August 2008 Archives by author

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Starting: Fri Aug 1 01:56:39 CDT 2008
Ending: Sun Aug 31 16:55:31 CDT 2008
Messages: 240

• [asterisk-dev] Voicemail has issues with DTMF   Max Alex
• [asterisk-dev] Blind Transfer is not working in incoming calls   Max Alex
• [asterisk-dev] Blind Transfer is not working in incoming calls   Max Alex
• [asterisk-dev] quotes in AEL for 1.6   Dmitry Andrianov
• [asterisk-dev] quotes in AEL for 1.6   Dmitry Andrianov
• [asterisk-dev] DTFM question   Dmitry Andrianov
• [asterisk-dev] ZAP channel hang   Anton
• [asterisk-dev] ZAP channel hang   Anton
• [asterisk-dev] ZAP channel hang   Anton
• [asterisk-dev] ZAP channel hang   Anton
• [asterisk-dev] [policy] Bug Tracker Workflow Discussion   Jay R. Ashworth
• [asterisk-dev] [policy] Bug Tracker Workflow Discussion   Jay R. Ashworth
• [asterisk-dev] [policy] Bug Tracker Workflow Discussion   Jay R. Ashworth
• [asterisk-dev] [policy] Bug Tracker Workflow Discussion   Jay R. Ashworth
• [asterisk-dev] Losing SIP TLS connection on spurious INVITE   Bruce Atherton
• [asterisk-dev] [asterisk-commits] kpfleming: trunk r135302 - /trunk/channels/chan_skinny.c   Michiel van Baak
• [asterisk-dev] [asterisk-commits] kpfleming: trunk r135302 - /trunk/channels/chan_skinny.c   Michiel van Baak
• [asterisk-dev] 56k SS7 using TE122.   Alex Balashov
• [asterisk-dev] realtime on 1.6   David A. Bandel
• [asterisk-dev] specific storage directory for CDR data instead of LOG_DIR   David A. Bandel
• [asterisk-dev] parse error in dahdi_cfg?   David A. Bandel
• [asterisk-dev] parse error in dahdi_cfg?   David A. Bandel
• [asterisk-dev] parse error in dahdi_cfg?   David A. Bandel
• [asterisk-dev] parse error in dahdi_cfg?   David A. Bandel
• [asterisk-dev] parse error in dahdi_cfg?   David A. Bandel
• [asterisk-dev] specific storage directory for CDR data instead of LOG_DIR   Caio Begotti
• [asterisk-dev] specific storage directory for CDR data instead of LOG_DIR   Caio Begotti
• [asterisk-dev] NOTICE[28890] chan_zap.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1   Nic Bellamy
• [asterisk-dev] compiling zaptel problems   Felipe Bergo
• [asterisk-dev] compiling zaptel problems   Felipe Bergo
• [asterisk-dev] Blind Transfer is not working in incoming calls   Serge Berney
• [asterisk-dev] automated response   Stefano Bodini
• [asterisk-dev] app_fax.so and addons beta4   Germán Aracil Boned
• [asterisk-dev] app_fax.so and addons beta4   Germán Aracil Boned
• [asterisk-dev] 1.6.1 ?   Sean Bright
• [asterisk-dev] Asterisk 1.4 with cdr_tds and FreeTDS 0.82   Sean Bright
• [asterisk-dev] r128242 "KILL THE USER!" kills sip realtime?   Sean Bright
• [asterisk-dev] [asterisk-commits] seanbright: branch 1.6.0 r138483 - in /branches/1.6.0: ./ main/features.c   Sean Bright
• [asterisk-dev] ExtensionStatus   Sean Bright
• [asterisk-dev] ExtensionStatus   Sean Bright
• [asterisk-dev] n_way conference call   Sean Bright
• [asterisk-dev] [asterisk-commits] kpfleming: trunk r135302 - /trunk/channels/chan_skinny.c   Russell Bryant
• [asterisk-dev] Attended transfers problem   Russell Bryant
• [asterisk-dev] custom text in voicemail   Russell Bryant
• [asterisk-dev] [policy] Bug Tracker Workflow Discussion   Russell Bryant
• [asterisk-dev] [policy] Bug Tracker Workflow Discussion   Russell Bryant
• [asterisk-dev] [policy] Bug Tracker Workflow Discussion   Russell Bryant
• [asterisk-dev] app_followme realtime   Russell Bryant
• [asterisk-dev] Asterisk 1.4 with cdr_tds and FreeTDS 0.82   Russell Bryant
• [asterisk-dev] sip_devicestate and queue problems   Russell Bryant
• [asterisk-dev] sip_devicestate and queue problems   Russell Bryant
• [asterisk-dev] Possible Deadlock?   Russell Bryant
• [asterisk-dev] n_way conference call   Russell Bryant
• [asterisk-dev] specific storage directory for CDR data instead of LOG_DIR   Russell Bryant
• [asterisk-dev] murf: branch 1.4 r139347 - in /branches/1.4: main/pbx.c res/res_features.c   Russell Bryant
• [asterisk-dev] app_fax.so and addons beta4   Russell Bryant
• [asterisk-dev] app_fax.so and addons beta4   Russell Bryant
• [asterisk-dev] Losing SIP TLS connection on spurious INVITE   Russell Bryant
• [asterisk-dev] Voicemail has issues with DTMF   Russell Bryant
• [asterisk-dev] Blind Transfer is not working in incoming calls   Russell Bryant
• [asterisk-dev] NOTICE[28890] chan_zap.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1   Russell Bryant
• [asterisk-dev] murf: branch 1.4 r139347 - in /branches/1.4: main/pbx.c res/res_features.c   Russell Bryant
• [asterisk-dev] murf: branch 1.4 r139347 - in /branches/1.4: main/pbx.c res/res_features.c   Russell Bryant
• [asterisk-dev] Two issues with Asterisk 1.4   Tzafrir Cohen
• [asterisk-dev] Two issues with Asterisk 1.4   Tzafrir Cohen
• [asterisk-dev] [policy] Bug Tracker Workflow Discussion   Tzafrir Cohen
• [asterisk-dev] asterisk fails to start with bri ptmp timing source   Tzafrir Cohen
• [asterisk-dev] asterisk fails to start with bri ptmp timing source   Tzafrir Cohen
• [asterisk-dev] app_fax.so and addons beta4   Tzafrir Cohen
• [asterisk-dev] "56k" in dahdi/system.conf [was Re: [svn-commits] mattf: tools/trunk r4818 - /tools/trunk/dahdi_cfg.c]   Tzafrir Cohen
• [asterisk-dev] ZAPTEL API Documentation   Tzafrir Cohen
• [asterisk-dev] parse error in dahdi_cfg?   Tzafrir Cohen
• [asterisk-dev] parse error in dahdi_cfg?   Tzafrir Cohen
• [asterisk-dev] parse error in dahdi_cfg?   Tzafrir Cohen
• [asterisk-dev] [policy] Bug Tracker Workflow Discussion   Joshua Colp
• [asterisk-dev] Off the wall idea - transmit wideband G722 or Speex over ISDN   Klaus Darilion
• [asterisk-dev] NOTICE[28890] chan_zap.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1   Klaus Darilion
• [asterisk-dev] Off the wall idea - transmit wideband G722 or Speex over ISDN   Stephen Davies
• [asterisk-dev] Off the wall idea - transmit wideband G722 or Speex over ISDN   Stephen Davies
• [asterisk-dev] cdr-custom rotate?   Brian Degenhardt
• [asterisk-dev] ZAP channel hang   Paradise Dove
• [asterisk-dev] Is there a way to terminate a sip channel when I know the CALL ID?   Johansson Olle E
• [asterisk-dev] r128242 "KILL THE USER!" kills sip realtime?   Johansson Olle E
• [asterisk-dev] r128242 "KILL THE USER!" kills sip realtime?   Johansson Olle E
• [asterisk-dev] custom text in voicemail   Johansson Olle E
• [asterisk-dev] [policy] Bug Tracker Workflow Discussion   Johansson Olle E
• [asterisk-dev] r128242 "KILL THE USER!" kills sip realtime?   Johansson Olle E
• [asterisk-dev] 1.6.1 ?   Johansson Olle E
• [asterisk-dev] 1.6.1 ?   Johansson Olle E
• [asterisk-dev] SIP <=> IAX - and RTP   Johansson Olle E
• [asterisk-dev] Voice parameters with RTP Init   Johansson Olle E
• [asterisk-dev] [policy] Bug Tracker Workflow Discussion   Carles Pina i Estany
• [asterisk-dev] SIP disconnect code   Carles Pina i Estany
• [asterisk-dev] H323 disconnect cause   Carles Pina i Estany
• [asterisk-dev] Warning message regarding t38   Thiago Fernandes
• [asterisk-dev] app_followme realtime   Daniel Ferrer
• [asterisk-dev] Submitted bug #13027 needs comment!   Daniel Ferrer
• [asterisk-dev] Submitted bug #13027 needs comment!   Daniel Ferrer
• [asterisk-dev] sip_devicestate and queue problems   Daniel Ferrer
• [asterisk-dev] sip_devicestate and queue problems   Daniel Ferrer
• [asterisk-dev] chan_sip.c:3707 sip_write: Asked to transmit frame type 64   Brad Finberg
• [asterisk-dev] [asterisk-commits] kpfleming: trunk r135302 - /trunk/channels/chan_skinny.c   Kevin P. Fleming
• [asterisk-dev] [asterisk-commits] eliel: branch group/appdocsxml r136850 - in /team/group/appdocsxml: apps/ cha...   Kevin P. Fleming
• [asterisk-dev] Asterisk 1.6 - Reading DTMF from AGI   Kevin P. Fleming
• [asterisk-dev] 1.6.1 ?   Kevin P. Fleming
• [asterisk-dev] asterisk fails to start with bri ptmp timing source   Kevin P. Fleming
• [asterisk-dev] asterisk fails to start with bri ptmp timing source   Kevin P. Fleming
• [asterisk-dev] "56k" in dahdi/system.conf [was Re: [svn-commits] mattf: tools/trunk r4818 - /tools/trunk/dahdi_cfg.c]   Kevin P. Fleming
• [asterisk-dev] Possible Deadlock?   Norman Franke
• [asterisk-dev] Possible Deadlock?   Norman Franke
• [asterisk-dev] Possible Deadlock?   Norman Franke
• [asterisk-dev] 56k SS7 using TE122.   Matthew Fredrickson
• [asterisk-dev] "56k" in dahdi/system.conf [was Re: [svn-commits] mattf: tools/trunk r4818 - /tools/trunk/dahdi_cfg.c]   Matthew Fredrickson
• [asterisk-dev] "56k" in dahdi/system.conf [was Re: [svn-commits] mattf: tools/trunk r4818 - /tools/trunk/dahdi_cfg.c]   Matthew Fredrickson
• [asterisk-dev] Warning message regarding t38   Denis Galvão
• [asterisk-dev] res_jabber call backs   Ruddy Gbaguidi
• [asterisk-dev] res_jabber call backs   Ruddy Gbaguidi
• [asterisk-dev] apprendre astérisk   Jeff Gehlbach
• [asterisk-dev] Blind Transfer is not working in incoming calls   Jeff Gehlbach
• [asterisk-dev] Asterisk build-environment in Xen-DomU   Thorolf Godawa
• [asterisk-dev] Push to talk over cellular?   Stefan Gofferje
• [asterisk-dev] 1.6 trunk - Crash because of MixMonitor (incl. log and dump)   Stefan Gofferje
• [asterisk-dev] kpfleming: branch 1.4 r134983 - /branches/1.4/main/http.c   Edwin Groothuis
• [asterisk-dev] NOTICE[28890] chan_zap.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1   Sumit Gulati
• [asterisk-dev] DTFM question   Mauricio Henriquez
• [asterisk-dev] DTFM question   Mauricio Henriquez
• [asterisk-dev] automated response   Steven Howes
• [asterisk-dev] Dear asterisk-dev at lists.digium.com , ® Official Site 2008!   Steven Howes
• [asterisk-dev] Does Asterisk 'queue' DTMFs?   Saúl Ibarra
• [asterisk-dev] Asterisk build-environment in Xen-DomU   Saúl Ibarra
• Dear asterisk-dev at lists.digium.com , ® Official Site 2008!   Viagra Inc
• [asterisk-dev] 56k SS7 using TE122.   Mark A Jenks
• [asterisk-dev] sip notify to reset cisco 79x1 phones   Pavel Jezek
• [asterisk-dev] [1.6.0] - chan_sip.c We could NOT get the channel lock   Pavel Jezek
• [asterisk-dev] apprendre astérisk   HONORA KOUADIO
• [asterisk-dev] 1.6 SetCallerPres()   Philipp Kempgen
• [asterisk-dev] 1.6 Privacy()   Philipp Kempgen
• [asterisk-dev] func_logic.c: IMPORT()   Philipp Kempgen
• [asterisk-dev] ExtensionStatus   Philipp Kempgen
• [asterisk-dev] ExtensionStatus   Philipp Kempgen
• [asterisk-dev] Attended transfers problem   Kaloyan Kovachev
• [asterisk-dev] Attended transfers problem   Kaloyan Kovachev
• [asterisk-dev] varget_helper and function   Kaloyan Kovachev
• [asterisk-dev] varget_helper and function   Kaloyan Kovachev
• [asterisk-dev] [policy] Bug Tracker Workflow Discussion   Tilghman Lesher
• [asterisk-dev] app_followme realtime   Tilghman Lesher
• [asterisk-dev] Asterisk 1.6 - Reading DTMF from AGI   Tilghman Lesher
• [asterisk-dev] ZAP channel hang   Tilghman Lesher
• [asterisk-dev] ZAP channel hang   Tilghman Lesher
• [asterisk-dev] varget_helper and function   Tilghman Lesher
• [asterisk-dev] cdr_mysql - do we really need "DESC cdr" every time?   Tilghman Lesher
• [asterisk-dev] Bounty   Tilghman Lesher
• [asterisk-dev] compiling zaptel problems   Lee Lundrigan
• [asterisk-dev] compiling zaptel problems   Lee Lundrigan
• [asterisk-dev] Turning off SSL in asterisk   Lee Lundrigan
• [asterisk-dev] Voicemail - IMAP Storage Problems - thanks for the help!   Lee Lundrigan
• [asterisk-dev] Curly Asterisk <-> Voxbone problem   Mark Lynch
• [asterisk-dev] Curly Asterisk <-> Voxbone problem   Mark Lynch
• [asterisk-dev] Curly Asterisk <-> Voxbone problem   Mark Lynch
• [asterisk-dev] AMI events for start/stop of MOH in 1.6   Chris Maciejewski
• [asterisk-dev] r128242 "KILL THE USER!" kills sip realtime?   Chris Maciejewski
• [asterisk-dev] r128242 "KILL THE USER!" kills sip realtime?   Chris Maciejewski
• [asterisk-dev] r128242 "KILL THE USER!" kills sip realtime?   Chris Maciejewski
• [asterisk-dev] r128242 "KILL THE USER!" kills sip realtime?   Chris Maciejewski
• [asterisk-dev] cdr_mysql - do we really need "DESC cdr" every time?   Chris Maciejewski
• [asterisk-dev] [1.6.0] - chan_sip.c We could NOT get the channel lock   Chris Maciejewski
• [asterisk-dev] [policy] Bug Tracker Workflow Discussion   Leif Madsen
• [asterisk-dev] [policy] Bug Tracker Workflow Discussion   Leif Madsen
• [asterisk-dev] [policy] Bug Tracker Workflow Discussion   Leif Madsen
• [asterisk-dev] [policy] Bug Tracker Workflow Discussion   Leif Madsen
• [asterisk-dev] [policy] Bug Tracker Workflow Discussion   Leif Madsen
• [asterisk-dev] [policy] Bug Tracker Workflow Discussion   Leif Madsen
• [asterisk-dev] Submitted bug #13027 needs comment!   Leif Madsen
• [asterisk-dev] Outgoing call file question   Steve Mathers
• [asterisk-dev] asterisk-dev Digest, Vol 49, Issue 33   Bill Michaelson
• [asterisk-dev] Turning off SSL in asterisk   Mark Michelson
• [asterisk-dev] Possible Deadlock?   Mark Michelson
• [asterisk-dev] Possible Deadlock?   Mark Michelson
• [asterisk-dev] cdr-custom rotate?   Tony Mountifield
• [asterisk-dev] quotes in AEL for 1.6   Steve Murphy
• [asterisk-dev] [policy] Bug Tracker Workflow Discussion   Steve Murphy
• [asterisk-dev] Memory Leaks (A janitor project?)   Steve Murphy
• [asterisk-dev] murf: branch 1.4 r139347 - in /branches/1.4: main/pbx.c res/res_features.c   Steve Murphy
• [asterisk-dev] [svn-commits] qwell: branch 1.4 r616 - /branches/1.4/Makefile   Jeffrey Ollie
• [asterisk-dev] parse error in dahdi_cfg?   Jason Parker
• [asterisk-dev] [svn-commits] qwell: branch 1.4 r616 - /branches/1.4/Makefile   Patrick
• [asterisk-dev] [svn-commits] qwell: branch 1.4 r616 - /branches/1.4/Makefile   Patrick
• [asterisk-dev] Asterisk 1.6 - Reading DTMF from AGI   Ralfe Poisson
• [asterisk-dev] Asterisk 1.6 - Reading DTMF from AGI   Ralfe Poisson
• [asterisk-dev] apprendre astérisk   Andre Courchesne - Prival
• [asterisk-dev] Not able to make outgoing call [Failed to authenticate on INVITE]   Sujit Das - R&D
• [asterisk-dev] app_fax.so and addons beta4   Matt Riddell
• [asterisk-dev] [asterisk-commits] eliel: branch group/appdocsxml r136850 - in /team/group/appdocsxml: apps/ cha...   Eliel Sardanons
• [asterisk-dev] [asterisk-commits] eliel: branch group/appdocsxml r136850 - in /team/group/appdocsxml: apps/ cha...   Eliel Sardañons
• [asterisk-dev] 1.6 Privacy()   Eliel Sardañons
• [asterisk-dev] SIP <=> IAX - and RTP   Darren Sessions
• [asterisk-dev] Possible Deadlock?   Moises Silva
• [asterisk-dev] Possible Deadlock?   Moises Silva
• [asterisk-dev] Possible Deadlock?   Moises Silva
• [asterisk-dev] kpfleming: branch 1.4 r134983 - /branches/1.4/main/http.c   Sergey Tamkovich
• [asterisk-dev] libpri versions 1.2.8 and 1.4.7, and libss7 version 1.0.1 released   Asterisk Development Team
• [asterisk-dev] Chan_ss7 use signal and voice in different span of E1   Nguyen Trung Thanh
• [asterisk-dev] frequent channel reset problem   Ming-Ching Tiew
• [asterisk-dev] zaptel/kernel/wct4xxp/base.c - FANCY_ALARM   Ming-Ching Tiew
• [asterisk-dev] cdr-custom rotate?   John Todd
• [asterisk-dev] cdr-custom rotate?   John Todd
• [asterisk-dev] res_jabber call backs   John Todd
• [asterisk-dev] Two issues with Asterisk 1.4   Steve Totaro
• [asterisk-dev] frequent channel reset problem   Steve Totaro
• [asterisk-dev] Dear asterisk-dev at lists.digium.com , (R) Official Site 2008!   Steve Totaro
• [asterisk-dev] Asterisk "moh reload" causes music to not workproperly until the next "moh reload"   Fernando Urzedo
• [asterisk-dev] Two issues with Asterisk 1.4   Fernando Urzedo
• [asterisk-dev] Is there a way to terminate a sip channel when I know the CALL ID?   Fernando Urzedo
• [asterisk-dev] Two issues with Asterisk 1.4   Fernando Urzedo
• [asterisk-dev] Two issues with Asterisk 1.4   Fernando Urzedo
• [asterisk-dev] Is there a way to terminate a sip channel when Iknow the CALL ID?   Fernando Urzedo
• [asterisk-dev] Strange DTMF received on a incoming call   Fernando Urzedo
• [asterisk-dev] Extesion check after a SIP REFER?   Fernando Urzedo
• [asterisk-dev] Blind Transfer is not working in incoming calls   Fernando Urzedo
• [asterisk-dev] cdr-custom rotate?   Venefax
• [asterisk-dev] Asterisk 1.4 with cdr_tds and FreeTDS 0.82   Venefax
• [asterisk-dev] Bounty   Venefax
• [asterisk-dev] Memory Leak bug 13235   Venefax
• [asterisk-dev] res_jabber call backs   BJ Weschke
• [asterisk-dev] Warning message regarding t38   Johan Wilfer
• [asterisk-dev] Not able to make outgoing call [Failed to authenticate on INVITE]   Will
• ® Official Update 2008!   Microsoft XP
• [asterisk-dev] H323 disconnect cause   denis.korshikov at ab-group.biz
• [asterisk-dev] Not able to make outgoing call [Failed to authenticate on INVITE]   anupam bairagi
• [asterisk-dev] n_way conference call   ameur brahim
• [asterisk-dev] Asterisk + OpenIMSCore   ims.asuser ims.asuser
• [asterisk-dev] ZAPTEL API Documentation   Hari kris
• [asterisk-dev] Voice parameters with RTP Init   Hari kris
• [asterisk-dev] ZAPTEL API Documentation   Hari kris
• [asterisk-dev] Dear asterisk-dev at lists.digium.com 87% 0FF   asterisk-dev at lists.digium.com
• [asterisk-dev] custom text in voicemail   nik600
• [asterisk-dev] murf: branch 1.4 r139347 - in /branches/1.4: main/pbx.c res/res_features.c   asterisk at ntplx.net
• [asterisk-dev] murf: branch 1.4 r139347 - in /branches/1.4: main/pbx.c res/res_features.c   asterisk at ntplx.net
• [asterisk-dev] Off the wall idea - transmit wideband G722 or Speex over ISDN   asterisk at ntplx.net
• [asterisk-dev] asterisk srtp config error:603   golge yolcu

Last message date: Sun Aug 31 16:55:31 CDT 2008
Archived on: Sun Aug 31 16:56:17 CDT 2008

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