October 2014 Archives by author

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Starting: Wed Oct 1 04:40:26 CDT 2014
Ending: Fri Oct 31 12:59:24 CDT 2014
Messages: 336

• [asterisk-users] Asterisk 11.9.0 crash and restart   為近 吉摩（情報システム本部）- Tamechika Yoshikiyo -
• [asterisk-users] Asterisk 11.9.0 crash and restart   為近 吉摩（情報システム本部）- Tamechika Yoshikiyo -
• [asterisk-users] Asterisk 12 CDR dst field empty   A.Santoro
• [asterisk-users] Asterisk 12 CDR dst field empty - cdrlog.txt (0/1)   A.Santoro
• [asterisk-users] Asterisk 12 CDR dst field empty - cdrlog.txt (1/1)   A.Santoro
• [asterisk-users] Video call   Brahim Abidar
• [asterisk-users] SIP over 3G Mobile Network using NAT   Chirag Ajmera
• [asterisk-users] SIP over 3G Mobile Network using NAT   Chirag Ajmera
• [asterisk-users] SIP over 3G Mobile Network using NAT   Chirag Ajmera
• [asterisk-users] AstriDevCon 2014: Agenda item Deprecate AMI/AGI (Ben Klang)   Paul Albrecht
• [asterisk-users] [asterisk-dev] AstriDevCon 2014: Agenda item Deprecate AMI/AGI(Ben Klang)   Paul Albrecht
• [asterisk-users] [asterisk-dev] AstriDevCon 2014: Agenda itemDeprecate AMI/AGI(Ben Klang)   Paul Albrecht
• [asterisk-users] [asterisk-dev] AstriDevCon 2014: Agenda item DeprecateAMI/AGI(Ben Klang)   Paul Albrecht
• [asterisk-users] AstriDevCon 2014: AgendaitemDeprecate AMI/AGI(Ben Klang)   Paul Albrecht
• [asterisk-users] AstriDevCon 2014: AgendaitemDeprecate AMI/AGI(Ben Klang)   Paul Albrecht
• [asterisk-users] AstriDevCon 2014:Agenda item Deprecate AMI/AGI(Ben Klang)   Paul Albrecht
• [asterisk-users] AstriDevCon 2014: Agenda item Deprecate AMI/AGI(Ben Klang)   Paul Albrecht
• [asterisk-users] AstriDevCon 2014: Agenda item Deprecate AMI/AGI(Ben Klang)   Paul Albrecht
• [asterisk-users] AstriDevCon 2014:Agenda item Deprecate AMI/AGI(Ben Klang)   Paul Albrecht
• [asterisk-users] AstriDevCon 2014:Agenda item Deprecate AMI/AGI (Ben Klang)   Paul Albrecht
• [asterisk-users] AppKonference 2.6   Paul Albrecht
• [asterisk-users] AstriDevCon 2014: Agenda item Deprecate AMI/AGI (Ben Klang)   Paul Albrecht
• [asterisk-users] AstriDevCon 2014: Agenda item Deprecate AMI/AGI (Ben Klang)   Paul Albrecht
• [asterisk-users] AstriDevCon 2014: Agenda item Deprecate AMI/AGI (Ben Klang)   Paul Albrecht
• [asterisk-users] AstriDevCon 2014: Agenda item Deprecate AMI/AGI(Ben Klang)   Paul Albrecht
• [asterisk-users] Asterisk and Kamailio Load Balancing   Mahmoud Ramadan Ali
• [asterisk-users] Asterisk registration with Dialogic HMP.   Shahnaz Ali
• [asterisk-users] Asterisk GOIP Outgoing Callerid not working   Stephan Alz
• [asterisk-users] Asterisk GOIP Outgoing Callerid not working   Stephan Alz
• [asterisk-users] Asterisk Phone ( Telecom feature )   Dania Asi
• [asterisk-users] Asterisk Phone ( Telecom feature )   Dania Asi
• [asterisk-users] Asterisk Phone ( Telecom feature )   Dania Asi
• [asterisk-users] Asterisk Phone ( Telecom feature )   Dania Asi
• [asterisk-users] PBX hacked: why hundred of calls to the same number ?   Gokan Atmaca
• [asterisk-users] AstLinux 1.2.0 Released   Gokan Atmaca
• [asterisk-users] Sent ami event from AGI?   Ilya Awesome
• [asterisk-users] Sent ami event from AGI?   Nick Awesome
• [asterisk-users] No chan_sip in compiled asterisk-11.13.0   Anthony Azzopardi
• [asterisk-users] No chan_sip in compiled asterisk-11.13.0   Anthony Azzopardi
• [asterisk-users] Asterisk LTS segment faults   Grant Bagdasarian
• [asterisk-users] Asterisk LTS segment faults   Grant Bagdasarian
• [asterisk-users] how to strip +1 out of incoming number   Chris Bagnall
• [asterisk-users] PBX hacked: why hundred of calls to the same number ?   Chris Bagnall
• [asterisk-users] Query on connecting 3G MSC with Asterisk PBX Server via SIP Interface.   Chris Bagnall
• [asterisk-users] Pjsip and regcontext (for DUNDi)   Dan Ballance
• [asterisk-users] Pjsip and regcontext (for DUNDi)   Dan Ballance
• [asterisk-users] Pjsip and regcontext (for DUNDi)   Dan Ballance
• [asterisk-users] Asterisk 12 - zombie processes   Paul Belanger
• [asterisk-users] ${HASH(SIP_CAUSE,<channel-name>)}   Paul Belanger
• [asterisk-users] ${HASH(SIP_CAUSE,<channel-name>)}   Gareth Blades
• [asterisk-users] PlayTones not working   Gareth Blades
• [asterisk-users] PlayTones while in call   Gareth Blades
• [asterisk-users] allo.com gsm card with AsteriskNOW   Brian
• [asterisk-users] Ubuntu 12.04 LTS / Asterisk / apt-get upgrade / exclude packages   Ethy H. Brito
• [asterisk-users] Asterisk Crashes Randomly with Cepstral Swift TTS   Bryan Burroughs
• [asterisk-users] Asterisk Crashes Randomly with Cepstral Swift TTS   Bryan Burroughs
• [asterisk-users] Debugging issues with setup   Marco Carvalho
• [asterisk-users] howto cancel simultaneous calls - dial(sip/phone1&sip/phone2)   Marek Cervenka
• [asterisk-users] Voice Mail Questions   Carlos Chavez
• [asterisk-users] Register multiple phones to a single AOR with PJSIP   Carlos Chavez
• [asterisk-users] Dahdi problem with dahdi_genconf   Tzafrir Cohen
• [asterisk-users] PBX hacked: why hundred of calls to the same number ?   Tzafrir Cohen
• [asterisk-users] bristuff-0.4.0-RC4-xr7   Tzafrir Cohen
• [asterisk-users] Asterisk removes ice lines in sdp when calling between webrtc clients   Joshua Colp
• [asterisk-users] Setting channel musicclass from AGI   Joshua Colp
• [asterisk-users] sdp_crypto_process: Crypto life time unsupported: crypto   Joshua Colp
• [asterisk-users] sdp_crypto_process: Crypto life time unsupported: crypto   Joshua Colp
• [asterisk-users] sdp_crypto_process: Crypto life time unsupported: crypto   Joshua Colp
• [asterisk-users] MWI publish VIA pjsip for non sip channels   Joshua Colp
• [asterisk-users] MWI publish VIA pjsip for non sip channels   Joshua Colp
• [asterisk-users] deactivate SRTP in asterisk 11   Conrad
• [asterisk-users] Astricom 2014 presentations   Bogdan Cristea
• [asterisk-users] Detect hangup due to RTP timeout   David Cunningham
• [asterisk-users] Voicemail message number off by one when using ODBC storage   Leandro Dardini
• [asterisk-users] Asterisk 12.6 and MWI, no more working   Leandro Dardini
• [asterisk-users] SPA504G auto answer   Leandro Dardini
• [asterisk-users] allo.com gsm card with AsteriskNOW   David Duffett
• [asterisk-users] Astricom 2014 presentations   David Duffett
• [asterisk-users] PBX hacked: why hundred of calls to the same number ?   Michelle Dupuis
• [asterisk-users] Reset calls processed counter   Michelle Dupuis
• [asterisk-users] how to strip +1 out of incoming number   Steve Edwards
• [asterisk-users] Asterisk Phone ( Telecom feature )   Steve Edwards
• [asterisk-users] Asterisk Phone ( Telecom feature )   Steve Edwards
• [asterisk-users] Debugging issues with setup   Steve Edwards
• [asterisk-users] OT: script to remove leading and trailing silence   Steve Edwards
• [asterisk-users] debugging T.38 issues   Frederic Van Espen
• [asterisk-users] PlayTones not working   Henry Fernandes
• [asterisk-users] PlayTones while in call   Henry Fernandes
• [asterisk-users] PlayTones while in call   Henry Fernandes
• [asterisk-users] Voicemail message number off by one when using ODBC storage   Marie Fischer
• [asterisk-users] how can queue agents choose which call to answer?   Marie Fischer
• [asterisk-users] how can queue agents choose which call to answer?   Marie Fischer
• [asterisk-users] res_fax T.38 Gateway with SpanDSP - Force ReINVITE?   Dave Fullerton
• [asterisk-users] Ast 13 beta 3 - Segfault when calling on pjsip trunk with directmedia=yes   Dave Fullerton
• [asterisk-users] Ast 13 beta 3 - Segfault when calling on pjsip trunk with directmedia=yes   Dave Fullerton
• [asterisk-users] SIP over 3G Mobile Network using NAT   Jose Flores Galicia
• [asterisk-users] AMI and CDR(answer)   Murthy Gandikota
• [asterisk-users] AMI and CDR(answer)   Murthy Gandikota
• [asterisk-users] AMI and CDR(answer)   Murthy Gandikota
• [asterisk-users] AMI and CDR(answer)   Murthy Gandikota
• [asterisk-users] AMI and CDR(answer)   Murthy Gandikota
• [asterisk-users] AMI and CDR(answer)   Murthy Gandikota
• [asterisk-users] AMI and CDR(answer)   Murthy Gandikota
• [asterisk-users] AMI and CDR(answer)   Murthy Gandikota
• [asterisk-users] AMI and CDR(answer)   Murthy Gandikota
• [asterisk-users] AMI and CDR(answer)   Murthy Gandikota
• [asterisk-users] Asterisk 12 Dialplan   Murthy Gandikota
• [asterisk-users] Asterisk 12 Dialplan   Murthy Gandikota
• [asterisk-users] Asterisk 12 Dialplan   Murthy Gandikota
• [asterisk-users] Asterisk 12 Dialplan   Murthy Gandikota
• [asterisk-users] Asterisk 12 Dialplan   Murthy Gandikota
• [asterisk-users] Asterisk 12 Dialplan   Murthy Gandikota
• [asterisk-users] Asterisk 12 Dialplan   Murthy Gandikota
• [asterisk-users] Do subroutines need their own h extension?   Daniel Gonzalez
• [asterisk-users] OpenSIPS Summit Oct 21st before Astricon   Alex Goulis
• [asterisk-users] Queue: Passing params to macros and gosubs   Ed Greenberg
• [asterisk-users] Sent ami event from AGI?   Scott Griepentrog
• [asterisk-users] Voice Mail Questions   Scott Griepentrog
• [asterisk-users] Issue playing high quality white noise   Scott Griepentrog
• [asterisk-users] [asterisk-dev] AstriDevCon 2014: Agenda itemDeprecate AMI/AGI(Ben Klang)   Scott Griepentrog
• [asterisk-users] PJSIP and NAT behind a dynamic IP address   Scott Griepentrog
• [asterisk-users] Call forwarding from Phones and getting the referrer IP   Scott Griepentrog
• [asterisk-users] dialplan reload context   Scott Griepentrog
• [asterisk-users] Register multiple phones to a single AOR with PJSIP   Scott Griepentrog
• [asterisk-users] Ubuntu 12.04 LTS / Asterisk / apt-get upgrade / exclude packages   Thorsten Göllner
• [asterisk-users] ConfBridge / internal_sample_rate=auto / warning   Thorsten Göllner
• [asterisk-users] Questions on musiconhold.conf custom mode   Thorsten Göllner
• [asterisk-users] Questions on musiconhold.conf custom mode   Thorsten Göllner
• [asterisk-users] make asterisk do something when an outgoing call is picked up   Thorsten Göllner
• [asterisk-users] AstLinux 1.2.0 Released   Darrick Hartman
• [asterisk-users] sip.conf to pjsip.conf conversion script   Kevin Harwell
• [asterisk-users] Asterisk removes ice lines in sdp when calling between webrtc clients   Olli Heiskanen
• [asterisk-users] Asterisk removes ice lines in sdp when calling between webrtc clients   Olli Heiskanen
• [asterisk-users] Asterisk removes ice lines in sdp when calling between webrtc clients   Olli Heiskanen
• [asterisk-users] Asterisk removes ice lines in sdp when calling between webrtc clients   Olli Heiskanen
• [asterisk-users] TLS on SIP trunk   Meadows Hoa
• [asterisk-users] MWI publish VIA pjsip for non sip channels   Matt Hoskins
• [asterisk-users] MWI publish VIA pjsip for non sip channels   Matt Hoskins
• [asterisk-users] MWI publish VIA pjsip for non sip channels   Matt Hoskins
• [asterisk-users] Asterisk Phone ( Telecom feature )   Steven Howes
• [asterisk-users] Asterisk 13 stable?   Steven Howes
• [asterisk-users] allo.com gsm card with AsteriskNOW   Nicolas Pham Van Huyen
• [asterisk-users] allo.com gsm card with AsteriskNOW   Nicolas Pham Van Huyen
• [asterisk-users] bristuff-0.4.0-RC4-xr7   Ray Image
• [asterisk-users] bristuff-0.4.0-RC4-xr7   Ray Image
• [asterisk-users] Asterisk removes ice lines in sdp when calling between webrtc clients   Matthew Jordan
• [asterisk-users] CALLERID(num) and CDR(clid) - originate   Matthew Jordan
• [asterisk-users] Setting channel musicclass from AGI   Matthew Jordan
• [asterisk-users] Pjsip and regcontext (for DUNDi)   Matthew Jordan
• [asterisk-users] Asterisk LTS segment faults   Matthew Jordan
• [asterisk-users] JABBER_STATUS CODE 7   Matthew Jordan
• [asterisk-users] Do subroutines need their own h extension?   Matthew Jordan
• [asterisk-users] Asterisk 12 CDR dst field empty   Matthew Jordan
• [asterisk-users] AMI and CDR(answer)   Matthew Jordan
• [asterisk-users] AMI and CDR(answer)   Matthew Jordan
• [asterisk-users] AMI and CDR(answer)   Matthew Jordan
• [asterisk-users] Asterisk 12 CDR dst field empty - cdrlog.txt (0/1)   Matthew Jordan
• [asterisk-users] AMI and CDR(answer)   Matthew Jordan
• [asterisk-users] [asterisk-dev] AstriDevCon 2014: Agenda item Deprecate AMI/AGI(Ben Klang)   Matthew Jordan
• [asterisk-users] [asterisk-dev] AstriDevCon 2014: Agenda itemDeprecate AMI/AGI(Ben Klang)   Matthew Jordan
• [asterisk-users] Ast 13 beta 3 - Segfault when calling on pjsip trunk with directmedia=yes   Matthew Jordan
• [asterisk-users] AstriDevCon 2014: Agenda item Deprecate AMI/AGI(Ben Klang)   Matthew Jordan
• [asterisk-users] Voicemail ODBC Storage   Matthew Jordan
• [asterisk-users] DTMF behavior in asterisk 12 with PJSIP   Matthew Jordan
• [asterisk-users] DTMF behavior in asterisk 12 with PJSIP   Matthew Jordan
• [asterisk-users] Setting Music on Hold with the Manager Interface   Matthew Jordan
• [asterisk-users] Asterisk 12 Dialplan   Matthew Jordan
• [asterisk-users] Asterisk 12 Dialplan   Matthew Jordan
• [asterisk-users] Asterisk 12 - zombie processes   Matthew Jordan
• [asterisk-users] Asterisk 13 stable?   Matthew Jordan
• [asterisk-users] sip.conf to pjsip.conf conversion script   Matthew Jordan
• [asterisk-users] Asterisk 12 - zombie processes   Matthew Jordan
• [asterisk-users] dialplan reload context   Matthew Jordan
• [asterisk-users] Asterisk 12 - zombie processes   Matthew Jordan
• [asterisk-users] Asterisk 12 Dialplan   Matthew Jordan
• [asterisk-users] Asterisk 13 : SILK codec ?   Matthew Jordan
• [asterisk-users] Register multiple phones to a single AOR with PJSIP   Matthew Jordan
• [asterisk-users] Paul Albrecht   Matthew Jordan
• [asterisk-users] Voicemail ODBC Storage   Dan Journo
• [asterisk-users] Grandstream GXP2160 + SRTP   Jonas Kellens
• [asterisk-users] Grandstream GXP2160 + SRTP   Jonas Kellens
• [asterisk-users] sdp_crypto_process: Crypto life time unsupported: crypto   Jonas Kellens
• [asterisk-users] sdp_crypto_process: Crypto life time unsupported: crypto   Jonas Kellens
• [asterisk-users] sdp_crypto_process: Crypto life time unsupported: crypto   Jonas Kellens
• [asterisk-users] dialplan reload context   Jonas Kellens
• [asterisk-users] ${HASH(SIP_CAUSE,<channel-name>)}   Jonas Kellens
• [asterisk-users] Asterisk Phone ( Telecom feature )   Don Kelly
• [asterisk-users] Asterisk Phone ( Telecom feature )   Don Kelly
• [asterisk-users] how to strip +1 out of incoming number   John Kiniston
• [asterisk-users] sip.conf to pjsip.conf conversion script   John Kiniston
• [asterisk-users] make asterisk do something when an outgoing call is picked up   John Kiniston
• [asterisk-users] PlayTones while in call   John Kiniston
• [asterisk-users] new app_swift is live   Jeremy Kister
• [asterisk-users] asterisk stun setup , not using public ip returned by stun server   Jeremy Kister
• [asterisk-users] asterisk stun setup , not using public ip returned by stun server   Jeremy Kister
• [asterisk-users] [asterisk-dev] AstriDevCon 2014: Agenda item Deprecate AMI/AGI (Ben Klang)   Ben Klang
• [asterisk-users] Astricom 2014 presentations   Jeff LaCoursiere
• [asterisk-users] Asterisk Crashes Randomly with Cepstral Swift TTS   Patrick Laimbock
• [asterisk-users] Asterisk 11.9.0 crash and restart   Patrick Laimbock
• [asterisk-users] bristuff-0.4.0-RC4-xr7   Patrick Laimbock
• [asterisk-users] Setting channel musicclass from AGI   James Lamanna
• [asterisk-users] Setting channel musicclass from AGI   James Lamanna
• [asterisk-users] Asterisk LTS segment faults   Kevin Larsen
• [asterisk-users] AstriDevCon 2014: AgendaitemDeprecate AMI/AGI(Ben Klang)   Kevin Larsen
• [asterisk-users] AstriDevCon 2014:Agenda item Deprecate AMI/AGI(Ben Klang)   Kevin Larsen
• [asterisk-users] Voice Mail Questions   Phil Ledon
• [asterisk-users] Asterisk Phone ( Telecom feature )   Mitul Limbani
• [asterisk-users] Asterisk Phone ( Telecom feature )   Mitul Limbani
• [asterisk-users] SIP over 3G Mobile Network using NAT   Mitul Limbani
• [asterisk-users] SIP over 3G Mobile Network using NAT   Mitul Limbani
• [asterisk-users] CALLERID(num) and CDR(clid) - originate   Gabriel Ortiz Lour
• [asterisk-users] CALLERID(num) and CDR(clid) - originate   Gabriel Ortiz Lour
• [asterisk-users] No chan_sip in compiled asterisk-11.13.0   Doug Lytle
• [asterisk-users] No chan_sip in compiled asterisk-11.13.0   Doug Lytle
• [asterisk-users] Call forwarding from Phones and getting the referrer IP   Ishfaq Malik
• [asterisk-users] Call forwarding from Phones and getting the referrer IP   Ishfaq Malik
• [asterisk-users] PBX hacked: why hundred of calls to the same number ?   Markus
• [asterisk-users] Voicemail ODBC Storage   Anthony Messina
• [asterisk-users] SIP over 3G Mobile Network using NAT   Larry Moore
• [asterisk-users] debugging T.38 issues   Larry Moore
• [asterisk-users] debugging T.38 issues   Larry Moore
• [asterisk-users] SPA504G auto answer   Larry Moore
• [asterisk-users] SPA504G auto answer   Larry Moore
• [asterisk-users] SPA504G auto answer   Larry Moore
• [asterisk-users] res_fax T.38 Gateway with SpanDSP - Force ReINVITE?   Larry Moore
• [asterisk-users] res_fax T.38 Gateway with SpanDSP - Force ReINVITE?   Larry Moore
• [asterisk-users] res_fax T.38 Gateway with SpanDSP - Force ReINVITE?   Larry Moore
• [asterisk-users] res_fax T.38 Gateway with SpanDSP - Force ReINVITE?   Larry Moore
• [asterisk-users] res_fax T.38 Gateway with SpanDSP - Force ReINVITE?   Larry Moore
• [asterisk-users] res_fax T.38 Gateway with SpanDSP - Force ReINVITE?   Larry Moore
• [asterisk-users] res_fax T.38 Gateway with SpanDSP - Force ReINVITE?   Larry Moore
• [asterisk-users] logger.conf   Richard Mudgett
• [asterisk-users] Asterisk 12 Dialplan   Richard Mudgett
• [asterisk-users] Query on connecting 3G MSC with Asterisk PBX Server via SIP Interface.   NACHIAPPAN, SUBBAIAH (SUBBAIAH)
• [asterisk-users] Query on connecting 3G MSC with Asterisk PBX Server via SIP Interface.   NACHIAPPAN, SUBBAIAH (SUBBAIAH)
• [asterisk-users] Query on connecting 3G MSC with Asterisk PBX Server via SIP Interface.   NACHIAPPAN, SUBBAIAH (SUBBAIAH)
• [asterisk-users] DTMF behavior in asterisk 12 with PJSIP   Yaron Nachum
• [asterisk-users] Port number in From URI on Asterisk 12 PJSIP   Yaron Nachum
• [asterisk-users] DTMF behavior in asterisk 12 with PJSIP   Yaron Nachum
• [asterisk-users] DTMF behavior in asterisk 12 with PJSIP   Yaron Nachum
• [asterisk-users] Asterisk 12 - zombie processes   Yaron Nachum
• [asterisk-users] Asterisk 12 - zombie processes   Yaron Nachum
• [asterisk-users] Asterisk 12 - zombie processes   Yaron Nachum
• [asterisk-users] Asterisk 12 - zombie processes   Yaron Nachum
• [asterisk-users] Asterisk 12 - zombie processes   Yaron Nachum
• [asterisk-users] Asterisk 12 - zombie processes   Yaron Nachum
• [asterisk-users] res_fax T.38 Gateway with SpanDSP - Force ReINVITE?   Tim Nelson
• [asterisk-users] res_fax T.38 Gateway with SpanDSP - Force ReINVITE?   Tim Nelson
• [asterisk-users] res_fax T.38 Gateway with SpanDSP - Force ReINVITE?   Tim Nelson
• [asterisk-users] res_fax T.38 Gateway with SpanDSP - Force ReINVITE?   Tim Nelson
• [asterisk-users] res_fax T.38 Gateway with SpanDSP - Force ReINVITE?   Tim Nelson
• [asterisk-users] res_fax T.38 Gateway with SpanDSP - Force ReINVITE?   Tim Nelson
• [asterisk-users] Reset calls processed counter   Rusty Newton
• [asterisk-users] howto cancel simultaneous calls - dial(sip/phone1&sip/phone2)   Rusty Newton
• [asterisk-users] PBX hacked: why hundred of calls to the same number ?   Olivier
• [asterisk-users] SPA112: one analog phone works, not the other   Olivier
• [asterisk-users] SPA112: one analog phone works, not the other   Olivier
• [asterisk-users] SIP dialing with authentication with dialstring and wothout sip; conf   Olivier
• [asterisk-users] Questions on musiconhold.conf custom mode   Olivier
• [asterisk-users] Questions on musiconhold.conf custom mode   Olivier
• [asterisk-users] Questions on musiconhold.conf custom mode   Olivier
• [asterisk-users] PJSIP and NAT behind a dynamic IP address   Jeffrey Ollie
• [asterisk-users] [asterisk-dev] AstriDevCon 2014:Agenda item Deprecate AMI/AGI(Ben Klang)   Jeffrey Ollie
• [asterisk-users] 11.13.1: unable to load sip.conf (or iax )   Jeffrey Ollie
• [asterisk-users] 11.13.1: unable to load sip.conf (or iax )   Jeffrey Ollie
• [asterisk-users] [asterisk-dev] AstriDevCon 2014:Agenda item Deprecate AMI/AGI (Ben Klang)   Jeffrey Ollie
• [asterisk-users] PBX hacked: why hundred of calls to the same number ?   Rainer Piper
• [asterisk-users] PBX hacked: why hundred of calls to the same number ?   Rainer Piper
• [asterisk-users] PBX hacked: why hundred of calls to the same number ?   Rainer Piper
• [asterisk-users] PBX hacked: why hundred of calls to the same number ?   Rainer Piper
• [asterisk-users] PBX hacked: why hundred of calls to the same number ?   Rainer Piper
• [asterisk-users] PBX hacked: why hundred of calls to the same number ?   Rainer Piper
• [asterisk-users] PBX hacked: why hundred of calls to the same number ?   Rainer Piper
• [asterisk-users] PBX hacked: why hundred of calls to the same number ?   Dave Platt
• [asterisk-users] [asterisk-user] Confbridge Kick Action   Shishir Pokharel
• [asterisk-users] [asterisk-user] Confbridge Kick Action   Shishir Pokharel
• [asterisk-users] Lost audio on forwarded calls   Todd R.
• [asterisk-users] Setting Music on Hold with the Manager Interface   Todd R.
• [asterisk-users] Setting Music on Hold with the Manager Interface   Todd R.
• [asterisk-users] Asterisk 13 stable?   Rafael dos Santos Saraiva
• [asterisk-users] Multicast AMI?   Sebastian
• [asterisk-users] asterisk-users Digest, Vol 123, Issue 38   Nitesh Sharma
• [asterisk-users] [asterisk-user] Confbridge Kick Action   Chandrakant Solanki
• [asterisk-users] [asterisk-user] Confbridge Kick Action   Chandrakant Solanki
• [asterisk-users] [asterisk-user] Confbridge Kick Action   Chandrakant Solanki
• [asterisk-users] [asterisk-user] Confbridge Kick Action   Chandrakant Solanki
• [asterisk-users] [asterisk-user] Confbridge Kick Action   Chandrakant Solanki
• [asterisk-users] Ubuntu 12.04 LTS / Asterisk / apt-get upgrade / exclude packages   A J Stiles
• [asterisk-users] Asterisk GOIP Outgoing Callerid not working   A J Stiles
• [asterisk-users] Query on connecting 3G MSC with Asterisk PBX Server via SIP Interface.   A J Stiles
• [asterisk-users] PBX hacked: why hundred of calls to the same number ?   Administrator TOOTAI
• [asterisk-users] Asterisk 1.8.28-cert2, 1.8.31.1, 11.6-cert7, 11.13.1, 12.6.1, 13.0.0-beta3 Now Available (Security Release)   Asterisk Development Team
• [asterisk-users] Asterisk 13.0.0 Now Available!   Asterisk Development Team
• [asterisk-users] AST-2014-011: Asterisk Susceptibility to POODLE Vulnerability   Asterisk Security Team
• [asterisk-users] logger.conf   Jared Terrell
• [asterisk-users] JABBER_STATUS CODE 7   Marcelo Terres
• [asterisk-users] JABBER_STATUS CODE 7   Marcelo Terres
• [asterisk-users] Lost audio on forwarded calls   Steve Totaro
• [asterisk-users] Auto video call hangup   Markos Vakondios
• [asterisk-users] Asterisk LTS segment faults   Ryan Wagoner
• [asterisk-users] My Asterisk can not send fax via T.38   Weiqi
• [asterisk-users] Asterisk Phone ( Telecom feature )   Ron Wheeler
• [asterisk-users] Asterisk removes ice lines in sdp when calling between webrtc clients   Eric Wieling
• [asterisk-users] how to strip +1 out of incoming number   Eric Wieling
• [asterisk-users] PBX hacked: why hundred of calls to the same number ?   Eric Wieling
• [asterisk-users] Lost audio on forwarded calls   Eric Wieling
• [asterisk-users] Ubuntu 12.04 LTS / Asterisk / apt-get upgrade / exclude packages   Linus Wiklund
• [asterisk-users] Asterisk Phone ( Telecom feature )   Hans Witvliet
• [asterisk-users] Astricom 2014 presentations   Bryant Zimmerman
• [asterisk-users] authentication time for asterisk server   rafa alfurqan
• [asterisk-users] how to strip +1 out of incoming number   motty cruz
• [asterisk-users] 11.13.1: unable to load sip.conf (or iax )   sean darcy
• [asterisk-users] 11.13.1: unable to load sip.conf (or iax )   sean darcy
• [asterisk-users] 11.13.1: unable to load sip.conf (or iax )   sean darcy
• [asterisk-users] 11.13.1: unable to load sip.conf (or iax )   sean darcy
• [asterisk-users] 11.13.1: unable to load sip.conf (or iax )   sean darcy
• [asterisk-users] Asterisk 13 : SILK codec ?   sean darcy
• [asterisk-users] Asterisk 13 : SILK codec ?   sean darcy
• [asterisk-users] Issue playing high quality white noise   asteriskusers at dovid.net
• [asterisk-users] JABBER_STATUS CODE 7   ricky gutierrez
• [asterisk-users] JABBER_STATUS CODE 7   ricky gutierrez
• [asterisk-users] JABBER_STATUS CODE 7   ricky gutierrez
• [asterisk-users] JABBER_STATUS CODE 7   ricky gutierrez
• [asterisk-users] JABBER_STATUS CODE 7   ricky gutierrez
• [asterisk-users] Asterisk Phone ( Telecom feature )   jg
• [asterisk-users] Asterisk Phone ( Telecom feature )   jg
• [asterisk-users] Ubuntu 12.04 LTS / Asterisk / apt-get upgrade / exclude packages   jg
• [asterisk-users] Ubuntu 12.04 LTS / Asterisk / apt-get upgrade / exclude packages   jg
• [asterisk-users] allo.com gsm card with AsteriskNOW   jg
• [asterisk-users] Asterisk GOIP Outgoing Callerid not working   jg
• [asterisk-users] Asterisk GOIP Outgoing Callerid not working   jg
• [asterisk-users] make asterisk do something when an outgoing call is picked up   lee
• [asterisk-users] make asterisk do something when an outgoing call is picked up   lee
• [asterisk-users] asterisk stun setup , not using public ip returned by stun server   chandapure shiva
• [asterisk-users] asterisk stun setup , not using public ip returned by stun server   chandapure shiva
• [asterisk-users] asterisk stun setup , not using public ip returned by stun server   chandapure shiva

Last message date: Fri Oct 31 12:59:24 CDT 2014
Archived on: Fri Oct 31 12:58:24 CDT 2014

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