June 2018 Archives by author

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Starting: Fri Jun 1 06:57:35 CDT 2018
Ending: Sat Jun 30 17:41:34 CDT 2018
Messages: 98

• [asterisk-users] shell dialplan application blocking   Matt Riddell (lists)
• [asterisk-users] Using ControlPlayback with AWS S3   Dovid Bender
• [asterisk-users] Using ControlPlayback with AWS S3   Dovid Bender
• [asterisk-users] Documentation for media caching   Dovid Bender
• [asterisk-users] Head request with curl in Asterisk   Dovid Bender
• [asterisk-users] Using ControlPlayback with AWS S3   Dovid Bender
• [asterisk-users] AMD min amount of words   Dovid Bender
• [asterisk-users] Best way to update ever changing dialplans   Dovid Bender
• [asterisk-users] Asterisk not matching longest prefix with include   Dovid Bender
• [asterisk-users] Asterisk crashing on AAAA lookup   Dovid Bender
• [asterisk-users] Asterisk not matching longest prefix with include   Dovid Bender
• [asterisk-users] Asterisk not matching longest prefix with include   Dovid Bender
• [asterisk-users] Asterisk not matching longest prefix with include   Dovid Bender
• [asterisk-users] Asterisk crashing on AAAA lookup   Dovid Bender
• [asterisk-users] Asterisk crashing on AAAA lookup   Dovid Bender
• [asterisk-users] Looking for better fax handling   D'Arcy Cain
• [asterisk-users] T-38 re-invite issue   D'Arcy Cain
• [asterisk-users] T-38 re-invite issue   D'Arcy Cain
• [asterisk-users] T-38 re-invite issue   D'Arcy Cain
• [asterisk-users] T-38 re-invite issue   D'Arcy Cain
• [asterisk-users] T-38 re-invite issue   D'Arcy Cain
• [asterisk-users] Busy indicator for FXO line or extension   Carlos Chavez
• [asterisk-users] T-38 re-invite issue   James Cloos
• [asterisk-users] T-38 re-invite issue   James Cloos
• [asterisk-users] Asterisk not matching longest prefix with include   Joshua Colp
• [asterisk-users] Best way to update ever changing dialplans   Steve Edwards
• [asterisk-users] Best way to update ever changing dialplans   Bruce Ferrell
• [asterisk-users] Asterisk Realtime PJSIP - slow output on "pjsip show xxxxx" commands   Floimair Florian
• [asterisk-users] remove   Matt Fredrickson
• [asterisk-users] Function CHANNELS   Matt Hamilton
• [asterisk-users] Recommended Linux version or how to compile DAHDI on Fedora?   Ira
• [asterisk-users] Questions about SIP From, P-Asserted-Id fields and Diversion headers ?   George Joseph
• [asterisk-users] Questions about SIP From, P-Asserted-Id fields and Diversion headers ?   George Joseph
• [asterisk-users] Questions about SIP From, P-Asserted-Id fields and Diversion headers ?   George Joseph
• [asterisk-users] Asterisk crashing on AAAA lookup   George Joseph
• [asterisk-users] Cancel   Khalil Khamlichi
• [asterisk-users] getting real sip status after dial   Khalil Khamlichi
• [asterisk-users] getting real sip status after dial   Khalil Khamlichi
• [asterisk-users] getting real sip status after dial   Khalil Khamlichi
• [asterisk-users] getting real sip status after dial   Khalil Khamlichi
• [asterisk-users] getting real sip status after dial   Khalil Khamlichi
• [asterisk-users] Start audio call and enable video later   John Kiniston
• [asterisk-users] Best way to update ever changing dialplans   Bertrand LUPART - Linkeo.com
• [asterisk-users] Voicemail Directory   Doug Lytle
• [asterisk-users] Asterisk not matching longest prefix with include   Doug Lytle
• [asterisk-users] Asterisk not matching longest prefix with include   Doug Lytle
• [asterisk-users] shell dialplan application blocking   Benjamin Marty
• [asterisk-users] shell dialplan application blocking   Benjamin Marty
• [asterisk-users] Dial to FastAGI application appears as 1-second CDR - how do I fix?   Tony Mountifield
• [asterisk-users] AMI manager logins - omitting from logging output?   Tony Mountifield
• [asterisk-users] Asterisk crashing on AAAA lookup   Richard Mudgett
• [asterisk-users] remove   David Mutterer
• [asterisk-users] Recommended Linux version or how to compile DAHDI on Fedora?   John Novack
• [asterisk-users] Questions about SIP From, P-Asserted-Id fields and Diversion headers ?   Olivier
• [asterisk-users] Questions about SIP From, P-Asserted-Id fields and Diversion headers ?   Olivier
• [asterisk-users] Questions about SIP From, P-Asserted-Id fields and Diversion headers ?   Olivier
• [asterisk-users] How to ignore REFER entirely with chan_sip or PJSIP ?   Olivier
• [asterisk-users] Do you set chan_sip's ignoresdpversion to true ?   Olivier
• [asterisk-users] How to execute priorities following a caller hangup in a successful Dial?   David P
• [asterisk-users] How to execute priorities following a caller hangup in a successful Dial?   David P
• [asterisk-users] How to execute priorities following a caller hangup in a successful Dial?   David P
• [asterisk-users] How to execute priorities following a caller hangup in a successful Dial?   David P
• [asterisk-users] How to execute priorities following a caller hangup in a successful Dial?   David P
• [asterisk-users] How to execute priorities following a caller hangup in a successful Dial?   David P
• [asterisk-users] Only 8kHz recorded after disallowing all but G722 codec on inbound   David P
• [asterisk-users] Only 8kHz recorded after disallowing all but G722 codec on inbound   David P
• [asterisk-users] Only 8kHz recorded after disallowing all but G722 codec on inbound   David P
• [asterisk-users] [Asterisk-video] (no subject)   Pankaj Pandey
• [asterisk-users] Asterisk receiving 415 Unsupported Media Type upon T.38 invite behaving absolutely weird.   Benoit Panizzon
• [asterisk-users] Asterisk kafka connector feedback   Alex Pappas
• [asterisk-users] Recommended Linux version or how to compile DAHDI on Fedora?   Kris Stark
• [asterisk-users] How to execute priorities following a caller hangup in a successful Dial?   Antony Stone
• [asterisk-users] Using ControlPlayback with AWS S3   Antony Stone
• [asterisk-users] Using ControlPlayback with AWS S3   Antony Stone
• [asterisk-users] AMI manager logins - omitting from logging output?   Antony Stone
• [asterisk-users] AMI manager logins - omitting from logging output?   Antony Stone
• [asterisk-users] Cancel   Karen Stroebele
• [asterisk-users] Certified Asterisk 13.21-cert1 Now Available   Asterisk Development Team
• [asterisk-users] Asterisk 15.4.1, 13.21.1, 14.7.7, 13.18-cert4 and 13.21-cert2 Now Available (Security)   Asterisk Development Team
• [asterisk-users] AST-2018-007: Infinite loop when reading iostreams   Asterisk Security Team
• [asterisk-users] AST-2018-008: PJSIP endpoint presence disclosure when using ACL   Asterisk Security Team
• [asterisk-users] Start audio call and enable video later   Stefan Tichy
• [asterisk-users] pjsip doesn't function   Marko Tirs
• [asterisk-users] Questions about SIP From, P-Asserted-Id fields and Diversion headers ?   Daniel Tryba
• [asterisk-users] How to ignore REFER entirely with chan_sip or PJSIP ?   Daniel Tryba
• [asterisk-users] Do you set chan_sip's ignoresdpversion to true ?   Daniel Tryba
• [asterisk-users] MixMonitor recording when in the holding bridge   Patrick Wakano
• [asterisk-users] AMI status events with res_fax_spandsp.so   Steven Wheeler
• [asterisk-users] How to execute priorities following a caller hangup in a successful Dial?   Eric Wieling
• [asterisk-users] How to execute priorities following a caller hangup in a successful Dial?   Eric Wieling
• [asterisk-users] How to execute priorities following a caller hangup in a successful Dial?   Eric Wieling
• [asterisk-users] getting real sip status after dial   Eric Wieling
• [asterisk-users] getting real sip status after dial   Eric Wieling
• [asterisk-users] getting real sip status after dial   Eric Wieling
• [asterisk-users] GSM card or GSM adaptor?   bilal ghayyad
• [asterisk-users] Busy indicator for FXO line or extension   bilal ghayyad
• [asterisk-users] Button for call forward and button for pickup call of another extension   bilal ghayyad
• [asterisk-users] Recommended Linux version or how to compile DAHDI on Fedora?   cvandesande at opendmz.com

Last message date: Sat Jun 30 17:41:34 CDT 2018
Archived on: Sat Jun 30 17:41:28 CDT 2018

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