using system(find args -exec rm {} \; )   bday at prosodiemail.com
• [Asterisk-Dev] Question on using system(find args -exec rm {} \; )   bday at prosodiemail.com
• [asterisk-dev] Asterisk 1.2.3 Released - Critical Update... Thanksfor the stability!   calvis
• [asterisk-dev] Debug Help   roberto caspa
• [Asterisk-Dev] snom programmable buttons   cfh
• [Asterisk-Dev] Asterisk 1.2.2 Released!   John covici
• [Asterisk-Dev] chan_sip.c - BUG REPORT: Request-URI not in compliance with RFC3261   george.robinson at softhome.net
• [Asterisk-Dev] Setting conference room timeout   ast guy
• [Asterisk-Dev] Setting conference room timeout   ast guy
• [asterisk-dev] rtp.c: Unable to allocate socket: Too many open files   ast guy
• [asterisk-dev] rtp.c: Unable to allocate socket: Too many open files   ast guy
• [asterisk-dev] rtp.c: Unable to allocate socket: Too many open files   ast guy
• [asterisk-dev] rtp.c: Unable to allocate socket: Too many open files   ast guy
• [asterisk-dev] Dail application, timeout not working...   ast guy
• [asterisk-dev] Dail application, timeout not working...   ast guy
• [Asterisk-Dev] ruby-agi-1.0.2 released !   info at beeplove.com
• [Asterisk-Dev] test   janvb at caselaboratories.com
• [Asterisk-Dev] moderation on asterisk-dev   janvb at caselaboratories.com
• [Asterisk-Dev] Hardware Manual   janvb at caselaboratories.com
• [Asterisk-Dev] More Zaptel/BRI questions   janvb at caselaboratories.com
• [asterisk-dev] SIP, H263, and ftmp   john97 at flatline.com
• [Asterisk-Dev] A few driver ideas...   kpj at junghanns.net
• [Asterisk-Dev] A few driver ideas...   kpj at junghanns.net
• [Asterisk-Dev] A few driver ideas...   kpj at junghanns.net
• [Asterisk-Dev] A few driver ideas...   kpj at junghanns.net
• [Asterisk-Dev] A few driver ideas...   kpj at junghanns.net
• [Asterisk-Dev] A few driver ideas...   kpj at junghanns.net
• [Asterisk-Dev] More Zaptel/BRI questions   kpj at junghanns.net
• [Asterisk-Dev] transfer() is not compliant with SIP RFC-3515 and RFC-3261   lea123 at wp.pl
• [Asterisk-Dev] REFER / NOTIFY methods according to RFC-3515   lea123 at wp.pl
• [Asterisk-Dev] transfer() is not compliant with SIP RFC-3515 and RFC-326 1   lea123 at wp.pl
• [Asterisk-Dev] Request-URI for SIP BYE method is also WRONG !!!   lea123 at wp.pl
• [asterisk-dev] OK, IAX2 - something is major wack - 1.2.3   mezzmor at aim.com
• [asterisk-dev] hints on an application?   mitch at thosesharps.net
• [Asterisk-Dev] *** Warning: "zt_register" [/usr/src/bristuff-0.3.0-PRE-1c/zaphfc/zaphfc.ko] undefined!   news.dalaidily news.dalaidily
• [Asterisk-Dev] Asterisk Port to Windows   tim panton
• [asterisk-dev] Asterisk 1.2.3 Released - Critical Update... Thanks for the stability!   tim panton
• [asterisk-dev] Passing AOC information across channels   tim panton
• [asterisk-dev] dial out and message playback   mirza sahib
• [Asterisk-Dev] potentially nasty bugs in string pools usage   steve at daviesfam.org
• [Asterisk-Dev] Asterisk segfault in pbx_builtin_setvar_helper   steve at daviesfam.org
• [Asterisk-Dev] [patch] Function to check passcode against vmpasscode   steve at daviesfam.org
• [Asterisk-Dev] Re: [Asterisk-Users] Disconnecting call 'SIP/X.X.X.X-085340d0' for lack of RTP activity in 11 seconds   steve at daviesfam.org
• [Asterisk-Dev] Asterisk Port to Windows   steve at daviesfam.org
• [asterisk-dev] ztdummy question   steve at daviesfam.org
• [asterisk-dev] app_queue: feature request and possible patch   steve at daviesfam.org
• [asterisk-dev] Dialstatus Oddity in 1.2   steve at daviesfam.org
• [asterisk-dev] app_queue: feature request and possible patch   steve at daviesfam.org
• [Asterisk-Dev] Deadlock?Thread create error.   tiszaii at t-online.hu
• [Asterisk-Dev] Threads create error.Deadlock?   tiszaii at t-online.hu
• [Asterisk-Dev] RE:"Failed to create update thread"   tiszaii at t-online.hu
• [Asterisk-Dev] Dialtone detection help needed   voip3 at nibble.net
• [Asterisk-Dev] Asterisk bounty PRI 2B channel transfer for NI2 PRI line   voip3 at nibble.net
• [Asterisk-Dev] Asterisk least cost routing expert needed   voip3 at nibble.net
• [asterisk-dev] bounty update $5000.00 - Asterisk bounty PRI 2B channel transfer for NI2 PRI line   voip3 at nibble.net

Last message date: Tue Jan 31 21:20:41 MST 2006
Archived on: Tue Sep 5 14:27:49 MST 2006

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