May 2017 Archives by subject

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Starting: Mon May 1 09:02:27 CDT 2017
Ending: Wed May 31 18:34:59 CDT 2017
Messages: 156

• [asterisk-users] [OT] Suggestion for VoIP-App   Luca Bertoncello
• [asterisk-users] AGICommand_exec remove my double quotation   Lợi Đặng
• [asterisk-users] AGICommand_exec remove my double quotation   Lợi Đặng
• [asterisk-users] AMI Originate not working   Thomas
• [asterisk-users] AMI Originate not working   Faheem Muhammad
• [asterisk-users] app_jack unavailable   andre castro
• [asterisk-users] app_jack unavailable   J Montoya or A J Stiles
• [asterisk-users] app_jack unavailable   andre castro
• [asterisk-users] app_jack unavailable   Joshua Colp
• [asterisk-users] app_jack unavailable   J Montoya or A J Stiles
• [asterisk-users] app_jack unavailable   andre castro
• [asterisk-users] AST-2017-002: Buffer Overrun in PJSIP transaction layer   Asterisk Security Team
• [asterisk-users] AST-2017-003: Crash in PJSIP multi-part body parser   Asterisk Security Team
• [asterisk-users] AST-2017-004: Memory exhaustion on short SCCP packets   Asterisk Security Team
• [asterisk-users] Asterisk 13.13-cert4, 13.15.1, 14.4.1 Now Available (Security Release)   Asterisk Team
• [asterisk-users] asterisk 13.15.0 stopping/crashing   marek cervenka
• [asterisk-users] asterisk 13.15.0 stopping/crashing   marek cervenka
• [asterisk-users] asterisk 13.15.0 stopping/crashing   marek cervenka
• [asterisk-users] asterisk 13.15.0 stopping/crashing   Joshua Colp
• [asterisk-users] Asterisk 13.16.0 Now Available   Asterisk Development Team
• [asterisk-users] Asterisk 13 queue and DND phones   Mike
• [asterisk-users] Asterisk 13 queue and DND phones   Héctor Royo
• [asterisk-users] Asterisk 13 queue and DND phones   Mike
• [asterisk-users] Asterisk 14.3.1 > 14.4.1 upgrade pjsip nat broken?   Christopher van de Sande
• [asterisk-users] Asterisk 14.3.1 > 14.4.1 upgrade pjsip nat broken?   Joshua Colp
• [asterisk-users] Asterisk 14.3.1 > 14.4.1 upgrade pjsip nat broken?   Joshua Colp
• [asterisk-users] Asterisk 14.3.1 > 14.4.1 upgrade pjsip nat broken?   Christopher van de Sande
• [asterisk-users] Asterisk 14.5.0 Now Available   Asterisk Development Team
• [asterisk-users] Asterisk 14 audio quality with remote files   Tiago Ferreira
• [asterisk-users] Asterisk 14 audio quality with remote files   Tiago Ferreira
• [asterisk-users] Asterisk 14 audio quality with remote files   Matthew Jordan
• [asterisk-users] Automatically dial a number, then an extension   Tech Support
• [asterisk-users] Automatically dial a number, then an extension   Administrator TOOTAI
• [asterisk-users] Automatically dial a number, then an extension   Tony Mountifield
• [asterisk-users] Automatically dial a number, then an extension   Victor Villarreal
• [asterisk-users] Automatically dial a number, then an extension   Tech Support
• [asterisk-users] Automatically dial a number, then an extension   Tech Support
• [asterisk-users] Automatically dial a number, then an extension   Antony Stone
• [asterisk-users] Automatically dial a number, then an extension   Tech Support
• [asterisk-users] Automatically dial a number, then an extension   Antony Stone
• [asterisk-users] Automatically dial a number, then an extension   Tech Support
• [asterisk-users] Automatically dial a number, then an extension   Mark Wiater
• [asterisk-users] Best way to know a call is being transfered   Jonas Kellens
• [asterisk-users] Best way to know a call is being transfered   Jonathan H
• [asterisk-users] Best way to know a call is being transfered   Jonas Kellens
• [asterisk-users] Best way to know a call is being transfered   Marcelo Terres
• [asterisk-users] Call does not go to voicemail   thelma at sys-concept.com
• [asterisk-users] Call does not go to voicemail   Tim S
• [asterisk-users] Call does not go to voicemail   thelma at sys-concept.com
• [asterisk-users] Call does not go voicemail   thelma at sys-concept.com
• [asterisk-users] Call does not go voicemail   Tim S
• [asterisk-users] Call does not go voicemail   thelma at sys-concept.com
• [asterisk-users] Call does not go voicemail   Tim S
• [asterisk-users] Callee id over chan_sip trunk   Dmitry Melekhov
• [asterisk-users] Callee id over chan_sip trunk   Sebastian Nielsen
• [asterisk-users] Can't compile asterisk-certified-11.6-cert16 on Ubuntu 16   Tech Support
• [asterisk-users] Can't compile asterisk-certified-11.6-cert16 on Ubuntu 16   Joshua Colp
• [asterisk-users] Cisco 7942G (SIP42.9-4-2) Failover Configuration [SEC=UNCLASSIFIED]   Calum Power
• [asterisk-users] cmd AGI(), maximum script time.   Dmitry Melekhov
• [asterisk-users] cmd AGI(), maximum script time.   Steve Edwards
• [asterisk-users] CM for menuselect choices   Richard Kenner
• [asterisk-users] CM for menuselect choices   Antony Stone
• [asterisk-users] CM for menuselect choices   Richard Kenner
• [asterisk-users] CM for menuselect choices   Antony Stone
• [asterisk-users] CM for menuselect choices   Tzafrir Cohen
• [asterisk-users] CM for menuselect choices   Richard Kenner
• [asterisk-users] connecting two asterisks - transfer=no   thelma at sys-concept.com
• [asterisk-users] Dial an extension to modify dialplan   Frank Vanoni
• [asterisk-users] Dial an extension to modify dialplan   Marcelo Terres
• [asterisk-users] Dial an extension to modify dialplan   J Montoya or A J Stiles
• [asterisk-users] Dial an extension to modify dialplan   John Kiniston
• [asterisk-users] Dial an extension to modify dialplan   Antony Stone
• [asterisk-users] Dial an extension to modify dialplan   Marcelo Terres
• [asterisk-users] Dial an extension to modify dialplan   Daniel Journo
• [asterisk-users] Dial an extension to modify dialplan   Héctor Royo
• [asterisk-users] Dial an extension to modify dialplan   Stefan Becker
• [asterisk-users] Dial an extension to modify dialplan   Frank Vanoni
• [asterisk-users] Dial an extension to modify dialplan   Frank Vanoni
• [asterisk-users] Feature Code to Meeting Room   Daniel Journo
• [asterisk-users] Feature Code to Meeting Room   Marcelo Terres
• [asterisk-users] hangup handlers & unwanted cdr   marek cervenka
• [asterisk-users] How to detect fake CallerID? (8xx?)   Steve Edwards
• [asterisk-users] How to detect fake CallerID? (8xx?)   Doug Lytle
• [asterisk-users] How to detect fake CallerID? (8xx?)   Adam Goldberg
• [asterisk-users] How to detect fake CallerID? (8xx?)   Andrew Latham
• [asterisk-users] How to detect fake CallerID? (8xx?)   J Montoya or A J Stiles
• [asterisk-users] How to detect fake CallerID? (8xx?)   Don Kelly
• [asterisk-users] How to detect fake CallerID? (8xx?)   Steve Edwards
• [asterisk-users] How to detect fake CallerID? (8xx?)   Sebastian Nielsen
• [asterisk-users] How to detect fake CallerID? (8xx?)   Don Kelly
• [asterisk-users] How to detect fake CallerID? (8xx?)   Sebastian Nielsen
• [asterisk-users] How to detect fake CallerID? (8xx?)   D'Arcy Cain
• [asterisk-users] How to detect fake CallerID? (8xx?)   Tim S
• [asterisk-users] How to detect fake CallerID? (8xx?)   J Montoya or A J Stiles
• [asterisk-users] How to detect fake CallerID? (8xx?)   Adam Goldberg
• [asterisk-users] How to detect fake CallerID? (8xx?)   Don Kelly
• [asterisk-users] How to detect fake CallerID? (8xx?)   Sebastian Nielsen
• [asterisk-users] How to detect fake CallerID? (8xx?)   Don Kelly
• [asterisk-users] iaxModem pickup problem   James B. Byrne
• [asterisk-users] iaxModem pickup problem   James B. Byrne
• [asterisk-users] iaxModem pickup problem   Telium Technical Support
• [asterisk-users] iaxModem pickup problem   James B. Byrne
• [asterisk-users] iaxModem pickup problem   Telium Technical Support
• [asterisk-users] iaxModem pickup problem   James B. Byrne
• [asterisk-users] JACK_HOOK Auto fallthrough   andre castro
• [asterisk-users] Need to restart Asterisk if remote server not working?   Luca Bertoncello
• [asterisk-users] Need to restart Asterisk if remote server not working?   Antony Stone
• [asterisk-users] Need to restart Asterisk if remote server not working?   Luca Bertoncello
• [asterisk-users] Need to restart Asterisk if remote server not working?   Antony Stone
• [asterisk-users] Need to restart Asterisk if remote server not working?   Max Grobecker
• [asterisk-users] Need to restart Asterisk if remote server not working?   Luca Bertoncello
• [asterisk-users] Need to restart Asterisk if remote server not working?   Max Grobecker
• [asterisk-users] Need to restart Asterisk if remote server not working?   Luca Bertoncello
• [asterisk-users] news from me   nick005
• [asterisk-users] operator panel   Kseniya Blashchuk
• [asterisk-users] OT: Want to capture all SIP messages   Steve Edwards
• [asterisk-users] OT: Want to capture all SIP messages   Mark Wiater
• [asterisk-users] OT: Want to capture all SIP messages   Daniel Tryba
• [asterisk-users] OT: Want to capture all SIP messages   Barry Flanagan
• [asterisk-users] OT: Want to capture all SIP messages   Barry Flanagan
• [asterisk-users] OT: Want to capture all SIP messages   Steve Edwards
• [asterisk-users] OT: Want to capture all SIP messages   Steve Edwards
• [asterisk-users] OT: Want to capture all SIP messages   Daniel Tryba
• [asterisk-users] OT: Want to capture all SIP messages   Steve Edwards
• [asterisk-users] OT: Want to capture all SIP messages   Matt Riddell
• [asterisk-users] OT: Want to capture all SIP messages   Steve Edwards
• [asterisk-users] OT: Want to capture all SIP messages   Jeff LaCoursiere
• [asterisk-users] OT: Want to capture all SIP messages   Steve Edwards
• [asterisk-users] OT: Want to capture all SIP messages   Pete Mundy
• [asterisk-users] OT: Want to capture all SIP messages   Steve Edwards
• [asterisk-users] Outbound T.38 via RTP with pjsip does not work as expected   Michael Maier
• [asterisk-users] pjsip: asterisk can't decide which codec to use   Michael Maier
• [asterisk-users] pjsip: asterisk can't decide which codec to use   Michael Maier
• [asterisk-users] pjsip: asterisk can't decide which codec to use   Joshua Colp
• [asterisk-users] pjsip: asterisk can't decide which codec to use   Michael Maier
• [asterisk-users] pjsip direct_media=yes and "unknown" endpoints   Daniel Tryba
• [asterisk-users] R: [OT] Suggestion for VoIP-App   michele.pinassi
• [asterisk-users] SIP Trunk over Proxy (matching ip of outbound proxy in incomming calls)   Benoit Panizzon
• [asterisk-users] SIP Trunk over Proxy (matching ip of outbound proxy in incomming calls)   Kseniya Blashchuk
• [asterisk-users] SIP Trunk over Proxy (matching ip of outbound proxy in incomming calls)   Dave Platt
• [asterisk-users] SNOM870 provisioning BLF settings   James B. Byrne
• [asterisk-users] Some questions regarding jitterbuffer in asterisk / pjsip   Michael Maier
• [asterisk-users] Surrogate channels   Patrick Wakano
• [asterisk-users] Surrogate channels   Richard Mudgett
• [asterisk-users] Surrogate channels   Patrick Wakano
• [asterisk-users] SwitchVox and Asterisk   Luca Pradovera
• [asterisk-users] SwitchVox and Asterisk   Antony Stone
• [asterisk-users] SwitchVox and Asterisk   Luca Pradovera
• [asterisk-users] Using queue priorities to add agents   Steve Davies
• [asterisk-users] Using queue priorities to add agents   Alexander Lopez
• [asterisk-users] Using queue priorities to add agents   John Kiniston
• [asterisk-users] Using queue priorities to add agents   Steve Davies
• [asterisk-users] Using queue priorities to add agents   Steve Davies
• [asterisk-users] wonderful stuff   Marco Signorini
• [asterisk-users] ☀Re: very useful information   Marco Signorini
• [asterisk-users] ✈Re: what a nice day   Marco Signorini

Last message date: Wed May 31 18:34:59 CDT 2017
Archived on: Wed May 31 18:35:06 CDT 2017

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