January 2010 Archives by author

• Messages sorted by: [ thread ] [ subject ] [ date ]
• More info on this list...

Starting: Sat Jan 2 00:23:37 CST 2010
Ending: Sun Jan 31 23:23:05 CST 2010
Messages: 508

• [asterisk-dev] jitterbuffer and PLC   nakaji at 02.246.ne.jp
• [asterisk-dev] /dev/zap/timer vs /dev/dahdi/timer   Dan Amarandei
• [asterisk-dev] /dev/zap/timer vs /dev/dahdi/timer   Dan Amarandei
• [asterisk-dev] Cannot load chan_dahdi.so   Dan Amarandei
• [asterisk-dev] Cannot load chan_dahdi.so   Dan Amarandei
• [asterisk-dev] Sub Address support for CDR logs   Benny Amorsen
• [asterisk-dev] Sub Address support for CDR logs   Benny Amorsen
• [asterisk-dev] Sub Address support for CDR logs   Benny Amorsen
• [asterisk-dev] SIP NOTIFY with 183 Ringing Body stays after call is tranfered   Karsten Asche
• [asterisk-dev] SIP invite to localhost/loopback   Alex Balashov
• [asterisk-dev] stringfields.h   Alex Balashov
• [asterisk-dev] Asterisk INVITE request problems   Josep Benavent
• [asterisk-dev] chan_sip: configurable peer username in digest authentication   Pietro Bertera
• [asterisk-dev] chan_sip: configurable peer username in digest authentication   Pietro Bertera
• [asterisk-dev] chan_sip: configurable peer username in digest authentication   Pietro Bertera
• [asterisk-dev] chan_sip: configurable peer username in digest authentication   Pietro Bertera
• [asterisk-dev] [Code Review] Fix reading samples from format_mp3 after ast_seekstream/ast_tellstream   Sean Bright
• [asterisk-dev] [Code Review] Fix ao2_callback when both OBJ_MULTIPLE and OBJ_NODATA are passed   Sean Bright
• [asterisk-dev] Muffins for new peer types   Sean Bright
• [asterisk-dev] [Code Review] Add notification of interrupted file and fix bug   Sean Bright
• [asterisk-dev] New Spandsp FAX Backend   Sean Bright
• [asterisk-dev] New Spandsp FAX Backend   Sean Bright
• [asterisk-dev] New Spandsp FAX Backend   Sean Bright
• [asterisk-dev] [Code Review] Group Variables   Sean Bright
• [asterisk-dev] [Code Review] [15609] Prevent a stream of warnings about the voice frame queue too long   Russell Bryant
• [asterisk-dev] Jabber more usefull received messages   Russell Bryant
• [asterisk-dev] Running a thread   Russell Bryant
• [asterisk-dev] FOSDEM 2010 - Who will be there?   Russell Bryant
• [asterisk-dev] [Code Review] Fix reading samples from format_mp3 after ast_seekstream/ast_tellstream   Russell Bryant
• [asterisk-dev] kmoore: branch 1.6.2 r237924 - /branches/1.6.2/apps/app_test.c   Russell Bryant
• [asterisk-dev] Running a thread   Russell Bryant
• [asterisk-dev] Asterisk dialplan application   Russell Bryant
• [asterisk-dev] [Code Review] Fix ao2_callback when both OBJ_MULTIPLE and OBJ_NODATA are passed   Russell Bryant
• [asterisk-dev] tilghman: trunk r239000 - in /trunk: Makefile main/asterisk.c   Russell Bryant
• [asterisk-dev] dvossel: trunk r239712 - in /trunk: ./ apps/ main/   Russell Bryant
• [asterisk-dev] lmadsen: trunk r239834 - /trunk/configs/extensions.conf.sample   Russell Bryant
• [asterisk-dev] dvossel: trunk r239712 - in /trunk: ./ apps/ main/   Russell Bryant
• [asterisk-dev] dvossel: trunk r239712 - in /trunk: ./ apps/ main/   Russell Bryant
• [asterisk-dev] signal   Russell Bryant
• [asterisk-dev] SMDI Module Requests   Russell Bryant
• [asterisk-dev] New Spandsp FAX Backend   Russell Bryant
• [asterisk-dev] oej: trunk r242043 - /trunk/main/acl.c   Russell Bryant
• [asterisk-dev] [Code Review] Convert test in test_sched.c to test API   Russell Bryant
• [asterisk-dev] [Code Review] Include test API example in test_skel.c   Russell Bryant
• [asterisk-dev] [Code Review] Convert test_substitution.c to unit test API   Russell Bryant
• [asterisk-dev] locking   Russell Bryant
• [asterisk-dev] oej: branch 1.4 r242850 - /branches/1.4/main/manager.c   Russell Bryant
• [asterisk-dev] [Code Review] Convert test_substitution.c to unit test API   Russell Bryant
• [asterisk-dev] Review board #464 - is it contributed with proper license   Russell Bryant
• [asterisk-dev] GSoC 2010   Russell Bryant
• [asterisk-dev] ast_module_user_add: question   Russell Bryant
• [asterisk-dev] stringfields.h   Russell Bryant
• [asterisk-dev] Asterisk application and SIP channel   Russell Bryant
• [asterisk-dev] Asterisk application and SIP channel   Russell Bryant
• [asterisk-dev] [Code Review] High Resolution Call Time for Billsec and Duration   Russell Bryant
• [asterisk-dev] [Code Review] Group Variables   Russell Bryant
• [asterisk-dev] [Code Review] Add support in AEL for macro return values and direct assignment of them to variables and functions.   Russell Bryant
• [asterisk-dev] [Code Review] directmediapermit/directmediadeny support to restrict which peers can do directmedia based on ip address   Russell Bryant
• [asterisk-dev] [Code Review] T.30 <-> T.38 Gateway application   Russell Bryant
• [asterisk-dev] [Code Review] High Resolution Call Time for Billsec and Duration   Russell Bryant
• [asterisk-dev] Pbx Developer   Gary Cardozo
• [asterisk-dev] Segfault under 1.4.23.1 and 1.4.23.2 (ast_do_masquerade)   Iñaki Baz Castillo
• [asterisk-dev] [Code Review] RFC compliant uri and display-name encode/decode   Iñaki Baz Castillo
• [asterisk-dev] [Code Review] RFC compliant uri and display-name encode/decode   Iñaki Baz Castillo
• [asterisk-dev] /dev/zap/timer vs /dev/dahdi/timer   Tzafrir Cohen
• [asterisk-dev] Moving call from channel x to channel x   Tzafrir Cohen
• [asterisk-dev] Cannot load chan_dahdi.so   Tzafrir Cohen
• [asterisk-dev] Cannot load chan_dahdi.so   Tzafrir Cohen
• [asterisk-dev] gmake[2]: crvs: Command not found   Tzafrir Cohen
• [asterisk-dev] realtime engine oracle   Tzafrir Cohen
• [asterisk-dev] realtime engine oracle   Tzafrir Cohen
• [asterisk-dev] New Spandsp FAX Backend   Tzafrir Cohen
• [asterisk-dev] GSoC 2010   Tzafrir Cohen
• [asterisk-dev] realtime engine oracle   Tzafrir Cohen
• [asterisk-dev] [code to use remote PBX's trunk]   Tzafrir Cohen
• [asterisk-dev] asterisk-addons   Tzafrir Cohen
• [asterisk-dev] asterisk-addons   Tzafrir Cohen
• [asterisk-dev] asterisk-addons   Tzafrir Cohen
• [asterisk-dev] Application that make sip call   Stephen Coles
• [asterisk-dev] Is this a Realtime SIP bug   Nic Colledge
• [asterisk-dev] rtp p2pbridge and rtcp   Joshua Colp
• [asterisk-dev] [Code Review] add silence gen to wait apps   Joshua Colp
• [asterisk-dev] srtp   Klaus Darilion
• [asterisk-dev] Regarding the "progressinband" SIP option   Klaus Darilion
• [asterisk-dev] 3rd party Client sending 5xx   Klaus Darilion
• [asterisk-dev] CSTA support for asterisk... or begining this module/project   Klaus Darilion
• [asterisk-dev] SIP invite to localhost/loopback   Klaus Darilion
• [asterisk-dev] fallback to audio faxing when T.38 INVITE fail with 488/606 Not acceptable   Klaus Darilion
• [asterisk-dev] fallback to audio faxing when T.38 INVITE fail with 488/606 Not acceptable   Klaus Darilion
• [asterisk-dev] Adventures in RTCP - a short report   Klaus Darilion
• [asterisk-dev] [Code Review] Add Congestion detail in CDR call logs when Congestion application is used.   Alec Davis
• [asterisk-dev] [Code Review] CDR: add Dialed Number Identifier field (DNID) field in record   Alec Davis
• [asterisk-dev] Sub Address support for CDR logs   Alec Davis
• [asterisk-dev] Sub Address support for CDR logs   Alec Davis
• [asterisk-dev] [Code Review] CDR: add Dialed Number Identifier field (DNID) field in record   Alec Davis
• [asterisk-dev] [Code Review] Add Congestion detail in CDR call logs when Congestion application is used.   Alec Davis
• [asterisk-dev] [Code Review] CDR: add Dialed Number Identifierfield (DNID) field in record   Alec Davis
• [asterisk-dev] [Code Review] CDR: add Dialed Number Identifier field (DNID) field in record   Alec Davis
• [asterisk-dev] [Code Review] CDR: add Dialed Number Identifier field (DNID) field in record   Alec Davis
• [asterisk-dev] [Code Review] CDR: add Dialed Number Identifier field (DNID) field in record   Alec Davis
• [asterisk-dev] [Code Review] CDR: Add Calling and Called Subaddress fields to CDR record   Alec Davis
• [asterisk-dev] [Code Review] CDR: Add Calling and Called Subaddress fields to CDR record   Alec Davis
• [asterisk-dev] [Code Review] CDR: Add Calling and Called Subaddress fields to CDR record   Alec Davis
• [asterisk-dev] [Code Review] CDR: Add Calling and Called Subaddress fields to CDR record   Alec Davis
• [asterisk-dev] [Code Review] CDR: Add Calling and Called Subaddress fields to CDR record   Alec Davis
• [asterisk-dev] [Code Review] CDR: Add Calling and Called Subaddress fields to CDR record   Alec Davis
• [asterisk-dev] Q.932 ROSE Support libpri   Degasperi, Matteo
• [asterisk-dev] Q.932 ROSE Support libpri   Degasperi, Matteo
• [asterisk-dev] Asterisk Mod Request   Glenn Elliott
• [asterisk-dev] [Code Review] rtp timestamp to timeval calculation fix   Dan Evans
• [asterisk-dev] [review-request] T.30 <-> T.38 Gateway application   Daniel Ferenci
• [asterisk-dev] app_alarmreceiver   Francisco Fernandes
• [asterisk-dev] app_alarmreceiver   Francisco Fernandes
• [asterisk-dev] app_alarmreceiver   Francisco Fernandes
• [asterisk-dev] Multiple authorization header for a SIP message. More info to the bug report.   Eduardo Ferro
• [asterisk-dev] Manager "sipchannel" command   Eduardo Ferro
• [asterisk-dev] [Code Review] Fix ao2_callback when both OBJ_MULTIPLE and OBJ_NODATA are passed   Kevin Fleming
• [asterisk-dev] [Code Review] rtp timestamp to timeval calculation fix   Kevin Fleming
• [asterisk-dev] [Code Review] High Resolution Call Time for Billsec and Duration   Kevin Fleming
• [asterisk-dev] AMI call on hold (new topic)   Kevin P. Fleming
• [asterisk-dev] AMI call on hold (new topic)   Kevin P. Fleming
• [asterisk-dev] fallback to audio faxing when T.38 INVITE fail with 488/606 Not acceptable   Kevin P. Fleming
• [asterisk-dev] fallback to audio faxing when T.38 INVITE fail with 488/606 Not acceptable   Kevin P. Fleming
• [asterisk-dev] dvossel: trunk r239712 - in /trunk: ./ apps/ main/   Kevin P. Fleming
• [asterisk-dev] dvossel: trunk r239712 - in /trunk: ./ apps/ main/   Kevin P. Fleming
• [asterisk-dev] Q.932 ROSE Support libpri   Kevin P. Fleming
• [asterisk-dev] New Spandsp FAX Backend   Kevin P. Fleming
• [asterisk-dev] Q.932 ROSE Support libpri   Kevin P. Fleming
• [asterisk-dev] asterisk 1.4 sent T.38 UDPTL only if DONT_OPTIMIZE is used   Kevin P. Fleming
• [asterisk-dev] New Spandsp FAX Backend   Kevin P. Fleming
• [asterisk-dev] [asterisk-commits] oej: branch 1.4 r242226 - /branches/1.4/channels/chan_sip.c   Kevin P. Fleming
• [asterisk-dev] [asterisk-commits] oej: branch 1.4 r242226 - /branches/1.4/channels/chan_sip.c   Kevin P. Fleming
• [asterisk-dev] [asterisk-commits] tilghman: branch 1.4 r242423 - /branches/1.4/pbx/Makefile   Kevin P. Fleming
• [asterisk-dev] [asterisk-commits] tilghman: branch 1.4 r242423 - /branches/1.4/pbx/Makefile   Kevin P. Fleming
• [asterisk-dev] 0015539: [patch]Registerrequestlinecontainswrong address when domain andregistrarhost differ   Kevin P. Fleming
• [asterisk-dev] Asterisk dialplan application   Salvatore Frandina
• [asterisk-dev] Asterisk application development   Salvatore Frandina
• [asterisk-dev] Application that make sip call   Salvatore Frandina
• [asterisk-dev] Ast_channel question   Salvatore Frandina
• [asterisk-dev] Ast_channel question   Salvatore Frandina
• [asterisk-dev] SIPp video support xml scenarios   Salvatore Frandina
• [asterisk-dev] Fwd: App_Conference: SIPp SDP and DTMF mode   Salvatore Frandina
• [asterisk-dev] ast_module_user_add: question   Salvatore Frandina
• [asterisk-dev] Ast_channel data structure   Salvatore Frandina
• [asterisk-dev] Asterisk audio/video RTP question   Salvatore Frandina
• [asterisk-dev] Ast_channel data structure   Salvatore Frandina
• [asterisk-dev] Ast_channel data structure   Salvatore Frandina
• [asterisk-dev] Asterisk application and SIP channel   Salvatore Frandina
• [asterisk-dev] Jabber more usefull received messages   Ruddy Gbaguidi
• [asterisk-dev] Fwd: res_pktccops   Nahuel Greco
• [asterisk-dev] Fwd: res_pktccops   Nahuel Greco
• [asterisk-dev] Fwd: Re: Katrina response, private and public   Alexander Harrowell
• [asterisk-dev] Sub Address support for CDR logs   Kai Hoerner
• [asterisk-dev] Sub Address support for CDR logs   Kai Hoerner
• [asterisk-dev] Asterisk dialplan application   Kai Hoerner
• [asterisk-dev] realtime engine oracle   Kai Hoerner
• [asterisk-dev] IMAP Invalid system flag UNSEEN   Hoggins!
• [asterisk-dev] [Code Review] directmediapermit/directmediadeny support to restrict which peers can do directmedia based on ip address   Olle E Johansson
• [asterisk-dev] [svn-commits] tilghman: trunk r237213 - /trunk/channels/chan_sip.c   Olle E. Johansson
• [asterisk-dev] Can't compile trunk chan_sip on Snow Leopard   Olle E. Johansson
• [asterisk-dev] Fwd: res_pktccops   Olle E. Johansson
• [asterisk-dev] Fwd: res_pktccops   Olle E. Johansson
• [asterisk-dev] Fwd: res_pktccops   Olle E. Johansson
• [asterisk-dev] srtp   Olle E. Johansson
• [asterisk-dev] Inquiry:Asterisk support KPML?   Olle E. Johansson
• [asterisk-dev] Inquiry:Asterisk support KPML?   Olle E. Johansson
• [asterisk-dev] AMI call on hold (new topic)   Olle E. Johansson
• [asterisk-dev] AMI call on hold (new topic)   Olle E. Johansson
• [asterisk-dev] Sub Address support for CDR logs   Olle E. Johansson
• [asterisk-dev] AMI call on hold (new topic)   Olle E. Johansson
• [asterisk-dev] Sub Address support for CDR logs   Olle E. Johansson
• [asterisk-dev] Sub Address support for CDR logs   Olle E. Johansson
• [asterisk-dev] rtp p2pbridge and rtcp   Olle E. Johansson
• [asterisk-dev] [svn-commits] dvossel: trunk r238405 - /trunk/channels/chan_sip.c   Olle E. Johansson
• [asterisk-dev] [svn-commits] dvossel: trunk r238405 - /trunk/channels/chan_sip.c   Olle E. Johansson
• [asterisk-dev] rtp p2pbridge and rtcp   Olle E. Johansson
• [asterisk-dev] SIP invite to localhost/loopback   Olle E. Johansson
• [asterisk-dev] Muffins for new peer types   Olle E. Johansson
• [asterisk-dev] Muffins for new peer types   Olle E. Johansson
• [asterisk-dev] fallback to audio faxing when T.38 INVITE fail with 488/606 Not acceptable   Olle E. Johansson
• [asterisk-dev] ast_rtp->altthem   Olle E. Johansson
• [asterisk-dev] ast_rtp->altthem   Olle E. Johansson
• [asterisk-dev] chan_datacard and RTCP   Olle E. Johansson
• [asterisk-dev] Multiple authorization header for a SIP message. More info to the bug report.   Olle E. Johansson
• [asterisk-dev] Manager "sipchannel" command   Olle E. Johansson
• [asterisk-dev] realtime engine oracle   Olle E. Johansson
• [asterisk-dev] AST_CONTROL_MUTE/UNMUTE   Olle E. Johansson
• [asterisk-dev] Retransmit: Manager "sipchannel" command   Olle E. Johansson
• [asterisk-dev] Manager "sipchannel" command   Olle E. Johansson
• [asterisk-dev] Manager "sipchannel" command   Olle E. Johansson
• [asterisk-dev] realtime engine oracle   Olle E. Johansson
• [asterisk-dev] Manipulating audio in asterisk   Olle E. Johansson
• [asterisk-dev] Manager "sipchannel" command   Olle E. Johansson
• [asterisk-dev] Please test my RTCP work!   Olle E. Johansson
• [asterisk-dev] Fwd: Can't compile trunk chan_sip on Snow Leopard   Olle E. Johansson
• [asterisk-dev] [asterisk-commits] oej: branch 1.4 r242226 - /branches/1.4/channels/chan_sip.c   Olle E. Johansson
• [asterisk-dev] [Code Review] RFC compliant uri and display-name encode/decode   Olle E. Johansson
• [asterisk-dev] Review board #464 - is it contributed with proper license   Olle E. Johansson
• [asterisk-dev] Review board #464 - is it contributed with proper license   Olle E. Johansson
• [asterisk-dev] 0015539: [patch] Register request line contains wrong address when domain and registrar host differ   Olle E. Johansson
• [asterisk-dev] 0015539: [patch] Register request line containswrong address when domain and registrar host differ   Olle E. Johansson
• [asterisk-dev] 0015539: [patch] Register request linecontainswrong address when domain and registrar host differ   Olle E. Johansson
• [asterisk-dev] 0015539: [patch] Register requestlinecontainswrong address when domain and registrar host differ   Olle E. Johansson
• [asterisk-dev] 0015539: [patch] Registerrequestlinecontainswrong address when domain and registrarhost differ   Olle E. Johansson
• [asterisk-dev] [Code Review] RFC compliant uri and display-name encode/decode   Olle E. Johansson
• [asterisk-dev] Ast_channel data structure   Olle E. Johansson
• [asterisk-dev] Ast_channel data structure   Olle E. Johansson
• [asterisk-dev] Ast_channel data structure   Olle E. Johansson
• [asterisk-dev] Adventures in RTCP - a short report   Olle E. Johansson
• [asterisk-dev] [svn-commits] jpeeler: trunk r243551 - in /trunk: ./ include/asterisk/ main/   Olle E. Johansson
• [asterisk-dev] Adventures in RTCP - a short report   Olle E. Johansson
• [asterisk-dev] Asterisk application and SIP channel   Olle E. Johansson
• [asterisk-dev] chan_sip: configurable peer username in digest authentication   Olle E. Johansson
• [asterisk-dev] Asterisk INVITE request problems   Olle E. Johansson
• [asterisk-dev] Problem with chan_local variables, optimization, and timing   Olle E. Johansson
• [asterisk-dev] Adventures in RTCP - a short report   Olle E. Johansson
• [asterisk-dev] chan_sip: configurable peer username in digest authentication   Olle E. Johansson
• [asterisk-dev] asterisk-addons   Olle E. Johansson
• [asterisk-dev] chan_sip.c realtime   Olle E. Johansson
• [asterisk-dev] chan_sip.c realtime   Olle E. Johansson
• [asterisk-dev] chan_sip.c realtime   Olle E. Johansson
• [asterisk-dev] [Code Review] CDR: add Dialed Number Identifier field (DNID) field in record   Kirill 'Big K' Katsnelson
• [asterisk-dev] [Code Review] CDR: add Dialed Number Identifierfield (DNID) field in record   Kirill 'Big K' Katsnelson
• [asterisk-dev] StopMonitor() reverts itself after SIP transfer: bug or feature?   Kirill 'Big K' Katsnelson
• [asterisk-dev] SIP invite to localhost/loopback   Kirill 'Big K' Katsnelson
• [asterisk-dev] SIP CHANNEL(rtpqos,audio,...) values   Kirill 'Big K' Katsnelson
• [asterisk-dev] gmake[2]: crvs: Command not found   Nathaniel L Keeling
• [asterisk-dev] Caller ID detection in India   Zahir Koradia
• [asterisk-dev] FOSDEM 2010 - Who will be there?   Stelios Koroneos
• [asterisk-dev] SIP invite to localhost/loopback   Kaloyan Kovachev
• [asterisk-dev] app_alarmreceiver   Kaloyan Kovachev
• [asterisk-dev] app_alarmreceiver   Kaloyan Kovachev
• [asterisk-dev] 3rd party Client sending 5xx   Aditya Kumar
• [asterisk-dev] asterisk-dev Digest, Vol 66, Issue 50   Mohammed Lambat
• [asterisk-dev] Out-of-order events from AMI interface - to be expected, or a bug?   Alex Villací­s Lasso
• [asterisk-dev] [Code Review] High Resolution Call Time for Billsec and Duration   Brad Latus
• [asterisk-dev] [Code Review] High Resolution Call Time for Billsec and Duration   Brad Latus
• [asterisk-dev] Fwd: res_pktccops   Tilghman Lesher
• [asterisk-dev] [Code Review] CDR: add Dialed Number Identifierfield (DNID) field in record   Tilghman Lesher
• [asterisk-dev] [Code Review] Provide method for testing GotoIfTime without changing localtime   Tilghman Lesher
• [asterisk-dev] [Code Review] CDR: add Dialed Number Identifier field (DNID) field in record   Tilghman Lesher
• [asterisk-dev] SIP invite to localhost/loopback   Tilghman Lesher
• [asterisk-dev] astmm   Tilghman Lesher
• [asterisk-dev] [Code Review] CDR: add Dialed Number Identifier field (DNID) field in record   Tilghman Lesher
• [asterisk-dev] [Code Review] Provide method for testing GotoIfTime without changing localtime   Tilghman Lesher
• [asterisk-dev] Out-of-order events from AMI interface - to be expected, or a bug?   Tilghman Lesher
• [asterisk-dev] [Code Review] Provide method for testing GotoIfTime without changing localtime   Tilghman Lesher
• [asterisk-dev] [Code Review] Provide method for testing GotoIfTime without changing localtime   Tilghman Lesher
• [asterisk-dev] [Code Review] add silence gen to wait apps   Tilghman Lesher
• [asterisk-dev] [Code Review] Missing new-message notification for urgent messages   Tilghman Lesher
• [asterisk-dev] [Code Review] High Resolution Call Time for Billsec and Duration   Tilghman Lesher
• [asterisk-dev] [Code Review] High Resolution Call Time for Billsec and Duration   Tilghman Lesher
• [asterisk-dev] [Code Review] High Resolution Call Time for Billsec and Duration   Tilghman Lesher
• [asterisk-dev] [Code Review] Added ability to perform SRV lookups for AGI URIs   Tilghman Lesher
• [asterisk-dev] realtime engine   Tilghman Lesher
• [asterisk-dev] signal   Tilghman Lesher
• [asterisk-dev] [Code Review] CDR: Add Calling and Called Subaddress fields to CDR record   Tilghman Lesher
• [asterisk-dev] [Code Review] Add command to send DTMF from ExternalIVR   Tilghman Lesher
• [asterisk-dev] [Code Review] High Resolution Call Time for Billsec and Duration   Tilghman Lesher
• [asterisk-dev] [Code Review] CDR: Add Calling and Called Subaddress fields to CDR record   Tilghman Lesher
• [asterisk-dev] [Code Review] Add command to send DTMF from ExternalIVR   Tilghman Lesher
• [asterisk-dev] Manager "sipchannel" command   Tilghman Lesher
• [asterisk-dev] Asterisk 1.6.1.13 crashes in masq_park_call()   Tilghman Lesher
• [asterisk-dev] [Code Review] Add support in AEL for macro return values and direct assignment of them to variables and functions.   Tilghman Lesher
• [asterisk-dev] [Code Review] RFC compliant uri and display-name encode/decode   Tilghman Lesher
• [asterisk-dev] [Code Review] RFC compliant uri and display-name encode/decode   Tilghman Lesher
• [asterisk-dev] [Code Review] High Resolution Call Time for Billsec and Duration   Tilghman Lesher
• [asterisk-dev] [Code Review] RFC compliant uri and display-name encode/decode   Tilghman Lesher
• [asterisk-dev] [Code Review] RFC compliant uri and display-name encode/decode   Tilghman Lesher
• [asterisk-dev] [asterisk-commits] tilghman: branch 1.4 r242423 - /branches/1.4/pbx/Makefile   Tilghman Lesher
• [asterisk-dev] [asterisk-commits] tilghman: branch 1.4 r242423 - /branches/1.4/pbx/Makefile   Tilghman Lesher
• [asterisk-dev] Review board #464 - is it contributed with proper license   Tilghman Lesher
• [asterisk-dev] [Code Review] High Resolution Call Time for Billsec and Duration   Tilghman Lesher
• [asterisk-dev] [Code Review] High Resolution Call Time for Billsec and Duration   Tilghman Lesher
• [asterisk-dev] AGI SAY DATETIME not working as expected   Tilghman Lesher
• [asterisk-dev] stringfields.h   Tilghman Lesher
• [asterisk-dev] AGI SAY DATETIME not working as expected   Tilghman Lesher
• [asterisk-dev] [Code Review] High Resolution Call Time for Billsec and Duration   Tilghman Lesher
• [asterisk-dev] cdr_radius.c thread safety   Tilghman Lesher
• [asterisk-dev] Muffins for new peer types   Nick Lewis
• [asterisk-dev] Muffins for new peer types   Nick Lewis
• [asterisk-dev] Muffins for new peer types   Nick Lewis
• [asterisk-dev] Muffins for new peer types   Nick Lewis
• [asterisk-dev] IMAP Invalid system flag UNSEEN   Nick Lewis
• [asterisk-dev] Problem FOP   Nick Lewis
• [asterisk-dev] Muffins for new peer types   Nick Lewis
• [asterisk-dev] AST_CONTROL_MUTE/UNMUTE   Nick Lewis
• [asterisk-dev] [Code Review] rtp timestamp to timevalcalculation fix   Nick Lewis
• [asterisk-dev] [Code Review] rtp timestamp to timeval calculation fix   Nick Lewis
• [asterisk-dev] [Code Review] RFC compliant uri and display-name encode/decode   Nick Lewis
• [asterisk-dev] [Code Review] RFC compliant uri and display-name encode/decode   Nick Lewis
• [asterisk-dev] [Code Review] RFC compliant uri and display-name encode/decode   Nick Lewis
• [asterisk-dev] [Code Review] RFC compliant uri and display-name encode/decode   Nick Lewis
• [asterisk-dev] 0015539: [patch] Register request line contains wrong address when domain and registrar host differ   Nick Lewis
• [asterisk-dev] 0015539: [patch] Register request line containswrong address when domain and registrar host differ   Nick Lewis
• [asterisk-dev] 0015539: [patch] Register request linecontainswrong address when domain and registrar host differ   Nick Lewis
• [asterisk-dev] 0015539: [patch] Register requestlinecontainswrong address when domain and registrar host differ   Nick Lewis
• [asterisk-dev] 0015539: [patch] Registerrequestlinecontainswrong address when domain and registrarhost differ   Nick Lewis
• [asterisk-dev] 0015539: [patch]Registerrequestlinecontainswrong address when domain andregistrarhost differ   Nick Lewis
• [asterisk-dev] [Code Review] RFC compliant uri and display-name encode/decode   Nick Lewis
• [asterisk-dev] [Code Review] RFC compliant uri and display-nameencode/decode   Nick Lewis
• [asterisk-dev] [Code Review] RFC compliant uri and display-nameencode/decode   Nick Lewis
• [asterisk-dev] realtime engine oracle   Nick Lewis
• [asterisk-dev] Adventures in RTCP - a short report   Nick Lewis
• [asterisk-dev] Adventures in RTCP - a short report   Nick Lewis
• [asterisk-dev] [Code Review] CDR: add Dialed Number Identifierfield (DNID) field in record   Kurt Lidl
• [asterisk-dev] Music on hold after an attended transfer to a queue.   Gabriel Ortiz Lour
• [asterisk-dev] Music on hold after an attended transfer to a queue.   Gabriel Ortiz Lour
• [asterisk-dev] A problem with attended transfer in 1.6.1.12   Leif Madsen
• [asterisk-dev] chan_datacard and RTCP   Artem Makhutov
• [asterisk-dev] chan_datacard and RTCP   Artem Makhutov
• [asterisk-dev] chan_datacard and RTCP   Artem Makhutov
• [asterisk-dev] chan_datacard and RTCP   Artem Makhutov
• [asterisk-dev] cdr_radius.c thread safety   Alex Massover
• [asterisk-dev] astmm   Bhrugu Mehta
• [asterisk-dev] Asterisk dialplan application   Bhrugu Mehta
• [asterisk-dev] astmm   Bhrugu Mehta
• [asterisk-dev] thread debug   Bhrugu Mehta
• [asterisk-dev] thread debug   Bhrugu Mehta
• [asterisk-dev] Pbx Developer   Bhrugu Mehta
• [asterisk-dev] thread debug   Bhrugu Mehta
• [asterisk-dev] signal   Bhrugu Mehta
• [asterisk-dev] realtime engine   Bhrugu Mehta
• [asterisk-dev] signal   Bhrugu Mehta
• [asterisk-dev] astmm   Bhrugu Mehta
• [asterisk-dev] realtime engine oracle   Bhrugu Mehta
• [asterisk-dev] realtime engine oracle   Bhrugu Mehta
• [asterisk-dev] realtime engine oracle   Bhrugu Mehta
• [asterisk-dev] realtime engine oracle   Bhrugu Mehta
• [asterisk-dev] locking   Bhrugu Mehta
• [asterisk-dev] locking   Bhrugu Mehta
• [asterisk-dev] stringfields.h   Bhrugu Mehta
• [asterisk-dev] GSoC 2010   Bhrugu Mehta
• [asterisk-dev] ast_module_user_add: question   Bhrugu Mehta
• [asterisk-dev] realtime engine oracle   Bhrugu Mehta
• [asterisk-dev] stringfields.h   Bhrugu Mehta
• [asterisk-dev] stringfields.h   Bhrugu Mehta
• [asterisk-dev] asterisk-addons   Bhrugu Mehta
• [asterisk-dev] asterisk.spec   Bhrugu Mehta
• [asterisk-dev] realtime engine oracle   Bhrugu Mehta
• [asterisk-dev] Asterisk, Bhrugu Mehta has invited you to open a Google mail account   Bhrugu Mehta
• [asterisk-dev] asterisk-addons   Bhrugu Mehta
• [asterisk-dev] asterisk-addons   Bhrugu Mehta
• [asterisk-dev] chan_sip.c realtime   Bhrugu Mehta
• [asterisk-dev] chan_sip.c realtime   Bhrugu Mehta
• [asterisk-dev] chan_sip.c realtime   Bhrugu Mehta
• [asterisk-dev] realtime engine oracle   Bhrugu Mehta
• [asterisk-dev] thread debug   Mark Michelson
• [asterisk-dev] ast_rtp->altthem   Mark Michelson
• [asterisk-dev] Ast_channel question   Mark Michelson
• [asterisk-dev] Requiring session timers and call drop   Mark Michelson
• [asterisk-dev] signal   Mark Michelson
• [asterisk-dev] Regarding asterisk source code contribution   Mark Michelson
• [asterisk-dev] thread debug   Mark Michelson
• [asterisk-dev] thread debug   Mark Michelson
• [asterisk-dev] [Code Review] code cleanup in app_senddtmf   Mark Michelson
• [asterisk-dev] Ast_channel data structure   Mark Michelson
• [asterisk-dev] [Code Review] T.30 <-> T.38 Gateway application   Mark Michelson
• [asterisk-dev] SIP invite to localhost/loopback   Geoffrey Mina
• [asterisk-dev] SIP invite to localhost/loopback   Geoffrey Mina
• [asterisk-dev] Problem with chan_local variables, optimization, and timing   Geoffrey Mina
• [asterisk-dev] Problem with chan_local variables, optimization, and timing   Geoffrey Mina
• [asterisk-dev] Running a thread   MohammedShehzad
• [asterisk-dev] Running a thread   MohammedShehzad
• [asterisk-dev] Running a thread   MohammedShehzad
• [asterisk-dev] ISDN Cause codes for unanswered calls   Steve Moran
• [asterisk-dev] Sub Address support for CDR logs   Richard Mudgett
• [asterisk-dev] Q.932 ROSE Support libpri   Richard Mudgett
• [asterisk-dev] [svn-commits] alecdavis: trunk r241097 - /trunk/channels/sig_pri.c   Richard Mudgett
• [asterisk-dev] Q.932 ROSE Support libpri   Richard Mudgett
• [asterisk-dev] CSTA support for asterisk... or begining this module/project   Chris Mylonas
• [asterisk-dev] AMI call on hold (new topic)   Chris Mylonas
• [asterisk-dev] AMI call on hold (new topic)   Chris Mylonas
• [asterisk-dev] Sub Address support for CDR logs   Chris Mylonas
• [asterisk-dev] AMI call on hold (new topic)   Chris Mylonas
• [asterisk-dev] CSTA support for asterisk... or begining this module/project   Chris Mylonas
• [asterisk-dev] Sub Address support for CDR logs   Chris Mylonas
• [asterisk-dev] CSTA support for asterisk... or begining this module/project   Chris Mylonas
• [asterisk-dev] Problem FOP   Naudit007
• [asterisk-dev] Moving call from channel x to channel x   Håkon Nessjøen
• [asterisk-dev] Moving call from channel x to channel x   Håkon Nessjøen
• [asterisk-dev] Moving call from channel x to channel x   Håkon Nessjøen
• [asterisk-dev] Moving call from channel x to channel x   Håkon Nessjøen
• [asterisk-dev] Moving call from channel x to channel x   Håkon Nessjøen
• [asterisk-dev] ISDN Cause codes for unanswered calls   Håkon Nessjøen
• [asterisk-dev] Moving call from channel x to channel x   Håkon Nessjøen
• [asterisk-dev] realtime engine oracle   Håkon Nessjøen
• [asterisk-dev] Fwd: Can't compile trunk chan_sip on Snow Leopard   Håkon Nessjøen
• [asterisk-dev] New Spandsp FAX Backend   Matthew Nicholson
• [asterisk-dev] New Spandsp FAX Backend   Matthew Nicholson
• [asterisk-dev] New Spandsp FAX Backend   Matthew Nicholson
• [asterisk-dev] New Spandsp FAX Backend   Matthew Nicholson
• [asterisk-dev] New Spandsp FAX Backend   Matthew Nicholson
• [asterisk-dev] New Spandsp FAX Backend   Matthew Nicholson
• [asterisk-dev] thread debug   Clod Patry
• [asterisk-dev] [Code Review] Extend the max number of callgroups/pickupgroups   Jeff Peeler
• [asterisk-dev] [Code Review] RFC compliant uri and display-name encode/decode   Jeff Peeler
• [asterisk-dev] [Code Review] When setting a soundfile for announce with a length longer then 80 chars a storage overlay happens   Jeff Peeler
• [asterisk-dev] [svn-commits] jpeeler: trunk r243551 - in /trunk: ./ include/asterisk/ main/   Jeff Peeler
• [asterisk-dev] Bug #0015997   Wolfgang Pichler
• [asterisk-dev] Please test my RTCP work!   Wolfgang Pichler
• [asterisk-dev] CSTA support for asterisk... or begining this module/project   Guilherme Rezende
• [asterisk-dev] CSTA support for asterisk... or begining this module/project   Guilherme Rezende
• [asterisk-dev] SIP invite to localhost/loopback   Matt Riddell
• [asterisk-dev] jitterbuffer and PLC   Matt Riddell
• [asterisk-dev] jitterbuffer and PLC   Matt Riddell
• [asterisk-dev] realtime engine oracle   Matt Riddell
• [asterisk-dev] Problem with chan_local variables, optimization, and timing   Tim Ringenbach
• [asterisk-dev] Bug #0015997   David Ruggles
• [asterisk-dev] Asterisk Mod Request   David Ruggles
• [asterisk-dev] Adding SendDTMF support to ExternalIVR   David Ruggles
• [asterisk-dev] Regarding asterisk source code contribution   David Ruggles
• [asterisk-dev] chan_sip: configurable peer username in digest authentication   David Ruggles
• [asterisk-dev] [regression] Anonymous SIP calls are hung up after 900 seconds (Asterisk v. 1.6.0.18)   Hans Petter Selasky
• [asterisk-dev] Asterisk sip.conf dtmfmode=info?   Nir Simionovich
• [asterisk-dev] Manipulating audio in asterisk   Slawek Sloma
• [asterisk-dev] Adventures in RTCP - a short report   Jared Smith
• [asterisk-dev] SMDI Module Requests   Bill Sofko
• [asterisk-dev] Asterisk Receives SIP 180 and then SIP 486 But Doesn't Play Busy Tones   Jamuel Starkey
• [asterisk-dev] asterisk.spec   Jamuel Starkey
• [asterisk-dev] Asterisk 1.4.29-rc1, 1.6.0.21-rc1, 1.6.1.13-rc1, and 1.6.2.1-rc1 Now Released   Asterisk Development Team
• [asterisk-dev] Asterisk 1.4.29 Now Available   Asterisk Development Team
• [asterisk-dev] Asterisk 1.6.0.21 Now Available   Asterisk Development Team
• [asterisk-dev] Asterisk 1.6.1.13 Now Available   Asterisk Development Team
• [asterisk-dev] Asterisk 1.6.2.1 Now Available   Asterisk Development Team
• [asterisk-dev] DAHDI-Linux 2.2.1 and DAHDI-Tools 2.2.1 Released   Asterisk Development Team
• [asterisk-dev] Unavailable peer not show in BLF   Fabio Terrile
• [asterisk-dev] Regarding asterisk source code contribution   Sherin Baby Thomas
• [asterisk-dev] Requiring session timers and call drop   John Todd
• [asterisk-dev] dvossel: trunk r239712 - in /trunk: ./ apps/ main/   John Todd
• [asterisk-dev] signal   John Todd
• [asterisk-dev] dvossel: trunk r239712 - in /trunk: ./ apps/ main/   John Todd
• [asterisk-dev] [Code Review] Add notification of interrupted file and fix bug   Chris Tooley
• [asterisk-dev] chan_datacard and RTCP   Steve Totaro
• [asterisk-dev] New Spandsp FAX Backend   Steve Totaro
• [asterisk-dev] A problem with attended transfer in 1.6.1.12   Pavel Troller
• [asterisk-dev] Inquiry:Asterisk support KPML?   Pavel Troller
• [asterisk-dev] Asterisk 1.6.1.13 crashes in masq_park_call()   Pavel Troller
• [asterisk-dev] New Spandsp FAX Backend   Steve Underwood
• [asterisk-dev] New Spandsp FAX Backend   Steve Underwood
• [asterisk-dev] [Code Review] ast_uri_encode() behavior change   David Vossel
• [asterisk-dev] [Code Review] parkinglots with different parking extensions   David Vossel
• [asterisk-dev] [Code Review] add silence gen to wait apps   David Vossel
• [asterisk-dev] [Code Review] add silence gen to wait apps   David Vossel
• [asterisk-dev] [Code Review] Provide method for testing GotoIfTime without changing localtime   David Vossel
• [asterisk-dev] [Code Review] Provide method for testing GotoIfTime without changing localtime   David Vossel
• [asterisk-dev] [Code Review] Provide method for testing GotoIfTime without changing localtime   David Vossel
• [asterisk-dev] [Code Review] rtp timestamp to timeval calculation fix   David Vossel
• [asterisk-dev] [Code Review] rtp timestamp to timeval calculation fix   David Vossel
• [asterisk-dev] [Code Review] directmediapermit/directmediadeny support to restrict which peers can do directmedia based on ip address   David Vossel
• [asterisk-dev] [Code Review] rtp timestamp to timeval calculation fix   David Vossel
• [asterisk-dev] [Code Review] RFC compliant uri and display-name encode/decode   David Vossel
• [asterisk-dev] [Code Review] Include test API example in test_skel.c   David Vossel
• [asterisk-dev] [Code Review] Convert test in test_sched.c to test API   David Vossel
• [asterisk-dev] [Code Review] RFC compliant uri and display-name encode/decode   David Vossel
• [asterisk-dev] [Code Review] RFC compliant uri and display-name encode/decode   David Vossel
• [asterisk-dev] [Code Review] RFC compliant uri and display-name encode/decode   David Vossel
• [asterisk-dev] [Code Review] RFC compliant uri and display-name encode/decode   David Vossel
• [asterisk-dev] [Code Review] RFC compliant uri and display-name encode/decode   David Vossel
• [asterisk-dev] [Code Review] RFC compliant uri and display-name encode/decode   David Vossel
• [asterisk-dev] [Code Review] RFC compliant uri and display-name encode/decode   David Vossel
• [asterisk-dev] [Code Review] Convert test_substitution.c to unit test API   David Vossel
• [asterisk-dev] [Code Review] RFC compliant uri and display-name encode/decode   David Vossel
• [asterisk-dev] [Code Review] RFC compliant uri and display-name encode/decode   David Vossel
• [asterisk-dev] [Code Review] RFC compliant uri and display-name encode/decode   David Vossel
• [asterisk-dev] [Code Review] SIP: refactoring parsing functions and the addition of new unit tests   David Vossel
• [asterisk-dev] fallback to audio faxing when T.38 INVITE fail with 488/606 Not acceptable   Kristijan Vrban
• [asterisk-dev] fallback to audio faxing when T.38 INVITE fail with 488/606 Not acceptable   Kristijan Vrban
• [asterisk-dev] fallback to audio faxing when T.38 INVITE fail with 488/606 Not acceptable   Kristijan Vrban
• [asterisk-dev] fallback to audio faxing when T.38 INVITE fail with 488/606 Not acceptable   Kristijan Vrban
• [asterisk-dev] New Spandsp FAX Backend   Kristijan Vrban
• [asterisk-dev] asterisk 1.4 sent T.38 UDPTL only if DONT_OPTIMIZE is used   Kristijan Vrban
• [asterisk-dev] New Spandsp FAX Backend   Kristijan Vrban
• [asterisk-dev] New Spandsp FAX Backend   Kristijan Vrban
• [asterisk-dev] New Spandsp FAX Backend   Kristijan Vrban
• [asterisk-dev] fallback to audio faxing when T.38 INVITE fail with 488/606 Not acceptable   Kristijan Vrban
• [asterisk-dev] one way call bridge   Tomáš Vrána
• [asterisk-dev] Sub Address support for CDR logs   Will
• [asterisk-dev] AGI SAY DATETIME not working as expected   Bryant Zimmerman
• [asterisk-dev] AGI SAY DATETIME not working as expected   Bryant Zimmerman
• [asterisk-dev] AGI SAY DATETIME not working as expected   Bryant Zimmerman
• [asterisk-dev] Inquiry:Asterisk support KPML?   Zoa
• [asterisk-dev] Inquiry:Asterisk support KPML?   Zoa
• [asterisk-dev] Adventures in RTCP - a short report   Zoa
• [asterisk-dev] chan_datacard and RTCP   santosh chintalwar
• [asterisk-dev] [Code Review] parkinglots with different parking extensions   mnick at digium.com
• [asterisk-dev] [Code Review] parkinglots with different parking extensions   mnick at digium.com
• [asterisk-dev] [Code Review] CDR: Add Calling and Called Subaddress fields to CDR record   rmudgett at digium.com
• [asterisk-dev] [Code Review] CDR: Add Calling and Called Subaddress fields to CDR record   rmudgett at digium.com
• [asterisk-dev] [Code Review] CDR: Add Calling and Called Subaddress fields to CDR record   rmudgett at digium.com
• [asterisk-dev] [Code Review] Add notificiation of intrupted file and fix bug   thedavidfactor at gmail.com
• [asterisk-dev] [Code Review] Add notificiation of intrupted file and fix bug   thedavidfactor at gmail.com
• [asterisk-dev] [Code Review] Add notification of interrupted file and fix bug   thedavidfactor at gmail.com
• [asterisk-dev] [Code Review] Add notification of interrupted file and fix bug   thedavidfactor at gmail.com
• [asterisk-dev] [Code Review] Add command to send DTMF from ExternalIVR   thedavidfactor at gmail.com
• [asterisk-dev] [Code Review] Add command to send DTMF from ExternalIVR   thedavidfactor at gmail.com
• [asterisk-dev] [Code Review] code cleanup in app_senddtmf   thedavidfactor at gmail.com
• [asterisk-dev] Inquiry:Asterisk sip ?   hadi motamedi
• [asterisk-dev] Inquiry:Asterisk support KPML?   hadi motamedi
• [asterisk-dev] Inquiry:Asterisk support KPML?   hadi motamedi
• [asterisk-dev] Inquiry:Asterisk support KPML?   hadi motamedi
• [asterisk-dev] Inquiry:Asterisk support KPML?   hadi motamedi
• [asterisk-dev] Asterisk sip.conf dtmfmode=info?   hadi motamedi
• [asterisk-dev] jitterbuffer and PLC   nakaji
• [asterisk-dev] Asterisk application and SIP channel   nedo nodo
• [asterisk-dev] [code to use remote PBX's trunk]   khalid touati
• [asterisk-dev] [code to use remote PBX's trunk]   khalid touati
• [asterisk-dev] Requiring session timers and call drop   김의경

Last message date: Sun Jan 31 23:23:05 CST 2010
Archived on: Sun Jan 31 23:23:19 CST 2010

• Messages sorted by: [ thread ] [ subject ] [ date ]
• More info on this list...