November 2014 Archives by date

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Starting: Sun Nov 2 10:35:38 CST 2014
Ending: Sat Nov 29 01:10:37 CST 2014
Messages: 210

• [asterisk-users] sslv3 alert handshake failure error   Atul Thosar
• [asterisk-users] sslv3 alert handshake failure error   Jeffrey Walton
• [asterisk-users] issue with NAT   Tom Braarup Cuykens
• [asterisk-users] issue with NAT   Rainer Piper
• [asterisk-users] issue with NAT   Rainer Piper
• [asterisk-users] sslv3 alert handshake failure error   Atul Thosar
• [asterisk-users] issue with NAT   Matthew Jordan
• [asterisk-users] Asterisk 12 - zombie processes   Yaron Nachum
• [asterisk-users] queue log realtime mysql   Jonas Kellens
• [asterisk-users] queue log realtime mysql   Ishfaq Malik
• [asterisk-users] queue log realtime mysql   Jonas Kellens
• [asterisk-users] SPA504G auto answer   Larry Moore
• [asterisk-users] Asterisk 12 - zombie processes   Matthew Jordan
• [asterisk-users] Asterisk SIP UUI Protocol   Gopalakrishnan N
• [asterisk-users] Hangup Chanel when a peer unregisters   Pat Collins
• [asterisk-users] Asterisk 12 - zombie processes   Yaron Nachum
• [asterisk-users] Hangup Chanel when a peer unregisters   Gareth Blades
• [asterisk-users] Asterisk 12 - zombie processes   Yaron Nachum
• [asterisk-users] queue log realtime mysql   Jonas Kellens
• [asterisk-users] Hangup Chanel when a peer unregisters   Pat Collins
• [asterisk-users] Configure Asterisk as SIP UA using NAT   Bogdan Cristea
• [asterisk-users] Function to get mailbox for a PJSIP Endpoint?   John Kiniston
• [asterisk-users] Function to get mailbox for a PJSIP Endpoint?   Joshua Colp
• [asterisk-users] Function to get mailbox for a PJSIP Endpoint?   John Kiniston
• [asterisk-users] detect volume level change on SIP channel   Ivan Vujisic
• [asterisk-users] Function to get mailbox for a PJSIP Endpoint?   Joshua Colp
• [asterisk-users] Asterisk and OSX   ilie neagu
• [asterisk-users] Asterisk 12 - MGCP realtime gw load   José Seabra
• [asterisk-users] Function to get mailbox for a PJSIP Endpoint?   George Joseph
• [asterisk-users] How to find RTP address of ongoing call?   Markus
• [asterisk-users] How to find RTP address of ongoing call?   Bryan Burroughs
• [asterisk-users] taskprocessor fails to allocate memory   CDR
• [asterisk-users] Asterisk 12 is broken   CDR
• [asterisk-users] One thread per peer   CDR
• [asterisk-users] One thread per peer   Matthew Jordan
• [asterisk-users] Subscribe event "ua-profile"   Norman Laidla
• [asterisk-users] Subscribe event "ua-profile"   Joshua Colp
• [asterisk-users] Subscribe event "ua-profile"   Norman Laidla
• [asterisk-users] Subscribe event "ua-profile"   Steven Howes
• [asterisk-users] Subscribe event "ua-profile"   Joshua Colp
• [asterisk-users] Function to get mailbox for a PJSIP Endpoint?   John Kiniston
• [asterisk-users] Webinar Gratuíto, Como evitar fraudes em telefonia   Flavio Goncalves
• [asterisk-users] Webinar Gratuíto, Como evitar fraudes em telefonia   Matthew Jordan
• [asterisk-users] Asterisk 1.8.32.0 Now Available   Asterisk Development Team
• [asterisk-users] Asterisk 11.14.0 Now Available   Asterisk Development Team
• [asterisk-users] Asterisk 12.7.0 Now Available   Asterisk Development Team
• [asterisk-users] odbc connection timeout varable   Israel Gottlieb
• [asterisk-users] odbc connection timeout varable   jg
• [asterisk-users] odbc connection timeout varable   Israel Gottlieb
• [asterisk-users] odbc connection timeout varable   jg
• [asterisk-users] odbc connection timeout varable   Tech Support
• [asterisk-users] odbc connection timeout varable   Ishfaq Malik
• [asterisk-users] odbc connection timeout varable   jg
• [asterisk-users] odbc connection timeout varable   Israel Gottlieb
• [asterisk-users] odbc connection timeout varable   jg
• [asterisk-users] odbc connection timeout varable   Israel Gottlieb
• [asterisk-users] odbc connection timeout varable   jg
• [asterisk-users] ITSP Gateway Solution?   Todd R.
• [asterisk-users] Incoming calls to a GSM gateway & "SIP/2.0 401 Unauthorized" response when dial 7777 to Asterisk   Luis Eduardo Cortes
• [asterisk-users] ITSP Gateway Solution?   Markos Vakondios
• [asterisk-users] ITSP Gateway Solution?   Mitul Limbani
• [asterisk-users] Asterisk 12 crashes on CANCEL received during ANSWER handlingl   Yaron Nachum
• [asterisk-users] Asterisk 12 crashes on CANCEL received during ANSWER handlingl   Joshua Colp
• [asterisk-users] E1 - Cisco - Asterisk and vice verso   Dmitriy Sirant
• [asterisk-users] odbc connection timeout varable   Matthew Jordan
• [asterisk-users] odbc connection timeout varable   Israel Gottlieb
• [asterisk-users] Asterisk 12 crashes on CANCEL received during ANSWER handlingl   Yaron Nachum
• [asterisk-users] Como unir webrtc con asterisk???   Dario Estupinan
• [asterisk-users] Webinar Gratuíto, Como evitar fraudes em telefonia   isamu
• [asterisk-users] [SOLVED] Re: Incoming calls to a GSM gateway & "SIP/2.0 401 Unauthorized" response when dial 7777 to Asterisk   Luis Eduardo Cortes
• [asterisk-users] Erratic calls through NAT-ed server   Norman Laidla
• [asterisk-users] pjsip phoneprov realtime?   John Kiniston
• [asterisk-users] pjsip phoneprov realtime?   George Joseph
• [asterisk-users] pjsip phoneprov realtime?   George Joseph
• [asterisk-users] pjsip phoneprov realtime?   John Kiniston
• [asterisk-users] pjsip phoneprov realtime?   George Joseph
• [asterisk-users] SLA (Shared Line Appearance) and realtime   Leandro Dardini
• [asterisk-users] Asterisk 13 confbridge recordings not working   Bill Barron
• [asterisk-users] Como unir webrtc con asterisk???   Rusty Newton
• [asterisk-users] Erratic calls through NAT-ed server   Rusty Newton
• [asterisk-users] E1 - Cisco - Asterisk and vice verso   Rusty Newton
• [asterisk-users] E1 - Cisco - Asterisk and vice verso   Kevin Larsen
• [asterisk-users] Como unir webrtc con asterisk???   symack
• [asterisk-users] E1 - Cisco - Asterisk and vice verso   Bruce Ferrell
• [asterisk-users] Sippy Cup   Murthy Gandikota
• [asterisk-users] E1 - Cisco - Asterisk and vice verso   Dmitriy Sirant
• [asterisk-users] from: rschroe at gmail.com   rschroe at gmail.com
• [asterisk-users] Erratic calls through NAT-ed server   Norman Laidla
• [asterisk-users] OT - Is T.38 possible on SPA8800 FXO port ?   Olivier
• [asterisk-users] OT - Is T.38 possible on SPA8800 FXO port ?   jg
• [asterisk-users] OT - Is T.38 possible on SPA8800 FXO port ?   Larry Moore
• [asterisk-users] OT - Is T.38 possible on SPA8800 FXO port ?   Larry Moore
• [asterisk-users] OT - Is T.38 possible on SPA8800 FXO port ?   Olivier
• [asterisk-users] Code send by asterisk when a peer hangs up   Bogdan Cristea
• [asterisk-users] OT - Is T.38 possible on SPA8800 FXO port ?   Larry Moore
• [asterisk-users] Get the status of a PJSIP endpoint?   John Kiniston
• [asterisk-users] motif and other xmpp   ricky gutierrez
• [asterisk-users] google voice   George Wu
• [asterisk-users] OT - Is T.38 possible on SPA8800 FXO port ?   Olivier
• [asterisk-users] OT - Is T.38 possible on SPA8800 FXO port ?   Kevin Larsen
• [asterisk-users] OT - Is T.38 possible on SPA8800 FXO port ?   Olivier
• [asterisk-users] Get the status of a PJSIP endpoint?   George Joseph
• [asterisk-users] Get the status of a PJSIP endpoint?   George Joseph
• [asterisk-users] Get the status of a PJSIP endpoint?   John Kiniston
• [asterisk-users] AGI and AMI in PHP -- What's current?   Steve Edwards
• [asterisk-users] AGI and AMI in PHP -- What's current?   Eric Wieling
• [asterisk-users] AGI and AMI in PHP -- What's current?   Steve Edwards
• [asterisk-users] Upgraded to 13 and now "Mailbox" is empty in sip show peers   Jayson Baker
• [asterisk-users] AGI and AMI in PHP -- What's current?   Eric Wieling
• [asterisk-users] AGI and AMI in PHP -- What's current?   Steve Edwards
• [asterisk-users] AGI and AMI in PHP -- What's current?   Johan Wilfer
• [asterisk-users] Upgraded to 13 and now "Mailbox" is empty in sip show peers   Matthew Jordan
• [asterisk-users] Upgraded to 13 and now "Mailbox" is empty in sip show peers   Jayson Baker
• [asterisk-users] Upgraded to 13 and now "Mailbox" is empty in sip show peers   Matthew Jordan
• [asterisk-users] Upgraded to 13 and now "Mailbox" is empty in sip show peers   Jayson Baker
• [asterisk-users] Upgraded to 13 and now "Mailbox" is empty in sip show peers   Jayson Baker
• [asterisk-users] Upgraded to 13 and now "Mailbox" is empty in sip show peers   Steve Edwards
• [asterisk-users] Upgraded to 13 and now "Mailbox" is empty in sip show peers   Jayson Baker
• [asterisk-users] Asterisk problems   Jerome SCHEVINGT
• [asterisk-users] Asterisk problems   Alonso Genis
• [asterisk-users] Upgraded to 13 and now "Mailbox" is empty in sip show peers   A J Stiles
• [asterisk-users] Upgraded to 13 and now "Mailbox" is empty in sip show peers   Jayson Baker
• [asterisk-users] Upgraded to 13 and now "Mailbox" is empty in sip show peers   A J Stiles
• [asterisk-users] Upgraded to 13 and now "Mailbox" is empty in sip show peers   Jayson Baker
• [asterisk-users] Upgraded to 13 and now "Mailbox" is empty in sip show peers   A J Stiles
• [asterisk-users] Upgraded to 13 and now "Mailbox" is empty in sip show peers   Eric Wieling
• [asterisk-users] Asterisk13 don't execute h exten inside macros   Alonso Genis
• [asterisk-users] Asterisk13 don't execute h exten inside macros   John Kiniston
• [asterisk-users] Error saving cdr at h exten in Asterisk13   Alonso Genis
• [asterisk-users] Error saving cdr at h exten in Asterisk13   Matthew Jordan
• [asterisk-users] Upgraded to 13 and now "Mailbox" is empty in sip show peers   Jayson Baker
• [asterisk-users] google voice   Rusty Newton
• [asterisk-users] Upgraded to 13 and now "Mailbox" is empty in sip show peers   John Novack
• [asterisk-users] Asterisk 1.8.28-cert3, 1.8.32.1, 11.6-cert8, 11.14.1, 12.7.1, 13.0.1 Now Available (Security Release)   Asterisk Development Team
• [asterisk-users] AST-2014-012: Mixed IP address families in access control lists may permit unwanted traffic.   Asterisk Security Team
• [asterisk-users] AST-2014-013: PJSIP ACLs are not loaded on startup   Asterisk Security Team
• [asterisk-users] AST-2014-014: High call load may result in hung channels in ConfBridge.   Asterisk Security Team
• [asterisk-users] AST-2014-015: Remote Crash Vulnerability in PJSIP channel driver   Asterisk Security Team
• [asterisk-users] AST-2014-016: Remote Crash Vulnerability in PJSIP channel driver   Asterisk Security Team
• [asterisk-users] AST-2014-017: <font size="3" style="font-size: 12pt">Permission escalation through ConfBridge actions/dialplan functions</font>   Asterisk Security Team
• [asterisk-users] AST-2014-018: AMI permission escalation through DB dialplan function   Asterisk Security Team
• [asterisk-users] Upgraded to 13 and now "Mailbox" is empty in sip show peers   A J Stiles
• [asterisk-users] One way audio internal   Andrew Colin
• [asterisk-users] Asterisk13 don't execute h exten inside macros   Alonso Genis
• [asterisk-users] Error saving cdr at h exten in Asterisk13   Alonso Genis
• [asterisk-users] One way audio internal   A J Stiles
• [asterisk-users] One way audio internal   Mitul Limbani
• [asterisk-users] One way audio internal   Andrew Colin
• [asterisk-users] One way audio internal   Mitul Limbani
• [asterisk-users] One way audio internal   Andrew Colin
• [asterisk-users] One way audio internal   Andrew Colin
• [asterisk-users] One way audio internal   A J Stiles
• [asterisk-users] High resident memory with 11.14.0 ?   James Lamanna
• [asterisk-users] Not able to register an Extension   akhilesh chand
• [asterisk-users] Not able to register an Extension   Alonso Genis
• [asterisk-users] Not able to register an Extension   akhilesh chand
• [asterisk-users] High resident memory with 11.14.0 ?   James Lamanna
• [asterisk-users] Not able to register an Extension   Alonso Genis
• [asterisk-users] SIP call drops after 32 seconds, but only when....   Yves A.
• [asterisk-users] SIP call drops after 32 seconds, but only when....   Andreas Sikkema
• [asterisk-users] SIP call drops after 32 seconds, but only when....   Yves A.
• [asterisk-users] Not able to register an Extension   akhilesh chand
• [asterisk-users] SIP call drops after 32 seconds, but only when....   Eric Wieling
• [asterisk-users] SIP call drops after 32 seconds, but only when....   Rafael Visser
• [asterisk-users] SIP call drops after 32 seconds, but only when....   Ron Wheeler
• [asterisk-users] High resident memory with 11.14.0 ?   Freddi Hansen
• [asterisk-users] SPA504G auto answer   Larry Moore
• [asterisk-users] Dahdi fxo vs sip blf   James Cloos
• [asterisk-users] SIP call drops after 32 seconds, but only when....   Yves A.
• [asterisk-users] SIP call drops after 32 seconds, but only when....   Yves A.
• [asterisk-users] "!" in dial-pattern not work with overlap dialing   s m
• [asterisk-users] how to set "timerb" in sip.conf   Yves A.
• [asterisk-users] Softphone signals busy although it isn´t   Yves A.
• [asterisk-users] High resident memory with 11.14.0 ?   James Lamanna
• [asterisk-users] asterisk and elastix   Salaheddine Elharit
• [asterisk-users] asterisk and elastix   Alex Villací­s Lasso
• [asterisk-users] "!" in dial-pattern not work with overlap dialing   Matthew Jordan
• [asterisk-users] High resident memory with 11.14.0 ?   James Lamanna
• [asterisk-users] Planned maintenance for community services, Tuesday, 25th of November 2014   Digium's Asterisk Development Team
• [asterisk-users] Size of frame.c cache in Asterisk 11?   James Lamanna
• [asterisk-users] High resident memory with 11.14.0 ?   Matthew Jordan
• [asterisk-users] park()-command always parks on default 701   Jonas Kellens
• [asterisk-users] High resident memory with 11.14.0 ?   James Lamanna
• [asterisk-users] asterisk prompt   Martin Vegter
• [asterisk-users] Prohibit transfer to one extension   Tom Peters
• [asterisk-users] motif and other xmpp   ricky gutierrez
• [asterisk-users] Strange Issue: asterisk deleted   Antoine Megalla
• [asterisk-users] Strange Issue: asterisk deleted   Thorsten Göllner
• [asterisk-users] Strange Issue: asterisk deleted   Tzafrir Cohen
• [asterisk-users] Strange Issue: asterisk deleted   Antoine Megalla
• [asterisk-users] High resident memory with 11.14.0 ?   James Lamanna
• [asterisk-users] Strange Issue: asterisk deleted   Chad Wallace
• [asterisk-users] Strange Issue: asterisk deleted   Marie Fischer
• [asterisk-users] SIP call drops after 32 seconds, but only when....   Marie Fischer
• [asterisk-users] SIP call drops after 32 seconds, but only when....   Amit Patkar
• [asterisk-users] Strange Issue: asterisk deleted   Thorsten Göllner
• [asterisk-users] Strange Issue: asterisk deleted   Antoine Megalla
• [asterisk-users] Strange Issue: asterisk deleted   A J Stiles
• [asterisk-users] Strange Issue: asterisk deleted   Antoine Megalla
• [asterisk-users] Strange Issue: asterisk deleted   John Novack
• [asterisk-users] day night service toggle   Control Oye
• [asterisk-users] Asterisk consuming high cpu usage   Mayank Kumar Gour
• [asterisk-users] Asterisk consuming high cpu usage   Mayank Kumar Gour
• [asterisk-users] ICE consuming High CPU   Mayank Kumar Gour
• [asterisk-users] day night service toggle   JB
• [asterisk-users] day night service toggle   A J Stiles
• [asterisk-users] day night service toggle   Steve Edwards
• [asterisk-users] High resident memory with 11.14.0 ?   James Lamanna
• [asterisk-users] RT voicemail greetings not played   Mike
• [asterisk-users] RT voicemail greetings not played   Mike

Last message date: Sat Nov 29 01:10:37 CST 2014
Archived on: Sat Nov 29 01:09:16 CST 2014

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