December 2005 Archives by date

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Starting: Thu Dec 1 00:09:58 MST 2005
Ending: Sat Dec 31 21:47:59 MST 2005
Messages: 491

• [Asterisk-Dev] asterisk 1.2 g729 compile errors   Diyanat Ali
• [Asterisk-Dev] check availability for several SIP and Zap channels by order   Guo-Wei Chiuan
• [Asterisk-Dev] check availability for several SIP and Zap channels by order   Kaloyan Kovachev
• [Asterisk-Dev] app_conference errors   Tzafrir Cohen
• [Asterisk-Dev] Help me out   Muhammad Asim Sajjad
• [Asterisk-Dev] Help me out   Giovanni Miano
• [Asterisk-Dev] Help me out   Muhammad Asim Sajjad
• [Asterisk-Dev] Help me out   Giovanni Miano
• [Asterisk-Dev] Help me out   Muhammad Asim Sajjad
• [Asterisk-Dev] app_conference errors   Diyanat Ali
• [Asterisk-Dev] Help me out   Steven
• [Asterisk-Dev] app_conference errors   Jerris, Michael MI
• [Asterisk-Dev] 1.2.0 Manager Action: Agents bug?   alan
• [Asterisk-Dev] AST_FLAG_DEFER_DTMF (would like dialogic-r4like semantics)   Greg Lim
• [Asterisk-Dev] AST_FLAG_DEFER_DTMF (would like dialogic-r4like semantics)   Steven Critchfield
• [Asterisk-Dev] AST_FLAG_DEFER_DTMF (would like dialogic-r4like semantics)   Kevin P. Fleming
• [Asterisk-Dev] AST_FLAG_DEFER_DTMF (would like dialogic-r4like semantics)   Tilghman Lesher
• [Asterisk-Dev] meetme enhancements to improve efficiency   Geoff Karl
• [Asterisk-Dev] meetme enhancements to improve efficiency   Kevin P. Fleming
• [Asterisk-Dev] meetme enhancements to improve efficiency   Geoff Karl
• [Asterisk-Dev] meetme enhancements to improve efficiency   SteveK
• [Asterisk-Dev] meetme enhancements to improve efficiency   Geoff Karl
• [Asterisk-Dev] meetme enhancements to improve efficiency   SteveK
• [Asterisk-Dev] Very Weird problem with MeetMe, SIP, Zap and the combo of the three   Nir Simionovich - CTO
• [Asterisk-Dev] meetme enhancements to improve efficiency   Jan Saell
• [Asterisk-Dev] SIP handling of Contact header with new port   Ed Greenberg
• [Asterisk-Dev] meetme enhancements to improve efficiency   Jeremy McNamara
• [Asterisk-Dev] meetme enhancements to improve efficiency   Darren Wiebe
• [Asterisk-Dev] New Jersey ATT Vocie T1 Asterisk Toll free not working   Charles Huang
• [Asterisk-Dev] meetme enhancements to improve efficiency   Greg Boehnlein
• [Asterisk-Dev] New Jersey ATT Vocie T1 Asterisk Toll free not working   BJ Weschke
• [Asterisk-Dev] meetme enhancements to improve efficiency   Dan Austin
• [Asterisk-Dev] SIP handling of Contact header with new port   Olle E. Johansson
• [Asterisk-Dev] meetme enhancements to improve efficiency   Olle E. Johansson
• [Asterisk-Dev] meetme enhancements to improve efficiency   Olle E. Johansson
• [Asterisk-Dev] meetme enhancements to improve efficiency   Geoff Karl
• [Asterisk-Dev] meetme enhancements to improve efficiency   Greg Boehnlein
• [Asterisk-Dev] meetme enhancements to improve efficiency   Kevin P. Fleming
• [Asterisk-Dev] Re: meetme enhancements to improve efficiency   Aidan Van Dyk
• [Asterisk-Dev] Branching/Merging page updated   Kevin P. Fleming
• [Asterisk-Dev] meetme enhancements to improve efficiency   Steve Kann
• [Asterisk-Dev] meetme enhancements to improve efficiency   Kevin P. Fleming
• [Asterisk-Dev] meetme enhancements to improve efficiency   Steve Kann
• [Asterisk-Dev] Adding new codec - should I write a new module?   Raúl
• [Asterisk-Dev] Adding new codec - should I write a new module?   Kevin P. Fleming
• [Asterisk-Dev] Adding new codec - should I write a new module?   Steve Kann
• [Asterisk-Dev] meetme enhancements to improve efficiency   Kevin P. Fleming
• [Asterisk-Dev] Adding new codec - should I write a new module?   Raúl
• [Asterisk-Dev] meetme enhancements to improve efficiency   Steve Kann
• [Asterisk-Dev] meetme enhancements to improve efficiency   Kevin P. Fleming
• [Asterisk-Dev] Re: Asterisk-Dev Digest, Vol 17, Issue 6   Tran Tony
• [Asterisk-Dev] CVS done?   asterisk at ntplx.net
• [Asterisk-Dev] CVS done?   Kevin P. Fleming
• [Asterisk-Dev] meetme enhancements to improve efficiency   David Pollak
• [Asterisk-Dev] asterisk 1.2 g729 compile errors   Celedonio Albarran
• [Asterisk-Dev] Codec comparisons   Hans Fugal
• [Asterisk-Dev] Codec comparisons   Kevin P. Fleming
• [Asterisk-Dev] DeadAGI problem   Abdul Lateef Khan
• [Asterisk-Dev] DeadAGI problem   Michiel van Baak
• [Asterisk-Dev] DeadAGI problem   Abdul Lateef Khan
• [Asterisk-Dev] Re: DeadAGI problem   Tony Mountifield
• [Asterisk-Dev] Re: DeadAGI problem   Michiel van Baak
• [Asterisk-Dev] DeadAGI problem   Abdul Lateef Khan
• [Asterisk-Dev] DeadAGI problem   Abdul Lateef Khan
• [Asterisk-Dev] DeadAGI problem   Michiel van Baak
• [Asterisk-Dev] DeadAGI problem   Abdul Lateef Khan
• [Asterisk-Dev] DeadAGI problem   Andrew Kohlsmith
• [Asterisk-Dev] No CID Info an TE405P with zaptel 1.2.0   BK
• [Asterisk-Dev] Verbose? Debug?   Andrew Latham
• [Asterisk-Dev] Verbose? Debug?   Daniel Swarbrick
• [Asterisk-Dev] Sipura 3000 Disconnect Singnel   Abdul Lateef Khan
• [Asterisk-Dev] Sipura 3000 Disconnect Singnel   Rich Adamson
• [Asterisk-Dev] Configuring asterisk for europe (UK)   Vizion
• [Asterisk-Dev] Verbose? Debug?   Andrew Latham
• [Asterisk-Dev] Configuring asterisk for europe (UK)   Alexander Lopez
• [Asterisk-Dev] Verbose? Debug?   Tilghman Lesher
• [Asterisk-Dev] Sipura 3000 Disconnect Singnel   Abdul Lateef Khan
• [Asterisk-Dev] Sipura 3000 Disconnect Singnel   Kaloyan Kovachev
• [Asterisk-Dev] MGCP problem   Alejandro Vargas
• [Asterisk-Dev] MGCP dropped calls   Alejandro Vargas
• [Asterisk-Dev] spandsp cisco t38   Ma Zhiyong
• [Asterisk-Dev] spandsp cisco t38   Torbjörn Abrahamsson
• [Asterisk-Dev] compile app_dial.c   Innocent Evil
• [Asterisk-Dev] compile app_dial.c   Jeremy McNamara
• [Asterisk-Dev] compile app_dial.c   Steven Critchfield
• [Asterisk-Dev] compile app_dial.c   Russell Bryant
• [Asterisk-Dev] monitoring a call and media path   Wolfgang S. Rupprecht
• [Asterisk-Dev] monitoring a call and media path   Kevin P. Fleming
• [Asterisk-Dev] OH323 user configuration   Abdul Lateef Khan
• [Asterisk-Dev] Re: 482 Loop Detected problem   snacktime
• [Asterisk-Dev] Re: 482 Loop Detected problem   snacktime
• [Asterisk-Dev] compile app_dial.c   Tzafrir Cohen
• [Asterisk-Dev] OH323 user configuration   Abdul Lateef Khan
• [Asterisk-Dev] subversion diff --exclude ?   Luigi Rizzo
• [Asterisk-Dev] compile app_dial.c   Tilghman Lesher
• [Asterisk-Dev] subversion diff --exclude ?   BJ Weschke
• [Asterisk-Dev] compile app_dial.c   Innocent Evil
• [Asterisk-Dev] compile app_dial.c   Innocent Evil
• [Asterisk-Dev] What would prevent logs from being recreated if they are deleted?   Chuck Bunn
• [Asterisk-Dev] OH323 user configuration   Steven Critchfield
• [Asterisk-Dev] meetme enhancements to improve efficiency   David Woodhouse
• [Asterisk-Dev] compile app_dial.c   Tzafrir Cohen
• [Asterisk-Dev] meetme enhancements to improve efficiency   Kevin P. Fleming
• [Asterisk-Dev] subversion diff --exclude ?   Steven Critchfield
• [Asterisk-Dev] compile app_dial.c   Tilghman Lesher
• [Asterisk-Dev] meetme enhancements to improve efficiency   David Woodhouse
• [Asterisk-Dev] meetme enhancements to improve efficiency   Kevin P. Fleming
• [Asterisk-Dev] Possible bug in realtime voicemail?   Saul Diaz
• [Asterisk-Dev] Verbose? Debug?   Andrew Latham
• [Asterisk-Dev] Re: 482 Loop Detected problem   Nishi Kant
• [Asterisk-Dev] Re: 482 Loop Detected problem   Nishi Kant
• [Asterisk-Dev] SRTP with keymanagement, SIP over TCP   Michael Prochaska
• [Asterisk-Dev] SRTP with keymanagement, SIP over TCP   Klaus Darilion
• [Asterisk-Dev] Re: 482 Loop Detected problem   Doug Meredith
• [Asterisk-Dev] compile app_dial.c   Moises Silva
• [Asterisk-Dev] svn commit problem   Avin Patel
• [Asterisk-Dev] svn commit problem   Avin Patel
• [Asterisk-Dev] Warnings to users of svn/asterisk/trunk   Russell Bryant
• [Asterisk-Dev] svn commit problem   Steven
• [Asterisk-Dev] compile app_dial.c   Tzafrir Cohen
• [Asterisk-Dev] Warnings to users of svn/asterisk/trunk   asterisk at ntplx.net
• [Asterisk-Dev] Warnings to users of svn/asterisk/trunk   Tzafrir Cohen
• [Asterisk-Dev] Warnings to users of svn/asterisk/trunk   Russell Bryant
• [Asterisk-Dev] Warnings to users of svn/asterisk/trunk   Russell Bryant
• [Asterisk-Dev] Warnings to users of svn/asterisk/trunk   Russell Bryant
• [Asterisk-Dev] svn commit problem   Kevin P. Fleming
• [Asterisk-Dev] Warnings to users of svn/asterisk/trunk   Tilghman Lesher
• [Asterisk-Dev] svn commit problem   Steven Critchfield
• [Asterisk-Dev] Asterisk 1.2.1 Released   Asterisk Development Team
• [Asterisk-Dev] RFC: simplifying sip configuration sections   Luigi Rizzo
• [Asterisk-Dev] RFC: simplifying sip configuration sections   Kevin P. Fleming
• [Asterisk-Dev] RFC: simplifying sip configuration sections   Luigi Rizzo
• [Asterisk-Dev] RFC: simplifying sip configuration sections   Russell Bryant
• [Asterisk-Dev] RFC: simplifying sip configuration sections   Luigi Rizzo
• [Asterisk-Dev] RFC: simplifying sip configuration sections   Kevin P. Fleming
• [Asterisk-Dev] RFC: simplifying sip configuration sections   Tilghman Lesher
• [Asterisk-Dev] FYI: New RFC for SIP Conferencing   Rod Dorman
• [Asterisk-Dev] RFC: simplifying sip configuration sections   Luigi Rizzo
• [Asterisk-Dev] SRTP with keymanagement, SIP over TCP   John Todd
• [Asterisk-Dev] svn commit problem   Ben Kramer
• [Asterisk-Dev] SRTP with keymanagement, SIP over TCP   Mikael Magnusson
• [Asterisk-Dev] SIP Format Bug??   Gene Willingham
• [Asterisk-Dev] SIP Format Bug??   Kevin P. Fleming
• [Asterisk-Dev] Queue and agent transfer   Tamas
• [Asterisk-Dev] Re: SRTP with keymanagement, SIP over TCP   Wolfgang S. Rupprecht
• [Asterisk-Dev] something wrong with variables, local channels, forwards and debug level   Sergio Chersovani
• [Asterisk-Dev] Re: SRTP with keymanagement, SIP over TCP   Rich Adamson
• [Asterisk-Dev] Re: SRTP with keymanagement, SIP over TCP   Mikael Magnusson
• [Asterisk-Dev] continue call for callee after caller hangup   Tristan Graham - Skymarket Ltd
• [Asterisk-Dev] continue call for callee after caller hangup   Jeremy McNamara
• [Asterisk-Dev] continue call for callee after caller hangup   Tristan Graham - Skymarket Ltd
• [Asterisk-Dev] chan_sip confused when distant end sends another port for contact info   Ed Greenberg
• [Asterisk-Dev] svn commit problem   Avin Patel
• [Asterisk-Dev] Unknown RTP codec 96 received   Javier Oviedo
• [Asterisk-Dev] run a command after Dial get answered   Ousmane Doukara
• [Asterisk-Dev] run a command after Dial get answered   Matthew Fredrickson
• [Asterisk-Dev] REFER with Replaces   Arnaud
• [Asterisk-Dev] asterisk code hacker   Bob Knight
• [Asterisk-Dev] Re: SRTP with keymanagement, SIP over TCP   John Todd
• [Asterisk-Dev] New app_groupcount.so in SVN-trunk-r7413 broken   Brian Capouch
• [Asterisk-Dev] New app_groupcount.so in SVN-trunk-r7413 broken   Steven
• [Asterisk-Dev] New app_groupcount.so in SVN-trunk-r7413 broken   Brian Capouch
• [Asterisk-Dev] New app_groupcount.so in SVN-trunk-r7413 broken   Brian Capouch
• [Asterisk-Dev] New app_groupcount.so in SVN-trunk-r7413 broken   Steven
• [Asterisk-Dev] RFC-2833 DTMF support bug in Asterisk 1.2.1   Michael Platov
• [Asterisk-Dev] C++ AGI debuggin   Danish Samad
• [Asterisk-Dev] C++ AGI debuggin   Tzafrir Cohen
• [Asterisk-Dev] Asterisk Manager encryption   John Todd
• [Asterisk-Dev] How Can i send Email using vc++   Muhammad Asim Sajjad
• [Asterisk-Dev] ebit, fas, crc4 counters for wct4xxp   Atif Rasheed
• [Asterisk-Dev] Asterisk Manager encryption   Tzafrir Cohen
• [Asterisk-Dev] How Can i send Email using vc++   Tzafrir Cohen
• [Asterisk-Dev] asterisk1.2.1+realtimedb+voicemail+contexts=bug   Frank Aartman
• [Asterisk-Dev] How to loop a zaptel channel   Kai Militzer
• [Asterisk-Dev] Asterisk Manager encryption   Paul
• [Asterisk-Dev] Asterisk Manager encryption   Kevin P. Fleming
• [Asterisk-Dev] Chan_sip version 1, 2 and NG: 3   Luigi Rizzo
• [Asterisk-Dev] asterisk1.2.1+realtimedb+voicemail+contexts=bug   Tilghman Lesher
• [Asterisk-Dev] Asterisk Manager encryption   Kristian Kielhofner
• [Asterisk-Dev] Asterisk Manager encryption   Brian Capouch
• [Asterisk-Dev] Asterisk Manager encryption   Kristian Kielhofner
• [Asterisk-Dev] Asterisk Manager encryption   Steven Critchfield
• [Asterisk-Dev] Asterisk Manager encryption   Kevin P. Fleming
• [Asterisk-Dev] Asterisk Manager encryption   Kristian Kielhofner
• [Asterisk-Dev] Asterisk Manager encryption   Mike Taht
• [Asterisk-Dev] Asterisk Manager encryption   John Todd
• [Asterisk-Dev] Asterisk Manager encryption   Kevin P. Fleming
• [Asterisk-Dev] Asterisk Manager encryption   Tzafrir Cohen
• [Asterisk-Dev] Asterisk Manager encryption   Kevin P. Fleming
• [Asterisk-Dev] fxs woes   saad
• [Asterisk-Dev] Asterisk Manager encryption   Andreas Sikkema
• [Asterisk-Dev] Asterisk Manager encryption   Mikael Magnusson
• [Asterisk-Dev] fxs woes...   saad
• [Asterisk-Dev] Asterisk Manager encryption   Kevin P. Fleming
• [Asterisk-Dev] The Zaptel init scripts must die!   Kevin P. Fleming
• [Asterisk-Dev] Re: The Zaptel init scripts must die!   Tony Mountifield
• [Asterisk-Dev] Re: The Zaptel init scripts must die!   Kevin P. Fleming
• [Asterisk-Dev] The Zaptel init scripts must die!   Greg Boehnlein
• [Asterisk-Dev] The Zaptel init scripts must die!   Kevin P. Fleming
• [Asterisk-Dev] newbie C programmer, problems with Malloc in Asterisk application   Moises Silva
• [Asterisk-Dev] newbie C programmer, problems with Malloc in Asterisk application   Marc Haisenko
• [Asterisk-Dev] The Zaptel init scripts must die!   Matt Riddell
• [Asterisk-Dev] What would prevent logs from being recreated if they are deleted?   Matt Riddell
• [Asterisk-Dev] newbie C programmer, problems with Malloc in Asterisk application   Moises Silva
• [Asterisk-Dev] Asterisk Feature Request: app_bridgeme   Nir Simionovich - CTO
• [Asterisk-Dev] The Zaptel init scripts must die!   Greg Boehnlein
• [Asterisk-Dev] What would prevent logs from being recreated i f they are deleted?   Colin Anderson
• [Asterisk-Dev] The Zaptel init scripts must die!   Jared Smith
• [Asterisk-Dev] r option issue in app_dial   Gil Kloepfer
• [Asterisk-Dev] The Zaptel init scripts must die!   Tilghman Lesher
• [Asterisk-Dev] ParkAndAnnounce() - trying to add var to indicate parked exten   Andrew Latham
• [Asterisk-Dev] ParkAndAnnounce() - trying to add var to indicate parked exten   Andrew Kohlsmith
• [Asterisk-Dev] Some Asterisk levity   Juan Carlos Castro y Castro
• [Asterisk-Dev] r option issue in app_dial   Armin Schindler
• [Asterisk-Dev] Some Asterisk levity   Juan Carlos Castro y Castro
• [Asterisk-Dev] The Zaptel init scripts must die!   Kevin P. Fleming
• [Asterisk-Dev] The Zaptel init scripts must die!   Kristian Kielhofner
• [Asterisk-Dev] The Zaptel init scripts must die!   Kevin P. Fleming
• [Asterisk-Dev] Some Asterisk levity   alex at pilosoft.com
• [Asterisk-Dev] ParkAndAnnounce() - trying to add var to indicate parked exten   Steve Blair
• [Asterisk-Dev] The Zaptel init scripts must die!   Tilghman Lesher
• [Asterisk-Dev] The Zaptel init scripts must die!   Andrew Kohlsmith
• [Asterisk-Dev] r option issue in app_dial   Denis Smirnov
• [Asterisk-Dev] Bug report   Alejandro Vargas
• [Asterisk-Dev] The Zaptel init scripts must die!   Kevin P. Fleming
• [Asterisk-Dev] newbie C programmer, problems with Malloc in Asterisk application   Chih-Wei Huang
• [Asterisk-Dev] The Zaptel init scripts must die!   Tzafrir Cohen
• [Asterisk-Dev] Re: The Zaptel init scripts must die!   Tzafrir Cohen
• [Asterisk-Dev] 408 Request Timeout vs. 403 Forbidden   Joseph Rothstein
• [Asterisk-Dev] The Zaptel init scripts must die!   Tzafrir Cohen
• [Asterisk-Dev] Re: The Zaptel init scripts must die!   Benny Amorsen
• [Asterisk-Dev] bug or feature ? extension not found...   Luigi Rizzo
• [Asterisk-Dev] What would prevent logs from being recreated if they are deleted?   Chuck Bunn
• [Asterisk-Dev] newbie C programmer, problems with Malloc in Asterisk application   Moises Silva
• [Asterisk-Dev] 408 Request Timeout vs. 403 Forbidden   Kevin P. Fleming
• [Asterisk-Dev] Re: The Zaptel init scripts must die!   Kevin P. Fleming
• [Asterisk-Dev] bug or feature ? extension not found...   Kevin P. Fleming
• [Asterisk-Dev] The Zaptel init scripts must die!   Kevin P. Fleming
• [Asterisk-Dev] bug or feature ? extension not found...   Luigi Rizzo
• [Asterisk-Dev] Blind transferred user does not hear phone ring while waiting for phone to be picked up.   Chuck Bunn
• [Asterisk-Dev] The Zaptel init scripts must die!   Kevin P. Fleming
• [Asterisk-Dev] bug or feature ? extension not found...   Kevin P. Fleming
• [Asterisk-Dev] defect in sip_chan.c?   Steve Murphy
• [Asterisk-Dev] Blind transferred user does not hear phone ring while waiting for phone to be picked up.   Chuck Bunn
• [Asterisk-Dev] r option issue in app_dial   Chuck Bunn
• [Asterisk-Dev] bug or feature ? extension not found...   Luigi Rizzo
• [Asterisk-Dev] What would prevent logs from being recreated if they are deleted?   Tilghman Lesher
• [Asterisk-Dev] defect in sip_chan.c?   Kevin P. Fleming
• [Asterisk-Dev] What would prevent logs from being recreated if they are deleted?   Matt Riddell
• [Asterisk-Dev] What would prevent logs from being recreated if they are deleted?   Chuck Bunn
• [Asterisk-Dev] The Zaptel init scripts must die!   Tzafrir Cohen
• [Asterisk-Dev] defect in sip_chan.c?   Rushowr
• [Asterisk-Dev] defect in sip_chan.c?   Luigi Rizzo
• [Asterisk-Dev] What would prevent logs from being recreated if they are deleted?   Chuck Bunn
• [Asterisk-Dev] What would prevent logs from being recreated if they are deleted?   Tzafrir Cohen
• [Asterisk-Dev] The Zaptel init scripts must die!   Kevin P. Fleming
• [Asterisk-Dev] What would prevent logs from being recreated ifthey are deleted?   Rushowr
• [Asterisk-Dev] What would prevent logs from being recreated if they are deleted?   Tilghman Lesher
• [Asterisk-Dev] What would prevent logs from being recreated if they are deleted?   Chuck Bunn
• [Asterisk-Dev] What would prevent logs from being recreated if they are deleted?   Chuck Bunn
• [Asterisk-Dev] What would prevent logs from being recreated ifthey are deleted?   Tilghman Lesher
• [Asterisk-Dev] The Zaptel init scripts must die!   Steven Critchfield
• [Asterisk-Dev] What would prevent logs from being recreated ifthey are deleted?   Tzafrir Cohen
• [Asterisk-Dev] bug or feature ? extension not found...   Luigi Rizzo
• [Asterisk-Dev] What would prevent logs from being recreated ifthey are deleted?   James Armstrong
• [Asterisk-Dev] What would prevent logs from being recreated ifthey are deleted?   Josh Roberson
• [Asterisk-Dev] Blind transferred user does not hear phone ring while waiting for phone to be picked up.   steve at daviesfam.org
• [Asterisk-Dev] The Zaptel init scripts must die!   Tzafrir Cohen
• [Asterisk-Dev] The Zaptel init scripts must die!   Tzafrir Cohen
• [Asterisk-Dev] What would prevent logs from being recreated ifthey are deleted?   Tzafrir Cohen
• [Asterisk-Dev] Help with mgcp   Alejandro Vargas
• [Asterisk-Dev] r option issue in app_dial   Eric "ManxPower" Wieling
• [Asterisk-Dev] Blind transferred user does not hear phone ring while waiting for phone to be picked up.   Chuck Bunn
• [Asterisk-Dev] The Zaptel init scripts must die!   Kevin P. Fleming
• [Asterisk-Dev] Help with mgcp   Kevin P. Fleming
• [Asterisk-Dev] Help with mgcp   mirza sahib
• [Asterisk-Dev] Help with mgcp   Steve Totaro
• [Asterisk-Dev] New app_groupcount.so in SVN-trunk-r7413 broken   Russell Bryant
• [Asterisk-Dev] RTP to IP Phone   ha i
• [Asterisk-Dev] Help with mgcp   Jeremy McNamara
• [Asterisk-Dev] RTP to IP Phone   Chris Teesdale
• [Asterisk-Dev] RTP to IP Phone   Steven Critchfield
• [Asterisk-Dev] New app_groupcount.so in SVN-trunk-r7413 broken   Brian Capouch
• [Asterisk-Dev] New app_groupcount.so in SVN-trunk-r7413 broken   Kevin P. Fleming
• [Asterisk-Dev] New app_groupcount.so in SVN-trunk-r7413 broken   Chris Parker
• [Asterisk-Dev] New app_groupcount.so in SVN-trunk-r7413 broken   Kevin P. Fleming
• [Asterisk-Dev] New app_groupcount.so in SVN-trunk-r7413 broken   Tzafrir Cohen
• [Asterisk-Dev] New app_groupcount.so in SVN-trunk-r7413 broken   Chris Parker
• [Asterisk-Dev] New app_groupcount.so in SVN-trunk-r7413 broken   Kevin P. Fleming
• [Asterisk-Dev] New app_groupcount.so in SVN-trunk-r7413 broken   Kevin P. Fleming
• [Asterisk-Dev] Help with mgcp   Alejandro Vargas
• [Asterisk-Dev] New app_groupcount.so in SVN-trunk-r7413 broken   Tilghman Lesher
• [Asterisk-Dev] New app_groupcount.so in SVN-trunk-r7413 broken   Kevin P. Fleming
• [Asterisk-Dev] What would prevent logs from being recreated ifthey are deleted?   tim panton
• [Asterisk-Dev] Help with mgcp   Kristian Kielhofner
• [Asterisk-Dev] New app_groupcount.so in SVN-trunk-r7413 broken   Steven Critchfield
• [Asterisk-Dev] New app_groupcount.so in SVN-trunk-r7413 broken   Steven Critchfield
• [Asterisk-Dev] PRI_SWITCH_EUROISDN_T1   janvb at caselaboratories.com
• [Asterisk-Dev] New app_groupcount.so in SVN-trunk-r7413 broken   Tzafrir Cohen
• [Asterisk-Dev] Help with mgcp   Alejandro Vargas
• [Asterisk-Dev] Is Asterisk a SIP proxy?   Steve Langstaff
• [Asterisk-Dev] dropped/ignored back-to-back dtmf ?   Luigi Rizzo
• [Asterisk-Dev] Asterisk Video Streaming   Himal
• [Asterisk-Dev] dropped/ignored back-to-back dtmf ?   Yaakov Menken
• [Asterisk-Dev] dropped/ignored back-to-back dtmf ?   Luigi Rizzo
• [Asterisk-Dev] Is Asterisk a SIP proxy?   Jeremy McNamara
• [Asterisk-Dev] Includes in realtime (ara) system   Steven Sokol
• [Asterisk-Dev] New app_groupcount.so in SVN-trunk-r7413 broken   Denis Smirnov
• [Asterisk-Dev] The Zaptel init scripts must die!   Denis Smirnov
• [Asterisk-Dev] improper locking in chan_sip:: struct sip_pvt's "packets" list ?   Luigi Rizzo
• [Asterisk-Dev] improper locking in chan_sip:: struct sip_pvt's "packets" list ?   Kevin P. Fleming
• [Asterisk-Dev] Registration of SIP accounts does not work well with dialup   Hans Petter Selasky
• [Asterisk-Dev] Registration of SIP accounts does not work well with dialup   North Antara
• [Asterisk-Dev] Possible deadlock issue   Tamas
• [Asterisk-Dev] asterisk audio conversion module   redice li
• [Asterisk-Dev] jingle: XMPP to VoIP. and asterisk?   Tzafrir Cohen
• [Asterisk-Dev] jingle: XMPP to VoIP. and asterisk?   Jeremy McNamara
• [Asterisk-Dev] callgroup and pickupgroup for IAX clients   kevin ling
• [Asterisk-Dev] Is Asterisk a SIP proxy?   Olle E Johansson
• [Asterisk-Dev] Is Asterisk a SIP proxy?   Klaus Darilion
• [Asterisk-Dev] no mutex_assert call ?   Luigi Rizzo
• [Asterisk-Dev] no mutex_assert call ?   Olle E Johansson
• [Asterisk-Dev] chan_iax2.c: is IAX_COMMAND_REGREL implementation working?   Eugene Prokopiev
• [Asterisk-Dev] no mutex_assert call ?   Kevin P. Fleming
• [Asterisk-Dev] chan_iax2.c: is IAX_COMMAND_REGREL implementation working?   Kevin P. Fleming
• [Asterisk-Dev] no mutex_assert call ?   Luigi Rizzo
• [Asterisk-Dev] no mutex_assert call ?   Kevin P. Fleming
• [Asterisk-Dev] no mutex_assert call ?   Luigi Rizzo
• [Asterisk-Dev] no mutex_assert call ?   Kevin P. Fleming
• [Asterisk-Dev] no mutex_assert call ?   Olle E Johansson
• [Asterisk-Dev] no mutex_assert call ?   Russell Bryant
• [Asterisk-Dev] default value of ast_opt_priority_jumping   Russell Bryant
• [Asterisk-Dev] meetme: codec_gsm.c errors when using user/admin menu   Gil Kloepfer
• [Asterisk-Dev] Called Number problem   dima_g at arcor.de
• [Asterisk-Dev] default value of ast_opt_priority_jumping   BJ Weschke
• [Asterisk-Dev] Asterisk::LDAP update   Ben Klang
• [Asterisk-Dev] Asterisk::LDAP update   Jeremy McNamara
• [Asterisk-Dev] no mutex_assert call ?   Kevin P. Fleming
• [Asterisk-Dev] default value of ast_opt_priority_jumping   Kevin P. Fleming
• [Asterisk-Dev] Asterisk::LDAP update   Ben Klang
• [Asterisk-Dev] Asterisk::LDAP update   Brian Capouch
• [Asterisk-Dev] Unacceptable delays in IAX channel.   Dmytro Mishchenko
• [Asterisk-Dev] Unacceptable delays in IAX channel.   Steve Kann
• [Asterisk-Dev] Called Number problem   Russell Bryant
• [Asterisk-Dev] Unacceptable delays in IAX channel.   tim panton
• [Asterisk-Dev] Asterisk::LDAP update   Ben Klang
• [Asterisk-Dev] Unacceptable delays in IAX channel.   Steve Kann
• [Asterisk-Dev] Help Debugging Dropped Call Audio   Matt Roth
• [Asterisk-Dev] Help Debugging Dropped Call Audio - Add'l Info   Matt Roth
• [Asterisk-Dev] meetme optimization   Geoff Karl
• [Asterisk-Dev] offer: packet cable for asterisk   plexorama
• [Asterisk-Dev] meetme optimization   Russell Bryant
• [Asterisk-Dev] make menuconfig   Russell Bryant
• [Asterisk-Dev] Unacceptable delays in IAX channel.   Kevin P. Fleming
• [Asterisk-Dev] Unacceptable delays in IAX channel.   Kevin P. Fleming
• [Asterisk-Dev] Help Debugging Dropped Call Audio - Add'l Info   Kevin P. Fleming
• [Asterisk-Dev] Change 'timelimit' on 'config' in a bridge.   Håkon Nessjøen
• [Asterisk-Dev] Change 'timelimit' on 'config' in a bridge.   Kaloyan Kovachev
• SV: [Asterisk-Dev] Change 'timelimit' on 'config' in a bridge.   Håkon Nessjøen
• [Asterisk-Dev] Module testing framework   Olle E Johansson
• [Asterisk-Dev] Need to talk to Twisted   John Martin
• [Asterisk-Dev] Need to talk to Twisted   Michael Krufky
• [Asterisk-Dev] Help Debugging Dropped Call Audio - Possibly Fixed   Matt Roth
• [Asterisk-Dev] Help Debugging Dropped Call Audio - Possibly Fixed   Kevin P. Fleming
• [Asterisk-Dev] Feature: Attendet transfer with original caller ID   Kib Eki
• SV: [Asterisk-Dev] Change 'timelimit' on 'config' in a bridge.   Kaloyan Kovachev
• [Asterisk-Dev] Change 'timelimit' on 'config' in a bridge.   Håkon Nessjøen
• [Asterisk-Dev] what might corrupt ulaw_encoder_pvt tail?   Goldfinger, Todd A
• [Asterisk-Dev] what might corrupt ulaw_encoder_pvt tail?   Tilghman Lesher
• [Asterisk-Dev] what might corrupt ulaw_encoder_pvt tail?   Goldfinger, Todd A
• [Asterisk-Dev] Help Debugging Dropped Call Audio - Possibly Fixed   Matt Roth
• [Asterisk-Dev] Help Debugging Dropped Call Audio - Possibly Fixed   Matt Roth
• [Asterisk-Dev] Help Debugging Dropped Call Audio - Possibly Fixed   Kevin P. Fleming
• [Asterisk-Dev] TDM400 answering POTS voicemail polarity reversal   asterisk at ntplx.net
• [Asterisk-Dev] TDM400 answering POTS voicemail polarity reversal   Rich Adamson
• [Asterisk-Dev] Help Debugging Dropped Call Audio - Possibly Fixed   Florian Overkamp
• [Asterisk-Dev] Help Debugging Dropped Call Audio - Possibly Fixed   Kevin P. Fleming
• [Asterisk-Dev] Help Debugging Dropped Call Audio - Possibly Fixed   SteveK
• [Asterisk-Dev] Help Debugging Dropped Call Audio - Possibly Fixed   Kevin P. Fleming
• [Asterisk-Dev] Help Debugging Dropped Call Audio - Possibly Fixed   tim panton
• [Asterisk-Dev] calloc vs malloc ?   Luigi Rizzo
• [Asterisk-Dev] calloc vs malloc ?   Tilghman Lesher
• [Asterisk-Dev] Help Debugging Dropped Call Audio - Possibly Fixed   Florian Overkamp
• [Asterisk-Dev] Help Debugging Dropped Call Audio - Possibly Fixed   Kevin P. Fleming
• [Asterisk-Dev] Help Debugging Dropped Call Audio - Possibly Fixed   Matt Roth
• [Asterisk-Dev] calloc vs malloc ?   Russell Bryant
• [Asterisk-Dev] Help Debugging Dropped Call Audio - Possibly Fixed   Matthew Roth
• [Asterisk-Dev] Help Debugging Dropped Call Audio - Possibly Fixed   Matthew Roth
• [Asterisk-Dev] Help Debugging Dropped Call Audio - Possibly Fixed   Kevin P. Fleming
• [Asterisk-Dev] Help Debugging Dropped Call Audio   Mike Benoit
• [Asterisk-Dev] ast_channel behaviour   Atif Rasheed
• [Asterisk-Dev] SQLite Realtime Driver   Steven Sokol
• [Asterisk-Dev] Help Debugging Dropped Call Audio   Matthew Roth
• [Asterisk-Dev] SQLite Realtime Driver   Steven Critchfield
• [Asterisk-Dev] Help Debugging Dropped Call Audio - Possibly Fixed   Matthew Roth
• [Asterisk-Dev] Help Debugging Dropped Call Audio - Possibly Fixed   Matthew Roth
• [Asterisk-Dev] Help Debugging Dropped Call Audio   Mike Benoit
• [Asterisk-Dev] Help Debugging Dropped Call Audio   Matthew Roth
• [Asterisk-Dev] Help Debugging Dropped Call Audio - Possibly Fixed   Kevin P. Fleming
• [Asterisk-Dev] Help Debugging Dropped Call Audio - Possibly Fixed   Kevin P. Fleming
• [Asterisk-Dev] ChanSpy() records files with funky permissions   Juan Carlos Castro y Castro
• [Asterisk-Dev] channel monitoring - use MixMonitor instead of Monitor   Wolfgang Pichler
• [Asterisk-Dev] channel monitoring - use MixMonitor instead of Monitor   BJ Weschke
• [Asterisk-Dev] channel monitoring - use MixMonitor instead of Monitor   Wolfgang Pichler
• [Asterisk-Dev] Help Debugging Dropped Call Audio   Mike Benoit
• [Asterisk-Dev] ChanSpy() records files with funky permissions   Alexander Lopez
• [Asterisk-Dev] ChanSpy() records files with funky permissions   Juan Carlos Castro y Castro
• [Asterisk-Dev] Re: [Asterisk-Users] Asterisk does not handle call from a Cisco IAD correctly   James Sizemore
• [Asterisk-Dev] Re: [Asterisk-Users] Asterisk does not handle call from a Cisco IAD correctly   Tilghman Lesher
• [Asterisk-Dev] Re: [Asterisk-Users] Asterisk does not handle call from a Cisco IAD correctly   Steven Critchfield
• [Asterisk-Dev] Re: [Asterisk-Users] Asterisk does not handle call from a Cisco IAD correctly   Tilghman Lesher
• [Asterisk-Dev] ChanSpy() records files with funky permissions   Alexander Lopez
• [Asterisk-Dev] Re: [Asterisk-Users] Asterisk does not handle call from a Cisco IAD correctly   Kevin P. Fleming
• [Asterisk-Dev] local channel dial status   Peng Yong
• [Asterisk-Dev] Asterisk extra logging to file   ast guy
• [Asterisk-Dev] Asterisk extra logging to file   BJ Weschke
• [Asterisk-Dev] Asterisk extra logging to file   ast guy
• [Asterisk-Dev] Asterisk extra logging to file   BJ Weschke
• [Asterisk-Dev] Re: [Asterisk-Users] Asterisk does not handle call from a Cisco IAD correctly   James Sizemore
• [Asterisk-Dev] asterisk 1.2 g729 compile errors   hemant surjuse
• [Asterisk-Dev] Help Debugging Dropped Call Audio   Matt Roth
• [Asterisk-Dev] asterisk 1.2 g729 compile errors   Steven Critchfield
• [Asterisk-Dev] Help Debugging Dropped Call Audio   Steven Critchfield
• [Asterisk-Dev] ztdummy? is it necessary?   Jason DiCioccio
• [Asterisk-Dev] ztdummy? is it necessary?   BJ Weschke
• [Asterisk-Dev] Help Debugging Dropped Call Audio - Possibly Fixed   Matt Roth
• [Asterisk-Dev] Re: ztdummy? is it necessary?   Jason DiCioccio
• [Asterisk-Dev] Re: ztdummy? is it necessary?   Dan Austin
• [Asterisk-Dev] Re: ztdummy? is it necessary?   Andrew Latham
• [Asterisk-Dev] Re: ztdummy? is it necessary?   Jason DiCioccio
• [Asterisk-Dev] any reason for #define FREE in the code ?   Luigi Rizzo
• [Asterisk-Dev] Re: ztdummy? is it necessary?   Steven Critchfield
• [Asterisk-Dev] Re: ztdummy? is it necessary?   Tzafrir Cohen
• [Asterisk-Dev] any reason for #define FREE in the code ?   Kevin P. Fleming
• [Asterisk-Dev] any reason for #define FREE in the code ?   Luigi Rizzo
• [Asterisk-Dev] any reason for #define FREE in the code ?   Kevin P. Fleming
• [Asterisk-Dev] any reason for #define FREE in the code ?   Luigi Rizzo
• [Asterisk-Dev] Re: ztdummy? is it necessary?   Eric "ManxPower" Wieling
• [Asterisk-Dev] Re: ztdummy? is it necessary?   Jason DiCioccio
• [Asterisk-Dev] Re: ztdummy? is it necessary?   North Antara
• [Asterisk-Dev] Re: ztdummy? is it necessary?   Olle E Johansson
• [Asterisk-Dev] Re: ztdummy? is it necessary?   Christian Richter
• [Asterisk-Dev] Re: ztdummy? is it necessary?   Kevin P. Fleming
• [Asterisk-Dev] ast_callerid_parse   Luigi Rizzo
• [Asterisk-Dev] Realtime call controll   Kaloyan Kovachev
• [Asterisk-Dev] Anybody experienced infinite loops in pbx_substitute_variables_helper_full?   Juan Carlos Castro y Castro
• SV: [Asterisk-Dev] Realtime call controll   Håkon Nessjøen
• SV: [Asterisk-Dev] Realtime call controll   Kaloyan Kovachev
• [Asterisk-Dev] I AM A DUMBASS (was: Anybody experienced infinite loops in...)   Juan Carlos Castro y Castro
• [Asterisk-Dev] I AM A DUMBASS (was: Anybody experienced infiniteloops in...)   Alexander Lopez
• SV: [Asterisk-Dev] Realtime call controll   Kaloyan Kovachev
• [Asterisk-Dev] C++ AGI debuggin   Matthew A. Nicholson
• [Asterisk-Dev] ast_callerid_parse   Brian Capouch
• [Asterisk-Dev] ast_callerid_parse   BJ Weschke
• [Asterisk-Dev] ast_callerid_parse   Kevin P. Fleming
• [Asterisk-Dev] ast_callerid_parse   Olle E Johansson
• [Asterisk-Dev] Packetization discussion   Dan Austin
• [Asterisk-Dev] Asterisk does not handle call from a Cisco IAD correctly   James Sizemore
• [Asterisk-Dev] Asterisk does not handle call from a Cisco IAD correctly   Kevin P. Fleming
• [Asterisk-Dev] ast_callerid_parse   Luigi Rizzo
• [Asterisk-Dev] RPID Issue   Ray Van Dolson
• [Asterisk-Dev] RPID Issue   Ray Van Dolson
• [Asterisk-Dev] ast_callerid_parse   Olle E Johansson
• [Asterisk-Dev] Race issue in channel.c involving uniqueint on Asterisk 1.2.1   Werner Johansson
• [Asterisk-Dev] Race issue in channel.c involving uniqueint on Asterisk 1.2.1   Olle E Johansson
• [Asterisk-Dev] Race issue in channel.c involving uniqueint on Asterisk 1.2.1 [bugid 6091]   Werner Johansson
• [Asterisk-Dev] Race issue in channel.c involving uniqueint on Asterisk 1.2.1   Luigi Rizzo
• [Asterisk-Dev] Problem on ZAP channel   rbrahmbhatt at adiance.com
• [Asterisk-Dev] Problem on ZAP channel   Steve Totaro
• [Asterisk-Dev] Problem on ZAP channel   Steven
• [Asterisk-Dev] Problem on ZAP channel   rbrahmbhatt at adiance.com
• [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 onAsterisk 1.2.1   Werner Johansson
• [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] proper use of ast_streamfile + ast_waitstream ?   Luigi Rizzo
• [Asterisk-Dev] Question on using system(find args -exec rm {} \;)   bday at prosodiemail.com
• [Asterisk-Dev] Problem on ZAP channel   Steve Totaro
• [Asterisk-Dev] Problem on ZAP channel   Steve Totaro
• [Asterisk-Dev] Race issue in channel.c involving uniqueint onAsterisk 1.2.1   imran ahmed
• [Asterisk-Dev] Coding Standard for Asterisk?   Steve Murphy
• [Asterisk-Dev] Race issue in channel.c involving uniqueint onAsterisk 1.2.1   Luigi Rizzo
• [Asterisk-Dev] chan_sip.c : ignoring domain part for incoming INVITE's causes conflicts between domains?   Bruno Rocha
• [Asterisk-Dev] Coding Standard for Asterisk?   Steven Critchfield
• [Asterisk-Dev] Fax Support   rbrahmbhatt at adiance.com
• [Asterisk-Dev] Coding Standard for Asterisk?   Steve Murphy
• [Asterisk-Dev] chan_sip.c : ignoring domain part for incoming INVITE's causes conflicts between domains?   Olle E Johansson
• [Asterisk-Dev] LOCAL_USER_ADD / LOCAL_USER_REMOVE semantics ?   Luigi Rizzo
• [Asterisk-Dev] LOCAL_USER_ADD / LOCAL_USER_REMOVE semantics ?   Russell Bryant
• [Asterisk-Dev] chan_sip.c : ignoring domain part for incomingINVITE's causes conflicts between domains?   Enzo Michelangeli
• [Asterisk-Dev] Voicemail through outlook   S.Ammad Jami

Last message date: Sat Dec 31 21:47:59 MST 2005
Archived on: Tue Sep 5 14:27:47 MST 2006

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