August 2012 Archives by author

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Starting: Wed Aug 1 00:34:57 CDT 2012
Ending: Fri Aug 31 21:36:57 CDT 2012
Messages: 394

• [asterisk-dev] Set and get Information Elements (IE)   Yves A.
• [asterisk-dev] Set and get Information Elements (IE)   Yves A.
• [asterisk-dev] Set and get Information Elements (IE)   Yves A.
• [asterisk-dev] Asterisk 11 configuration files   Benny Amorsen
• [asterisk-dev] Asterisk 11 configuration files   Benny Amorsen
• [asterisk-dev] [Code Review]: Extend the max number of callgroups/pickupgroups   Thomas Arimont
• [asterisk-dev] [Code Review]: SIP NOTIFY dialog event infos: Number & Name Presentation for BLF signaling and pickup offers   Thomas Arimont
• [asterisk-dev] [Code Review]: SIP NOTIFY dialog event infos: Number & Name Presentation for BLF signaling and pickup offers   Thomas Arimont
• [asterisk-dev] [Code Review]: SIP NOTIFY dialog event infos: Number & Name Presentation for BLF signaling and pickup offers   Thomas Arimont
• [asterisk-dev] [Code Review]: SIP NOTIFY dialog event infos: Number & Name Presentation for BLF signaling and pickup offers   Thomas Arimont
• [asterisk-dev] Asterisk - binary naming convention is not approperiate   Bruce B
• [asterisk-dev] Asterisk - binary naming convention is not approperiate   Bruce B
• [asterisk-dev] [svn-commits] mmichelson: branch 1.8 r370769 - in /branches/1.8/channels: ./ sip/ sip/include/   Paul Belanger
• [asterisk-dev] [asterisk-users] asterisk debian package and digium repository   Paul Belanger
• [asterisk-dev] [Code Review]: Ensure that CDRs for a caller in a Queue who is not answered is NO ANSWER   Paul Belanger
• [asterisk-dev] Removing res_jabber, chan_gtalk, chan_jingle?   Paul Belanger
• [asterisk-dev] Removing res_jabber, chan_gtalk, chan_jingle?   Paul Belanger
• [asterisk-dev] Removing res_jabber, chan_gtalk, chan_jingle?   Paul Belanger
• [asterisk-dev] Provider requires Delay between OK and ReINVITE   Paul Belanger
• [asterisk-dev] [svn-commits] file: branch 11 r371619 - /branches/11/channels/chan_motif.c   Paul Belanger
• [asterisk-dev] [svn-commits] file: branch 11 r371619 - /branches/11/channels/chan_motif.c   Paul Belanger
• [asterisk-dev] [svn-commits] file: branch 11 r371619 - /branches/11/channels/chan_motif.c   Paul Belanger
• [asterisk-dev] Asterisk 11 configuration files   Paul Belanger
• [asterisk-dev] Asterisk 11 configuration files   Paul Belanger
• [asterisk-dev] Error compile asterisk 11   Paul Belanger
• [asterisk-dev] libasteriskssl.so.1   Paul Belanger
• [asterisk-dev] libasteriskssl.so.1   Paul Belanger
• [asterisk-dev] Asterisk 11 configuration files   Paul Belanger
• [asterisk-dev] Status of OPUS codec work   Paul Belanger
• [asterisk-dev] User management   Paul Belanger
• [asterisk-dev] User management   Paul Belanger
• [asterisk-dev] [Code Review] Scoped locks   Russell Bryant
• [asterisk-dev] Removing res_jabber, chan_gtalk, chan_jingle?   Russell Bryant
• [asterisk-dev] Removing res_jabber, chan_gtalk, chan_jingle?   Russell Bryant
• [asterisk-dev] distributed device state: xmpp and res_corosync   Russell Bryant
• [asterisk-dev] Asterisk 11 configuration files   Russell Bryant
• [asterisk-dev] libasteriskssl.so.1   Russell Bryant
• [asterisk-dev] DAHDIRAS   Rodolfo Carvalho
• [asterisk-dev] [asterisk-users] asterisk debian package and digium repository   Tzafrir Cohen
• [asterisk-dev] Asterisk 11 configuration files   Tzafrir Cohen
• [asterisk-dev] Asterisk 11 configuration files   Tzafrir Cohen
• [asterisk-dev] Asterisk 11 configuration files   Tzafrir Cohen
• [asterisk-dev] Starting development   Tzafrir Cohen
• [asterisk-dev] Asterisk 11.0.0-beta1 Now Available!   Joshua Colp
• [asterisk-dev] [Code Review] Add callid logging support to chan_motif   Joshua Colp
• [asterisk-dev] [Code Review] Add callid logging support to chan_motif   Joshua Colp
• [asterisk-dev] [Code Review] Asterisk WebRTC Wiki Page   Joshua Colp
• [asterisk-dev] [Code Review]: Asterisk WebRTC Wiki Page   Joshua Colp
• [asterisk-dev] [Code Review]: Asterisk WebRTC Wiki Page   Joshua Colp
• [asterisk-dev] [Code Review]: Asterisk WebRTC Wiki Page   Joshua Colp
• [asterisk-dev] [Code Review]: Asterisk WebRTC Wiki Page   Joshua Colp
• [asterisk-dev] [Code Review]: Asterisk WebRTC Wiki Page   Joshua Colp
• [asterisk-dev] [svn-commits] file: branch 11 r371619 - /branches/11/channels/chan_motif.c   Joshua Colp
• [asterisk-dev] User group in Utah   Tracee Comstock
• [asterisk-dev] Provider requires Delay between OK and ReINVITE   John R Covert
• [asterisk-dev] Provider requires Delay between OK and ReINVITE   John R. Covert
• [asterisk-dev] Provider requires Delay between OK and ReINVITE   John R. Covert
• [asterisk-dev] distributed device state: xmpp and res_corosync   Klaus Darilion
• [asterisk-dev] [Code Review] refactor dsp.c that can no longer detect a quick DTMF sequence.   Alec Davis
• [asterisk-dev] [Code Review] refactor dsp.c that can no longer detect a quick DTMF sequence.   Alec Davis
• [asterisk-dev] [Code Review] refactor dsp.c that can no longer detect a quick DTMF sequence.   Alec Davis
• [asterisk-dev] [Code Review] refactor dsp.c that can no longer detect a quick DTMF sequence.   Alec Davis
• [asterisk-dev] [Code Review] dsp.c: dtmf_detect, Fix multiple issues when no-interdigit delay is present.   Alec Davis
• [asterisk-dev] [asterisk-commits] alecdavis: branch 1.8 r371662 -/branches/1.8/main/dsp.c   Alec Davis
• [asterisk-dev] [Code Review] dsp.c: dtmf_detect, Fix multiple issues when no-interdigit delay is present.   Alec Davis
• [asterisk-dev] [Code Review] Asterisk 11 presence state wiki page   Alec Davis
• [asterisk-dev] [Code Review] dsp.c: dtmf_detect, Fix multiple issues when no-interdigit delay is present.   Alec Davis
• [asterisk-dev] libasteriskssl.so.1   Alec Davis
• [asterisk-dev] IAX support for SIP URI as called number   Alec Davis
• [asterisk-dev] IAX support for SIP URI as called number   Alec Davis
• [asterisk-dev] [Code Review]: refactor dsp.c that can no longer detect a quick DTMF sequence.   Alec Davis
• [asterisk-dev] [Code Review]: dsp.c: dtmf_detect, Fix multiple issues when no-interdigit delay is present.   Alec Davis
• [asterisk-dev] [Code Review]: dsp.c: dtmf_detect, Fix multiple issues when no-interdigit delay is present.   Alec Davis
• [asterisk-dev] [Code Review]: dsp.c: dtmf_detect, Fix multiple issues when no-interdigit delay is present.   Alec Davis
• [asterisk-dev] [Code Review]: dsp.c: dtmf_detect, Fix multiple issues when no-interdigit delay is present.   Alec Davis
• [asterisk-dev] [Code Review] dsp.c: dtmf_detect, Fix multiple issues when no-interdigit delay is present.   Alec Davis
• [asterisk-dev] [Code Review] dsp.c: dtmf_detect, Fix multiple issues when no-interdigit delay is present.   Alec Davis
• [asterisk-dev] [Code Review] dsp.c: dtmf_detect, Fix multiple issues when no-interdigit delay is present.   Alec Davis
• [asterisk-dev] [Code Review]: dsp.c: dtmf_detect, Fix multiple issues when no-interdigit delay is present.   Alec Davis
• [asterisk-dev] [Code Review]: dsp.c: dtmf_detect, Fix multiple issues when no-interdigit delay is present.   Alec Davis
• [asterisk-dev] Problem doing SIP/RTP with an Asterisk to another Asterisk that has PSTN card   Abelenda Diego
• [asterisk-dev] [solved ?] Problem doing SIP/RTP with an Asterisk to another Asterisk that has PSTN card   Abelenda Diego
• [asterisk-dev] Asterisk 11 configuration files   Rod Dorman
• [asterisk-dev] Asterisk 11 configuration files   Steve Edwards
• [asterisk-dev] Need to run an AGI from dialplan only upon call hold   Steve Edwards
• [asterisk-dev] Need to run an AGI from dialplan only upon call hold   Steve Edwards
• [asterisk-dev] Need to run an AGI from dialplan only upon call hold   Steve Edwards
• [asterisk-dev] Need to run an AGI from dialplan only upon call hold   Lenz Emilitri
• [asterisk-dev] Need to run an AGI from dialplan only upon call hold   Lenz Emilitri
• [asterisk-dev] Persistent SIP Subscriptions   Kevin P. Fleming
• [asterisk-dev] Developer Needed   Rick Hall
• [asterisk-dev] [Code Review] Add support for ${} parsing in sip_notify.conf   Olle E Johansson
• [asterisk-dev] [Code Review] app_alarmreceiver new features and fixes   Olle E Johansson
• [asterisk-dev] Persistent SIP Subscriptions   Olle E. Johansson
• [asterisk-dev] Fwd: [svn-commits] kmoore: trunk r370851 - /trunk/main/channel.c   Olle E. Johansson
• [asterisk-dev] Asterisk 11.0.0-beta1 Now Available!   Olle E. Johansson
• [asterisk-dev] Asterisk 11.0.0-beta1 Now Available!   Olle E. Johansson
• [asterisk-dev] Multiple Contacts in 302 Message   Olle E. Johansson
• [asterisk-dev] Removing res_jabber, chan_gtalk, chan_jingle?   Olle E. Johansson
• [asterisk-dev] Removing res_jabber, chan_gtalk, chan_jingle?   Olle E. Johansson
• [asterisk-dev] Removing res_jabber, chan_gtalk, chan_jingle?   Olle E. Johansson
• [asterisk-dev] Removing res_jabber, chan_gtalk, chan_jingle?   Olle E. Johansson
• [asterisk-dev] Removing res_jabber, chan_gtalk, chan_jingle?   Olle E. Johansson
• [asterisk-dev] [svn-commits] jrose: branch 1.8 r371337 - /branches/1.8/channels/chan_sip.c   Olle E. Johansson
• [asterisk-dev] [svn-commits] jrose: branch 1.8 r371337 - /branches/1.8/channels/chan_sip.c   Olle E. Johansson
• [asterisk-dev] Provider requires Delay between OK and ReINVITE   Olle E. Johansson
• [asterisk-dev] Asterisk 11 configuration files   Olle E. Johansson
• [asterisk-dev] [svn-commits] jrose: trunk r371889 - in /trunk: UPGRADE.txt channels/chan_sip.c   Olle E. Johansson
• [asterisk-dev] IAX support for SIP URI as called number   Olle E. Johansson
• [asterisk-dev] Status of OPUS codec work   Olle E. Johansson
• [asterisk-dev] Status of OPUS codec work   Olle E. Johansson
• [asterisk-dev] Need to run an AGI from dialplan only upon call hold   Olle E. Johansson
• [asterisk-dev] Need to run an AGI from dialplan only upon call hold   Olle E. Johansson
• [asterisk-dev] Status of OPUS codec work   Olle E. Johansson
• [asterisk-dev] [Code Review]: Extend the max number of callgroups/pickupgroups   Matt Jordan
• [asterisk-dev] [Code Review]: Initial Bridging Test Object and sample tests   Matt Jordan
• [asterisk-dev] [Code Review] Initial Bridging Test Object and sample tests   Matt Jordan
• [asterisk-dev] [Code Review] Ensure that CDRs for a caller in a Queue who is not answered is NO ANSWER   Matt Jordan
• [asterisk-dev] [Code Review]: Ensure that CDRs for a caller in a Queue who is not answered is NO ANSWER   Matt Jordan
• [asterisk-dev] [Code Review]: Ensure that CDRs for a caller in a Queue who is not answered is NO ANSWER   Matt Jordan
• [asterisk-dev] [Code Review]: Changes from Mantis ID 13495 in trunk.   Matt Jordan
• [asterisk-dev] [Code Review] MixMonitor Leaves Empty Audio Files Behind   Matt Jordan
• [asterisk-dev] [Code Review]: MixMonitor Leaves Empty Audio Files Behind   Matt Jordan
• [asterisk-dev] [Code Review] Properly detect overflow on int32 and uint32 types in ast_parse_arg   Matt Jordan
• [asterisk-dev] [Code Review] Ensure that CDRs for a caller in a Queue who is not answered is NO ANSWER   Matt Jordan
• [asterisk-dev] [Code Review]: Make callbackextension work for realtime and also match on callbackextension when multiple peers with the same host have differing callbackextensions   Matt Jordan
• [asterisk-dev] [Code Review] Deprecate chan_gtalk, chan_jingle, and res_jabber in preference of chan_motif and res_xmpp   Matt Jordan
• [asterisk-dev] [Code Review]: Make callbackextension work for realtime and also match on callbackextension when multiple peers with the same host have differing callbackextensions   Matt Jordan
• [asterisk-dev] [Code Review] Revamp app_confbridge into something resembling a state machine   Matt Jordan
• [asterisk-dev] [Code Review]: Fix segfault relating to use of uninitialized aco objects / Fix misrepresentation of confbridge elements that fail to load   Matt Jordan
• [asterisk-dev] [Bamboo] Matthew Jordan commented on Asterisk Testing - Asterisk Trunk 570   Matthew Jordan
• [asterisk-dev] [Bamboo] Matthew Jordan commented on Asterisk Testing - Asterisk Trunk 570   Matthew Jordan
• [asterisk-dev] [svn-commits] mjordan: branch 11 r371121 - /branches/11/   Matthew Jordan
• [asterisk-dev] Repotools Update   Matthew Jordan
• [asterisk-dev] Asterisk 11 / Trunk Review with Diffs   Matthew Jordan
• [asterisk-dev] Multiple Contacts in 302 Message   Matthew Jordan
• [asterisk-dev] [Bamboo] Matthew Jordan commented on Asterisk Testing - Asterisk 11 Branch 5   Matthew Jordan
• [asterisk-dev] Removing res_jabber, chan_gtalk, chan_jingle?   Matthew Jordan
• [asterisk-dev] Removing res_jabber, chan_gtalk, chan_jingle?   Matthew Jordan
• [asterisk-dev] Removing res_jabber, chan_gtalk, chan_jingle?   Matthew Jordan
• [asterisk-dev] [svn-commits] file: branch 11 r371619 - /branches/11/channels/chan_motif.c   Matthew Jordan
• [asterisk-dev] [svn-commits] file: branch 11 r371619 - /branches/11/channels/chan_motif.c   Matthew Jordan
• [asterisk-dev] Asterisk 11 configuration files   Matthew Jordan
• [asterisk-dev] [asterisk-commits] alecdavis: branch 1.8 r371662 - /branches/1.8/main/dsp.c   Matthew Jordan
• [asterisk-dev] Asterisk 11 configuration files   Matthew Jordan
• [asterisk-dev] Asterisk 11 configuration files   Matthew Jordan
• [asterisk-dev] Need to run an AGI from dialplan only upon call hold   Matthew Jordan
• [asterisk-dev] Asterisk 11 configuration files   Matthew Jordan
• [asterisk-dev] Starting development   Matthew Jordan
• [asterisk-dev] libasteriskssl.so.1   Matthew Jordan
• [asterisk-dev] libasteriskssl.so.1   Matthew Jordan
• [asterisk-dev] Error compile asterisk 11   Matthew Jordan
• [asterisk-dev] libasteriskssl.so.1   Matthew Jordan
• [asterisk-dev] IAX support for SIP URI as called number   Matthew Jordan
• [asterisk-dev] [svn-commits] jrose: trunk r371889 - in /trunk: UPGRADE.txt channels/chan_sip.c   Matthew Jordan
• [asterisk-dev] User management   Matthew Jordan
• [asterisk-dev] Need to run an AGI from dialplan only upon call hold   Matthew Jordan
• [asterisk-dev] [Code Review] Bring the (missing) changes from Mantis ID 13495 in trunk.   KNK
• [asterisk-dev] [Code Review] Changes from Mantis ID 13495 in trunk.   KNK
• [asterisk-dev] [Code Review]: Bring the (missing) changes from Mantis ID 13495 in trunk.   KNK
• [asterisk-dev] [Code Review]: Changes from Mantis ID 13495 in trunk.   KNK
• [asterisk-dev] [Code Review] Changes from Mantis ID 13495 in trunk.   KNK
• [asterisk-dev] [Code Review] Changes from Mantis ID 13495 in trunk.   KNK
• [asterisk-dev] [Code Review] app_alarmreceiver cleanup and fixes   KNK
• [asterisk-dev] [Code Review] app_alarmreceiver cleanup and fixes   KNK
• [asterisk-dev] [Code Review] app_alarmreceiver cleanup and fixes   KNK
• [asterisk-dev] [Code Review] app_alarmreceiver cleanup and fixes   KNK
• [asterisk-dev] [Code Review]: Changes from Mantis ID 13495 in trunk.   KNK
• [asterisk-dev] [Code Review] Changes from Mantis ID 13495 in trunk.   KNK
• [asterisk-dev] [Code Review]: app_alarmreceiver cleanup and fixes   KNK
• [asterisk-dev] [Code Review]: app_alarmreceiver cleanup and fixes   KNK
• [asterisk-dev] [Code Review] app_alarmreceiver cleanup and fixes   KNK
• [asterisk-dev] [Code Review] app_alarmreceiver cleanup and fixes   KNK
• [asterisk-dev] [Code Review] app_alarmreceiver new features and fixes   KNK
• [asterisk-dev] [Code Review]: app_alarmreceiver cleanup and fixes   KNK
• [asterisk-dev] [Code Review]: Changes from Mantis ID 13495 in trunk.   KNK
• [asterisk-dev] [Code Review]: Changes from Mantis ID 13495 in trunk.   KNK
• [asterisk-dev] [Code Review] Changes from Mantis ID 13495 in trunk.   KNK
• [asterisk-dev] [Code Review]: SIP NOTIFY dialog event infos: Number & Name Presentation for BLF signaling and pickup offers   Guenther Kelleter
• [asterisk-dev] [Code Review] SIP NOTIFY dialog event infos: Number & Name Presentation for BLF signaling and pickup offers   Guenther Kelleter
• [asterisk-dev] [Code Review]: SIP NOTIFY dialog event infos: Number & Name Presentation for BLF signaling and pickup offers   Guenther Kelleter
• [asterisk-dev] [Code Review] SIP NOTIFY dialog event infos: Number & Name Presentation for BLF signaling and pickup offers   Guenther Kelleter
• [asterisk-dev] [Code Review]: SIP NOTIFY dialog event infos: Number & Name Presentation for BLF signaling and pickup offers   Guenther Kelleter
• [asterisk-dev] Makefile Help   Sean Kelly
• [asterisk-dev] Makefile Help   Sean Kelly
• [asterisk-dev] App Alarm Receiver   Pedro Kiefer
• [asterisk-dev] [Code Review]: app_alarmreceiver cleanup and fixes   Pedro Kiefer
• [asterisk-dev] [svn-commits] mjordan: branch 11 r371121 - /branches/11/   Andrew Latham
• [asterisk-dev] [svn-commits] mjordan: branch 11 r371121 - /branches/11/   Andrew Latham
• [asterisk-dev] Asterisk 11 / Trunk Review with Diffs   Andrew Latham
• [asterisk-dev] Asterisk 11 / Trunk Review with Diffs   Andrew Latham
• [asterisk-dev] Removing res_jabber, chan_gtalk, chan_jingle?   Andrew Latham
• [asterisk-dev] Removing res_jabber, chan_gtalk, chan_jingle?   Andrew Latham
• [asterisk-dev] Making Doxygen Happy - Issue and Diff   Andrew Latham
• [asterisk-dev] load_module Documentation   Andrew Latham
• [asterisk-dev] Developer Needed   Andrew Latham
• [asterisk-dev] https://wiki.asterisk.org/ Down?   Andrew Latham
• [asterisk-dev] https://wiki.asterisk.org/ Down?   Andrew Latham
• [asterisk-dev] [Code Review] Deprecate chan_gtalk, chan_jingle, and res_jabber in preference of chan_motif and res_xmpp   Andrew Latham
• [asterisk-dev] Asterisk 11 configuration files   Andrew Latham
• [asterisk-dev] Asterisk 11 configuration files   Andrew Latham
• [asterisk-dev] Asterisk 11 configuration files   Andrew Latham
• [asterisk-dev] Status of OPUS codec work   Andrew Latham
• [asterisk-dev] [Code Review]: Ensure that CDRs for a caller in a Queue who is not answered is NO ANSWER   Tilghman Lesher
• [asterisk-dev] [Code Review]: Ensure that CDRs for a caller in a Queue who is not answered is NO ANSWER   Tilghman Lesher
• [asterisk-dev] Repotools Update   Tilghman Lesher
• [asterisk-dev] [Code Review]: Changes from Mantis ID 13495 in trunk.   Tilghman Lesher
• [asterisk-dev] [Code Review]: app_alarmreceiver cleanup and fixes   Tilghman Lesher
• [asterisk-dev] link asterisk to dmalloc   Tilghman Lesher
• [asterisk-dev] Asterisk 11 configuration files   Tilghman Lesher
• [asterisk-dev] Asterisk 11 configuration files   Tilghman Lesher
• [asterisk-dev] Asterisk 11 configuration files   Tilghman Lesher
• [asterisk-dev] Asterisk 11 configuration files   Tilghman Lesher
• [asterisk-dev] Starting development   Tilghman Lesher
• [asterisk-dev] libasteriskssl.so.1   Tilghman Lesher
• [asterisk-dev] Asterisk 11 configuration files   Tilghman Lesher
• [asterisk-dev] Asterisk 11 configuration files   Tilghman Lesher
• [asterisk-dev] Need to run an AGI from dialplan only upon call hold   Tilghman Lesher
• [asterisk-dev] Need to run an AGI from dialplan only upon call hold   Tilghman Lesher
• [asterisk-dev] Behaviour in trunk's pbx_config.c, app_stack.c and users.conf   Tilghman Lesher
• [asterisk-dev] Asterisk 11 configuration files   JP Loh
• [asterisk-dev] Asterisk GUI   Shitian Long
• [asterisk-dev] Developer Needed   Shitian Long
• [asterisk-dev] Asterisk GUI   Giuseppe Longo
• [asterisk-dev] Asterisk GUI   Giuseppe Longo
• [asterisk-dev] Starting development   Giuseppe Longo
• [asterisk-dev] Starting development   Giuseppe Longo
• [asterisk-dev] Error compile asterisk 11   Giuseppe Longo
• [asterisk-dev] Error compile asterisk 11   Giuseppe Longo
• [asterisk-dev] Starting development   Giuseppe Longo
• [asterisk-dev] libasteriskssl.so.1   Giuseppe Longo
• [asterisk-dev] libasteriskssl.so.1   Giuseppe Longo
• [asterisk-dev] libasteriskssl.so.1   Giuseppe Longo
• [asterisk-dev] Starting development   Giuseppe Longo
• [asterisk-dev] Starting development   Giuseppe Longo
• [asterisk-dev] User management   Giuseppe Longo
• [asterisk-dev] User management   Giuseppe Longo
• [asterisk-dev] User management   Giuseppe Longo
• [asterisk-dev] https://wiki.asterisk.org/ Down?   Doug Lytle
• [asterisk-dev] https://wiki.asterisk.org/ Down?   Doug Lytle
• [asterisk-dev] [Code Review]: Ensure that CDRs for a caller in a Queue who is not answered is NO ANSWER   Leif Madsen
• [asterisk-dev] [Code Review] Deprecate chan_gtalk, chan_jingle, and res_jabber in preference of chan_motif and res_xmpp   Leif Madsen
• [asterisk-dev] [Code Review] Deprecate chan_gtalk, chan_jingle, and res_jabber in preference of chan_motif and res_xmpp   Leif Madsen
• [asterisk-dev] [Code Review] Deprecate chan_gtalk, chan_jingle, and res_jabber in preference of chan_motif and res_xmpp   Leif Madsen
• [asterisk-dev] Asterisk 11 configuration files   Leif Madsen
• [asterisk-dev] [Code Review] Deprecate chan_gtalk, chan_jingle, and res_jabber in preference of chan_motif and res_xmpp   Leif Madsen
• [asterisk-dev] Asterisk 11 configuration files   Leif Madsen
• [asterisk-dev] link asterisk to dmalloc   Tomas Maldonado
• [asterisk-dev] link asterisk to dmalloc   Tomas Maldonado
• [asterisk-dev] HANGUPCAUSE variable wrong   Mark Michelson
• [asterisk-dev] [Code Review]: Initial Bridging Test Object and sample tests   Mark Michelson
• [asterisk-dev] [Code Review]: Initial Bridging Test Object and sample tests   Mark Michelson
• [asterisk-dev] [Code Review]: SIP NOTIFY dialog event infos: Number & Name Presentation for BLF signaling and pickup offers   Mark Michelson
• [asterisk-dev] [Code Review]: SIP NOTIFY dialog event infos: Number & Name Presentation for BLF signaling and pickup offers   Mark Michelson
• [asterisk-dev] [Code Review] Extended maximum number of pickupgroups (callgroup/pickupgroup)   Mark Michelson
• [asterisk-dev] [Code Review]: SIP NOTIFY dialog event infos: Number & Name Presentation for BLF signaling and pickup offers   Mark Michelson
• [asterisk-dev] [Code Review] SIP NOTIFY dialog event infos: Number & Name Presentation for BLF signaling and pickup offers   Mark Michelson
• [asterisk-dev] [Code Review] possible fix for broken SIP/peer/exten/IPorHost dial syntax   Mark Michelson
• [asterisk-dev] [svn-commits] mmichelson: branch 1.8 r370769 - in /branches/1.8/channels: ./ sip/ sip/include/   Mark Michelson
• [asterisk-dev] [Code Review] Initial Bridging Test Object and sample tests   Mark Michelson
• [asterisk-dev] [Code Review] Initial Bridging Test Object and sample tests   Mark Michelson
• [asterisk-dev] [Code Review] Initial Bridging Test Object and sample tests   Mark Michelson
• [asterisk-dev] [Code Review] Scoped locks   Mark Michelson
• [asterisk-dev] [Code Review] Scoped locks   Mark Michelson
• [asterisk-dev] [Code Review] More bridge tests; Add CDR and CEL checking to bridge tests   Mark Michelson
• [asterisk-dev] [Code Review] More bridge tests; Add CDR and CEL checking to bridge tests   Mark Michelson
• [asterisk-dev] [Code Review] dialing through chan_local breaks t38 fax   Mark Michelson
• [asterisk-dev] [Code Review] Add HANGUPCAUSE information to called channels   Mark Michelson
• [asterisk-dev] [Code Review] Fixes dealing with misuse of asprintf()   Mark Michelson
• [asterisk-dev] [Code Review] Properly detect overflow on int32 and uint32 types in ast_parse_arg   Mark Michelson
• [asterisk-dev] [Code Review] Fixes dealing with misuse of asprintf()   Mark Michelson
• [asterisk-dev] [Code Review] app_alarmreceiver cleanup and fixes   Mark Michelson
• [asterisk-dev] [Code Review] Add callid logging support to chan_motif   Mark Michelson
• [asterisk-dev] [Code Review] Revamp app_confbridge into something resembling a state machine   Mark Michelson
• [asterisk-dev] [Code Review] Asterisk WebRTC Wiki Page   Mark Michelson
• [asterisk-dev] Provider requires Delay between OK and ReINVITE   Mark Michelson
• [asterisk-dev] Provider requires Delay between OK and ReINVITE   Mark Michelson
• [asterisk-dev] [Code Review]: astobj2 API container enhancements   Mark Michelson
• [asterisk-dev] Asterisk 11 configuration files   Mark Michelson
• [asterisk-dev] Asterisk 11 configuration files   Mark Michelson
• [asterisk-dev] [Code Review] Fix segfault relating to use of uninitialized aco objects / Fix misrepresentation of confbridge elements that fail to load   Mark Michelson
• [asterisk-dev] [Code Review] Asterisk 11 presence state wiki page   Mark Michelson
• [asterisk-dev] [Code Review]: Asterisk 11 presence state wiki page   Mark Michelson
• [asterisk-dev] [Code Review]: Asterisk 11 presence state wiki page   Mark Michelson
• [asterisk-dev] [Code Review] Test device state notifications in chan_sip   Mark Michelson
• [asterisk-dev] [Code Review] Wiki page for direct media glare avoidance in Asterisk 11   Mark Michelson
• [asterisk-dev] [Code Review] Fix SEQNO on retransmitted DTMF event END packets not being incremented   Mark Michelson
• [asterisk-dev] [Code Review] app_alarmreceiver cleanup and fixes   Mark Michelson
• [asterisk-dev] Fwd: [svn-commits] kmoore: trunk r370851 - /trunk/main/channel.c   Kinsey Moore
• [asterisk-dev] Set and get Information Elements (IE)   Richard Mudgett
• [asterisk-dev] Set and get Information Elements (IE)   Richard Mudgett
• [asterisk-dev] Behaviour in trunk's pbx_config.c, app_stack.c and users.conf   Richard Mudgett
• [asterisk-dev] Asterisk - binary naming convention is not approperiate   Jason Parker
• [asterisk-dev] Persistent SIP Subscriptions   Wolfgang Pichler
• [asterisk-dev] Multiple Contacts in 302 Message   Andaleeb Roomy
• [asterisk-dev] Asterisk GUI   Jonathan Rose
• [asterisk-dev] Asterisk GUI   Jonathan Rose
• [asterisk-dev] [svn-commits] jrose: branch 1.8 r371337 - /branches/1.8/channels/chan_sip.c   Jonathan Rose
• [asterisk-dev] [svn-commits] jrose: branch 1.8 r371337 - /branches/1.8/channels/chan_sip.c   Jonathan Rose
• [asterisk-dev] Starting development   Jonathan Rose
• [asterisk-dev] Asterisk GUI   Jonathan Rose
• [asterisk-dev] Behaviour in trunk's pbx_config.c, app_stack.c and users.conf   Octavio Ruiz
• [asterisk-dev] Behaviour in trunk's pbx_config.c, app_stack.c and users.conf   Octavio Ruiz
• [asterisk-dev] Starting development   Torrey Searle
• [asterisk-dev] Need to run an AGI from dialplan only upon call hold   Marco Signorini
• [asterisk-dev] Planned service outage for community services on August 2, 2012   Asterisk Development Team
• [asterisk-dev] Unplanned Asterisk community service outage   Asterisk Development Team
• [asterisk-dev] Unplanned Asterisk community service outage   Asterisk Development Team
• [asterisk-dev] Scheduled Maintenance for Asterisk Project community services   Asterisk Development Team
• [asterisk-dev] Asterisk 11.0.0-beta1 Now Available!   Asterisk Development Team
• [asterisk-dev] code.asterisk.org migration - service outage on Tues, Aug 14th   Asterisk Development Team
• [asterisk-dev] Asterisk community services - Old Mantis instance to be shutdown on Aug 28th, 2012   Asterisk Development Team
• [asterisk-dev] Asterisk 1.8.11-cert7, 1.8.15.1, 10.7.1, 10.7.1-digiumphones Now Available (Security Release)   Asterisk Development Team
• [asterisk-dev] AST-2012-012: Asterisk Manager User Unauthorized Shell Access   Asterisk Security Team
• [asterisk-dev] AST-2012-013: ACL rules ignored when placing outbound calls by certain IAX2 users   Asterisk Security Team
• [asterisk-dev] HANGUPCAUSE variable wrong   Pavel Troller
• [asterisk-dev] Asterisk 11 configuration files   Michel Verbraak
• [asterisk-dev] Asterisk 11 configuration files   Michel Verbraak
• [asterisk-dev] [Code Review] Revamp app_confbridge into something resembling a state machine   Terry Wilson
• [asterisk-dev] [Code Review] Revamp app_confbridge into something resembling a state machine   Terry Wilson
• [asterisk-dev] [Code Review] Properly detect overflow on int32 and uint32 types in ast_parse_arg   Terry Wilson
• [asterisk-dev] [Code Review] Revamp app_confbridge into something resembling a state machine   Terry Wilson
• [asterisk-dev] [Code Review] Revamp app_confbridge into something resembling a state machine   Terry Wilson
• [asterisk-dev] [Code Review]: Revamp app_confbridge into something resembling a state machine   Terry Wilson
• [asterisk-dev] [Code Review] Revamp app_confbridge into something resembling a state machine   Terry Wilson
• [asterisk-dev] [Code Review]: Revamp app_confbridge into something resembling a state machine   Terry Wilson
• [asterisk-dev] Call progress analysis , Voicemail beep / tone detection   tahir almas
• [asterisk-dev] [Code Review] MixMonitor Leaves Empty Audio Files Behind   elguero
• [asterisk-dev] [Code Review]: MixMonitor Leaves Empty Audio Files Behind   elguero
• [asterisk-dev] [Code Review] MixMonitor Leaves Empty Audio Files Behind   elguero
• [asterisk-dev] [Code Review]: MixMonitor Leaves Empty Audio Files Behind   elguero
• [asterisk-dev] [Code Review] Fix SEQNO on retransmitted DTMF event END packets not being incremented   elguero
• [asterisk-dev] [Code Review] Fix SEQNO on retransmitted DTMF event END packets not being incremented   elguero
• [asterisk-dev] [Code Review] Add support for ${} parsing in sip_notify.conf   gareth
• [asterisk-dev] Asterisk GUI   giuseppelng at gmail.com
• [asterisk-dev] [Code Review] MixMonitor Leaves Empty Audio Files Behind   jrose
• [asterisk-dev] [Code Review] Add callid logging support to chan_motif   jrose
• [asterisk-dev] [Code Review] app_queue: Don't log PAUSEALL/UNPAUSEALL for an interface if it doesn't exist in any queue when pause   jrose
• [asterisk-dev] [Code Review] Add callid logging support to chan_motif   jrose
• [asterisk-dev] [Code Review] Revamp app_confbridge into something resembling a state machine   jrose
• [asterisk-dev] [Code Review]: Revamp app_confbridge into something resembling a state machine   jrose
• [asterisk-dev] [Code Review] Asterisk WebRTC Wiki Page   lathama
• [asterisk-dev] [Code Review]: Asterisk WebRTC Wiki Page   lathama
• [asterisk-dev] [Code Review] Asterisk WebRTC Wiki Page   lathama
• [asterisk-dev] [Code Review]: Asterisk WebRTC Wiki Page   lathama
• [asterisk-dev] [Code Review] Asterisk WebRTC Wiki Page   lathama
• [asterisk-dev] [Code Review]: Asterisk WebRTC Wiki Page   lathama
• [asterisk-dev] [Code Review] Deprecate chan_gtalk, chan_jingle, and res_jabber in preference of chan_motif and res_xmpp   lathama
• [asterisk-dev] [Code Review] Fix segfault relating to use of uninitialized aco objects / Fix misrepresentation of confbridge elements that fail to load   lathama
• [asterisk-dev] [Code Review] Make callbackextension work for realtime and also match on callbackextension when multiple peers with the same host have differing callbackextensions   nixon
• [asterisk-dev] [Code Review]: Make callbackextension work for realtime and also match on callbackextension when multiple peers with the same host have differing callbackextensions   nixon
• [asterisk-dev] [Code Review]: Make callbackextension work for realtime and also match on callbackextension when multiple peers with the same host have differing callbackextensions   nixon
• [asterisk-dev] [Code Review] Scoped locks   opticron
• [asterisk-dev] [Code Review]: Revise skinny debugging messages   opticron
• [asterisk-dev] [Code Review] Revise skinny debugging messages   opticron
• [asterisk-dev] [Code Review] Add HANGUPCAUSE information to called channels   opticron
• [asterisk-dev] [Code Review] Scoped locks   opticron
• [asterisk-dev] [Code Review]: Initial Bridging Test Object and sample tests   rmudgett
• [asterisk-dev] [Code Review] Changes from Mantis ID 13495 in trunk.   rmudgett
• [asterisk-dev] [Code Review] Bring the (missing) changes from Mantis ID 13495 in trunk.   rmudgett
• [asterisk-dev] [Code Review]: Bring the (missing) changes from Mantis ID 13495 in trunk.   rmudgett
• [asterisk-dev] [Code Review] Changes from Mantis ID 13495 in trunk.   rmudgett
• [asterisk-dev] [Code Review] Fixes dealing with misuse of asprintf()   rmudgett
• [asterisk-dev] [Code Review] Fixes dealing with misuse of asprintf()   rmudgett
• [asterisk-dev] [Code Review] Changes from Mantis ID 13495 in trunk.   rmudgett
• [asterisk-dev] [Code Review] astobj2 API container enhancements   rmudgett
• [asterisk-dev] [Code Review] astobj2 API container enhancements   rmudgett
• [asterisk-dev] [Code Review]: app_alarmreceiver cleanup and fixes   rmudgett
• [asterisk-dev] [Code Review] astobj2 API container enhancements   rmudgett
• [asterisk-dev] [Code Review] astobj2 API container enhancements   rmudgett
• [asterisk-dev] [Code Review] Asterisk 11 presence state wiki page   rmudgett
• [asterisk-dev] [Code Review] Changes from Mantis ID 13495 in trunk.   rmudgett
• [asterisk-dev] [Code Review] refactor dsp.c that can no longer detect a quick DTMF sequence.   rmudgett
• [asterisk-dev] [Code Review] dsp.c: dtmf_detect, Fix multiple issues when no-interdigit delay is present.   rmudgett
• [asterisk-dev] [Code Review]: refactor dsp.c that can no longer detect a quick DTMF sequence.   rmudgett
• [asterisk-dev] [Code Review]: Changes from Mantis ID 13495 in trunk.   rmudgett
• [asterisk-dev] [Code Review]: dsp.c: dtmf_detect, Fix multiple issues when no-interdigit delay is present.   rmudgett
• [asterisk-dev] [Code Review]: dsp.c: dtmf_detect, Fix multiple issues when no-interdigit delay is present.   rmudgett
• [asterisk-dev] [Code Review] Changes from Mantis ID 13495 in trunk.   rmudgett
• [asterisk-dev] [Code Review] Changes from Mantis ID 13495 in trunk.   rmudgett
• [asterisk-dev] [Code Review] dsp.c: dtmf_detect, Fix multiple issues when no-interdigit delay is present.   rmudgett
• [asterisk-dev] [Code Review] dsp.c: dtmf_detect, Fix multiple issues when no-interdigit delay is present.   rmudgett
• [asterisk-dev] Need to run an AGI from dialplan only upon call hold   uma udupi
• [asterisk-dev] Need to run an AGI from dialplan only upon call hold   uma udupi
• [asterisk-dev] Need to run an AGI from dialplan only upon call hold   uma udupi
• [asterisk-dev] Need to run an AGI from dialplan only upon call hold   uma udupi
• [asterisk-dev] Need to run an AGI from dialplan only upon call hold   uma udupi
• [asterisk-dev] Need to run an AGI from dialplan only upon call hold   uma udupi
• [asterisk-dev] Need to run an AGI from dialplan only upon call hold   uma udupi
• [asterisk-dev] Need to run an AGI from dialplan only upon call hold   uma udupi
• [asterisk-dev] [Code Review] dialing through chan_local breaks t38 fax   wdoekes
• [asterisk-dev] [Code Review]: Make callbackextension work for realtime and also match on callbackextension when multiple peers with the same host have differing callbackextensions   wdoekes
• [asterisk-dev] List of Asterisk Hardening.   white.heron white
• [asterisk-dev] Voice Encryption for Asterisk VoIP Server.   white.heron white

Last message date: Fri Aug 31 21:36:57 CDT 2012
Archived on: Fri Aug 31 21:50:57 CDT 2012

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