February 2005 Archives by author

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Starting: Tue Feb 1 07:30:47 MST 2005
Ending: Mon Feb 28 20:40:21 MST 2005
Messages: 619

• [Asterisk-Dev] reload and realtime   Frank (INMTE)
• [Asterisk-Dev] ExtensionState problems using Manager.conf API   Nicolás Gudiño
• [Asterisk-Dev] include the results of an executable file (*.conf)   Nicolás Gudiño
• [Asterisk-Dev] .call-file handling suggestion   Daniel Nyström
• [Asterisk-Dev] Asterisk - failover from g729 to gsm capable?   Rich Adamson
• [Asterisk-Dev] Dev Conf 2pm CST   Rich Adamson
• [Asterisk-Dev] Voicemail not working properly   Kamran Ahmad
• [Asterisk-Dev] calling problem in cvs verison on fedora core2   Kamran Ahmad
• [Asterisk-Dev] Function used in Register User Agent   Kamran Ahmad
• [Asterisk-Dev] difference of ast_channel struct in cvs and stable   Kamran Ahmad
• [Asterisk-Dev] Re: difference of ast_channel struct in cvs and stable   Kamran Ahmad
• [Asterisk-Dev] problem in loading module with asterisk   Kamran Ahmad
• [Asterisk-Dev] how to load application with asterisk   Kamran Ahmad
• [Asterisk-Dev] calling stored procedure   Kamran Ahmad
• [Asterisk-Dev] moving to another priority   Kamran Ahmad
• [Asterisk-Dev] calling one application from other application   Kamran Ahmad
• [Asterisk-Dev] Re: calling one application from other application   Kamran Ahmad
• [Asterisk-Dev] Re: calling one application from other application   Kamran Ahmad
• [Asterisk-Dev] Re: calling one application from other application   Kamran Ahmad
• [Asterisk-Dev] how to use ast_channel_setwhentohangup to allocate maximum time for call   Kamran Ahmad
• [Asterisk-Dev] Re: how to use ast_channel_setwhentohangup to allocate maximum time for call   Kamran Ahmad
• [Asterisk-Dev] Re: how to use ast_channel_setwhentohangup to allocat maximum time for call   Kamran Ahmad
• [Asterisk-Dev] Too many open files   Federico Alves
• [Asterisk-Dev] dev conf topic: better CDRs   Asterisk
• [Asterisk-Dev] Wildcard TE410P Question   Domjan Attila
• [Asterisk-Dev] manager interface, get callerid number??   Atuc
• [Asterisk-Dev] Asterisk compliant JAIN API?   Robert Augustyn
• [Asterisk-Dev] Asterisk compliant JAIN API?   Robert Augustyn
• [Asterisk-Dev] Asterisk use cases   Robert Augustyn
• [Asterisk-Dev] I'm writing chan_gsm   Igor - BZ
• [Asterisk-Dev] $1500 Bounty for Dictation Module   Nick Bachmann
• [Asterisk-Dev] Dev Conf 2pm CST   Nick Bachmann
• [Asterisk-Dev] Dev Conf 2pm CST   Nick Bachmann
• [Asterisk-Dev] Codec = RAW ?   Nick Bachmann
• [Asterisk-Dev] chan_sip   Nick Bachmann
• [Asterisk-Dev] Dev Conf 2pm CST   Nick Bachmann
• [Asterisk-Dev] DTMF inter-digit delay   Nick Bachmann
• [Asterisk-Dev] DTMF inter-digit delay   Nick Bachmann
• [Asterisk-Dev] Packet Cable NCS Support   Fernando Berretta
• [Asterisk-Dev] Asterisk Memory Leak   Sachin Bhatia
• [Asterisk-Dev] Asterisk Memory Leak   Sachin Bhatia
• [Asterisk-Dev] Asterisk Memory Leak   Sachin Bhatia
• [Asterisk-Dev] changing codec during call   Daniel Bichara
• [Asterisk-Dev] libiax2 OK for production?   Matthew Boehm
• [Asterisk-Dev] Bounty IAX Fax Tone Detection   Matthew Boehm
• [Asterisk-Dev] deadlock / can't debug   Matthew Boehm
• [Asterisk-Dev] reload and realtime   Matthew Boehm
• [Asterisk-Dev] Dev Conf 2pm CST   Matthew Boehm
• [Asterisk-Dev] splitting users/peers?   Matthew Boehm
• [Asterisk-Dev] Dev Conf 2pm CST   Matthew Boehm
• [Asterisk-Dev] Developer with asterisk experience wanted forfull-time position   Matthew Boehm
• [Asterisk-Dev] Wanted: ast_category_merge   Matthew Boehm
• [Asterisk-Dev] Wanted: ast_category_merge   Matthew Boehm
• [Asterisk-Dev] app_directory - Now with RealTime Inside!   Matthew Boehm
• [Asterisk-Dev] app_directory - Now with RealTime Inside!   Matthew Boehm
• [Asterisk-Dev] dev conf topic: better CDRs   Matthew Boehm
• [Asterisk-Dev] dev conf topic: better CDRs   Matthew Boehm
• [Asterisk-Dev] dev conf topic: better CDRs   Matthew Boehm
• [Asterisk-Dev] dev conf topic: better CDRs   Matthew Boehm
• [Asterisk-Dev] Re: asterisk/doc README.realtime,NONE,1.1   Matthew Boehm
• [Asterisk-Dev] Re: asterisk/doc README.realtime,NONE,1.1   Matthew Boehm
• [Asterisk-Dev] Refined Voice CallerID Announce & Partial Apologies to Steven Critchfield ... Still an ass   Greg Boehnlein
• [Asterisk-Dev] Asterisk Memory Leak   Russell Bryant
• [Asterisk-Dev] Newbie   Russell Bryant
• [Asterisk-Dev] [Asterisk-Users] Asterisk 1.0.6   Russell Bryant
• [Asterisk-Dev] UltraSparc hardware, Linux and X100P REVISITED   Robert Burcham
• [Asterisk-Dev] UltraSparc hardware, Linux and X100P REVISITED   Robert Burcham
• [Asterisk-Dev] $1500 Bounty for Dictation Module   Ray Burkholder
• *SPAM*[Asterisk-Dev] $1500 Bounty for Dictation Module   Ray Burkholder
• [Asterisk-Dev] Wildcard TE410P Question   Ray Burkholder
• [Asterisk-Dev] TE410P B-channel restart   CW_ASN
• [Asterisk-Dev] TE410P B-channel restart   CW_ASN
• [Asterisk-Dev] Revised Call Queue Addition Ideas/Request   CW_ASN
• [Asterisk-Dev] Revised Call Queue Addition Ideas/Request   CW_ASN
• [Asterisk-Dev] TE410P B-channel restart   CW_ASN
• [Asterisk-Dev] Revised Call Queue Addition Ideas/Request   CW_ASN
• [Asterisk-Dev] Going CrAzY Call Queue Addition Help   CW_ASN
• [Asterisk-Dev] Going CrAzY Call Queue Addition Help   CW_ASN
• [Asterisk-Dev] [Asterisk-Users] How to monitor Agen Voice channal?   CW_ASN
• [Asterisk-Dev] Wildcard TE410P Question   Paul Cadach
• [Asterisk-Dev] help needed implementing H324M (UMTS Video)   Paul Cadach
• [Asterisk-Dev] CDRs with ms and not just seconds   Paul Cadach
• [Asterisk-Dev] TE410P init problem   Paul Cadach
• [Asterisk-Dev] TE410P init problem   Paul Cadach
• [Asterisk-Dev] I'm writing chan_gsm   Paul Cadach
• [Asterisk-Dev] dev conf topic: better CDRs   Paul Cadach
• [Asterisk-Dev] Re: rfc2833 DTMFs sent with bad timestamps (patch)   Paul Cadach
• [Asterisk-Dev] Silence suppression in asterisk/chan_sip?   Paul Cadach
• [Asterisk-Dev] New jitterbuffer and Packet Loss Concealment preview/prototype patch available in tracker.   Brian Capouch
• [Asterisk-Dev] Dev Conference   Brian Capouch
• [Asterisk-Dev] Dev Conference   Brian Capouch
• [Asterisk-Dev] Refined Voice CallerID Announce & Partial Apologies to Steven Critchfield ... Still an ass   Brian Capouch
• [Asterisk-Dev] Dev Con: Time to drop the 2 in IAX2?   Brian Capouch
• [Asterisk-Dev] Channel module question - setstate vs queue_control   Bob Carlson
• [Asterisk-Dev] Asterisk Compile Problem on Red Hat 9   Bob Carlson
• [Asterisk-Dev] Newbie   Daniel del Castillo
• [Asterisk-Dev] [Asterisk-Users] VXML and *   Daniel del Castillo
• [Asterisk-Dev] Disabling "!"   Tzafrir Cohen
• [Asterisk-Dev] Transcription...services...client.   Brian Connelly
• [Asterisk-Dev] ENUM multiple records handling   Conroy, Lawrence (SMTP)
• [Asterisk-Dev] Wanted: ast_category_merge   Leif Madsen - Independent Asterisk Consultant
• [Asterisk-Dev] MS-ADPCM in format_wav?   Steven Critchfield
• [Asterisk-Dev] Re: MS-ADPCM in format_wav?   Steven Critchfield
• [Asterisk-Dev] behaviour of ast_writefile (file.c) and wav_rewrite (format_wav.c)   Steven Critchfield
• Attention Bug Marshal.... Was Re: [Asterisk-Dev] behaviour of ast_writefile (file.c) and wav_rewrite (format_wav.c)   Steven Critchfield
• *SPAM*[Asterisk-Dev] $1500 Bounty for Dictation Module   Steven Critchfield
• [Asterisk-Dev] Question about VoIP Solution   Steven Critchfield
• [Asterisk-Dev] Asterisk Memory Leak   Steven Critchfield
• [Asterisk-Dev] Question about X100P card   Steven Critchfield
• [Asterisk-Dev] Problem with my perl applications.   Steven Critchfield
• [Asterisk-Dev] calling problem in cvs verison on fedora core2   Steven Critchfield
• [Asterisk-Dev] Basically a C AGI app.   Steven Critchfield
• [Asterisk-Dev] Basically a C AGI app.   Steven Critchfield
• [Asterisk-Dev] Asterisk not accepting multiple SIP phone logins   Steven Critchfield
• [Asterisk-Dev] C AGI clarification   Steven Critchfield
• [Asterisk-Dev] ChangeLog file messed up in HEAD?   Steven Critchfield
• [Asterisk-Dev] problem in loading module with asterisk   Steven Critchfield
• [Asterisk-Dev] Refined Voice CallerID Announce   Steven Critchfield
• [Asterisk-Dev] Refined Voice CallerID Announce   Steven Critchfield
• [Asterisk-Dev] Refined Voice CallerID Announce   Steven Critchfield
• [Asterisk-Dev] variable sample period?   Steven Critchfield
• [Asterisk-Dev] ClueCon in Chicago June 8th to the 10th.   Steven Critchfield
• [Asterisk-Dev] Receive calls without be registered   Steven Critchfield
• [Asterisk-Dev] file descriptors per call?   Steven Critchfield
• [Asterisk-Dev] file descriptors per call?   Steven Critchfield
• [Asterisk-Dev] help needed implementing H324M (UMTS Video)   Klaus Darilion
• [Asterisk-Dev] help needed implementing H324M (UMTS Video)   Klaus Darilion
• [Asterisk-Dev] help needed implementing H324M (UMTS Video)   Klaus Darilion
• [Asterisk-Dev] help needed implementing H324M (UMTS Video)   Klaus Darilion
• [Asterisk-Dev] help needed implementing H324M (UMTS Video)   Klaus Darilion
• [Asterisk-Dev] ENUM multiple records handling   Klaus Darilion
• [Asterisk-Dev] ENUM multiple records handling   Klaus Darilion
• [Asterisk-Dev] dev conf topic: better CDRs   Derrick D. Daugherty
• [Asterisk-Dev] ExtensionState problems using Manager.conf API   Dewbank
• [Asterisk-Dev] Refined Voice CallerID Announce & Partial Apologiesto Steven Critchfield ... Still an ass   Tom Dickenson
• [Asterisk-Dev] Refined Voice CallerID Announce   Tom Dickenson
• [Asterisk-Dev] Re: calling one application from other application   Tom Dickenson
• [Asterisk-Dev] dev conf topic: better CDRs   Tom Dickenson
• [Asterisk-Dev] dev conf topic: better CDRs   Tom Dickenson
• [Asterisk-Dev] Openh323 question   Diseyi Diffa
• [Asterisk-Dev] rfc2833 DTMFs sent with bad timestamps (patch)   Frank van Dijk
• [Asterisk-Dev] Re: rfc2833 DTMFs sent with bad timestamps (patch)   Frank van Dijk
• [Asterisk-Dev] rfc2833 DTMFs sent with bad timestamps (patch)   Frank van Dijk
• [Asterisk-Dev] rfc2833 DTMFs sent with bad timestamps (patch)   Frank van Dijk
• [Asterisk-Dev] rfc2833 DTMFs sent with bad timestamps (patch)   Frank van Dijk
• [Asterisk-Dev] rfc2833 DTMFs sent with bad timestamps (patch)   Frank van Dijk
• [Asterisk-Dev] ASTERISK + Q931   Geniatakis Dimitrios
• [Asterisk-Dev] Refined Voice CallerID Announce   Shawn L. Djernes
• [Asterisk-Dev] Basically a C AGI app.   Christopher Dobbs
• [Asterisk-Dev] Basically a C AGI app.   Christopher Dobbs
• [Asterisk-Dev] Dev Conf 2pm CST   Christopher Dobbs
• [Asterisk-Dev] Developer with asterisk experience wanted for full-time position   Steve Dolloff
• [Asterisk-Dev] libiax2 OK for production?   Michael Van Donselaar
• [Asterisk-Dev] Wildcard TE410P Question   Richard Dutton
• [Asterisk-Dev] [aidan@ifax.com: Re: E&M Wink problems]   Aidan Van Dyk
• [Asterisk-Dev] Re: [aidan@ifax.com: Re: E&M Wink problems]   Aidan Van Dyk
• [Asterisk-Dev] Question about VoIP Solution   Egypt.com
• [Asterisk-Dev] ChangeLog file messed up in HEAD?   Bryan Elliot
• [Asterisk-Dev] ChangeLog file messed up in HEAD?   Bryan Elliot
• [Asterisk-Dev] rtp.c process_rfc2833 - why no sequence number checking?   Dan Evans
• [Asterisk-Dev] help needed implementing H324M (UMTS Video)   Nicolas FOURNIL
• [Asterisk-Dev] IntraFrame request on SIP channel ? New feature in chan_sip & app_voicemail ?   Nicolas FOURNIL
• [Asterisk-Dev] Question about VoIP Solution   Yousri Farouk
• [Asterisk-Dev] Question about VoIP Solution   Yousri Farouk
• [Asterisk-Dev] Question about X100P card   Yousri Farouk
• [Asterisk-Dev] RE: Asterisk-Dev Digest, Vol 7, Issue 58   Constantine Filin
• [Asterisk-Dev] Wildcard TE410P Question   Kevin P. Fleming
• [Asterisk-Dev] include the results of an executable file (*.conf)   Kevin P. Fleming
• [Asterisk-Dev] include the results of an executable file (*.conf)   Kevin P. Fleming
• [Asterisk-Dev] reload and realtime   Kevin P. Fleming
• [Asterisk-Dev] Asterisk Compile Problem on Red Hat 9   Kevin P. Fleming
• [Asterisk-Dev] Dev Conf 2pm CST   Kevin P. Fleming
• [Asterisk-Dev] splitting users/peers?   Kevin P. Fleming
• [Asterisk-Dev] chan_sip   Kevin P. Fleming
• [Asterisk-Dev] chan_sip   Kevin P. Fleming
• [Asterisk-Dev] app_queue.c when agent prematurely hangs up   Kevin P. Fleming
• [Asterisk-Dev] TCP/IP Gigabit Ethernet accel. question   Kevin P. Fleming
• [Asterisk-Dev] Out-of-tree channel maintainers: bug 3573 needs your attention   Kevin P. Fleming
• [Asterisk-Dev] Core Dumped on CVS-HEAD-02/09/05-13:12:07   Kevin P. Fleming
• [Asterisk-Dev] Audio delay in MeetMe using SIP when not 'q' mode   Kevin P. Fleming
• [Asterisk-Dev] Re: Audio delay in MeetMe using SIP when not 'q' mode   Kevin P. Fleming
• [Asterisk-Dev] SIP/IAX repacking cpu cost?   Kevin P. Fleming
• [Asterisk-Dev] SIP/IAX repacking cpu cost?   Kevin P. Fleming
• [Asterisk-Dev] 1.0.5, SIP, possible bug: "From:" field wrong in authentication?   Kevin P. Fleming
• [Asterisk-Dev] Wanted: ast_category_merge   Kevin P. Fleming
• [Asterisk-Dev] Wanted: ast_category_merge   Kevin P. Fleming
• [Asterisk-Dev] Potential bug in caller ID setting; comments requested   Kevin P. Fleming
• [Asterisk-Dev] Dev Conference   Kevin P. Fleming
• OT: Re: [Asterisk-Dev] Going CrAzY Call Queue Addition Help   Kevin P. Fleming
• OT: Re: [Asterisk-Dev] Going CrAzY Call Queue Addition Help   Kevin P. Fleming
• [Asterisk-Dev] Misbehaviour in ast_variable_browse   Kevin P. Fleming
• [Asterisk-Dev] gcc 2.96 compatibility   Kevin P. Fleming
• [Asterisk-Dev] dev conf topic: better CDRs   Kevin P. Fleming
• [Asterisk-Dev] RE: Asterisk-Dev Digest, Vol 7, Issue 58   Kevin P. Fleming
• [Asterisk-Dev] dev conf topic: better CDRs   Kevin P. Fleming
• [Asterisk-Dev] dev conf topic: better CDRs   Kevin P. Fleming
• [Asterisk-Dev] dev conf topic: better CDRs   Kevin P. Fleming
• [Asterisk-Dev] changing codec during call   Kevin P. Fleming
• [Asterisk-Dev] Keeping queue status between reloads?   Kevin P. Fleming
• [Asterisk-Dev] RTP not sending UDP checksums?   Kevin P. Fleming
• [Asterisk-Dev] file descriptors per call?   Kevin P. Fleming
• [Asterisk-Dev] setting up fromuser   Kevin P. Fleming
• [Asterisk-Dev] setting up fromuser   Kevin P. Fleming
• [Asterisk-Dev] RTP not sending UDP checksums?   Kevin P. Fleming
• [Asterisk-Dev] setting up fromuser   Kevin P. Fleming
• [Asterisk-Dev] Silence suppression in asterisk/chan_sip?   Kevin P. Fleming
• [Asterisk-Dev] Silence suppression in asterisk/chan_sip?   Kevin P. Fleming
• [Asterisk-Dev] RTP not sending UDP checksums?   Kevin P. Fleming
• [Asterisk-Dev] Disabling "!"   Alessio Focardi
• [Asterisk-Dev] Bristuff and Realtime   Alessio Focardi
• [Asterisk-Dev] Incoming calls in H323 always going to default context   Danny Froberg
• [Asterisk-Dev] (no subject)   Danny Froberg
• [Asterisk-Dev] libiax2   Hans Fugal
• [Asterisk-Dev] Error in Res make file   Claus Futtrup
• [Asterisk-Dev] Re: Error in Res make file   Claus Futtrup
• [Asterisk-Dev] Call Queue Addition to MOH   Jon Gabrielson
• [Asterisk-Dev] Developer with asterisk experience wanted forfull-time position   Preston Garrison
• [Asterisk-Dev] How to monitor Agen Voice channal?   Preston Garrison
• [Asterisk-Dev] Zaptel - "dacs" not quite ...   Gary
• [Asterisk-Dev] Codec = RAW ?   Gary
• [Asterisk-Dev] Codec = RAW ?   Gary
• [Asterisk-Dev] Codec = RAW ?   Gary
• [Asterisk-Dev] Asterisk Compilation using ARM GCC   Geetha
• [Asterisk-Dev] Asterisk Compilation using ARM GCC   Geetha
• [Asterisk-Dev] Asterisk Compilation using ARM GCC   Geetha
• [Asterisk-Dev] .call-file handling suggestion   Michael Giagnocavo
• [Asterisk-Dev] Basically a C AGI app.   Michael Giagnocavo
• [Asterisk-Dev] Basically a C AGI app.   Michael Giagnocavo
• [Asterisk-Dev] Basically a C AGI app.   Michael Giagnocavo
• [Asterisk-Dev] Basically a C AGI app.   Michael Giagnocavo
• [Asterisk-Dev] Basically a C AGI app.   Michael Giagnocavo
• [Asterisk-Dev] dev conf topic: better CDRs   Michael Giagnocavo
• [Asterisk-Dev] dev conf topic: better CDRs   Michael Giagnocavo
• [Asterisk-Dev] changing codec during call   Michael Giagnocavo
• [Asterisk-Dev] ENUM multiple records handling   Adam Goryachev
• [Asterisk-Dev] Refined Voice CallerID Announce   Adam Goryachev
• [Asterisk-Dev] Refined Voice CallerID Announce   Adam Goryachev
• [Asterisk-Dev] 1.0.5, SIP, possible bug: "From:" field wrong in authentication?   Andreas Greulich
• [Asterisk-Dev] Alcatel UA Protocol   Edwin Groothuis
• [Asterisk-Dev] ENUM multiple records handling   Edwin Groothuis
• [Asterisk-Dev] ENUM multiple records handling   Edwin Groothuis
• [Asterisk-Dev] ENUM multiple records handling   Edwin Groothuis
• [Asterisk-Dev] ENUM multiple records handling   Edwin Groothuis
• [Asterisk-Dev] ENUM multiple records handling   Edwin Groothuis
• [Asterisk-Dev] dev conf topic: better CDRs   Clint Guillot
• [Asterisk-Dev] Problem with my perl applications.   Anis Hachemi
• [Asterisk-Dev] Problem of the Hung up after 2 minutes   Anis Hachemi
• [Asterisk-Dev] Problem of the Hung up after 2 minutes   Anis Hachemi
• [Asterisk-Dev] app_queue.c when agent prematurely hangs up   Michael Haigh
• [Asterisk-Dev] Modification of app_dial to use long tone to terminate or transfer call   Trevor G. Hammonds
• [Asterisk-Dev] gcc 2.96 compatibility   Steve Hanselman
• [Asterisk-Dev] gcc 2.96 compatibility   Steve Hanselman
• [Asterisk-Dev] libiax2   Adam Hart
• [Asterisk-Dev] Dev Conference   Nicholas Hart
• [Asterisk-Dev] New answering machine detection app   Ben Hencke
• [Asterisk-Dev] Bounty IAX Fax Tone Detection   Lee Howard
• [Asterisk-Dev] Bounty IAX Fax Tone Detection   Lee Howard
• [Asterisk-Dev] Bounty IAX Fax Tone Detection   Lee Howard
• [Asterisk-Dev] H323 Trunking   Huddleston, Robert
• [Asterisk-Dev] API docs?   Robert Jackson
• [Asterisk-Dev] $1500 Bounty for Dictation Module   Bill Jennings
• *SPAM*[Asterisk-Dev] $1500 Bounty for Dictation Module   Jerris, Michael MI
• [Asterisk-Dev] API docs?   Jerris, Michael MI
• [Asterisk-Dev] API docs?   Jerris, Michael MI
• [Asterisk-Dev] can't enable trunking   Jerris, Michael MI
• [Asterisk-Dev] -user questions in -dev   Jerris, Michael MI
• [Asterisk-Dev] ENUM multiple records handling   Olle E. Johansson
• [Asterisk-Dev] chan_sip   Olle E. Johansson
• [Asterisk-Dev] Peer weekend improvements   Olle E. Johansson
• [Asterisk-Dev] Patch to test: Voicemail distribution lists   Olle E. Johansson
• [Asterisk-Dev] Voicemail and busy tone   Olle E. Johansson
• [Asterisk-Dev] how to bridge iaxtel calls to PSTN?   Olle E. Johansson
• [Asterisk-Dev] Re: asterisk/doc README.realtime,NONE,1.1   Olle E. Johansson
• [Asterisk-Dev] "click to dial extension number" functionality ?   Olle E. Johansson
• [Asterisk-Dev] setting up fromuser   Olle E. Johansson
• [Asterisk-Dev] Silence suppression in asterisk/chan_sip?   Olle E. Johansson
• [Asterisk-Dev] Asterisk not accepting multiple SIP phone logins   Juki
• [Asterisk-Dev] changing codec during call   Jesse Kaijen
• [Asterisk-Dev] changing codec during call   Jesse Kaijen
• [Asterisk-Dev] changing codec during call   Jesse Kaijen
• [Asterisk-Dev] changing codec during call   Jesse Kaijen
• [Asterisk-Dev] asterisk softphone source code   Steve Kann
• [Asterisk-Dev] .call-file handling suggestion   Steve Kann
• [Asterisk-Dev] IAX2 trunk really should send timestamps as part of iax2_meta_trunk_entry..   Steve Kann
• [Asterisk-Dev] IAX2 trunk really should send timestamps as part of iax2_meta_trunk_entry..   Steve Kann
• [Asterisk-Dev] IAX2 trunk really should send timestamps as part of iax2_meta_trunk_entry..   Steve Kann
• [Asterisk-Dev] IAX2 trunk really should send timestamps as part of iax2_meta_trunk_entry..   Steve Kann
• [Asterisk-Dev] IAX2 trunk really should send timestamps as part of iax2_meta_trunk_entry..   Steve Kann
• [Asterisk-Dev] Dev Conf 2pm CST   Steve Kann
• [Asterisk-Dev] New jitterbuffer and Packet Loss Concealment preview/prototype patch available in tracker.   Steve Kann
• [Asterisk-Dev] Codec = RAW ?   Steve Kann
• [Asterisk-Dev] SIP/IAX repacking cpu cost?   Steve Kann
• [Asterisk-Dev] new jitterbuffer in 1.2?   Steve Kann
• [Asterisk-Dev] Re: variable sample period?   Steve Kann
• [Asterisk-Dev] new jitterbuffer in 1.2?   Steve Kann
• [Asterisk-Dev] dev conf topic: better CDRs   Steve Kann
• [Asterisk-Dev] new jitterbuffer in 1.2?   Steve Kann
• [Asterisk-Dev] dev conf topic: better CDRs   Steve Kann
• [Asterisk-Dev] changing codec during call   Steve Kann
• [Asterisk-Dev] changing codec during call   Steve Kann
• [Asterisk-Dev] changing codec during call   Steve Kann
• [Asterisk-Dev] changing codec during call   Steve Kann
• [Asterisk-Dev] changing codec during call   Steve Kann
• [Asterisk-Dev] changing codec during call   Steve Kann
• [Asterisk-Dev] changing codec during call   Steve Kann
• [Asterisk-Dev] Caller ID Variable Change?   Nate Kapi
• [Asterisk-Dev] upper- and lowercase on variables?   Roy Sigurd Karlsbakk
• [Asterisk-Dev] SIP/IAX repacking cpu cost?   Roy Sigurd Karlsbakk
• [Asterisk-Dev] new jitterbuffer in 1.2?   Roy Sigurd Karlsbakk
• [Asterisk-Dev] variable sample period?   Roy Sigurd Karlsbakk
• [Asterisk-Dev] Keeping queue status between reloads?   Roy Sigurd Karlsbakk
• [Asterisk-Dev] file descriptors per call?   Roy Sigurd Karlsbakk
• [Asterisk-Dev] RTP not sending UDP checksums?   Roy Sigurd Karlsbakk
• [Asterisk-Dev] file descriptors per call?   Roy Sigurd Karlsbakk
• [Asterisk-Dev] Silence suppression in asterisk/chan_sip?   Roy Sigurd Karlsbakk
• [Asterisk-Dev] Silence suppression in asterisk/chan_sip?   Roy Sigurd Karlsbakk
• [Asterisk-Dev] RTP not sending UDP checksums?   Roy Sigurd Karlsbakk
• [Asterisk-Dev] Asterisk Compilation using ARM GCC   Vikramsinh Katkar
• [Asterisk-Dev] Dev Conf 2pm CST   Kristian Kielhofner
• [Asterisk-Dev] "click to dial extension number" functionality ?   Kristian Kielhofner
• [Asterisk-Dev] Channel module question - setstate vs queue_control   Matt Klein
• [Asterisk-Dev] Bounty IAX Fax Tone Detection   Andrew Kohlsmith
• [Asterisk-Dev] Bounty IAX Fax Tone Detection   Andrew Kohlsmith
• [Asterisk-Dev] IAX2 trunk really should send timestamps as part of iax2_meta_trunk_entry..   Andrew Kohlsmith
• [Asterisk-Dev] New jitterbuffer and Packet Loss Concealment preview/prototype patch available in tracker.   Andrew Kohlsmith
• [Asterisk-Dev] New jitterbuffer and Packet Loss Concealment preview/prototype patch available in tracker.   Andrew Kohlsmith
• [Asterisk-Dev] iax2 bridge optimization screwing with timestamps   Andrew Kohlsmith
• [Asterisk-Dev] iax2 bridge optimization screwing with timestamps   Andrew Kohlsmith
• [Asterisk-Dev] iax2 bridge optimization screwing with timestamps   Andrew Kohlsmith
• [Asterisk-Dev] SIP/IAX repacking cpu cost?   Andrew Kohlsmith
• [Asterisk-Dev] new jitterbuffer in 1.2?   Andrew Kohlsmith
• [Asterisk-Dev] new jitterbuffer in 1.2?   Andrew Kohlsmith
• [Asterisk-Dev] new jitterbuffer in 1.2?   Andrew Kohlsmith
• [Asterisk-Dev] Asterisk Compilation using ARM GCC   Andrew Kohlsmith
• [Asterisk-Dev] -user questions in -dev   Andrew Kohlsmith
• [Asterisk-Dev] how to design digim clone card?   Andrew Kohlsmith
• [Asterisk-Dev] Question about compiling a module for Asterisk with CPP stuff   Sergey Kuznetsov
• [Asterisk-Dev] RE: Asterisk-Dev Digest, Vol 7, Issue 58   Sergey Kuznetsov
• [Asterisk-Dev] RE: Asterisk-Dev Digest, Vol 7, Issue 58   Sergey Kuznetsov
• [Asterisk-Dev] Asterisk Dialplan command "PPPD" released   Tilghman Lesher
• [Asterisk-Dev] Channel module question - setstate vs queue_control   Tilghman Lesher
• [Asterisk-Dev] CDRs with ms and not just seconds   Tilghman Lesher
• [Asterisk-Dev] RE: RE: Basically a C AGI app.   Tilghman Lesher
• [Asterisk-Dev] question about agi handle_exec   Tilghman Lesher
• [Asterisk-Dev] TE410P B-channel restart   Tilghman Lesher
• [Asterisk-Dev] ClueCon in Chicago June 8th to the 10th.   Tilghman Lesher
• [Asterisk-Dev] dev conf topic: better CDRs   Tilghman Lesher
• [Asterisk-Dev] dev conf topic: better CDRs   Tilghman Lesher
• [Asterisk-Dev] dev conf topic: better CDRs   Tilghman Lesher
• [Asterisk-Dev] dev conf topic: better CDRs   Tilghman Lesher
• [Asterisk-Dev] Re: Audio delay in MeetMe using SIP when not 'q' mode   Andrew Lindh
• [Asterisk-Dev] RTP not sending UDP checksums?   Andrew Lindh
• [Asterisk-Dev] RTP not sending UDP checksums?   Andrew Lindh
• [Asterisk-Dev] Bounty IAX Fax Tone Detection   Simon Lockhart
• [Asterisk-Dev] RTP not sending UDP checksums?   Michael Loftis
• [Asterisk-Dev] Re: New Channel for alsa   Jaime Lopez
• [Asterisk-Dev] Asterisk registration requirements API   Sadie Louise
• [Asterisk-Dev] Asterisk registration requirements API   Sadie Louise
• [Asterisk-Dev] Asterisk registration requirements API   Sadie Louise
• [Asterisk-Dev] Receive calls without be registered   Sadie Louise
• [Asterisk-Dev] Receive calls without be registered   Sadie Louise
• [Asterisk-Dev] can't enable trunking   Muhammad Muzzamil Luqman
• OT: Re: [Asterisk-Dev] Going CrAzY Call Queue Addition Help   Richard Lyman
• [Asterisk-Dev] Refined Voice CallerID Announce   Richard Lyman
• [Asterisk-Dev] API docs?   Leif Madsen
• [Asterisk-Dev] Core Dumped on CVS-HEAD-02/09/05-13:12:07   Diego Magalhaes
• [Asterisk-Dev] Thoughts on call progress detection   C. Maj
• [Asterisk-Dev] ASTERISK + Q931   C. Maj
• [Asterisk-Dev] dev conf topic: better CDRs   C. Maj
• [Asterisk-Dev] dev conf topic: better CDRs   C. Maj
• [Asterisk-Dev] dev conf topic: better CDRs   C. Maj
• [Asterisk-Dev] h323 and oh323 codec support   Michael Manousos
• [Asterisk-Dev] h323 and oh323 codec support   Michael Manousos
• [Asterisk-Dev] oh323 invalid array index   Michael Manousos
• [Asterisk-Dev] Temporary Help Needed   Christopher McBee
• [Asterisk-Dev] Zultys Paging - Found Solution!   Mathew McKernan
• [Asterisk-Dev] Call Queue Addition to MOH   Steve McMahon
• [Asterisk-Dev] Call Queue Addition to MOH   Steve McMahon
• [Asterisk-Dev] Revised Call Queue Addition Ideas/Request   Steve McMahon
• [Asterisk-Dev] Revised Call Queue Addition Ideas/Request   Steve McMahon
• [Asterisk-Dev] Revised Call Queue Addition Ideas/Request   Steve McMahon
• [Asterisk-Dev] Going CrAzY Call Queue Addition Help   Steve McMahon
• OT: Re: [Asterisk-Dev] Going CrAzY Call Queue Addition Help   Steve McMahon
• [Asterisk-Dev] Re: New Channel for alsa   Steve McMahon
• [Asterisk-Dev] Voice CallerID Over Phone before pickup feature request   Steve McMahon
• [Asterisk-Dev] Refined Voice CallerID Announce   Steve McMahon
• [Asterisk-Dev] Refined Voice CallerID Announce   Steve McMahon
• [Asterisk-Dev] Refined Voice CallerID Announce & Partial Apologies to Steven Critchfield ... Still an ass   Steve McMahon
• [Asterisk-Dev] stable chan_skinny crash + fix   Jeremy McNamara
• [Asterisk-Dev] Question about compiling a module for Asterisk with CPP stuff   Jeremy McNamara
• [Asterisk-Dev] Openh323 question   Jeremy McNamara
• [Asterisk-Dev] Asterisk compliant JAIN API?   Jeremy McNamara
• [Asterisk-Dev] h323 and oh323 codec support   Jeremy McNamara
• [Asterisk-Dev] gcc 2.96 compatibility   Jeremy McNamara
• [Asterisk-Dev] gcc 2.96 compatibility   Jeremy McNamara
• [Asterisk-Dev] Re: calling one application from other application   Jeremy McNamara
• [Asterisk-Dev] rfc2833 DTMFs sent with bad timestamps (patch)   Jeremy McNamara
• [Asterisk-Dev] Question about VoIP Solution   Jim Van Meggelen
• [Asterisk-Dev] Asterisk IAX registration refresh   Liaan vd Merwe
• [Asterisk-Dev] IAX2 definitions   Liaan vd Merwe
• [Asterisk-Dev] dev conf topic: better CDRs   James Middleton
• [Asterisk-Dev] dev conf topic: better CDRs   James Middleton
• [Asterisk-Dev] dev conf topic: better CDRs   James Middleton
• [Asterisk-Dev] CDRs with ms and not just seconds   Kai Militzer
• [Asterisk-Dev] New feature patch for SIP Receiving Distinctive Ring bug: 3608   Ben Miller
• [Asterisk-Dev] New feature patch for SIP Receiving Distinctive Ringbug: 3608   Ben Miller
• [Asterisk-Dev] Question about compiling a module for Asterisk with CPP stuff   Leo Moll
• [Asterisk-Dev] SOLVED! Question about compiling a module for Asterisk with CPP stuff   Leo Moll
• [Asterisk-Dev] MS-ADPCM in format_wav?   Tony Mountifield
• [Asterisk-Dev] Re: MS-ADPCM in format_wav?   Tony Mountifield
• [Asterisk-Dev] Audio delay in MeetMe using SIP when not 'q' mode   Tony Mountifield
• [Asterisk-Dev] Re: Audio delay in MeetMe using SIP when not 'q' mode   Tony Mountifield
• [Asterisk-Dev] Re: Audio delay in MeetMe using SIP when not 'q' mode   Tony Mountifield
• [Asterisk-Dev] Re: Audio delay in MeetMe using SIP when not 'q' mode   Tony Mountifield
• [Asterisk-Dev] Re: variable sample period?   Tony Mountifield
• [Asterisk-Dev] Re: variable sample period?   Tony Mountifield
• [Asterisk-Dev] Re: Error in Res make file   Tony Mountifield
• [Asterisk-Dev] "click to dial extension number" functionality ?   Terje Myhre
• [Asterisk-Dev] Deadlock ? app_queue / chan_agent   NRB
• [Asterisk-Dev] Deadlock ? app_queue / chan_agent   NRB
• [Asterisk-Dev] Deadlock ? app_queue / chan_agent   NRB
• [Asterisk-Dev] Re: h323 and oh323 codec support   Peter Nixon
• [Asterisk-Dev] Re: difference of ast_channel struct in cvs and stable   Peter Nixon
• [Asterisk-Dev] Re: Asterisk registration requirements API   Peter Nixon
• [Asterisk-Dev] dev conf topic: better CDRs   Chris Parker
• [Asterisk-Dev] Core Dumped on CVS-HEAD-02/09/05-13:12:07   Trevor Peirce
• [Asterisk-Dev] dev conf topic: better CDRs   Trevor Peirce
• [Asterisk-Dev] calling one application from other application   Wolfgang Pichler
• [Asterisk-Dev] h323-channel dynamic endpoint registration   Michael Platov
• [Asterisk-Dev] h323-channel dynamic endpoint registration   Michael Platov
• [Asterisk-Dev] TE410P init problem   Daniel Pocock
• [Asterisk-Dev] TE410P init problem   Daniel Pocock
• [Asterisk-Dev] oh323 invalid array index   Daniel Pocock
• [Asterisk-Dev] h323 and oh323 codec support   Daniel Pocock
• [Asterisk-Dev] FXO line in use detection   Glenn Powers
• [Asterisk-Dev] Helps needed for Tethereal and sip   Andrew Pyles
• [Asterisk-Dev] Strcit Routing vs Loose Routing   Chuck Ramirez
• [Asterisk-Dev] RE: Bounty IAX Fax Tone Detection   Matt Riddell
• [Asterisk-Dev] chan_sip goes "deaf" - SIP monitor thread stops   Matt Riddell
• [Asterisk-Dev] Pocket pc wireless internet applications   Robert Kim, Wireless Internet Consultant
• [Asterisk-Dev] app_directory - Now with RealTime Inside!   Robert Kim, Wireless Internet / Wifi Hotspot Advisor
• [Asterisk-Dev] bugs #2700 and #2721 (chan_agent transfers)   Andreas Roedl
• [Asterisk-Dev] wctdm driver broken?   Fernando Romo
• [Asterisk-Dev] Making a "whispering" mode in meetme   Fernando Romo
• [Asterisk-Dev] Call Queue Addition to MOH   Brian Roy
• [Asterisk-Dev] TCP/IP Gigabit Ethernet accel. question   Wolfgang S. Rupprecht
• [Asterisk-Dev] asterisk softphone source code   Mouhamed Mahi SY
• [Asterisk-Dev] Asterisk Dialplan command "PPPD" released   Oskar Senft
• [Asterisk-Dev] Asterisk Dialplan command "PPPD" released   Oskar Senft
• [Asterisk-Dev] ENUM multiple records handling   Andreas Sikkema
• [Asterisk-Dev] Refined Voice CallerID Announce   Andreas Sikkema
• [Asterisk-Dev] RTP not sending UDP checksums?   Andreas Sikkema
• [Asterisk-Dev] Alcatel UA Protocol   Thomas Sillaber
• [Asterisk-Dev] RE: Bounty IAX Fax Tone Detection   Matthew Simpson
• [Asterisk-Dev] variable sample period?   Jared Smith
• [Asterisk-Dev] rfc2833 DTMFs sent with bad timestamps (patch)   Jared Smith
• [Asterisk-Dev] Re: MS-ADPCM in format_wav?   Roger Smith
• [Asterisk-Dev] Openh323 question   Craig Southeren
• [Asterisk-Dev] Unified Parsing Stuff   Mark Spencer
• [Asterisk-Dev] Manual Bridge Application   Robert Spielmann
• [Asterisk-Dev] Manual Bridge Application   Robert Spielmann
• [Asterisk-Dev] API docs?   Robert Spielmann
• [Asterisk-Dev] API docs?   Robert Spielmann
• [Asterisk-Dev] Basically a C AGI app.   Robert Spielmann
• [Asterisk-Dev] Disabling "!"   Robert Spielmann
• [Asterisk-Dev] TE410P B-channel restart   Peter Svensson
• [Asterisk-Dev] Call Queue Addition to MOH   Peter Svensson
• [Asterisk-Dev] TE410P B-channel restart   Peter Svensson
• [Asterisk-Dev] variable sample period?   Peter Svensson
• [Asterisk-Dev] stable chan_skinny crash + fix   Vincent Sweeney
• [Asterisk-Dev] Asterisk Compilation using ARM GCC   Mike Taht
• [Asterisk-Dev] new jitterbuffer in 1.2?   Mike Taht
• [Asterisk-Dev] new jitterbuffer in 1.2?   Mike Taht
• [Asterisk-Dev] Thoughts on call progress detection   Samuel Tardieu
• [Asterisk-Dev] Re: Thoughts on call progress detection   Samuel Tardieu
• [Asterisk-Dev] Re: IAX2 trunk really should send timestamps as part of iax2_meta_trunk_entry..   Samuel Tardieu
• [Asterisk-Dev] [Asterisk-Users] How to monitor Agen Voice channal?   Aram Ter-Martirosyan
• [Asterisk-Dev] IAX2 trunk really should send timestamps as part of iax2_meta_trunk_entry..   Andrew Thompson
• [Asterisk-Dev] include the results of an executable file (*.conf)   Andrew Thompson
• [Asterisk-Dev] include the results of an executable file (*.conf)   Andrew Thompson
• [Asterisk-Dev] include the results of an executable file (*.conf)   Andrew Thompson
• [Asterisk-Dev] potential buffer overflow (minor problem)   Todd
• [Asterisk-Dev] question about agi handle_exec   Todd
• [Asterisk-Dev] TCP/IP Gigabit Ethernet accel. question   John Todd
• [Asterisk-Dev] TCP/IP Gigabit Ethernet accel. question   John Todd
• [Asterisk-Dev] Potential bug in caller ID setting; comments requested   John Todd
• [Asterisk-Dev] RE: Bounty IAX Fax Tone Detection   Steve Underwood
• [Asterisk-Dev] Wildcard TE410P Question   Steve Underwood
• [Asterisk-Dev] IAX2 trunk really should send timestamps as part of iax2_meta_trunk_entry..   Steve Underwood
• [Asterisk-Dev] IAX2 trunk really should send timestamps as part of iax2_meta_trunk_entry..   Steve Underwood
• [Asterisk-Dev] wctdm driver broken?   Steve Underwood
• [Asterisk-Dev] help needed implementing H324M (UMTS Video)   Steve Underwood
• [Asterisk-Dev] changing codec during call   Steve Underwood
• [Asterisk-Dev] changing codec during call   Steve Underwood
• [Asterisk-Dev] changing codec during call   Steve Underwood
• [Asterisk-Dev] Voice CallerID Over Phone before pickup featurerequest   Joel Vandal
• [Asterisk-Dev] Incoming calls in H323 always going to defaultcontext   Race Vanderdecken
• [Asterisk-Dev] Debugging: selectable feature?   Race Vanderdecken
• [Asterisk-Dev] Receive calls without be registered   Race Vanderdecken
• [Asterisk-Dev] RE: Asterisk-Dev Digest, Vol 7, Issue 58   Race Vanderdecken
• [Asterisk-Dev] dev conf topic: better CDRs   Race Vanderdecken
• [Asterisk-Dev] dev conf topic: better CDRs   Race Vanderdecken
• [Asterisk-Dev] dev conf topic: better CDRs   Race Vanderdecken
• [Asterisk-Dev] dev conf topic: better CDRs   Race Vanderdecken
• [Asterisk-Dev] changing codec during call   Race Vanderdecken
• [Asterisk-Dev] changing codec during call   Race Vanderdecken
• [Asterisk-Dev] changing codec during call   Race Vanderdecken
• [Asterisk-Dev] Helps needed for Tethereal and sip   Race Vanderdecken
• [Asterisk-Dev] Misbehaviour in ast_variable_browse   Luis Vazquez
• [Asterisk-Dev] Misbehaviour in ast_variable_browse   Luis Vazquez
• [Asterisk-Dev] Misbehaviour in ast_variable_browse   Luis Vazquez
• [Asterisk-Dev] Misbehaviour in ast_variable_browse   Luis Vazquez
• [Asterisk-Dev] Misbehaviour in ast_variable_browse   Luis Vazquez
• [Asterisk-Dev] Modification of app_dial to use long tone to terminate or transfer call   Chris Wade
• [Asterisk-Dev] dev conf topic: better CDRs   Chris Wade
• [Asterisk-Dev] dev conf topic: better CDRs   Chris Wade
• [Asterisk-Dev] dev conf topic: better CDRs   Chris Wade
• [Asterisk-Dev] Dev Conference   Chris Wade
• [Asterisk-Dev] RE: RE: Basically a C AGI app.   Christopher L. Wade
• [Asterisk-Dev] Message: We hit our IOCTL   Robert Webb
• [Asterisk-Dev] Manual Bridge Application   Brian West
• [Asterisk-Dev] $1500 Bounty for Dictation Module   Brian West
• [Asterisk-Dev] $1500 Bounty for Dictation Module   Brian West
• [Asterisk-Dev] MS-ADPCM in format_wav?   Brian West
• [Asterisk-Dev] $1500 Bounty for Dictation Module   Brian West
• [Asterisk-Dev] Dev Conf 2pm CST   Brian West
• [Asterisk-Dev] Dev Conf 2pm CST   Brian West
• [Asterisk-Dev] Dev Conf 2pm CST   Brian West
• [Asterisk-Dev] Problem with too many files open   Brian West
• [Asterisk-Dev] Problem of the Hung up after 2 minutes   Brian West
• [Asterisk-Dev] Dev Conf 2PM CST, FEB 17th   Brian West
• [Asterisk-Dev] Dev Conference   Brian West
• [Asterisk-Dev] Dev Conference   Brian West
• [Asterisk-Dev] Misbehaviour in ast_variable_browse   Brian West
• [Asterisk-Dev] calling stored procedure   Brian West
• [Asterisk-Dev] Misbehaviour in ast_variable_browse   Brian West
• [Asterisk-Dev] UltraSparc hardware, Linux and X100P REVISITED   Brian West
• [Asterisk-Dev] ClueCon in Chicago June 8th to the 10th.   Brian West
• [Asterisk-Dev] variable sample period?   Brian West
• [Asterisk-Dev] Re: calling one application from other application   Brian West
• [Asterisk-Dev] ClueCon in Chicago June 8th to the 10th.   Brian West
• [Asterisk-Dev] ClueCon in Chicago June 8th to the 10th.   Brian West
• [Asterisk-Dev] Speaking in Chicago   Brian West
• [Asterisk-Dev] Re: rfc2833 DTMFs sent with bad timestamps (patch)   Brian West
• [Asterisk-Dev] rfc2833 DTMFs sent with bad timestamps (patch)   Brian West
• [Asterisk-Dev] rfc2833 DTMFs sent with bad timestamps (patch)   Brian West
• [Asterisk-Dev] rfc2833 DTMFs sent with bad timestamps (patch)   Brian West
• [Asterisk-Dev] rfc2833 DTMFs sent with bad timestamps (patch)   Brian West
• [Asterisk-Dev] rfc2833 DTMFs sent with bad timestamps (patch)   Brian West
• [Asterisk-Dev] rfc2833 DTMFs sent with bad timestamps (patch)   Brian West
• [Asterisk-Dev] rfc2833 DTMFs sent with bad timestamps (patch)   Brian West
• [Asterisk-Dev] .call-file handling suggestion   Eric Wieling
• [Asterisk-Dev] .call-file handling suggestion   Eric Wieling
• [Asterisk-Dev] DTMF inter-digit delay   Eric Wieling
• [Asterisk-Dev] Bounty IAX Fax Tone Detection   Gene Willingham
• [Asterisk-Dev] RE: Bounty IAX Fax Tone Detection   Gene Willingham
• [Asterisk-Dev] RE: Bounty IAX Fax Tone Detection   Gene Willingham
• [Asterisk-Dev] RE: RE: Bounty IAX Fax Tone Detection (Matt Riddell   Gene Willingham
• [Asterisk-Dev] RE: Bounty IAX Fax Tone Detection   Gene Willingham
• [Asterisk-Dev] Dynamic Gain Control   Jamie Yukes
• [Asterisk-Dev] TE410P B-channel restart   Ma Zhiyong
• [Asterisk-Dev] 404 Not Found and illegal SIP URI?   Dan Zhou
• [Asterisk-Dev] Bounty IAX Fax Tone Detection   alex at pilosoft.com
• [Asterisk-Dev] Bounty IAX Fax Tone Detection   alex at pilosoft.com
• [Asterisk-Dev] RE: Bounty IAX Fax Tone Detection   alex at pilosoft.com
• [Asterisk-Dev] TCP/IP Gigabit Ethernet accel. question   alex at pilosoft.com
• [Asterisk-Dev] TCP/IP Gigabit Ethernet accel. question   alex at pilosoft.com
• [Asterisk-Dev] Asterisk registration requirements API   alex at pilosoft.com
• [Asterisk-Dev] Asterisk registration requirements API   alex at pilosoft.com
• [Asterisk-Dev] changing codec during call   alex at pilosoft.com
• [Asterisk-Dev] Message: We hit our IOCTL   creslin at digium.com
• [Asterisk-Dev] Wildcard TE410P Question   creslin at digium.com
• [Asterisk-Dev] Question about compiling a module for Asterisk with CPP stuff   creslin at digium.com
• [Asterisk-Dev] Dev Conf 2pm CST   creslin at digium.com
• [Asterisk-Dev] Codec = RAW ?   creslin at digium.com
• [Asterisk-Dev] how to design digim clone card?   dev2003 at mail.ustc.edu.cn
• [Asterisk-Dev] UltraSparc hardware, Linux and X100P REVISITED   dking at pimpsoft.com
• [Asterisk-Dev] AMP set httpd to asterisk user   ePonkFiria
• 1 [Asterisk-Dev] Asterisk - libunicall - MFCr2 *** AGI application ***   kaws elchamal
• [Asterisk-Dev] chan_sip   harry gaillac
• [Asterisk-Dev] chan_sip   harry gaillac
• [Asterisk-Dev] chan_sip   harry gaillac
• [Asterisk-Dev] chan_sip   harry gaillac
• [Asterisk-Dev] Openh323 question   giancarlo garavaglia
• [Asterisk-Dev] what is the masquerade in Asterisk ???   oleg gibayev
• [Asterisk-Dev] Web based IAX softphone   ht at phonitel.com
• [Asterisk-Dev] Incoming calls in H323 always going to default context   ht at phonitel.com
• [Asterisk-Dev] (no subject)   ht at phonitel.com
• [Asterisk-Dev] how to bridge iaxtel calls to PSTN?   info at cybernergies.com
• [Asterisk-Dev] Modification of app_dial to use long tone to terminate or transfer call   izo
• [Asterisk-Dev] rfc2833 DTMFs sent with bad timestamps (patch)   lconroy
• [Asterisk-Dev] Request for feedback: Overriding codec in dialplan   niels at wxn.nl
• [Asterisk-Dev] How to Set up an voip infrastructure using Open source   phu at bevertec.com
• [Asterisk-Dev] Asterisk compliant JAIN API?   phu at bevertec.com
• [Asterisk-Dev] Helps needed for Tethereal and sip   phu at bevertec.com
• [Asterisk-Dev] Problem with too many files open   roberto
• [Asterisk-Dev] Problem with too many files open   roberto
• [Asterisk-Dev] CCBS / CCNR integration into asterisk   roman.sidler
• [Asterisk-Dev] Asterisk - failover from g729 to gsm capable?   rsenykoff at harrislogic.com
• [Asterisk-Dev] New jitterbuffer and Packet Loss Concealment preview/prototype patch available in tracker.   rsenykoff at harrislogic.com
• [Asterisk-Dev] New jitterbuffer and Packet Loss Concealment preview/prototype patch available in tracker.   rsenykoff at harrislogic.com
• [Asterisk-Dev] new jitterbuffer in 1.2?   rsenykoff at harrislogic.com
• [Asterisk-Dev] new jitterbuffer in 1.2?   rsenykoff at harrislogic.com
• [Asterisk-Dev] DTMF bugs in Asterisk ?   scm-j at nuntius.com
• [Asterisk-Dev] IAX2 trunk really should send timestamps as part of iax2_meta_trunk_entry..   steve at daviesfam.org
• [Asterisk-Dev] IAX2 trunk really should send timestamps as part of iax2_meta_trunk_entry..   steve at daviesfam.org
• [Asterisk-Dev] IAX2 trunk really should send timestamps as part of iax2_meta_trunk_entry..   steve at daviesfam.org
• [Asterisk-Dev] IAX2 trunk really should send timestamps as part of iax2_meta_trunk_entry..   steve at daviesfam.org
• [Asterisk-Dev] Re: IAX2 trunk really should send timestamps as part of iax2_meta_trunk_entry..   steve at daviesfam.org
• [Asterisk-Dev] chan_sip goes "deaf" - SIP monitor thread stops   steve at daviesfam.org
• [Asterisk-Dev] Basically a C AGI app.   taintedham-mailinglists at yahoo.com
• [Asterisk-Dev] RE: RE: Basically a C AGI app.   taintedham-mailinglists at yahoo.com
• [Asterisk-Dev] C AGI clarification   taintedham-mailinglists at yahoo.com
• [Asterisk-Dev] Web based IAX softphone   timebandit001 at gmail.com
• [Asterisk-Dev] Voicemail and busy tone   tr
• [Asterisk-Dev] setting up fromuser   usman at user.iphonica.net
• [Asterisk-Dev] setting up fromuser   usman at user.iphonica.net
• [Asterisk-Dev] setting up fromuser   usman at user.iphonica.net
• [Asterisk-Dev] Asterisk Compile Problem on Red Hat 9   vdasilva

Last message date: Mon Feb 28 20:40:21 MST 2005
Archived on: Tue Sep 5 14:27:16 MST 2006

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