September 2010 Archives by author

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Starting: Wed Sep 1 03:01:24 CDT 2010
Ending: Thu Sep 30 16:22:23 CDT 2010
Messages: 480

• [asterisk-dev] Fwd: T38 and codecs negotiation   Mike Dixon (STS)
• [asterisk-dev] __login_exec() + "agent-alreadyon"   ob .
• [asterisk-dev] ast_config and character escaping   Alexander
• [asterisk-dev] Asterisk support for media through TCP   Muhammad Ali
• [asterisk-dev] How to improve astdb? Any ideas?   Benny Amorsen
• [asterisk-dev] How to improve astdb? Any ideas?   Benny Amorsen
• [asterisk-dev] How to improve astdb? Any ideas?   Benny Amorsen
• [asterisk-dev] Arbitrary hangup handlers [patch]   Benny Amorsen
• [asterisk-dev] Arbitrary hangup handlers [patch]   Benny Amorsen
• [asterisk-dev] How to improve astdb? Any ideas?   Ron Arts
• [asterisk-dev] How to improve astdb? Any ideas?   Ron Arts
• [asterisk-dev] How to improve astdb? Any ideas?   Ron Arts
• [asterisk-dev] MYSQL(Set timeout x) not working   F B
• [asterisk-dev] rebuilding kmod-dahdi-linux i686 vs i386   Bob Beers
• [asterisk-dev] rebuilding kmod-dahdi-linux i686 vs i386   Bob Beers
• [asterisk-dev] Voice Mail System   Paul Belanger
• [asterisk-dev] Asterisk 1.6.2.11 possible bug   Paul Belanger
• [asterisk-dev] [Code Review] Rate limit astdb->sync() calls   Paul Belanger
• [asterisk-dev] MYSQL(Set timeout x) not working   Paul Belanger
• [asterisk-dev] downloads.asterisk.org not working on IPv6   Paul Belanger
• [asterisk-dev] OpenCL for improved performance transcoding and conferencing   Paul Belanger
• [asterisk-dev] New Feature Idea   Paul Belanger
• [asterisk-dev] New Feature Idea   Paul Belanger
• [asterisk-dev] Fwd: T38 and codecs negotiation   Theo Belder
• [asterisk-dev] Asterisk 1.8 and T.38 gatewaying   Niccolò Belli
• [asterisk-dev] Asterisk 1.8 and T.38 gatewaying   Niccolò Belli
• [asterisk-dev] Hi, no more astertest project ??   Benaiad
• [asterisk-dev] [Video / SIP] Codec negotiation passthrough ?   Nicolas Bourbaki
• [asterisk-dev] [Video / SIP] Codec negotiation passthrough ?   Nicolas Bourbaki
• [asterisk-dev] [Video / SIP] Codec negotiation passthrough ?   Nicolas Bourbaki
• [asterisk-dev] [Code Review] 1.4 manager configuration reset to default values before config files are reloaded   Brett Bryant
• [asterisk-dev] [Code Review] New testsuite test for call files and pbx_spool   Brett Bryant
• [asterisk-dev] [Code Review] Music on hold in 1.8 doesn't recover cleanly when a moh file fails to load   Brett Bryant
• [asterisk-dev] [Code Review] New testsuite test for call files and pbx_spool   Brett Bryant
• [asterisk-dev] [Code Review] New testsuite test for call files and pbx_spool   Brett Bryant
• [asterisk-dev] [Code Review] VERBOSE message shows up on console when 'debug' enabled in logger.conf   Brett Bryant
• [asterisk-dev] [Code Review] VERBOSE message shows up on console when 'debug' enabled in logger.conf   Brett Bryant
• [asterisk-dev] [Code Review] VERBOSE message shows up on console when 'debug' enabled in logger.conf   Brett Bryant
• [asterisk-dev] [Code Review] VERBOSE message shows up on console when 'debug' enabled in logger.conf   Brett Bryant
• [asterisk-dev] [Code Review] VERBOSE message shows up on console when 'debug' enabled in logger.conf   Brett Bryant
• [asterisk-dev] [Code Review] VERBOSE message shows up on console when 'debug' enabled in logger.conf   Brett Bryant
• [asterisk-dev] [Code Review] VERBOSE message shows up on console when 'debug' enabled in logger.conf   Brett Bryant
• [asterisk-dev] [Code Review] VERBOSE message shows up on console when 'debug' enabled in logger.conf   Brett Bryant
• [asterisk-dev] [Code Review] Newchannel AMI event is missing during masquerading process.   Brett Bryant
• [asterisk-dev] [Code Review] Newchannel AMI event is missing during masquerading process.   Brett Bryant
• [asterisk-dev] [Code Review] Newchannel AMI event is missing during masquerading process.   Brett Bryant
• [asterisk-dev] [Code Review] res_agi.c:handle_getvariablefull() could recursively lock a channel and not release it if an argument is the current channel's name.   Brett Bryant
• [asterisk-dev] [Code Review] Fix SRTP for changing SSRC and multiple a=crypto SDP lines   Russell Bryant
• [asterisk-dev] [Code Review] ADSI and Crypto backwards compatibility with 1.6.2   Russell Bryant
• [asterisk-dev] [Code Review] Add parking extension for non-default parking lots   Russell Bryant
• [asterisk-dev] [Code Review] SIP: authenticate OPTIONS requests just like we would an INVITE   Russell Bryant
• [asterisk-dev] [Code Review] 1.4 manager configuration reset to default values before config files are reloaded   Russell Bryant
• [asterisk-dev] [Code Review] handle_request_subscribe is too slow cause of iterating through all sip dialogs   Russell Bryant
• [asterisk-dev] [Code Review] Allow inheritance of CHANNEL() write changes for Local/n queue members   Russell Bryant
• [asterisk-dev] [Code Review] Allow inheritance of CHANNEL() write changes for Local/n queue members   Russell Bryant
• [asterisk-dev] [Code Review] New testsuite test for call files and pbx_spool   Russell Bryant
• [asterisk-dev] [Code Review] Allow inheritance of CHANNEL() write changes for Local/n queue members   Russell Bryant
• [asterisk-dev] [Code Review] New testsuite test for call files and pbx_spool   Russell Bryant
• [asterisk-dev] [Code Review] handle_request_subscribe is too slow cause of iterating through all sip dialogs   Russell Bryant
• [asterisk-dev] [Code Review] handle_request_subscribe is too slow cause of iterating through all sip dialogs   Russell Bryant
• [asterisk-dev] [Code Review] Fix how mutexes are created/destroyed on platforms which need constructors   Russell Bryant
• [asterisk-dev] [Code Review] Add parking extension for non-default parking lots   Russell Bryant
• [asterisk-dev] How to improve astdb? Any ideas?   Russell Bryant
• [asterisk-dev] [Code Review] Rate limit astdb->sync() calls   Russell Bryant
• [asterisk-dev] How to improve astdb? Any ideas?   Russell Bryant
• [asterisk-dev] [Code Review] Rate limit astdb->sync() calls   Russell Bryant
• [asterisk-dev] [Code Review] Music on hold in 1.8 doesn't recover cleanly when a moh file fails to load   Russell Bryant
• [asterisk-dev] [Code Review] Rate limit astdb->sync() calls   Russell Bryant
• [asterisk-dev] [Code Review] Music on hold in 1.8 doesn't recover cleanly when a moh file fails to load   Russell Bryant
• [asterisk-dev] [Code Review] New testsuite test for call files and pbx_spool   Russell Bryant
• [asterisk-dev] [Code Review] Make sure that we handle possible format error in ast_openstream_full() - channel.c   Russell Bryant
• [asterisk-dev] [Code Review] Rate limit astdb->sync() calls   Russell Bryant
• [asterisk-dev] [Code Review] Rate limit astdb->sync() calls   Russell Bryant
• [asterisk-dev] [Code Review] Rate limit astdb->sync() calls   Russell Bryant
• [asterisk-dev] [Code Review] Rate limit astdb->sync() calls   Russell Bryant
• [asterisk-dev] [Code Review] Make sure that we handle possible format error in ast_openstream_full() - channel.c   Russell Bryant
• [asterisk-dev] Hi all, is struct ast_frame encoded by any codec?   Russell Bryant
• [asterisk-dev] confuse no the method audiohook mix read side and write side data   Russell Bryant
• [asterisk-dev] [Code Review] Allow inheritance of CHANNEL() write changes for Local/n queue members   Russell Bryant
• [asterisk-dev] [Code Review] Allow inheritance of CHANNEL() write changes for Local/n queue members   Russell Bryant
• [asterisk-dev] [Code Review] VERBOSE message shows up on console when 'debug' enabled in logger.conf   Russell Bryant
• [asterisk-dev] [Code Review] chan_iax2: load config before registering any functions   Russell Bryant
• [asterisk-dev] [Code Review] Allow inheritance of CHANNEL() write changes for Local/n queue members   Russell Bryant
• [asterisk-dev] [Code Review] chan_iax2: load config before registering any functions   Russell Bryant
• [asterisk-dev] [Code Review] chan_iax2: load config before registering any functions   Russell Bryant
• [asterisk-dev] Question about #17888 and why it has been blocked   Russell Bryant
• [asterisk-dev] Question about #17888 and why it has been blocked   Russell Bryant
• [asterisk-dev] confuse no the method audiohook mix read side and write side data   Russell Bryant
• [asterisk-dev] 1.8 Beta 5 and g729 codec?   Russell Bryant
• [asterisk-dev] 1.8 Beta 5 and g729 codec?   Russell Bryant
• [asterisk-dev] [Code Review] VERBOSE message shows up on console when 'debug' enabled in logger.conf   Russell Bryant
• [asterisk-dev] [Code Review] Add parking extension for non-default parking lots   Russell Bryant
• [asterisk-dev] [Code Review] VERBOSE message shows up on console when 'debug' enabled in logger.conf   Russell Bryant
• [asterisk-dev] [Code Review] Add parking extension for non-default parking lots   Russell Bryant
• [asterisk-dev] [Code Review] VERBOSE message shows up on console when 'debug' enabled in logger.conf   Russell Bryant
• [asterisk-dev] [Code Review] Add parking extension for non-default parking lots   Russell Bryant
• [asterisk-dev] Change Control Procedures   Russell Bryant
• [asterisk-dev] Analyze inband DTMF in channel driver   Russell Bryant
• [asterisk-dev] REFER and NOTIFY behavior   Russell Bryant
• [asterisk-dev] [Code Review] Set the queues.conf autofill option default to "yes"   Russell Bryant
• [asterisk-dev] [Code Review] Set the queues.conf autofill option default to "yes"   Russell Bryant
• [asterisk-dev] [Code Review] Set the queues.conf autofill option default to "yes"   Russell Bryant
• [asterisk-dev] [Code Review] Set the queues.conf autofill option default to "yes"   Russell Bryant
• [asterisk-dev] REFER and NOTIFY behavior   Russell Bryant
• [asterisk-dev] [Code Review] VERBOSE message shows up on console when 'debug' enabled in logger.conf   Russell Bryant
• [asterisk-dev] [Code Review] awesome review   Russell Bryant
• [asterisk-dev] [Code Review] Don't ast_strdupa() from within the arguments to a function   Russell Bryant
• [asterisk-dev] [Code Review] awesome review   Russell Bryant
• [asterisk-dev] [Code Review] chan_iax2 owner channel locking and memory leak   Russell Bryant
• [asterisk-dev] [Code Review] VERBOSE message shows up on console when 'debug' enabled in logger.conf   Russell Bryant
• [asterisk-dev] [Code Review] VERBOSE message shows up on console when 'debug' enabled in logger.conf   Russell Bryant
• [asterisk-dev] [Code Review] Prevent 'Bad Magic Number' caused when a channel is optimized out by masquerade   Russell Bryant
• [asterisk-dev] [Code Review] awesome review   Russell Bryant
• [asterisk-dev] [Code Review] res_jabber: Disable debug by default   Russell Bryant
• [asterisk-dev] [Code Review] Avoid infinite loop with certain local channel connected line updates   Russell Bryant
• [asterisk-dev] [Code Review] Avoid infinite loop with certain local channel connected line updates   Russell Bryant
• [asterisk-dev] [Code Review] Avoid infinite loop with certain local channel connected line updates   Russell Bryant
• [asterisk-dev] [Code Review] Avoid infinite loop with certain local channel connected line updates   Russell Bryant
• [asterisk-dev] encounter "no reply to our critical packet" on 1.8.0beta5   Russell Bryant
• [asterisk-dev] Asterisk 1.8 and T.38 gatewaying   Russell Bryant
• [asterisk-dev] Asterisk 1.8 and T.38 gatewaying   Russell Bryant
• [asterisk-dev] [Code Review] Don't let a Local channel get bridged to itself   Russell Bryant
• [asterisk-dev] [Code Review] Newchannel AMI event is missing during masquerading process.   Russell Bryant
• [asterisk-dev] [Code Review] Manager listens even when not enabled.   Russell Bryant
• [asterisk-dev] [Code Review] Newchannel AMI event is missing during masquerading process.   Russell Bryant
• [asterisk-dev] [Code Review] Newchannel AMI event is missing during masquerading process.   Russell Bryant
• [asterisk-dev] 16khz/8khz Translation path   Russell Bryant
• [asterisk-dev] New Feature Idea   Russell Bryant
• [asterisk-dev] [Code Review] Use paths from ./configure instead of overriding in Makefile   Russell Bryant
• [asterisk-dev] [Code Review] Use paths from ./configure instead of overriding in Makefile   Russell Bryant
• [asterisk-dev] Dadhi_dummy disabled   Russell Bryant
• [asterisk-dev] [Code Review] res_agi.c:handle_getvariablefull() could recursively lock a channel and not release it if an argument is the current channel's name.   Russell Bryant
• [asterisk-dev] [Code Review] Don't crash on a bogus Set()   Russell Bryant
• [asterisk-dev] [Code Review] Add a demo utility for rapidly sending manager commands   Russell Bryant
• [asterisk-dev] [Code Review] Add CLI-command "cdr pgsql status" based on "cdr mysql status"   Russell Bryant
• [asterisk-dev] [Code Review] Change RFC2833 DTMF event duration on end to report actual elapsed time   Russell Bryant
• [asterisk-dev] [Code Review] Change RFC2833 DTMF event duration on end to report actual elapsed time   Russell Bryant
• [asterisk-dev] [Code Review] Add CLI-command "cdr pgsql status" based on "cdr mysql status"   Russell Bryant
• [asterisk-dev] [Code Review] Reduce references to global "chans" array in dahdi-base.c   Tzafrir Cohen
• [asterisk-dev] How to improve astdb? Any ideas?   Tzafrir Cohen
• [asterisk-dev] New Feature Idea   Tzafrir Cohen
• [asterisk-dev] DAHDI FXO port only recognizes the "S" extension   Tzafrir Cohen
• [asterisk-dev] OpenCL for improved performance transcoding and conferencing   Chris Coleman
• [asterisk-dev] OpenCL for improved performance transcoding and conferencing   Chris Coleman
• [asterisk-dev] OpenCL for improved performance transcoding and conferencing   Chris Coleman
• [asterisk-dev] OpenCL for improved performance transcoding and conferencing   Chris Coleman
• [asterisk-dev] OpenCL for improved performance, transcoding and conferencing   Chris Coleman
• [asterisk-dev] OpenCL for improved performance transcoding and conferencing   Chris Coleman
• [asterisk-dev] OpenCL for improved performance transcoding and conferencing   Chris Coleman
• [asterisk-dev] Problems compiling Asterisk   Saúl Ibarra Corretgé
• [asterisk-dev] [Code Review] Rate limit astdb->sync() calls   Saúl Ibarra Corretgé
• [asterisk-dev] 1.8 Beta 5 and addons   Saúl Ibarra Corretgé
• [asterisk-dev] Compiling a c++ function module   Andre Courchesne
• [asterisk-dev] [Code Review] SIP: authenticate OPTIONS requests just like we would an INVITE   Klaus Darilion
• [asterisk-dev] [Code Review] PineTree:: Matching SIP peers beyond SIP proxy   Klaus Darilion
• [asterisk-dev] [Code Review] PineTree:: Matching SIP peers beyond SIP proxy   Klaus Darilion
• [asterisk-dev] [Code Review] SIP: authenticate OPTIONS requests just like we would an INVITE   Klaus Darilion
• [asterisk-dev] REFER and NOTIFY behavior   Klaus Darilion
• [asterisk-dev] REFER and NOTIFY behavior   Klaus Darilion
• [asterisk-dev] [Code Review] awesome review   Klaus Darilion
• [asterisk-dev] Asterisk Cnfiguration and Open VPN Asterisk Setup   Klaus Darilion
• [asterisk-dev] REFER and NOTIFY behavior   Klaus Darilion
• [asterisk-dev] [Code Review] Prevent 'Bad Magic Number' caused when a channel is optimized out by masquerade   Alec Davis
• [asterisk-dev] [Code Review] Prevent 'Bad Magic Number' caused when a channel is optimized out by masquerade   Alec Davis
• [asterisk-dev] [Code Review] Prevent 'Bad Magic Number' caused when a channel is optimized out by masquerade   Alec Davis
• [asterisk-dev] [Code Review] Don't let a Local channel get bridged to itself   Alec Davis
• [asterisk-dev] [Code Review] Don't let a Local channel get bridged to itself   Alec Davis
• [asterisk-dev] Troubleshooting compile of dahdi-linux-complete-2.4.0/tools/xpp/astribank   Gabriel Dungan
• [asterisk-dev] [Code Review] Fix how mutexes are created/destroyed on platforms which need constructors   Kevin Fleming
• [asterisk-dev] [Code Review] Eliminate memmove from pbx_spool implementation   Kevin Fleming
• [asterisk-dev] [Code Review] Eliminate memmove from pbx_spool implementation   Kevin Fleming
• [asterisk-dev] [Code Review] SIP: proper handling of forked outbound INVITE requests.   Kevin Fleming
• [asterisk-dev] [Video / SIP] Codec negotiation passthrough ?   Kevin P. Fleming
• [asterisk-dev] How to improve astdb? Any ideas?   Kevin P. Fleming
• [asterisk-dev] [Code Review] Rate limit astdb->sync() calls   Kevin P. Fleming
• [asterisk-dev] t.38 passthrough filters redudant packages ?   Kevin P. Fleming
• [asterisk-dev] t.38 passthrough filters redudant packages ?   Kevin P. Fleming
• [asterisk-dev] t.38 passthrough filters redudant packages ?   Kevin P. Fleming
• [asterisk-dev] Compiling a c++ function module   Kevin P. Fleming
• [asterisk-dev] gsmtolin translator seems miss frameout function.   Kevin P. Fleming
• [asterisk-dev] [Code Review] Append two container for dialogs to delete and for rtp timeout checks to replace all dialogs container in ao2 callback for dialogs_needdestroy   Kevin P. Fleming
• [asterisk-dev] [Code Review] awesome review   Kevin P. Fleming
• [asterisk-dev] downloads.asterisk.org not working on IPv6   Kevin P. Fleming
• [asterisk-dev] Asterisk 1.8 and T.38 gatewaying   Kevin P. Fleming
• [asterisk-dev] OpenCL for improved performance transcoding and conferencing   Kevin P. Fleming
• [asterisk-dev] OpenCL for improved performance, transcoding and conferencing   Kevin P. Fleming
• [asterisk-dev] 16khz/8khz Translation path   Kevin P. Fleming
• [asterisk-dev] New Feature Idea   Kevin P. Fleming
• [asterisk-dev] humble request for review - cdr pgsql status CLI command   Kevin P. Fleming
• [asterisk-dev] [Code Review] Use paths from ./configure instead of overriding in Makefile   Kevin P. Fleming
• [asterisk-dev] Voice Mail System   Frenette, Rob
• [asterisk-dev] Help me Out!!!!   Rob Fugina
• [asterisk-dev] [Code Review] Make sure that we handle possible format error in ast_openstream_full() - channel.c   Olle E Johansson
• [asterisk-dev] [Code Review] Make sure that we handle possible format error in ast_openstream_full() - channel.c   Olle E Johansson
• [asterisk-dev] [Code Review] Add CLI-command "cdr pgsql status" based on "cdr mysql status"   Olle E Johansson
• [asterisk-dev] [Code Review] Add CLI-command "cdr pgsql status" based on "cdr mysql status"   Olle E Johansson
• [asterisk-dev] [Code Review] New Manager Action: MeetmeListRooms   Olle E Johansson
• [asterisk-dev] [Code Review] handle_request_subscribe is too slow cause of iterating through all sip dialogs   Olle E. Johansson
• [asterisk-dev] [Code Review] PineTree:: Matching SIP peers beyond SIP proxy   Olle E. Johansson
• [asterisk-dev] How to improve astdb? Any ideas?   Olle E. Johansson
• [asterisk-dev] How to improve astdb? Any ideas?   Olle E. Johansson
• [asterisk-dev] [EDVX] r506 - in asterisk-1.4: asterisk-1.4-edv asterisk-1.4-edv/channels asterisk-1.4-edv/main diffs   Olle E. Johansson
• [asterisk-dev] REFER and NOTIFY behavior   Olle E. Johansson
• [asterisk-dev] [Code Review] SIP: for an INVITE transaction, treat all 2XX responses the same as a 200   Olle E. Johansson
• [asterisk-dev] humble request for review - cdr pgsql status CLI command   Olle E. Johansson
• [asterisk-dev] [Code Review] SIP: for an INVITE transaction, treat all 2XX responses the same as a 200   Olle E. Johansson
• [asterisk-dev] humble request for review - cdr pgsql status CLI command   Olle E. Johansson
• [asterisk-dev] [Code Review] SIP: for an INVITE transaction, treat all 2XX responses the same as a 200   Olle E. Johansson
• [asterisk-dev] [Code Review] Use paths from ./configure instead of overriding in Makefile   Olle E. Johansson
• [asterisk-dev] RTCP question!   Olle E. Johansson
• [asterisk-dev] Dadhi_dummy disabled   Olle E. Johansson
• [asterisk-dev] Dadhi_dummy disabled   Olle E. Johansson
• [asterisk-dev] Dadhi_dummy disabled   Olle E. Johansson
• [asterisk-dev] Dadhi_dummy disabled - init.d script   Olle E. Johansson
• [asterisk-dev] [Code Review] SIP: proper handling of forked outbound INVITE requests.   Olle E. Johansson
• [asterisk-dev] SIP Registration Failing randomly (analyzed)   Olle E. Johansson
• [asterisk-dev] Fwd: [Code Review] Add CLI-command "cdr pgsql status" based on "cdr mysql status"   Olle E. Johansson
• [asterisk-dev] [Code Review] Add CLI-command "cdr pgsql status" based on "cdr mysql status"   Olle E. Johansson
• [asterisk-dev] Channel Signalling   Arnaldo Giacomitti Junior
• [asterisk-dev] dahdi-tools rev. 9067   Max Khon
• [asterisk-dev] dahdi-tools rev. 9067   Max Khon
• [asterisk-dev] [Code Review] awesome review   Vadim Lebedev
• [asterisk-dev] Hi, no more astertest project ??   Min Lee
• [asterisk-dev] [Code Review] Fix how mutexes are created/destroyed on platforms which need constructors   Tilghman Lesher
• [asterisk-dev] [Code Review] Fix how mutexes are created/destroyed on platforms which need constructors   Tilghman Lesher
• [asterisk-dev] [Code Review] Fix how mutexes are created/destroyed on platforms which need constructors   Tilghman Lesher
• [asterisk-dev] ast_config and character escaping   Tilghman Lesher
• [asterisk-dev] ast_config and character escaping   Tilghman Lesher
• [asterisk-dev] [Code Review] Eliminate memmove from pbx_spool implementation   Tilghman Lesher
• [asterisk-dev] [Code Review] Eliminate memmove from pbx_spool implementation   Tilghman Lesher
• [asterisk-dev] [Code Review] Eliminate memmove from pbx_spool implementation   Tilghman Lesher
• [asterisk-dev] [Code Review] Eliminate memmove from pbx_spool implementation   Tilghman Lesher
• [asterisk-dev] Hi all, is struct ast_frame encoded by any codec?   Tilghman Lesher
• [asterisk-dev] 1.8 Beta 5 and g729 codec?   Tilghman Lesher
• [asterisk-dev] Arbitrary hangup handlers [patch]   Tilghman Lesher
• [asterisk-dev] Arbitrary hangup handlers [patch]   Tilghman Lesher
• [asterisk-dev] confuse with this syntax: int gsm_decode P3((s, c, target), gsm s, gsm_byte * c, gsm_signal * target)   Tilghman Lesher
• [asterisk-dev] Arbitrary hangup handlers [patch]   Tilghman Lesher
• [asterisk-dev] [Code Review] Don't ast_strdupa() from within the arguments to a function   Tilghman Lesher
• [asterisk-dev] [Code Review] Use paths from ./configure instead of overriding in Makefile   Tilghman Lesher
• [asterisk-dev] [Code Review] Use paths from ./configure instead of overriding in Makefile   Tilghman Lesher
• [asterisk-dev] [Code Review] Use paths from ./configure instead of overriding in Makefile   Tilghman Lesher
• [asterisk-dev] [Code Review] Use paths from ./configure instead of overriding in Makefile   Tilghman Lesher
• [asterisk-dev] [Code Review] Add CLI-command "cdr pgsql status" based on "cdr mysql status"   Tilghman Lesher
• [asterisk-dev] Asterisk 1.6.2.11 possible bug   Andre Luis
• [asterisk-dev] Problems compiling Asterisk   Doug Lytle
• [asterisk-dev] Turning on mode=new by default in say.conf   Leif Madsen
• [asterisk-dev] New Feature Idea   Leif Madsen
• [asterisk-dev] How commiting new features to code   Leif Madsen
• [asterisk-dev] DAHDI FXO port only recognizes the "S" extension   Leif Madsen
• [asterisk-dev] Analyze inband DTMF in channel driver   Artem Makhutov
• [asterisk-dev] sip_pvt call origin information   Mark Michelson
• [asterisk-dev] Code review: app_queue: Give members a penalty time for not answering   Mark Michelson
• [asterisk-dev] Asterisk Cnfiguration and Open VPN Asterisk Setup   Ahmed Muddasir
• [asterisk-dev] How to improve astdb? Any ideas?   Mark Murawski
• [asterisk-dev] How to improve astdb? Any ideas?   Mark Murawski
• [asterisk-dev] How to improve astdb? Any ideas?   Mark Murawski
• [asterisk-dev] How to improve astdb? Any ideas?   Mark Murawski
• [asterisk-dev] How to improve astdb? Any ideas?   Mark Murawski
• [asterisk-dev] Arbitrary hangup handlers [patch]   Mark Murawski
• [asterisk-dev] Arbitrary hangup handlers [patch]   Mark Murawski
• [asterisk-dev] Arbitrary hangup handlers [patch]   Mark Murawski
• [asterisk-dev] Arbitrary hangup handlers [patch]   Mark Murawski
• [asterisk-dev] Arbitrary hangup handlers [patch]   Mark Murawski
• [asterisk-dev] [Code Review] awesome review   Mark Murawski
• [asterisk-dev] Code review: app_queue: Give members a penalty time for not answering   Håkon Nessjøen
• [asterisk-dev] Code review: app_queue: Give members a penalty time for not answering   Håkon Nessjøen
• [asterisk-dev] Code review: app_queue: Give members a penalty time for not answering   Håkon Nessjøen
• [asterisk-dev] [Code Review] New Manager Action: MeetmeListRooms   Matthew Nicholson
• [asterisk-dev] [Code Review] enable SMS polling in chan_mobile   Matthew Nicholson
• [asterisk-dev] Unsolicited MWI   Kevin Pare
• [asterisk-dev] [Code Review] Add parking extension for non-default parking lots   Jeff Peeler
• [asterisk-dev] [Code Review] Add LWP for core show threads output   Jeff Peeler
• [asterisk-dev] [Code Review] Add parking extension for non-default parking lots   Jeff Peeler
• [asterisk-dev] [Code Review] Check if message can be forwarded to mailbox before prompting for prepending   Jeff Peeler
• [asterisk-dev] [Code Review] Add parking extension for non-default parking lots   Jeff Peeler
• [asterisk-dev] [Code Review] Add parking extension for non-default parking lots   Jeff Peeler
• [asterisk-dev] [Code Review] Add parking extension for non-default parking lots   Jeff Peeler
• [asterisk-dev] [Code Review] Change RFC2833 DTMF event duration on end to report actual elapsed time   Jeff Peeler
• [asterisk-dev] [Code Review] Change RFC2833 DTMF event duration on end to report actual elapsed time   Jeff Peeler
• [asterisk-dev] [Code Review] Change RFC2833 DTMF event duration on end to report actual elapsed time   Jeff Peeler
• [asterisk-dev] [Code Review] Change RFC2833 DTMF event duration on end to report actual elapsed time   Jeff Peeler
• [asterisk-dev] [Video / SIP] Codec negotiation passthrough ?   Simon Perreault
• [asterisk-dev] How to improve astdb? Any ideas?   Simon Perreault
• [asterisk-dev] [Code Review] Rate limit astdb->sync() calls   Simon Perreault
• [asterisk-dev] [Code Review] Rate limit astdb->sync() calls   Simon Perreault
• [asterisk-dev] [Code Review] SIP: for an INVITE transaction, treat all 2XX responses the same as a 200   Simon Perreault
• [asterisk-dev] [Code Review] SIP: for an INVITE transaction, treat all 2XX responses the same as a 200   Simon Perreault
• [asterisk-dev] IAX2 call leg optimization behind firewall   Wolfgang Pichler
• [asterisk-dev] Q931_IE_TIME_DATE in libpri - only dump of IE available   Wolfgang Pichler
• [asterisk-dev] Q931_IE_TIME_DATE in libpri - only dump of IE available   Wolfgang Pichler
• [asterisk-dev] Q931_IE_TIME_DATE in libpri - only dump of IE available   Wolfgang Pichler
• [asterisk-dev] How to improve astdb? Any ideas?   Matt Riddell
• [asterisk-dev] dahdi_device representation   Shaun Ruffell
• [asterisk-dev] [Code Review] Reduce references to global "chans" array in dahdi-base.c   Shaun Ruffell
• [asterisk-dev] [Code Review] Reduce references to global "chans" array in dahdi-base.c   Shaun Ruffell
• [asterisk-dev] [Code Review] Reduce references to global "chans" array in dahdi-base.c   Shaun Ruffell
• [asterisk-dev] [Code Review] Remove 'chans' array from dahdi-base.c   Shaun Ruffell
• [asterisk-dev] [Code Review] Mainly remove bigzaplock, teach core timing about DAHDI_CHUNKSIZE > 8, and friends.   Shaun Ruffell
• [asterisk-dev] Troubleshooting compile of dahdi-linux-complete-2.4.0/tools/xpp/astribank   Shaun Ruffell
• [asterisk-dev] Dadhi_dummy disabled   Shaun Ruffell
• [asterisk-dev] Dadhi_dummy disabled   Shaun Ruffell
• [asterisk-dev] [DAHDI-linux 0014862]: When using DESTDIR, depmod is run on the host, not on the staging directory   Shaun Ruffell
• [asterisk-dev] ExternalIvr documentation fix   David Ruggles
• [asterisk-dev] How commiting new features to code   David Ruggles
• [asterisk-dev] question about handle_request_subscribe in chan_sip.c V1.6.2   Stefan Schmidt
• [asterisk-dev] question about handle_request_subscribe in chan_sip.c V1.6.2   Stefan Schmidt
• [asterisk-dev] [Code Review] handle_request_subscribe is too slow cause of iterating through all sip dialogs   Stefan Schmidt
• [asterisk-dev] How to improve astdb? Any ideas?   Stefan Schmidt
• [asterisk-dev] How to improve astdb? Any ideas?   Stefan Schmidt
• [asterisk-dev] How to improve astdb? Any ideas?   Stefan Schmidt
• [asterisk-dev] How to improve astdb? Any ideas?   Stefan Schmidt
• [asterisk-dev] How to improve astdb? Any ideas?   Stefan Schmidt
• [asterisk-dev] How to improve astdb? Any ideas?   Stefan Schmidt
• [asterisk-dev] How to improve astdb? Any ideas?   Stefan Schmidt
• [asterisk-dev] [Code Review] Rate limit astdb->sync() calls   Stefan Schmidt
• [asterisk-dev] Question about #17888 and why it has been blocked   Stefan Schmidt
• [asterisk-dev] Question about #17888 and why it has been blocked   Stefan Schmidt
• [asterisk-dev] [Code Review] Prevent 'Bad Magic Number' caused when a channel is optimized out by masquerade   Stefan Schmidt
• [asterisk-dev] OpenCL for improved performance, transcoding and conferencing   Stefan Schmidt
• [asterisk-dev] [patch] Allow user more control over parked calls that expire from a one touch park   Mitch Sharp
• [asterisk-dev] [feedback request] Allow user more control over parked callsthat expire from a one touch park   Mitch Sharp
• [asterisk-dev] How commiting new features to code   M Shokuie
• [asterisk-dev] How commiting new features to code   M Shokuie
• [asterisk-dev] Channel Signalling   Moises Silva
• [asterisk-dev] 16khz/8khz Translation path   Moises Silva
• [asterisk-dev] New Feature Idea   Nir Simionovich
• [asterisk-dev] New Feature Idea   Nir Simionovich
• [asterisk-dev] New Feature Idea   Nir Simionovich
• [asterisk-dev] New Feature Idea   Nir Simionovich
• [asterisk-dev] New Feature Idea   Nir Simionovich
• [asterisk-dev] New Feature Idea   Nir Simionovich
• [asterisk-dev] New Feature Idea   Nir Simionovich
• [asterisk-dev] Asterisk Cnfiguration and Open VPN Asterisk Setup   Jamuel Starkey
• [asterisk-dev] libpri 1.4.11.4 Now Available   Asterisk Development Team
• [asterisk-dev] libpri 1.4.12-beta2 Now Available   Asterisk Development Team
• [asterisk-dev] [draft] DAHDI-linux & DAHDI-tools 2.4.0 Release Announcement   Asterisk Development Team
• [asterisk-dev] Asterisk 1.8.0-beta5 Now Available   Asterisk Development Team
• [asterisk-dev] Asterisk 1.4.36 Now Available   Asterisk Development Team
• [asterisk-dev] Asterisk 1.6.2.12 Now Available   Asterisk Development Team
• [asterisk-dev] Asterisk 1.6.2.13 Now Available (Re-Releast of 1.6.2.12)   Asterisk Development Team
• [asterisk-dev] Asterisk 1.4.37-rc1 Now Available   Asterisk Development Team
• [asterisk-dev] Asterisk 1.6.2.14-rc1 Now Available   Asterisk Development Team
• [asterisk-dev] Asterisk 1.8.0 Release Candidate 2 Now Available   Asterisk Development Team
• [asterisk-dev] Unscheduled service outage for various Asterisk community services   Asterisk Development Team
• [asterisk-dev] SIP Registration Failing randomly (analyzed)   Timo Teräs
• [asterisk-dev] SIP Registration Failing randomly (analyzed)   Timo Teräs
• [asterisk-dev] Developers: Hey there! AstriDevCon signups, please.   John Todd
• [asterisk-dev] Enhanced SS7 support in Asterisk-1.8 ?   Pavel Troller
• [asterisk-dev] t.38 passthrough filters redudant packages ?   Steve Underwood
• [asterisk-dev] OpenCL for improved performance transcoding and conferencing   Steve Underwood
• [asterisk-dev] OpenCL for improved performance transcoding and conferencing   Steve Underwood
• [asterisk-dev] OpenCL for improved performance transcoding and conferencing   Steve Underwood
• [asterisk-dev] OpenCL for improved performance, transcoding and conferencing   Steve Underwood
• [asterisk-dev] [Code Review] SIP: authenticate OPTIONS requests just like we would an INVITE   David Vossel
• [asterisk-dev] [Code Review] Append two container for dialogs to delete and for rtp timeout checks to replace all dialogs container in ao2 callback for dialogs_needdestroy   David Vossel
• [asterisk-dev] [Code Review] Append two container for dialogs to delete and for rtp timeout checks to replace all dialogs container in ao2 callback for dialogs_needdestroy   David Vossel
• [asterisk-dev] [Code Review] Append two container for dialogs to delete and for rtp timeout checks to replace all dialogs container in ao2 callback for dialogs_needdestroy   David Vossel
• [asterisk-dev] [Code Review] Append two container for dialogs to delete and for rtp timeout checks to replace all dialogs container in ao2 callback for dialogs_needdestroy   David Vossel
• [asterisk-dev] [Code Review] Append two container for dialogs to delete and for rtp timeout checks to replace all dialogs container in ao2 callback for dialogs_needdestroy   David Vossel
• [asterisk-dev] [Code Review] awesome review   David Vossel
• [asterisk-dev] [Code Review] awesome review   David Vossel
• [asterisk-dev] [Code Review] Append two container for dialogs to delete and for rtp timeout checks to replace all dialogs container in ao2 callback for dialogs_needdestroy   David Vossel
• [asterisk-dev] [Code Review] Append two container for dialogs to delete and for rtp timeout checks to replace all dialogs container in ao2 callback for dialogs_needdestroy   David Vossel
• [asterisk-dev] [Code Review] awesome review   David Vossel
• [asterisk-dev] [Code Review] awesome review   David Vossel
• [asterisk-dev] [Code Review] awesome review   David Vossel
• [asterisk-dev] [Code Review] awesome review   David Vossel
• [asterisk-dev] [Code Review] Append two container for dialogs to delete and for rtp timeout checks to replace all dialogs container in ao2 callback for dialogs_needdestroy   David Vossel
• [asterisk-dev] [Code Review] SIP: for an INVITE transaction, treat all 2XX responses the same as a 200   David Vossel
• [asterisk-dev] [Code Review] SIP: for an INVITE transaction, treat all 2XX responses the same as a 200   David Vossel
• [asterisk-dev] [Code Review] handle_request_subscribe is too slow cause of iterating through all sip dialogs   David Vossel
• [asterisk-dev] [Code Review] handle_request_subscribe is too slow cause of iterating through all sip dialogs   David Vossel
• [asterisk-dev] [Code Review] Newchannel AMI event is missing during masquerading process.   David Vossel
• [asterisk-dev] [Code Review] Newchannel AMI event is missing during masquerading process.   David Vossel
• [asterisk-dev] 16khz/8khz Translation path   David Vossel
• [asterisk-dev] 16khz/8khz Translation path   David Vossel
• [asterisk-dev] RTCP question!   David Vossel
• [asterisk-dev] [Code Review] SIP: for an INVITE transaction, treat all 2XX responses the same as a 200   David Vossel
• [asterisk-dev] [Code Review] SIP: proper handling of forked outbound INVITE requests.   David Vossel
• [asterisk-dev] [Code Review] SIP: proper handling of forked outbound INVITE requests.   David Vossel
• [asterisk-dev] [Code Review] Change RFC2833 DTMF event duration on end to report actual elapsed time   David Vossel
• [asterisk-dev] t.38 passthrough filters redudant packages ?   Kristijan Vrban
• [asterisk-dev] t.38 passthrough filters redudant packages ?   Kristijan Vrban
• [asterisk-dev] t.38 passthrough filters redudant packages ?   Kristijan Vrban
• [asterisk-dev] Change Control Procedures   Dave WOOLLEY
• [asterisk-dev] Help me Out!!!! (SCAM)   Dave WOOLLEY
• [asterisk-dev] [DAHDI-linux 0014862]: When using DESTDIR, depmod is run on the host, not on the staging directory   Dave WOOLLEY
• [asterisk-dev] dahdi-tools rev. 9067   Will
• [asterisk-dev] dahdi-tools rev. 9067   Will
• [asterisk-dev] [Code Review] Support for calendar events priorities and categories   Terry Wilson
• [asterisk-dev] [Code Review] Allow inheritance of CHANNEL() write changes for Local/n queue members   Terry Wilson
• [asterisk-dev] [Code Review] Allow inheritance of CHANNEL() write changes for Local/n queue members   Terry Wilson
• [asterisk-dev] [Code Review] Allow inheritance of CHANNEL() write changes for Local/n queue members   Terry Wilson
• [asterisk-dev] [Code Review] Allow inheritance of CHANNEL() write changes for Local/n queue members   Terry Wilson
• [asterisk-dev] [Code Review] Allow inheritance of CHANNEL() write changes for Local/n queue members   Terry Wilson
• [asterisk-dev] [Code Review] Allow inheritance of CHANNEL() write changes for Local/n queue members   Terry Wilson
• [asterisk-dev] [Code Review] Allow inheritance of CHANNEL() write changes for Local/n queue members   Terry Wilson
• [asterisk-dev] [Code Review] Allow inheritance of CHANNEL() write changes for Local/n queue members   Terry Wilson
• [asterisk-dev] [Code Review] Merging events for Exchange web service doesn't work as expected, resulting in only one event in calendar   Terry Wilson
• [asterisk-dev] [Code Review] HTTP redirect support for calendars   Terry Wilson
• [asterisk-dev] [Code Review] Avoid infinite loop with certain local channel connected line updates   Terry Wilson
• [asterisk-dev] [Code Review] Avoid infinite loop with certain local channel connected line updates   Terry Wilson
• [asterisk-dev] [Code Review] Avoid infinite loop with certain local channel connected line updates   Terry Wilson
• [asterisk-dev] [Code Review] Avoid infinite loop with certain local channel connected line updates   Terry Wilson
• [asterisk-dev] [Code Review] Avoid infinite loop with certain local channel connected line updates   Terry Wilson
• [asterisk-dev] [Code Review] Avoid infinite loop with certain local channel connected line updates   Terry Wilson
• [asterisk-dev] [Code Review] Don't let a Local channel get bridged to itself   Terry Wilson
• [asterisk-dev] [Code Review] Manager listens even when not enabled.   Terry Wilson
• [asterisk-dev] [Code Review] Add a demo utility for rapidly sending manager commands   Terry Wilson
• [asterisk-dev] Hi all, is struct ast_frame encoded by any codec?   John Wu
• [asterisk-dev] Hi all, is struct ast_frame encoded by any codec?   John Wu
• [asterisk-dev] Hi all, is struct ast_frame encoded by any codec?   John Wu
• [asterisk-dev] confuse no the method audiohook mix read side and write side data   John Wu
• [asterisk-dev] confuse no the method audiohook mix read side and write side data   John Wu
• [asterisk-dev] confuse no the method audiohook mix read side and write side data   John Wu
• [asterisk-dev] confuse with this syntax: int gsm_decode P3((s, c, target), gsm s, gsm_byte * c, gsm_signal * target)   John Wu
• [asterisk-dev] confuse with this syntax: int gsm_decode P3((s, c, target), gsm s, gsm_byte * c, gsm_signal * target)   John Wu
• [asterisk-dev] gsmtolin translator seems miss frameout function.   John Wu
• [asterisk-dev] gsmtolin translator seems miss frameout function.   John Wu
• [asterisk-dev] encounter "no reply to our critical packet" on 1.8.0beta5   John Wu
• [asterisk-dev] DAHDI FXO port only recognizes the "S" extension   Songtao Yu
• [asterisk-dev] [Code Review] Fix SRTP for changing SSRC and multiple a=crypto SDP lines   Bryant Zimmerman
• [asterisk-dev] 1.6.2.12-rc1 Audio prompts randmonly stopping   Bryant Zimmerman
• [asterisk-dev] 1.6.2.12-rc1 Audio prompts randmonly stopping   Bryant Zimmerman
• [asterisk-dev] 1.8 Beta 5 and addons   Bryant Zimmerman
• [asterisk-dev] 1.8 Beta 5 and addons   Bryant Zimmerman
• [asterisk-dev] 1.8 Beta 5 and g729 codec?   Bryant Zimmerman
• [asterisk-dev] New Feature Idea   Andrew Kohlsmith (mailing lists account)
• [asterisk-dev] Fwd: T38 and codecs negotiation   federico cabiddu
• [asterisk-dev] Fwd: T38 and codecs negotiation   federico cabiddu
• [asterisk-dev] [Code Review] Document CONNECTEDLINE and REDIRECTING useage.   rmudgett at digium.com
• [asterisk-dev] [Code Review] Document CONNECTEDLINE and REDIRECTING useage.   rmudgett at digium.com
• [asterisk-dev] [Code Review] chan_iax2 owner channel locking and memory leak   rmudgett at digium.com
• [asterisk-dev] [Code Review] Avoid infinite loop with certain local channel connected line updates   rmudgett at digium.com
• [asterisk-dev] [Code Review] sig_pri and chan_misdn do not update connected line after call transfer   rmudgett at digium.com
• [asterisk-dev] [Code Review] Merging events for Exchange web service doesn't work as expected, resulting in only one event in calendar   pitlicek at gmail.com
• [asterisk-dev] [Code Review] HTTP redirect support for calendars   pitlicek at gmail.com
• [asterisk-dev] [Code Review] Add LWP for core show threads output   tim.ringenbach at gmail.com
• [asterisk-dev] 16khz/8khz Translation path   asterisk at ntplx.net
• [asterisk-dev] [Code Review] chan_iax2: load config before registering any functions   paul.belanger at polybeacon.com
• [asterisk-dev] [Code Review] chan_iax2: load config before registering any functions   paul.belanger at polybeacon.com
• [asterisk-dev] [Code Review] chan_iax2: load config before registering any functions   paul.belanger at polybeacon.com
• [asterisk-dev] [Code Review] res_jabber: Disable debug by default   paul.belanger at polybeacon.com
• [asterisk-dev] [Code Review] handle_request_subscribe is too slow cause of iterating through all sip dialogs   sst at sil.at
• [asterisk-dev] [Code Review] handle_request_subscribe is too slow cause of iterating through all sip dialogs   sst at sil.at
• [asterisk-dev] [Code Review] handle_request_subscribe is too slow cause of iterating through all sip dialogs   sst at sil.at
• [asterisk-dev] [Code Review] handle_request_subscribe is too slow cause of iterating through all sip dialogs   sst at sil.at
• [asterisk-dev] [Code Review] handle_request_subscribe is too slow cause of iterating through all sip dialogs   sst at sil.at
• [asterisk-dev] [Code Review] handle_request_subscribe is too slow cause of iterating through all sip dialogs   sst at sil.at
• [asterisk-dev] [Code Review] Rate limit astdb->sync() calls   sst at sil.at
• [asterisk-dev] [Code Review] Rate limit astdb->sync() calls   sst at sil.at
• [asterisk-dev] [Code Review] Rate limit astdb->sync() calls   sst at sil.at
• [asterisk-dev] [Code Review] Rate limit astdb->sync() calls   sst at sil.at
• [asterisk-dev] [Code Review] Rate limit astdb->sync() calls   sst at sil.at
• [asterisk-dev] [Code Review] Rate limit astdb->sync() calls   sst at sil.at
• [asterisk-dev] [Code Review] Append two container for dialogs to delete and for rtp timeout checks to replace all dialogs container in ao2 callback for dialogs_needdestroy   sst at sil.at
• [asterisk-dev] [Code Review] Append two container for dialogs to delete and for rtp timeout checks to replace all dialogs container in ao2 callback for dialogs_needdestroy   sst at sil.at
• [asterisk-dev] [Code Review] Append two container for dialogs to delete and for rtp timeout checks to replace all dialogs container in ao2 callback for dialogs_needdestroy   sst at sil.at
• [asterisk-dev] [Code Review] Append two container for dialogs to delete and for rtp timeout checks to replace all dialogs container in ao2 callback for dialogs_needdestroy   sst at sil.at
• [asterisk-dev] [Code Review] Append two container for dialogs to delete and for rtp timeout checks to replace all dialogs container in ao2 callback for dialogs_needdestroy   sst at sil.at
• [asterisk-dev] [Code Review] Append two container for dialogs to delete and for rtp timeout checks to replace all dialogs container in ao2 callback for dialogs_needdestroy   sst at sil.at
• [asterisk-dev] [Code Review] Append two container for dialogs to delete and for rtp timeout checks to replace all dialogs container in ao2 callback for dialogs_needdestroy   sst at sil.at
• [asterisk-dev] [Code Review] Append two container for dialogs to delete and for rtp timeout checks to replace all dialogs container in ao2 callback for dialogs_needdestroy   sst at sil.at
• [asterisk-dev] [Code Review] Append two container for dialogs to delete and for rtp timeout checks to replace all dialogs container in ao2 callback for dialogs_needdestroy   sst at sil.at
• [asterisk-dev] [Code Review] Append two container for dialogs to delete and for rtp timeout checks to replace all dialogs container in ao2 callback for dialogs_needdestroy   sst at sil.at
• [asterisk-dev] [Code Review] Append two container for dialogs to delete and for rtp timeout checks to replace all dialogs container in ao2 callback for dialogs_needdestroy   sst at sil.at
• [asterisk-dev] [Code Review] Append two container for dialogs to delete and for rtp timeout checks to replace all dialogs container in ao2 callback for dialogs_needdestroy   sst at sil.at
• [asterisk-dev] [Code Review] Append two container for dialogs to delete and for rtp timeout checks to replace all dialogs container in ao2 callback for dialogs_needdestroy   sst at sil.at
• [asterisk-dev] [Code Review] Append two container for dialogs to delete and for rtp timeout checks to replace all dialogs container in ao2 callback for dialogs_needdestroy   sst at sil.at
• [asterisk-dev] [Code Review] Append two container for dialogs to delete and for rtp timeout checks to replace all dialogs container in ao2 callback for dialogs_needdestroy   sst at sil.at
• [asterisk-dev] [Code Review] SIP: for an INVITE transaction, treat all 2XX responses the same as a 200   sst at sil.at
• [asterisk-dev] [Code Review] SIP: for an INVITE transaction, treat all 2XX responses the same as a 200   sst at sil.at
• [asterisk-dev] [Code Review] handle_request_subscribe is too slow cause of iterating through all sip dialogs   sst at sil.at
• [asterisk-dev] sip_pvt call origin information   st41ker
• [asterisk-dev] sip_pvt call origin information   st41ker
• [asterisk-dev] ast_config and character escaping   st41ker
• [asterisk-dev] ast_config and character escaping   st41ker at st41ker.net
• [asterisk-dev] RTP conference   sarvesh telang
• [asterisk-dev] Problems compiling Asterisk   tomyee3 at verizon.net

Last message date: Thu Sep 30 16:22:23 CDT 2010
Archived on: Thu Sep 30 16:22:25 CDT 2010

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