November 2013 Archives by author

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Starting: Fri Nov 1 05:02:49 CDT 2013
Ending: Sat Nov 30 02:25:43 CST 2013
Messages: 276

• [asterisk-users] Redirect a GSM call through Wifi to a SIP phone   adamk at 3a.hu
• [asterisk-users] Asterisk 1.8.20 crashing   Amit Patkar | ATPL
• [asterisk-users] sip show channelstats shows all 0   Ezequiel R. Achenbach
• [asterisk-users] Communicate with barge agent   Shahbaz Afzal
• [asterisk-users] Movistar sip Mexico   Alyed
• [asterisk-users] Movistar sip Mexico   Alyed
• [asterisk-users] Movistar sip Mexico   Alyed
• [asterisk-users] Asterisk 11.6.0 not starting up   Daniel - Asterisk
• [asterisk-users] Asterisk 11.6.0 not starting up   Daniel - Asterisk
• [asterisk-users] Asterisk 11.6.0 not starting up   Daniel - Asterisk
• [asterisk-users] combine external video source and audio call to make SIP video call?   Brandon B.
• [asterisk-users] SIP Mass exodus   Chris Bagnall
• [asterisk-users] Asterisk 11.6.0 not starting up   Bakko
• [asterisk-users] Asterisk 11.6.0 not starting up   Bakko
• [asterisk-users] Communicate with barge agent   Satish Barot
• [asterisk-users] Unix connections not always disconnecting   Paul Belanger
• [asterisk-users] Capture dead phone?   Paul Belanger
• [asterisk-users] calendar.conf include   Paul Belanger
• [asterisk-users] CEL for attented transfer   Paul Belanger
• [asterisk-users] Asterisk uses 105% CPU   Paul Belanger
• [asterisk-users] issue with speech in IVR   Paul Belanger
• [asterisk-users] Asterisk uses 105% CPU   Paul Belanger
• [asterisk-users] issue with speech in IVR   Paul Belanger
• [asterisk-users] Res corosync.   Slava Bendersky
• [asterisk-users] Res corosync.   Slava Bendersky
• [asterisk-users] Asterisk RTP Questions   James Bensley
• [asterisk-users] Asterisk Realtime Static Voicemail   John T. Bittner
• [asterisk-users] Asterisk Real-time Static Voicemail   John T. Bittner
• [asterisk-users] Unix connections not always disconnecting   Gareth Blades
• [asterisk-users] Ast11: How to see call progress like in Ast <= 1.8   Gareth Blades
• [asterisk-users] Asterisk RTP Questions   Gareth Blades
• [asterisk-users] Amazon, Asterisk and reliability beyond a hobby system?   Brian
• [asterisk-users] Question about Management Interface   CDR
• [asterisk-users] Caller's phone keeps ringing after 200 OK   Nick Cameo
• [asterisk-users] Realtime Call Files   Carlos Chavez
• [asterisk-users] terminating the call, when transferer hangs up the call during attended transfer   Nikola Ciprich
• [asterisk-users] terminating the call, when transferer hangs up the call during attended transfer   Nikola Ciprich
• [asterisk-users] terminating the call, when transferer hangs up the call during attended transfer   Nikola Ciprich
• [asterisk-users] terminating the call, when transferer hangs up the call during attended transfer   Nikola Ciprich
• [asterisk-users] terminating the call, when transferer hangs up the call during attended transfer   Nikola Ciprich
• [asterisk-users] Capture dead phone?   Mitch Claborn
• [asterisk-users] Capture dead phone?   Mitch Claborn
• [asterisk-users] Asterisk uses 105% CPU   Andrew Colin
• [asterisk-users] No matching peers message has gone (1.8.23.1)   Joshua Colp
• [asterisk-users] Question about Management Interface   Joshua Colp
• [asterisk-users] how to answer a Panasonic PBX extension with Asterisk?   Eric Cooper
• [asterisk-users] combine external video source and audio call to make SIP video call?   Eric Cooper
• [asterisk-users] combine external video source and audio call to make SIP video call?   Eric Cooper
• [asterisk-users] Adding SIP method MESSAGE to Allow header   Daniel
• [asterisk-users] Asterisk Realtime Static Voicemail   Leandro Dardini
• [asterisk-users] SIP Presence across two servers   Leandro Dardini
• [asterisk-users] SIP Presence across two servers   Leandro Dardini
• [asterisk-users] Queue linear "unordered" feature when using realtime   Leandro Dardini
• [asterisk-users] Dialing directly with username and password   Leandro Dardini
• [asterisk-users] Asterisk 11.6.0 not starting up   Leandro Dardini
• [asterisk-users] Answering agent   Leandro Dardini
• [asterisk-users] Help - DTMF relay in meetme is not reliable   Rajib Deka
• [asterisk-users] DTMF relay in meetme is not reliable   Rajib Deka
• [asterisk-users] MCID   Deka, Rajib IN MAA SL
• [asterisk-users] Asterisk is delaying DTMF INFO in meetme   Deka, Rajib IN MAA SL
• [asterisk-users] SIP Mass exodus   Mike Diehl
• [asterisk-users] Capture dead phone?   John Doe
• [asterisk-users] Bulk forwarding to another Asterisk   Doug
• [asterisk-users] Bulk forwarding to another Asterisk   Doug
• [asterisk-users] AMI version vs. AST version   Michelle Dupuis
• [asterisk-users] Bulk forwarding to another Asterisk   Steve Edwards
• [asterisk-users] Call files without permission for asterisk to read   Steve Edwards
• [asterisk-users] Call files without permission for asterisk to read   Steve Edwards
• [asterisk-users] issue with speech in IVR   Salaheddine Elharit
• [asterisk-users] issue with speech in IVR   Salaheddine Elharit
• [asterisk-users] issue with speech in IVR   Salaheddine Elharit
• [asterisk-users] issue with speech in IVR   Salaheddine Elharit
• [asterisk-users] issue with speech in IVR   Salaheddine Elharit
• [asterisk-users] issue with speech in IVR   Salaheddine Elharit
• [asterisk-users] Fwd: issue with speech in IVR   Salaheddine Elharit
• [asterisk-users] Add SIP Header for 1 SIP peer when calling a group of SIP peers   Barry Flanagan
• [asterisk-users] Sangoma transcoding card bug - drops audio samples   Grzegorz Garlewicz
• [asterisk-users] Sangoma transcoding card bug - drops audio samples   Grzegorz Garlewicz
• [asterisk-users] Make phone ring through webserver using Asterisk   Dominik George
• [asterisk-users] DAHDI with (CDR(userfield)   Michael Gilleran
• [asterisk-users] CEL for attented transfer   Jean-Denis Girard
• [asterisk-users] CEL for attented transfer   Jean-Denis Girard
• [asterisk-users] CEL for attented transfer   Jean-Denis Girard
• [asterisk-users] CEL for attented transfer   Jean-Denis Girard
• [asterisk-users] CEL for attented transfer   Jean-Denis Girard
• [asterisk-users] Movistar sip Mexico   Damian Gonzalez
• [asterisk-users] Movistar sip Mexico   Damian Gonzalez
• [asterisk-users] Movistar sip Mexico   Damian Gonzalez
• [asterisk-users] Movistar sip Mexico   Damian Gonzalez
• [asterisk-users] Call files without permission for asterisk to read   Rizwan Hisham
• [asterisk-users] Call files without permission for asterisk to read   Rizwan Hisham
• [asterisk-users] Unix connections not always disconnecting   Ioan Indreias
• [asterisk-users] Call files without permission for asterisk to read   Ioan Indreias
• [asterisk-users] combine external video source and audio call to make SIP video call?   Ioan Indreias
• [asterisk-users] DAHDI-Linux and DAHDI-Tools 2.8.0-rc2 Now Available   Ira
• [asterisk-users] CEL for attented transfer   Jairo
• [asterisk-users] CEL for attented transfer   Jairo
• [asterisk-users] [asterisk-dev] how determine mandatory modules to slimming asterisk   Matthew Jordan
• [asterisk-users] Asterisk 1.8.20 crashing   Matthew Jordan
• [asterisk-users] Asterisk 10 EOL Approaching   Matthew Jordan
• [asterisk-users] Voicemail greeting playback issues?   Matthew Jordan
• [asterisk-users] userfield not logged to CDR   Dan Journo
• [asterisk-users] userfield not logged to CDR   Dan Journo
• [asterisk-users] userfield not logged to CDR   Dan Journo
• [asterisk-users] VoIP sound quality : highroad sound   Jonas Kellens
• [asterisk-users] VoIP sound quality : highroad sound   Jonas Kellens
• [asterisk-users] VoIP sound quality : highroad sound   Jonas Kellens
• [asterisk-users] VoIP sound quality : highroad sound   Jonas Kellens
• [asterisk-users] VoIP sound quality : highroad sound   Jonas Kellens
• [asterisk-users] calendar.conf include   Jonas Kellens
• [asterisk-users] Add SIP Header for 1 SIP peer when calling a group of SIP peers   Jonas Kellens
• [asterisk-users] Asterisk 1.8.24 : illegal instruction   Jonas Kellens
• [asterisk-users] Asterisk 1.8.24 : illegal instruction   Jonas Kellens
• [asterisk-users] Asterisk 1.8.24 : illegal instruction   Jonas Kellens
• [asterisk-users] Asterisk 1.8.24 : illegal instruction   Jonas Kellens
• [asterisk-users] Asterisk 1.8.24 : illegal instruction   Jonas Kellens
• [asterisk-users] Asterisk 1.8.24 : illegal instruction   Jonas Kellens
• [asterisk-users] Asterisk 1.8.24 : illegal instruction   Jonas Kellens
• [asterisk-users] Asterisk 1.8.24 : illegal instruction   Jonas Kellens
• [asterisk-users] Asterisk uses 105% CPU   Jonas Kellens
• [asterisk-users] Asterisk uses 105% CPU   Jonas Kellens
• [asterisk-users] Asterisk uses 105% CPU   Jonas Kellens
• [asterisk-users] Direct Media and message "SIP/SipAgent-00000bf9 requested media update control 26, passing it to SIP/ead14-00000bfb"   Jonas Kellens
• [asterisk-users] RTP packets send, but no audio   Jonas Kellens
• [asterisk-users] RTP packets send, but no audio   Jonas Kellens
• [asterisk-users] terminating the call, when transferer hangs up the call during attended transfer   Don Kelly
• [asterisk-users] Movistar sip Mexico   Kristian Kielhofner
• [asterisk-users] recieve fax from PRI using spandsp 65% failure rate   Justin Killen
• [asterisk-users] receive fax from PRI using spandsp 65% failure rate   Justin Killen
• [asterisk-users] SIP Presence across two servers   Lincoln King-Cliby
• [asterisk-users] Add SIP Header for 1 SIP peer when calling a group of SIP peers   John Kiniston
• [asterisk-users] Recurring SIP problem with asterisk 11.6 & 11.7   Jeremy Kister
• [asterisk-users] Recurring SIP problem with asterisk 11.6 & 11.7   Jeremy Kister
• [asterisk-users] Recurring SIP problem with asterisk 11.6 & 11.7   Jeremy Kister
• [asterisk-users] T.38 termination   Jeff LaCoursiere
• [asterisk-users] Monitor extension status   Eduardo Leones
• [asterisk-users] TE420, is it possible do disable span (red blinking)?   Mitul Limbani
• [asterisk-users] Asterisk 1.8.22   Mitul Limbani
• [asterisk-users] Bulk forwarding to another Asterisk   Mitul Limbani
• [asterisk-users] issue with speech in IVR   Mitul Limbani
• [asterisk-users] issue with speech in IVR   Mitul Limbani
• [asterisk-users] Capture dead phone?   Mikhail Lischuk
• [asterisk-users] Voicemail greeting playback issues?   Patrick Lists
• [asterisk-users] CallerID settings   Gabriel Ortiz Lour
• [asterisk-users] userfield not logged to CDR   Doug Lytle
• [asterisk-users] Voicemail greeting playback issues?   Doug Lytle
• [asterisk-users] Voicemail greeting playback issues?   Doug Lytle
• [asterisk-users] SIP FXS ATA with Gigabit ethernet bridge port,   Isamar Maia
• [asterisk-users] No matching peers message has gone (1.8.23.1)   Ishfaq Malik
• [asterisk-users] No matching peers message has gone (1.8.23.1)   Ishfaq Malik
• [asterisk-users] No matching peers message has gone (1.8.23.1)   Ishfaq Malik
• [asterisk-users] Unix connections not always disconnecting   Ishfaq Malik
• [asterisk-users] Unix connections not always disconnecting   Ishfaq Malik
• [asterisk-users] SIP Mass exodus   Markus
• [asterisk-users] Redirect a GSM call through Wifi to a SIP phone   Silvère Maugain
• [asterisk-users] Redirect a GSM call through Wifi to a SIP phone   Silvère Maugain
• [asterisk-users] TE420, is it possible do disable span (red blinking)?   Dmitry Melekhov
• [asterisk-users] e1 , hdlc data link?   Dmitry Melekhov
• [asterisk-users] overlapdialing and no digits in setup problem   Dmitry Melekhov
• [asterisk-users] CONNECTEDLINE and panasonic 500   Dmitry Melekhov
• [asterisk-users] CONNECTEDLINE and panasonic 500   Dmitry Melekhov
• [asterisk-users] e1 , hdlc data link?   Dmitry Melekhov
• [asterisk-users] e1 , hdlc data link?   Dmitry Melekhov
• [asterisk-users] e1 , hdlc data link?   Dmitry Melekhov
• [asterisk-users] Outgoing phone calls "muffled"   Eddie Mikell
• [asterisk-users] Outgoing phone calls muffled   Eddie Mikell
• [asterisk-users] Asterisk 1.8.24 : illegal instruction   Asghar Mohammad
• [asterisk-users] How to enable T.38 between SPA3102 PSTN Line port and ReceiveFAX app ?   Larry Moore
• [asterisk-users] How to enable T.38 between SPA3102 PSTN Line port and ReceiveFAX app ?   Larry Moore
• [asterisk-users] Movistar sip Mexico   Larry Moore
• [asterisk-users] Movistar sip Mexico   Larry Moore
• [asterisk-users] Integration with NEC DSX - help with dial line   Stephen More
• [asterisk-users] Integration with NEC DSX - help with dial line   Stephen More
• [asterisk-users] 11.5.0 - SIP registration not retrying after failures   Tony Mountifield
• [asterisk-users] 11.5.0 - SIP registration not retrying after failures   Tony Mountifield
• [asterisk-users] CONNECTEDLINE and panasonic 500   Richard Mudgett
• [asterisk-users] SaySentence/SoundPack Proposal   Steve Murphy
• [asterisk-users] issue with speech in IVR   Steve Murphy
• [asterisk-users] Channel not releasing immediately for Attended Transfer   Gopalakrishnan N
• [asterisk-users] Sangoma transcoding card bug - drops audio samples   Gopalakrishnan N
• [asterisk-users] AGI Script not working   Gopalakrishnan N
• [asterisk-users] Answering agent   Gopalakrishnan N
• [asterisk-users] Integration with NEC DSX - help with dial line   John Novack
• [asterisk-users] How to enable T.38 between SPA3102 PSTN Line port and ReceiveFAX app ?   Olivier
• [asterisk-users] How to enable T.38 between SPA3102 PSTN Line port and ReceiveFAX app ?   Olivier
• [asterisk-users] Automated Call Testing - end-to-end - SIP Provider   Positively Optimistic
• [asterisk-users] fraud detection   Positively Optimistic
• [asterisk-users] Asterisk 1.4 and DAHDI 2.7   Rodrigo Borges Pereira
• [asterisk-users] Asterisk 1.4 and DAHDI 2.7   Rodrigo Borges Pereira
• [asterisk-users] How to check ISDN / MFCR2 Trunk Status   Nicolás Podrojsky
• [asterisk-users] AMI version vs. AST version   Shishir Pokharel
• [asterisk-users] Make phone ring through webserver using Asterisk   Todd R.
• [asterisk-users] Amazon, Asterisk and reliability beyond a hobby system?   Todd R.
• [asterisk-users] Amazon, Asterisk and reliability beyond a hobby system?   Todd R.
• [asterisk-users] Amazon, Asterisk and reliability beyond a hobby system?   Todd R.
• [asterisk-users] Amazon, Asterisk and reliability beyond a hobby system?   Todd R.
• [asterisk-users] Amazon, Asterisk and reliability beyond a hobby system?   Todd R.
• [asterisk-users] Answering agent   Todd R.
• [asterisk-users] Ast11: How to see call progress like in Ast <= 1.8   Bas Rijniersce
• [asterisk-users] Asterisk 1.4 and DAHDI 2.7   Shaun Ruffell
• [asterisk-users] e1 , hdlc data link?   Shaun Ruffell
• [asterisk-users] DAHDI Missing '/sys/bus/astribanks/drivers/xppdrv/sync'   Shaun Ruffell
• [asterisk-users] Redirecting a channel to Meetme fails with Hangup.   S, Kantharuban IN MAA SL
• [asterisk-users] Amazon, Asterisk and reliability beyond a hobby system?   James Sharp
• [asterisk-users] Amazon, Asterisk and reliability beyond a hobby system?   James Sharp
• [asterisk-users] Movistar sip Mexico   Andreas Sikkema
• [asterisk-users] Redirect a GSM call through Wifi to a SIP phone   Sil
• [asterisk-users] Automated Call Testing - end-to-end - SIP Provider   St_Dwarf
• [asterisk-users] No matching peers message has gone (1.8.23.1)   Arthur J. Stanfield
• [asterisk-users] Redirect a GSM call through Wifi to a SIP phone   A J Stiles
• [asterisk-users] two steps when calling from web!   A J Stiles
• [asterisk-users] two steps when calling from web!   A J Stiles
• [asterisk-users] Recurring SIP problem with asterisk 11.6 & 11.7   A J Stiles
• [asterisk-users] Make phone ring through webserver using Asterisk   A J Stiles
• [asterisk-users] Asterisk 1.8.24 : illegal instruction   A J Stiles
• [asterisk-users] issue with speech in IVR   A J Stiles
• [asterisk-users] issue with speech in IVR   A J Stiles
• [asterisk-users] issue with speech in IVR   A J Stiles
• [asterisk-users] DAHDI-Linux and DAHDI-Tools 2.8.0-rc2 Now Available   Asterisk Development Team
• [asterisk-users] Asterisk 12.0.0-beta2 Now Available!   Asterisk Development Team
• [asterisk-users] DAHDI Missing '/sys/bus/astribanks/drivers/xppdrv/sync'   Joseph Towery
• [asterisk-users] Recurring SIP problem with asterisk 11.6 & 11.7   Duncan Turnbull
• [asterisk-users] Welcome to the "asterisk-users" mailing list   Roel Wagenaar
• [asterisk-users] SIP Presence across two servers   Ryan Wagoner
• [asterisk-users] Asterisk 1.8.24 : illegal instruction   Ron Wheeler
• [asterisk-users] Asterisk 1.8.24 : illegal instruction   Ron Wheeler
• [asterisk-users] Amazon, Asterisk and reliability beyond a hobby system?   Ron Wheeler
• [asterisk-users] Queue linear "unordered" feature when using realtime   Steven Wheeler
• [asterisk-users] Capture dead phone?   Eric Wieling
• [asterisk-users] Call files without permission for asterisk to read   Eric Wieling
• [asterisk-users] Sangoma transcoding card bug - drops audio samples   Eric Wieling
• [asterisk-users] VoIP sound quality : highroad sound   Johan Wilfer
• [asterisk-users] VoIP sound quality : highroad sound   Johan Wilfer
• [asterisk-users] dahdi fax catch-22 [SOLVED]   Greg Woods
• [asterisk-users] 11.5.0 - SIP registration not retrying after failures   Michael L. Young
• [asterisk-users] Movistar sip Mexico   Bryant Zimmerman
• [asterisk-users] 11.6 voicemail message cropped off?   Bryant Zimmerman
• [asterisk-users] 11.6 voicemail message cropped off?   Bryant Zimmerman
• [asterisk-users] Voicemail greeting playback issues?   Bryant Zimmerman
• [asterisk-users] Voicemail greeting playback issues?   Bryant Zimmerman
• [asterisk-users] Voicemail greeting playback issues?   Bryant Zimmerman
• [asterisk-users] Voicemail greeting playback issues?   Bryant Zimmerman
• [asterisk-users] Register Sip extension with out Sip phone   $$ dave cantera (android asus)
• [asterisk-users] Amazon, Asterisk and reliability beyond a hobby system?   covici at ccs.covici.com
• [asterisk-users] Register Sip extension with out Sip phone   akhilesh chand
• [asterisk-users] two steps when calling from web!   akhilesh chand
• [asterisk-users] two steps when calling from web!   akhilesh chand
• [asterisk-users] two steps when calling from web!   akhilesh chand
• [asterisk-users] Make phone ring through webserver using Asterisk   akhilesh chand
• [asterisk-users] Communicate with barge agent   akhilesh chand
• [asterisk-users] Communicate with barge agent   akhilesh chand
• [asterisk-users] Asterisk 1.8.22   motty cruz
• [asterisk-users] Asterisk 1.8.22   motty cruz
• [asterisk-users] Movistar sip Mexico   dotnetdub
• [asterisk-users] issue with speech in IVR   emilianovazquez at gmail.com
• [asterisk-users] issue with speech in IVR   isrlgb at gmail.com
• [asterisk-users] CallerID settings   jg
• [asterisk-users] VoIP sound quality : highroad sound   jg
• [asterisk-users] Recurring SIP problem with asterisk 11.6 & 11.7   jg
• [asterisk-users] VoIP sound quality : highroad sound   jg
• [asterisk-users] AMI version vs. AST version   jg
• [asterisk-users] Call files without permission for asterisk to read   jg
• [asterisk-users] Call files without permission for asterisk to read   jg
• [asterisk-users] terminating the call, when transferer hangs up the call during attended transfer   jg
• [asterisk-users] terminating the call, when transferer hangs up the call during attended transfer   jg
• [asterisk-users] terminating the call, when transferer hangs up the call during attended transfer   jg
• [asterisk-users] terminating the call, when transferer hangs up the call during attended transfer   jg
• [asterisk-users] terminating the call, when transferer hangs up the call during attended transfer   jg
• [asterisk-users] Outgoing phone calls "muffled"   jg
• [asterisk-users] Outgoing phone calls muffled   jg
• [asterisk-users] set different codec for different sip calls   s m
• [asterisk-users] how determine mandatory modules to slimming asterisk   s m
• [asterisk-users] Redirect a GSM call through Wifi to a SIPphone   mahendra_mahendra
• [asterisk-users] DAHDI with (CDR(userfield)   troxlinux
• [asterisk-users] DAHDI with (CDR(userfield)   troxlinux
• [asterisk-users] app_swift on centos 6 X64   troxlinux

Last message date: Sat Nov 30 02:25:43 CST 2013
Archived on: Sat Nov 30 02:21:59 CST 2013

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