February 2009 Archives by date

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Starting: Sun Feb 1 04:26:19 CST 2009
Ending: Sat Feb 28 22:21:44 CST 2009
Messages: 1797

• [asterisk-users] Need some information on SS7 parameters   research at businesstz.com
• [asterisk-users] Strange Packet Behavior   Elliot Murdock
• [asterisk-users] iax clients were unregistered after 30sec   Pezhman Lali
• [asterisk-users] Quiet 24 port POE gig switch   Chris Bagnall
• [asterisk-users] Quiet 24 port POE gig switch   OCG Technical Support
• [asterisk-users] Quiet 24 port POE gig switch   Ian Cowley
• [asterisk-users] Quiet 24 port POE gig switch   Bernd Felsche
• [asterisk-users] Quiet 24 port POE gig switch   Olivier
• [asterisk-users] asterisk-users Digest, Vol 54, Issue 109   bilal ghayyad
• [asterisk-users] Sending Calls via SIP trunk from two different IP addresses from same Asterisk Machine   bilal ghayyad
• [asterisk-users] Sending Calls via SIP trunk from two different IP addresses from same Asterisk Machine   Jeff LaCoursiere
• [asterisk-users] Sending Calls via SIP trunk from two different IP addresses from same Asterisk Machine   Mike
• [asterisk-users] Quiet 24 port POE gig switch   Sam Tam
• [asterisk-users] Quiet 24 port POE gig switch   Wilton Helm
• [asterisk-users] Sending Calls via SIP trunk from two different IP addresses from same Asterisk Machine   Tilghman Lesher
• [asterisk-users] Sending Calls via SIP trunk from two different IP addresses from same Asterisk Machine   bilal ghayyad
• [asterisk-users] Sending Calls via SIP trunk from two different IP addresses from same Asterisk Machine   Mike
• [asterisk-users] Sending Calls via SIP trunk from two different IP addresses from same Asterisk Machine   Jeff LaCoursiere
• [asterisk-users] Sending Calls via SIP trunk from two different IP addresses from same Asterisk Machine   Mike
• [asterisk-users] Sending Calls via SIP trunk from two different IP addresses from same Asterisk Machine   Geraint Lee
• [asterisk-users] iChat voice (and maybe video?)   Andreas Anderson
• [asterisk-users] Sending Calls via SIP trunk from two different IP addresses from same Asterisk Machine   Jeff LaCoursiere
• [asterisk-users] Sending Calls via SIP trunk from two different IP addresses from same Asterisk Machine   Jeff LaCoursiere
• [asterisk-users] Sending Calls via SIP trunk from two different IP addresses from same Asterisk Machine   Tilghman Lesher
• [asterisk-users] Sending Calls via SIP trunk from two different IP addresses from same Asterisk Machine   Tilghman Lesher
• [asterisk-users] Quiet 24 port POE gig switch   Paul Hales
• [asterisk-users] Quiet 24 port POE gig switch   Steve Underwood
• [asterisk-users] Quiet 24 port POE gig switch   OCG Technical Support
• [asterisk-users] Sending Calls via SIP trunk from two different IP addresses from same Asterisk Machine   Mike
• [asterisk-users] Sending Calls via SIP trunk from two different IP addresses from same Asterisk Machine   Tilghman Lesher
• [asterisk-users] Music On Hold   Ex Vito
• [asterisk-users] early dial: asterisk and ATA   Ex Vito
• [asterisk-users] Managing codecs   Ex Vito
• [asterisk-users] Trunk with Polocom Video Conferencing Unit   Daniel Harper
• [asterisk-users] Quiet 24 port POE gig switch   D Tucny
• [asterisk-users] Trunk with Polocom Video Conferencing Unit   D Tucny
• [asterisk-users] SIP.Conf - bindaddr per peer?   Johansson Olle E
• [asterisk-users] EVRC support   Max Alex
• [asterisk-users] Music On Hold   Idris AVCI
• [asterisk-users] Quiet 24 port POE gig switch   Gordon Henderson
• [asterisk-users] Sending Calls via SIP trunk from two different IP addresses from same Asterisk Machine   Benny Amorsen
• [asterisk-users] Quiet 24 port POE gig switch   Benny Amorsen
• [asterisk-users] Preferred Clock   Chris Knipe
• [asterisk-users] Sending Calls via SIP trunk from two different IP addresses from same Asterisk Machine   Steve Howes
• [asterisk-users] Quiet 24 port POE gig switch   Steve Underwood
• [asterisk-users] Duplicate Radius accounting in Asterisk   Ricardo Martinez
• [asterisk-users] dundi negative caching   Klaus Darilion
• [asterisk-users] Quiet 24 port POE gig switch   Wilton Helm
• [asterisk-users] Asterisk - Trixbox   Mike Hammett
• [asterisk-users] Configuring Patton SmartNode with ISDN2e and Asterisk   Phil Knighton
• [asterisk-users] Asterisk - Trixbox   Steve Totaro
• [asterisk-users] Quiet 24 port POE gig switch   Singer XJ Wang
• [asterisk-users] Quiet 24 port POE gig switch   Singer XJ Wang
• [asterisk-users] Asterisk - Trixbox   Mike Hammett
• [asterisk-users] Quiet 24 port POE gig switch   Steve Underwood
• [asterisk-users] Configuring Patton SmartNode with ISDN2e and Asterisk   Gordon Henderson
• [asterisk-users] Quiet 24 port POE gig switch   Wilton Helm
• [asterisk-users] Quiet 24 port POE gig switch   Singer XJ Wang
• [asterisk-users] Need some information on SS7 parameters   research at businesstz.com
• [asterisk-users] "No Reply to Our Critical Packet" SIP Calls Dropped in Voicemail   Lincoln King-Cliby
• [asterisk-users] "No Reply to Our Critical Packet" SIP Calls Dropped in Voicemail   Steve Totaro
• [asterisk-users] "No Reply to Our Critical Packet" SIP Calls Dropped in Voicemail   David Gibbons
• [asterisk-users] "No Reply to Our Critical Packet" SIP Calls Dropped in Voicemail   Alex Balashov
• [asterisk-users] Quiet 24 port POE gig switch   Bill Michaelson
• [asterisk-users] "No Reply to Our Critical Packet" SIP Calls Dropped in Voicemail   Lincoln King-Cliby
• [asterisk-users] Sending Calls via SIP trunk from two different IP addresses from same Asterisk Machine   bilal ghayyad
• [asterisk-users] SIP presence sample script   Vieri
• [asterisk-users] Quiet 24 port POE gig switch   Christian Victor
• [asterisk-users] Sending Calls via SIP trunk from two different IP addresses from same Asterisk Machine   David Gibbons
• [asterisk-users] ChanSpy or other variant   Nicholas Blasgen
• [asterisk-users] Quiet 24 port POE gig switch   Singer XJ Wang
• [asterisk-users] Sending Calls via SIP trunk from two different IP addresses from same Asterisk Machine   bilal ghayyad
• [asterisk-users] Sending Calls via SIP trunk from two different IP addresses from same Asterisk Machine   David Gibbons
• [asterisk-users] ChanSpy or other variant   Mark Michelson
• [asterisk-users] Invalid Extension   David at ULC
• [asterisk-users] Sending Calls via SIP trunk from two different IP addresses from same Asterisk Machine   Steve Totaro
• [asterisk-users] Invalid Extension   Philipp Kempgen
• [asterisk-users] ChanSpy or other variant   Nicholas Blasgen
• [asterisk-users] Asterisk - Trixbox   Paul Hales
• [asterisk-users] Sending Calls via SIP trunk from two different IP addresses from same Asterisk Machine   Tilghman Lesher
• [asterisk-users] Sending Calls via SIP trunk from two different IP addresses from same Asterisk Machine   Tilghman Lesher
• [asterisk-users] Videoconference one-to-many   Alejandro Cabrera
• [asterisk-users] Quiet 24 port POE gig switch   Bernd Felsche
• [asterisk-users] Quiet 24 port POE gig switch   Bernd Felsche
• [asterisk-users] Music On Hold   Ex Vito
• [asterisk-users] Quiet 24 port POE gig switch   Bernd Felsche
• [asterisk-users] GTalk Channel   GNUbie
• [asterisk-users] RBS T1 DID issue   Jeff LaCoursiere
• [asterisk-users] Quiet 24 port POE gig switch   Michael
• [asterisk-users] RBS T1 DID issue   Barton Fisher
• [asterisk-users] Which policy for ISDN BRI support in NT/PtMP ?   Ex Vito
• [asterisk-users] Patch to dahdi Chans.pm   Steve Johnson
• [asterisk-users] dialstatus through a call file   Ex Vito
• [asterisk-users] Quiet 24 port POE gig switch   Paul Hales
• [asterisk-users] RBS T1 DID issue   Jeff LaCoursiere
• [asterisk-users] Which policy for ISDN BRI support in NT/PtMP ?   Olivier
• [asterisk-users] Sending Calls via SIP trunk from two different IP addresses from same Asterisk Machine   bilal ghayyad
• [asterisk-users] "No Reply to Our Critical Packet" SIP Calls Dropped in Voicemail   Steve J. Douglas
• [asterisk-users] Videoconference one-to-many   C. Savinovich
• [asterisk-users] dialstatus through a call file   Johansson Olle E
• [asterisk-users] Videoconference one-to-many   Hans Witvliet
• [asterisk-users] What's the difference between the Jabber Client Mode And Component Mode?   tony luo
• [asterisk-users] analysing tools   Pezhman Lali
• [asterisk-users] Sending Calls via SIP trunk from two different IP addresses from same Asterisk Machine   Benny Amorsen
• [asterisk-users] may convert SIP call in H.323 to words terminator??   MianAsif
• [asterisk-users] Can't compile on NSLU2 LE   Gunner
• [asterisk-users] dialstatus through a call file   Pascal Bruno
• [asterisk-users] Can't compile on NSLU2 LE   Tzafrir Cohen
• [asterisk-users] Can't compile on NSLU2 LE   Gunner
• [asterisk-users] Can't compile on NSLU2 LE   Drew Gibson
• [asterisk-users] Can't compile on NSLU2 LE   Danny Nicholas
• [asterisk-users] Problem with building dahdi-linux RPM   bee-beeep
• [asterisk-users] Contact lookup   Geoff Lane
• [asterisk-users] n-way conferencing   Jeff LaCoursiere
• [asterisk-users] Contact lookup   Danny Nicholas
• [asterisk-users] n-way conferencing   Kevin P. Fleming
• [asterisk-users] n-way conferencing   Dean Collins
• [asterisk-users] fake ring again when using SIP   wassim Darwish
• [asterisk-users] Contact lookup   Gordon Henderson
• [asterisk-users] early dial: asterisk and ATA   Vieri
• [asterisk-users] n-way conferencing   Gordon Henderson
• [asterisk-users] Contact lookup   OCG Technical Support
• [asterisk-users] dahdi-linux 2.1.0.4 released   Asterisk Team
• [asterisk-users] Warning in CLI   Mike
• [asterisk-users] n-way conferencing   Jeff LaCoursiere
• [asterisk-users] Quiet 24 port POE gig switch   Wilton Helm
• [asterisk-users] Can't compile on NSLU2 LE   Gunner
• [asterisk-users] "No Reply to Our Critical Packet" SIP Calls Dropped in Voicemail   Lincoln King-Cliby
• [asterisk-users] "No Reply to Our Critical Packet" SIP Calls Dropped in Voicemail   Wilton Helm
• [asterisk-users] Warning in CLI   Mark Michelson
• [asterisk-users] Warning in CLI   Mike
• [asterisk-users] app_rxfax.c: Channel T30 DONE < 0 -- incommplete fax reception.   Ken D'Ambrosio
• [asterisk-users] "No Reply to Our Critical Packet" SIP Calls Dropped in Voicemail   Mark Wiater
• [asterisk-users] app_rxfax.c: Channel T30 DONE < 0 -- incommpletefax reception.   Danny Nicholas
• [asterisk-users] dahdi-linux 2.1.0.4 released   Thomas Kenyon
• [asterisk-users] dahdi-linux 2.1.0.4 released   Shaun Ruffell
• [asterisk-users] "No Reply to Our Critical Packet" SIP Calls Dropped in Voicemail   Lincoln King-Cliby
• [asterisk-users] Problem with building dahdi-linux RPM   Axel Thimm
• [asterisk-users] Broken Pipe error while using UpdateConfig command   Jose P. Espinal
• [asterisk-users] Broken Pipe error while using UpdateConfig command   Remco Barendse
• [asterisk-users] Broken Pipe error while using UpdateConfig command   Mark Michelson
• [asterisk-users] dahdi-linux 2.1.0.4 released   Mr. James W. Laferriere
• [asterisk-users] How to set udptl.conf ?   Olivier
• [asterisk-users] dahdi-linux 2.1.0.4 released   Thomas Kenyon
• [asterisk-users] dahdi-linux 2.1.0.4 released   Shaun Ruffell
• [asterisk-users] dahdi-linux 2.1.0.4 released   Shaun Ruffell
• [asterisk-users] some kind of timeout problem in pbx_spool.c   Jerry Geis
• [asterisk-users] dahdi-linux 2.1.0.4 released   Thomas Kenyon
• [asterisk-users] some kind of timeout problem in pbx_spool.c   Tilghman Lesher
• [asterisk-users] How to set udptl.conf ?   Mark Michelson
• [asterisk-users] some kind of timeout problem in pbx_spool.c   Philipp Kempgen
• [asterisk-users] Warnings during a compile   Robert Boardman
• [asterisk-users] Warnings during a compile   Mark Michelson
• [asterisk-users] Warnings during a compile   Robert Boardman
• [asterisk-users] Out of Office: Re: some kind of timeout problem in pbx_spool.c   gustavo at utopixnetworks.com
• [asterisk-users] Out of Office: Out of Office: Re: some kind of t imeout problem in pbx_spool.c   gustavo at utopixnetworks.com
• [asterisk-users] some kind of timeout problem in pbx_spool.c   Tilghman Lesher
• [asterisk-users] Out of Office: Out of Office: Out of Office: Re : some kind of t imeout problem in pbx_spool.c   gustavo at utopixnetworks.com
• [asterisk-users] Out of Office: Out of Office: Out of Office: Ou t of Office: Re : some kind of t imeout problem in pbx_spool.c   gustavo at utopixnetworks.com
• [asterisk-users] Out of Office: Out of Office: Out of Office: Ou t of Office: Ou t of Office: Re : some kind of t imeout problem in pbx_spool.c   gustavo at utopixnetworks.com
• [asterisk-users] Out of Office: Out of Office: Out of Office: Ou t of Office: Ou t of Office: Ou t of Office: Re : some kind of t imeout problem in pbx _spool.c   gustavo at utopixnetworks.com
• [asterisk-users] Out of Office: Out of Office: Out of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Re : some kind of t i meout problem in pbx _spool.c   gustavo at utopixnetworks.com
• [asterisk-users] Out of Office: Out of Office: Out of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Re : some kind of t i meout problem in pbx _sp ool.c   gustavo at utopixnetworks.com
• [asterisk-users] Out of Office: Out of Office: Out of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Re : some kind of t i meou t problem in pbx _sp ool.c   gustavo at utopixnetworks.com
• [asterisk-users] Out of Office: Out of Office: Out of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Re : som e kind of t i meou t problem in pbx _sp ool.c   gustavo at utopixnetworks.com
• [asterisk-users] Contact lookup   D Tucny
• [asterisk-users] Out of Office: Out of Office: Out of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Re : some kind of t i meou t problem in pbx _sp ool.c   Philipp Kempgen
• [asterisk-users] Out of Office: Out of Office: Out of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Re : som e kind of t i meou t problem in pbx _sp ool.c   D Tucny
• [asterisk-users] How to set udptl.conf ?   Steve Underwood
• [asterisk-users] some kind of timeout problem in pbx_spool.c   Philipp Kempgen
• [asterisk-users] Out of Office: Re: Trunk with Polocom Video Conferencing Unit   gustavo at utopixnetworks.com
• [asterisk-users] Out of Office: Out of Office: Re: Trunk with Pol ocom Video Conferencing Unit   gustavo at utopixnetworks.com
• [asterisk-users] Trunk with Polocom Video Conferencing Unit   Alexander Lopez
• [asterisk-users] Out of Office: Out of Office: Out of Office: Re : Trunk with Pol ocom Video Conferencing Unit   gustavo at utopixnetworks.com
• [asterisk-users] Out of Office: Out of Office: Out of Office: Ou t of Office: Re : Trunk with Pol ocom Video Con ferencing Unit   gustavo at utopixnetworks.com
• [asterisk-users] Out of Office: Out of Office: Out of Office: Ou t of Office: Ou t of Office: Re : Trunk with Pol ocom Video Con ferencing Unit   gustavo at utopixnetworks.com
• [asterisk-users] Out of Office: Out of Office: Out of Office: Ou t of Office: Ou t of Office: Ou t of Office: Re : Trunk with Pol ocom Video Con ferencin g Unit   gustavo at utopixnetworks.com
• [asterisk-users] Out of Office: Out of Office: Out of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Re : Trunk with Pol o com Video Con ferencin g Unit   gustavo at utopixnetworks.com
• [asterisk-users] Out of Office: Out of Office: Out of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Re : Trunk with Pol o com Video Con ferencin g Un it   gustavo at utopixnetworks.com
• [asterisk-users] Out of Office: Out of Office: Out of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Re : Trunk with Pol o com Video Con ferencin g Un it   gustavo at utopixnetworks.com
• [asterisk-users] Out of Office: Out of Office: Out of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Re : Tru nk with Pol o com Video Con ferencin g Un it   gustavo at utopixnetworks.com
• [asterisk-users] Out of Office (last message repeated 20 times)   Philipp Kempgen
• [asterisk-users] Out of Office (last message repeated 20 times)   Tilghman Lesher
• [asterisk-users] Need some information on SS7 parameters   Matthew Fredrickson
• [asterisk-users] "No Reply to Our Critical Packet" SIP Calls Dropped in Voicemail   Steven J. Douglas
• [asterisk-users] AOC-E pass through   Jean-Denis Girard
• [asterisk-users] Out of Office: Out of Office: Out of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Re : Tru nk with Pol o com Video Con ferencin g Un it   Alex Balashov
• [asterisk-users] Need some information on SS7 parameters   research at businesstz.com
• [asterisk-users] Stopping chanspy   Jim Dickenson
• [asterisk-users] Audio lag on SIP connections...   Gordon Henderson
• [asterisk-users] Problem with building dahdi-linux RPM   bee-beeep
• [asterisk-users] Contact lookup   Geoff Lane
• [asterisk-users] BerkeleyTIP Feb 7 Sat Global Meeting - Ekiga3, Asterisk, KDE, GPGPU, Debian Edu, GStreamer   john_re
• [asterisk-users] escaping regular expression   Klaus Darilion
• [asterisk-users] AOC-E pass through   Klaus Darilion
• [asterisk-users] TAPI and Asterisk   Klaus Darilion
• [asterisk-users] question on originate call   Jerry Geis
• [asterisk-users] escaping regular expression   D Tucny
• [asterisk-users] escaping regular expression   Klaus Darilion
• [asterisk-users] siemens hipath 4000   Jerry Geis
• [asterisk-users] Call parking   Jeremy G. Gault
• [asterisk-users] AOC-E pass through   Jean-Denis Girard
• [asterisk-users] escaping regular expression   D Tucny
• [asterisk-users] siemens hipath 4000   Josué Conti
• [asterisk-users] Call parking   Danny Nicholas
• [asterisk-users] T1, FoneBRIDGE, and dropped D-Channel   Gleim, Jason
• [asterisk-users] Call parking   Jeremy G. Gault
• [asterisk-users] Call parking   Steven C. Blair
• [asterisk-users] Call parking   Danny Nicholas
• [asterisk-users] Call parking   Mike
• [asterisk-users] Call parking   Jeremy G. Gault
• [asterisk-users] Stopping chanspy followup   Jim Dickenson
• [asterisk-users] siemens hipath 4000   Jerry Geis
• [asterisk-users] question on originate call   Jerry Geis
• [asterisk-users] Problem with MOH and streaming music on 1.6.0.5   Jonn Taylor
• [asterisk-users] Problems with 9133i config   David Ruggles
• [asterisk-users] [asterisk-dev] RFC 2833 DTMF w/ Level 3 Sonus   Kristian Kielhofner
• [asterisk-users] early dial: asterisk and ATA   Ex Vito
• [asterisk-users] question on originate call   Ex Vito
• [asterisk-users] Contact lookup   Ex Vito
• [asterisk-users] siemens hipath 4000   Ex Vito
• [asterisk-users] Stopping chanspy followup   Anthony Francis
• [asterisk-users] question on originate call - solved   Jerry Geis
• [asterisk-users] "No Reply to Our Critical Packet" SIP Calls Dropped in Voicemail   Lincoln King-Cliby
• [asterisk-users] hardware that can accomondate 2 TDM24   Kelvin Chan
• [asterisk-users] hardware that can accomondate 2 TDM24   Lincoln King-Cliby
• [asterisk-users] hardware that can accomondate 2 TDM24   George Pajari
• [asterisk-users] Problem with building dahdi-linux RPM   Axel Thimm
• [asterisk-users] Autodialler query   Sriram
• [asterisk-users] Autodialler query   david at iaxtalk.com
• [asterisk-users] TDM400P Circuit/channel congestion problem   Asfihani
• [asterisk-users] Contact lookup   Geoff Lane
• [asterisk-users] Autodialler query   Kinjal Dixit
• [asterisk-users] escaping regular expression   Klaus Darilion
• [asterisk-users] Configure Asterisk to preserve SIP header?   Scott McNab
• [asterisk-users] sendto syscall: EPERM (Operation not permitted)   John Morris
• [asterisk-users] extensions ending with "#"...   flux at hotbox.ru
• [asterisk-users] TDM400P Circuit/channel congestion problem   Tzafrir Cohen
• [asterisk-users] no need to dial areacode   Ralf Träskman
• [asterisk-users] Configure Asterisk to preserve SIP header?   Benny Amorsen
• [asterisk-users] SIP Authentication only on auth-user?   Klaus Darilion
• [asterisk-users] no need to dial areacode   Gordon Henderson
• [asterisk-users] no need to dial areacode   Ralf Träskman
• [asterisk-users] SIP Authentication only on auth-user? (solved)   Klaus Darilion
• [asterisk-users] musiconhold realtime queue   Ralf Träskman
• [asterisk-users] no need to dial areacode   Danny Nicholas
• [asterisk-users] Configure Asterisk to preserve SIP header?   Danny Nicholas
• [asterisk-users] no need to dial areacode   Geoff Lane
• [asterisk-users] musiconhold realtime queue   David fire
• [asterisk-users] musiconhold realtime queue   Ralf Träskman
• [asterisk-users] Pattom M-ATA, T.38 and Asterisk 1.4. Canreinvite=yes ?   Olivier
• [asterisk-users] musiconhold realtime queue   David fire
• [asterisk-users] Crash Hard, Crash Often   Josiah Bryan
• [asterisk-users] musiconhold realtime queue   Ralf Träskman
• [asterisk-users] Crash Hard, Crash Often   Danny Nicholas
• [asterisk-users] Crash Hard, Crash Often   Doug Lytle
• [asterisk-users] Crash Hard, Crash Often   David Gibbons
• [asterisk-users] Crash Hard, Crash Often   Roderick A. Anderson
• [asterisk-users] Crash Hard, Crash Often   Josiah Bryan
• [asterisk-users] Crash Hard, Crash Often   Josiah Bryan
• [asterisk-users] Crash Hard, Crash Often   Josiah Bryan
• [asterisk-users] registration problem using asterisk 1.6   Laurent Bonny
• [asterisk-users] manager API   Jerry Geis
• [asterisk-users] Crash Hard, Crash Often   Danny Nicholas
• [asterisk-users] manager API   Jerry Geis
• [asterisk-users] Crash Hard, Crash Often   Russell Bryant
• [asterisk-users] Friday Feb 6th at 12 Noon EST: Polycom and Application Development   randulo
• [asterisk-users] Amazon Flexible Payment System - micropayments finally cracked?   Dean Collins
• [asterisk-users] extensions ending with "#"...   Benoit
• [asterisk-users] musiconhold realtime queue   David fire
• [asterisk-users] Pattom M-ATA, T.38 and Asterisk 1.4. Canreinvite=yes ? [SOLVED]   Olivier
• [asterisk-users] How to set udptl.conf ?   Olivier
• [asterisk-users] asterisk-radius   Enrique
• [asterisk-users] Crash Hard, Crash Often   Josiah Bryan
• [asterisk-users] Crash Hard, Crash Often   Doug Lytle
• [asterisk-users] Autodialler query   Sriram
• [asterisk-users] Crash Hard, Crash Often   Josiah Bryan
• [asterisk-users] Crash Hard, Crash Often   Wilton Helm
• [asterisk-users] Crash Hard, Crash Often   Josiah Bryan
• [asterisk-users] Amazon Flexible Payment System - micropayments finally cracked?   James Moore
• [asterisk-users] Amazon Flexible Payment System - micropaymentsfinally cracked?   Dean Collins
• [asterisk-users] Stopping chanspy followup   Jim Dickenson
• [asterisk-users] hardware that can accomondate 2 TDM24   Kelvin Chan
• [asterisk-users] Contact lookup   Ex Vito
• [asterisk-users] Patton M-ATA and T.38   Olivier
• [asterisk-users] Newbie query: how to write priority n+101   Geoff Lane
• [asterisk-users] Newbie query: how to write priority n+101   Mark Michelson
• [asterisk-users] Newbie query: how to write priority n+101   Geoff Lane
• [asterisk-users] Newbie query: how to write priority n+101   Philipp Kempgen
• [asterisk-users] Newbie query: how to write priority n+101   Doug Lytle
• [asterisk-users] Newbie query: how to write priority n+101   Geoff Lane
• [asterisk-users] laptop with modem -> PSTN line -> Asterisk server -> Server's broadband   Enrique
• [asterisk-users] Newbie query: how to write priority n+101   Tilghman Lesher
• [asterisk-users] Voicemail post-processing   Adam Robins
• [asterisk-users] Newbie query: how to write priority n+101   Geoff Lane
• [asterisk-users] Newbie query: how to write priority n+101   Philipp Kempgen
• [asterisk-users] Newbie query: how to write priority n+101   Mike
• [asterisk-users] Voicemail post-processing   Philipp Kempgen
• [asterisk-users] Voicemail post-processing   Alex Balashov
• [asterisk-users] Voicemail post-processing   Alex Balashov
• [asterisk-users] Newbie query: how to write priority n+101   Mark Michelson
• [asterisk-users] RBS T1 DID issue   Jeff LaCoursiere
• [asterisk-users] Voicemail post-processing   Paul Chambers
• [asterisk-users] Voicemail post-processing   Philipp Kempgen
• [asterisk-users] Autodialler query   Jeff LaCoursiere
• [asterisk-users] Voicemail post-processing   Tzafrir Cohen
• [asterisk-users] Crash Hard, Crash Often   Paul Chambers
• [asterisk-users] Incoming fax detection on mISDN hfcmulti B410P card   Ex Vito
• [asterisk-users] TDM400P Circuit/channel congestion problem   Asfihani
• [asterisk-users] Crash Hard, Crash Often   Wilton Helm
• [asterisk-users] Getting DIALSTATUS from SIP provider   Jim Dickenson
• [asterisk-users] extensions ending with "#"...   flux at hotbox.ru
• [asterisk-users] extensions ending with "#"...   flux at hotbox.ru
• [asterisk-users] Configure Asterisk to preserve SIP header?   Scott McNab
• [asterisk-users] Java IAX Implementation   Wolfgang Pichler
• [asterisk-users] Newbie query: how to write priority n+101   Geoff Lane
• [asterisk-users] asterisk and DNS   Julian Lyndon-Smith
• [asterisk-users] Monitor and SIP transfers (SIP REFER)   Gunnar Schaller
• [asterisk-users] Rewriting numbers while processing dial plan?   Jan-Aage Frydenbø-Bruvoll
• [asterisk-users] asterisk and DNS   Giorgio Incantalupo
• [asterisk-users] asterisk and DNS   Jan-Aage Frydenbø-Bruvoll
• [asterisk-users] set caller id on outgoing calls through BRI ISDN lines   Vieri
• [asterisk-users] Incoming fax detection on mISDN hfcmulti B410P card   Vieri
• [asterisk-users] asterisk and DNS   Julian Lyndon-Smith
• [asterisk-users] Incoming fax detection on mISDN hfcmulti B410Pcard   Andrew Thomas
• [asterisk-users] asterisk and DNS   Jan-Aage Frydenbø-Bruvoll
• [asterisk-users] set caller id on outgoing calls through BRI ISDNlines   Andrew Thomas
• [asterisk-users] Problem with building dahdi-linux RPM   bee-beeep
• [asterisk-users] Crash Hard, Crash Often   Josiah Bryan
• [asterisk-users] Voicemail post-processing   Adam Robins
• [asterisk-users] set caller id on outgoing calls through BRI ISDNlines   Vieri
• [asterisk-users] Amazon Flexible Payment System - micropayments finally cracked?   SIP
• [asterisk-users] set caller id on outgoing calls through BRIISDNlines   Andrew Thomas
• [asterisk-users] asterisk and DNS   Tzafrir Cohen
• [asterisk-users] Add-on for SRTP and SSIP   Mike
• [asterisk-users] [Asterisk-Users] Asterisk as a dial in server for internet   Enrique
• [asterisk-users] asterisk-radius   Philipp Kempgen
• [asterisk-users] dial in server   Enrique
• [asterisk-users] H323 stress test   Mindaugas Kezys
• [asterisk-users] asterisk and DNS   Paul Chambers
• [asterisk-users] Monitor and SIP transfers (SIP REFER)   Mark Michelson
• [asterisk-users] asterisk and DNS   Dave Cotton
• [asterisk-users] dial in server   Steve Howes
• [asterisk-users] upgrade from 1.4.22-rc5 to 1.4.23.1: crash when transferring a call   Giorgio Incantalupo
• [asterisk-users] set caller id on outgoing calls through BRIISDNlines   Vieri
• [asterisk-users] [asterisk-user] $100USD for anyone who can install Chan_SCCP for me   Sam Tam
• [asterisk-users] Newbie query: how to write priority n+101   Leif Madsen
• [asterisk-users] upgrade from 1.4.22-rc5 to 1.4.23.1: crash when transferring a call   Jonn Taylor
• [asterisk-users] Newbie query: how to write priority n+101   Jared Smith
• [asterisk-users] upgrade from 1.4.22-rc5 to 1.4.23.1: crash when transferring a call   Mark Michelson
• [asterisk-users] Add-on for SRTP and SSIP   Olivier
• [asterisk-users] AgentCallBackLogin no longer works after installing asterisk 1.6   oumar ndiaye
• [asterisk-users] AgentCallBackLogin no longer works after installing asterisk 1.6   Robert Broyles
• [asterisk-users] Security issue   oumar ndiaye
• [asterisk-users] Security issue   Danny Nicholas
• [asterisk-users] AgentCallBackLogin no longer works after installing asterisk 1.6   Anthony Francis
• [asterisk-users] Newbie query: how to write priority n+101   Matthew Nicholson
• [asterisk-users] AgentCallBackLogin no longer works after installing asterisk 1.6   Danny Nicholas
• [asterisk-users] AgentCallBackLogin no longer works after installing asterisk 1.6   Philipp Kempgen
• [asterisk-users] Credit Card processing machines   Jeff LaCoursiere
• [asterisk-users] Newbie query: how to write priority n+101   Hans Witvliet
• [asterisk-users] Credit Card processing machines   Jeff LaCoursiere
• [asterisk-users] AgentCallBackLogin no longer works after installing asterisk 1.6   Anthony Francis
• [asterisk-users] AgentCallBackLogin no longer works after installing asterisk 1.6   Robert Broyles
• [asterisk-users] AgentCallBackLogin no longer works after installing asterisk 1.6   Philipp Kempgen
• [asterisk-users] AgentCallBackLogin no longer works after installing asterisk 1.6   Philipp Kempgen
• [asterisk-users] Allow-Automated-Answer (was: Re: AgentCallBackLogin no longer works after installing asterisk 1.6)   Philipp Kempgen
• [asterisk-users] Allow-Automated-Answer   Philipp Kempgen
• [asterisk-users] AgentCallBackLogin no longer works after installing asterisk 1.6   Robert Broyles
• [asterisk-users] AgentCallBackLogin no longer works after installing asterisk 1.6   Rob Hillis
• [asterisk-users] AgentCallBackLogin no longer works after installing asterisk 1.6   Philipp Kempgen
• [asterisk-users] Rewriting numbers while processing dial plan?   Laurent
• [asterisk-users] Running asterisk on ARM (TS-7800) 1.4.23.1   James Lamanna
• [asterisk-users] Running asterisk on ARM (TS-7800) 1.4.23.1   Julien Claassen
• [asterisk-users] AgentCallBackLogin no longer works after installing asterisk 1.6   oumar ndiaye
• [asterisk-users] AgentCallBackLogin no longer works after installing asterisk 1.6   Robert Broyles
• [asterisk-users] VPN and Asterisk   David at ULC
• [asterisk-users] VPN and Asterisk   Jose P. Espinal
• [asterisk-users] put the hostname of asterisk in the callerid uri to avoid NAT problems   nik600
• [asterisk-users] AOC-E pass through   Olivier
• [asterisk-users] A Simple Asterisk Based Toll Fraud Prevention Script   J. Oquendo
• [asterisk-users] Security issue   David fire
• [asterisk-users] Asterisk 3rd party developed commercial software sales licensing platform   Dean Collins
• [asterisk-users] GROUP() decrement   Thomas Winter
• [asterisk-users] One way audio after IVR tree   James Lamanna
• [asterisk-users] VPN and Asterisk   Dave Platt
• [asterisk-users] VPN and Asterisk   Bill Michaelson
• [asterisk-users] Minimum version for asterisk and iaxmodem   James Lamanna
• [asterisk-users] Minimum version for asterisk and iaxmodem   Lee Howard
• [asterisk-users] Minimum version for asterisk and iaxmodem   James Lamanna
• [asterisk-users] Minimum version for asterisk and iaxmodem   James Lamanna
• [asterisk-users] can anybody tell me how Magic jack can be so cheap ????   Ignacio Ortega A.
• [asterisk-users] Minimum version for asterisk and iaxmodem   Tzafrir Cohen
• [asterisk-users] Security issue   oumar ndiaye
• [asterisk-users] Security issue   Eric Fort
• [asterisk-users] can anybody tell me how Magic jack can be so cheap ????   C F
• [asterisk-users] can anybody tell me how Magic jack can be so cheap ????   Forrest W Christian
• [asterisk-users] can anybody tell me how Magic jack can be so cheap ????   k4rjj at bellsouth.net
• [asterisk-users] can anybody tell me how Magic jack can be so cheap ????   Sam
• [asterisk-users] can anybody tell me how Magic jack can be so cheap ????   Kinjal Dixit
• [asterisk-users] Security issue   Jim DeVito
• [asterisk-users] Security issue   David fire
• [asterisk-users] Streaming meetings vs conference hardware   Michael Graves
• [asterisk-users] meetme application   邱磊
• [asterisk-users] chan_oss.c:585 setformat: Unable to re-open DSP device   David at ULC
• [asterisk-users] Asterisk + voxbone ==> Failed to authenticate user   Johan Dindaine - Asterisk
• [asterisk-users] meetme application   Giedrius Augys
• [asterisk-users] Asterisk and CIsco 1760 SIP ?   Phibee Network Operation Center
• [asterisk-users] meetme application   D Tucny
• [asterisk-users] can anybody tell me how Magic jack can be so cheap ????   andrej at teksavvy.com
• [asterisk-users] Asterisk + voxbone ==> Failed to authenticate user   Tobias Wolf
• [asterisk-users] Security issue   Grygoriy Dobrovolskyy
• [asterisk-users] Security issue   Gordon Henderson
• [asterisk-users] Security issue   Geraint Lee
• [asterisk-users] Security issue   Tzafrir Cohen
• [asterisk-users] Asterisk + voxbone ==> Failed to authenticate user   hh174
• [asterisk-users] InUse&Ringing   Andrew Thomas
• [asterisk-users] InUse&Ringing   Philipp Kempgen
• [asterisk-users] Security issue   Geraint Lee
• [asterisk-users] Security issue   Gordon Henderson
• [asterisk-users] Security issue   Tilghman Lesher
• [asterisk-users] InUse&Ringing   Andrew Thomas
• [asterisk-users] Michael Graves post   Dean Collins
• [asterisk-users] Michael Graves post   Steve Howes
• [asterisk-users] reinvite   Jeff LaCoursiere
• [asterisk-users] Monitor and SIP transfers (SIP REFER)   Gunnar Schaller
• [asterisk-users] Michael Graves post   Dean Collins
• [asterisk-users] Michael Graves post   Geraint Lee
• [asterisk-users] Noisy Ring Back Tone with TE205P card   Imanol Pardavila
• [asterisk-users] Michael Graves post   Philipp Kempgen
• [asterisk-users] Michael Graves post   Steve Howes
• [asterisk-users] Michael Graves post   Michael Graves
• [asterisk-users] AgentCallBackLogin no longer works after installing asterisk 1.6   Anthony Francis
• [asterisk-users] Michael Graves post   Michael Graves
• [asterisk-users] AgentCallBackLogin no longer works after installing asterisk 1.6   Anthony Francis
• [asterisk-users] AgentCallBackLogin no longer works after installing asterisk 1.6   Anthony Francis
• [asterisk-users] Problem with AMI originate   Jim Dickenson
• [asterisk-users] problem getting asterisk behind NAT to run with sipproxd   Tamer Higazi
• [asterisk-users] Transfer Asterisk 1.6 Telephone IP   Daviramos Roussenq Fortunato
• [asterisk-users] USA BRI -- any hope at all?   Wilton Helm
• [asterisk-users] How to make the Asterisk-GUI work with DAHDI..please??   Enrique
• [asterisk-users] Call drops after a minute on 1.6.0.5   Carlos Chavez
• [asterisk-users] Problem with upper case extension names   Jim Dickenson
• [asterisk-users] How to make the Asterisk-GUI work withDAHDI..please??   Danny Nicholas
• [asterisk-users] Call drops after a minute on 1.6.0.5   Carlos Chavez
• [asterisk-users] Call drops after a minute on 1.6.0.5   Danny Nicholas
• [asterisk-users] Hangup extensions via CLI?   Tim Nelson
• [asterisk-users] [?? Probable Spam] Re: How to make the Asterisk-GUI workwithDAHDI..please??   Enrique
• [asterisk-users] SIP password encryption   Chris Rowson
• [asterisk-users] SIP password encryption   Chris Rowson
• [asterisk-users] What t38pt_udptl is ? Explain T.38 in 1.4   Olivier
• [asterisk-users] asterisk registered as UA   Szasz Szabolcs
• [asterisk-users] SIP password encryption   Kevin P. Fleming
• [asterisk-users] Is "a=fmtp:101 0-15" a legal option in SDP ?   Olivier
• [asterisk-users] [asterisk-dev] 1.4 and CDRs -- The Breaking Point   Steve Murphy
• [asterisk-users] SIP password encryption   Chris Rowson
• [asterisk-users] asterisk registered as UA   Matthew Nicholson
• [asterisk-users] Is "a=fmtp:101 0-15" a legal option in SDP ?   Raj Jain
• [asterisk-users] asterisk registered as UA   Anthony Francis
• [asterisk-users] Hangup extensions via CLI?   Alexander Lopez
• [asterisk-users] Audiocodes - Disconnect Supervision   Nabeel Jafferali
• [asterisk-users] [SOLVED] Re: Hangup extensions via CLI?   Tim Nelson
• [asterisk-users] SMS /w Asterisk   Mike Diehl
• [asterisk-users] USA BRI -- any hope at all?   Eric Chamberlain
• [asterisk-users] USA BRI -- any hope at all?   peder at networkoblivion.com
• [asterisk-users] USA BRI -- any hope at all?   Wilton Helm
• [asterisk-users] What t38pt_udptl is ? Explain T.38 in 1.4   David Backeberg
• [asterisk-users] USA BRI -- any hope at all?   Joe Greco
• [asterisk-users] USA BRI -- any hope at all?   Joe Greco
• [asterisk-users] Asterisk - Trixbox   Mike Hammett
• [asterisk-users] hosted voip?   Dean Collins
• [asterisk-users] What t38pt_udptl is ? Explain T.38 in 1.4   Olivier
• [asterisk-users] meetme application   邱磊
• [asterisk-users] SMS /w Asterisk   randulo
• [asterisk-users] What do you use? .conf or AEL?   Alan Lord (News)
• [asterisk-users] meetme application   Lee, John (Sydney)
• [asterisk-users] What do you use? .conf or AEL?   Lee, John (Sydney)
• [asterisk-users] USA BRI -- any hope at all?   Tzafrir Cohen
• [asterisk-users] [NO ANSWER] Re: Asterisk and CIsco 1760 SIP ?   Phibee Network Operation Center
• [asterisk-users] Asterisk how many calls handle using H.323 to SIP conversion?   MianAsif
• [asterisk-users] Disabling Echo Cancellation on a per Call basis   Tobias Wolf
• [asterisk-users] put the hostname of asterisk in the callerid uri to avoid NAT problems   nik600
• [asterisk-users] USA BRI -- any hope at all?   Philipp Kempgen
• [asterisk-users] Skip password option in voicemail.conf   Olivier
• [asterisk-users] Disabling Echo Cancellation on a per Call basis   Olivier
• [asterisk-users] Asterisk how many calls handle using H.323 to SIP conversion?   David fire
• [asterisk-users] Asterisk - Trixbox   Steve Totaro
• [asterisk-users] [NO ANSWER] Re: Asterisk and CIsco 1760 SIP ?   David Gibbons
• [asterisk-users] SMS /w Asterisk   Steve Totaro
• [asterisk-users] [NO ANSWER] Re: Asterisk and CIsco 1760 SIP ?   Phibee Network Operation Center
• [asterisk-users] [ANSWER] Re: Asterisk and CIsco 1760 SIP ?   peder at networkoblivion.com
• [asterisk-users] What do you use? .conf or AEL?   Danny Nicholas
• [asterisk-users] SMS /w Asterisk   Remco Barendse
• [asterisk-users] SMS /w Asterisk   Steve Totaro
• [asterisk-users] Disabling Echo Cancellation on a per Call basis   Tobias Wolf
• [asterisk-users] SMS /w Asterisk   Steve Totaro
• [asterisk-users] What do you use? .conf or AEL?   Dave Fullerton
• [asterisk-users] Asterisk - Trixbox   Mike Hammett
• [asterisk-users] Asterisk - Trixbox   Steve Totaro
• [asterisk-users] unistim and transfer calls   Ralf Träskman
• [asterisk-users] Asterisk - Trixbox   Mike Hammett
• [asterisk-users] What do you use? .conf or AEL?   Tzafrir Cohen
• [asterisk-users] SMS /w Asterisk   Remco Barendse
• [asterisk-users] What t38pt_udptl is ? Explain T.38 in 1.4   David Backeberg
• [asterisk-users] Asterisk - Trixbox   Mike Hammett
• [asterisk-users] SMS /w Asterisk   Steve Totaro
• [asterisk-users] put the hostname of asterisk in the callerid uri to avoid NAT problems   Tilghman Lesher
• [asterisk-users] Asterisk - Trixbox   Steve Totaro
• [asterisk-users] Asterisk - Trixbox   Steve Totaro
• [asterisk-users] Asterisk - Trixbox   Mike Hammett
• [asterisk-users] USA BRI -- any hope at all?   Wilton Helm
• [asterisk-users] What do you use? .conf or AEL?   Gordon Henderson
• [asterisk-users] Transfer Asterisk 1.6 Telephone IP   Daviramos Roussenq Fortunato
• [asterisk-users] connecting 66 analog phones to asterisk - hardware suggestions   Erick Perez
• [asterisk-users] Asterisk AGX addons compile issues   Olivier
• [asterisk-users] asterisk registered as UA   Szasz Szabolcs
• [asterisk-users] What do you use? .conf or AEL?   Geraint Lee
• [asterisk-users] chan_oss.c:585 setformat: Unable to re-open DSP device   David at ULC
• [asterisk-users] connecting 66 analog phones to asterisk - hardware suggestions   John Balogh
• [asterisk-users] What do you use? .conf or AEL?   Mik Cheez
• [asterisk-users] Rewriting numbers while processing dial plan?   Martin Lima
• [asterisk-users] Aastra phone crashes with Asterisk 1.6   Carlos Chavez
• [asterisk-users] Asterisk + voxbone ==> Failed to authenticate user   Johan Dindaine - Asterisk
• [asterisk-users] Digium TE220 card partially detected   Maxim Litnitskiy
• [asterisk-users] Digium TE220 card partially detected   Kevin P. Fleming
• [asterisk-users] Max person in meetme conference   Emmanuel Bruno
• [asterisk-users] Max person in meetme conference   Nhadie
• [asterisk-users] Max person in meetme conference   Emmanuel Bruno
• [asterisk-users] put the hostname of asterisk in the callerid uri to avoid NAT problems   Steven J. Douglas
• [asterisk-users] Aastra phone crashes with Asterisk 1.6   Philipp Kempgen
• [asterisk-users] What do you use? .conf or AEL?   Alan Lord (News)
• [asterisk-users] unistim and transfer calls   Ralf Träskman
• [asterisk-users] Is "a=fmtp:101 0-15" a legal option in SDP ?   Johansson Olle E
• [asterisk-users] put the hostname of asterisk in the callerid uri to avoid NAT problems   nik600
• [asterisk-users] OPTIONS packets   michel freiha
• [asterisk-users] OPTIONS packets   Alex Balashov
• [asterisk-users] call forward all except the extension it is forwarded to   Vieri
• [asterisk-users] call forward all except the extension it is forwarded to   Philipp Kempgen
• [asterisk-users] Hangup extensions via CLI?   Lenz Emilitri
• [asterisk-users] call forward all except the extension it is forwarded to   Vieri
• [asterisk-users] call forward all except the extension it is forwarded to   Philipp Kempgen
• [asterisk-users] What do you use? .conf or AEL?   Gordon Henderson
• [asterisk-users] call forward all except the extension it is forwarded to   Vieri
• [asterisk-users] What do you use? .conf or AEL?   Philipp Kempgen
• [asterisk-users] call forward all except the extension it is forwarded to   Vieri
• [asterisk-users] connecting 66 analog phones to asterisk - hardware suggestions   Jeff LaCoursiere
• [asterisk-users] Asterisk AGX addons compile issues   Andrew Thomas
• [asterisk-users] What do you use? .conf or AEL?   Gordon Henderson
• [asterisk-users] Billing and Soft Switch.   David at ULC
• [asterisk-users] Looking for 'remote Asterisk hands' support in Mexico   Jeff Verheyen
• [asterisk-users] Expressions, CUT(), ... (was: Re: call forward all except the extension it is forwarded to)   Philipp Kempgen
• [asterisk-users] DTMF tones mid conversation   Andrew Thomas
• [asterisk-users] Billing and Soft Switch.   Philipp Kempgen
• [asterisk-users] DTMF tones mid conversation   Paulo Santos
• [asterisk-users] connecting 66 analog phones to asterisk - hardware suggestions   Heath Roberts
• [asterisk-users] Aastra phone crashes with Asterisk 1.6   OCG Technical Support
• [asterisk-users] Aastra phone crashes with Asterisk 1.6   Steve Davies
• [asterisk-users] What do you use? .conf or AEL?   Tilghman Lesher
• [asterisk-users] Billing and Soft Switch.   Steve Howes
• [asterisk-users] DTMF tones mid conversation   F6HQZ
• [asterisk-users] Billing and Soft Switch.   Alex Balashov
• [asterisk-users] What do you use? .conf or AEL?   C F
• [asterisk-users] ChanSpy problem   Jim Dickenson
• [asterisk-users] What do you use? .conf or AEL?   Matthew Nicholson
• [asterisk-users] asterisk across a firewall   Erick Perez
• [asterisk-users] asterisk across a firewall   Tim Nelson
• [asterisk-users] asterisk across a firewall   Alex Balashov
• [asterisk-users] Intercom/Doorbell Integration   Gavin Lewandowski
• [asterisk-users] asterisk across a firewall   Gordon Henderson
• [asterisk-users] What do you use? .conf or AEL?   Terry Wilson
• [asterisk-users] What do you use? .conf or AEL?   Gordon Henderson
• [asterisk-users] [OT] Re: What do you use? .conf or AEL?   Philipp Kempgen
• [asterisk-users] The download link, why server down?   bilal ghayyad
• [asterisk-users] The download link, why server down?   Tilghman Lesher
• [asterisk-users] What do you use? .conf or AEL?   Tilghman Lesher
• [asterisk-users] What do you use? .conf or AEL?   Philipp Kempgen
• [asterisk-users] What do you use? .conf or AEL?   Tilghman Lesher
• [asterisk-users] What do you use? .conf or AEL?   Tzafrir Cohen
• [asterisk-users] call picking and transfers   Jeff LaCoursiere
• [asterisk-users] call picking and transfers   Philipp Kempgen
• [asterisk-users] Problem with AMI action userevent   Jim Dickenson
• [asterisk-users] WiFi SIP phone w/VPN?   Ken D'Ambrosio
• [asterisk-users] connecting 66 analog phones to asterisk - hardware suggestions   C F
• [asterisk-users] Strange dialplan matching issue   Chris Bagnall
• [asterisk-users] Strange dialplan matching issue   Jose P. Espinal
• [asterisk-users] IDAP T1   Lee, John (Sydney)
• [asterisk-users] Strange dialplan matching issue   Nabeel Jafferali
• [asterisk-users] Strange dialplan matching issue   Nabeel Jafferali
• [asterisk-users] Strange dialplan matching issue   Stephen Davies
• [asterisk-users] connecting 66 analog phones to asterisk - hardware suggestions   Stephen Davies
• [asterisk-users] How to make the Asterisk-GUI workwithDAHDI..please??   Dovid Bender
• [asterisk-users] Invalid Extension   Dovid Bender
• [asterisk-users] connecting 66 analog phones to asterisk - hardware suggestions   Steve Totaro
• [asterisk-users] What do you use? .conf or AEL?   Louis-David Mitterrand
• [asterisk-users] What do you use? .conf or AEL?   Gordon Henderson
• [asterisk-users] Friday the 13th Muhahaha Allison Smith and more on the Polycom Applications   randulo
• [asterisk-users] Keep your passwords secure .. (VoIP hacker news)   Gordon Henderson
• [asterisk-users] Siemens Hipath PRI to Asterisk Call Routing?   joekane at gmail.com
• [asterisk-users] Siemens Hipath PRI to Asterisk Call Routing?   Rob Hillis
• [asterisk-users] Siemens Hipath PRI to Asterisk Call Routing?   turby at canistec.com
• [asterisk-users] OPTIONS packets   Johansson Olle E
• [asterisk-users] Hangup extensions via CLI?   Helius Ferreira
• [asterisk-users] OPTIONS packets   michel freiha
• [asterisk-users] g723 llicense   Nhadie
• [asterisk-users] Disabling Echo Cancellation on a per Call basis   Tobias Wolf
• [asterisk-users] Keep your passwords secure .. (VoIP hacker news)   Steve Totaro
• [asterisk-users] After Monitor() files disappear   Gunnar Schaller
• [asterisk-users] Strange dialplan matching issue   OCG Technical Support
• [asterisk-users] Keep your passwords secure .. (VoIP hacker news)   Jeff LaCoursiere
• [asterisk-users] g723 llicense   Kevin P. Fleming
• [asterisk-users] Disabling Echo Cancellation on a per Call basis   Kevin P. Fleming
• [asterisk-users] Hangup extensions via CLI?   Danny Nicholas
• [asterisk-users] Multiple caller id ...   Julian Lyndon-Smith
• [asterisk-users] Problem with parking   Örn Arnarson
• [asterisk-users] Keep your passwords secure .. (VoIP hacker news)   Steve Totaro
• [asterisk-users] How to make the Asterisk-GUIworkwithDAHDI..please??   Danny Nicholas
• [asterisk-users] Hangup extensions via CLI?   Lukas Rypl
• [asterisk-users] Problem with parking   Örn Arnarson
• [asterisk-users] OSLEC not being loaded on Ubuntu Intrepid   Alan Lord (News)
• [asterisk-users] What do you use? .conf or AEL?   Matthew Nicholson
• [asterisk-users] Configuring Patton SmartNode with ISDN2e and Asterisk   Olivier
• [asterisk-users] DTMF tones mid conversation   Andrew Thomas
• [asterisk-users] Caller ID replacement   David Ruggles
• [asterisk-users] Caller ID replacement   Barry L. Kline
• [asterisk-users] Caller ID replacement   Matthew Nicholson
• [asterisk-users] Asterisk Queue and URL Calling   Mohsen
• [asterisk-users] OSLEC not being loaded on Ubuntu Intrepid [SOLVED]   Alan Lord (News)
• [asterisk-users] Caller ID replacement   David Ruggles
• [asterisk-users] Disabling Echo Cancellation on a per Call basis   Tobias Wolf
• [asterisk-users] Caller ID replacement   David Ruggles
• [asterisk-users] reinvite   Jeff LaCoursiere
• [asterisk-users] reinvite   Mark Michelson
• [asterisk-users] OSLEC not being loaded on Ubuntu Intrepid   Tzafrir Cohen
• [asterisk-users] 1.6.1-rc1 errors   Carlos Chavez
• [asterisk-users] Disabling Echo Cancellation on a per Call basis   Kevin P. Fleming
• [asterisk-users] Skype beta news ?   Casey Boone
• [asterisk-users] 1.6.1-rc1 errors   Mark Michelson
• [asterisk-users] Disabling Echo Cancellation on a per Call basis   Tobias Wolf
• [asterisk-users] WiFi SIP phone w/VPN?   Dave Platt
• [asterisk-users] What t38pt_udptl is ? Explain T.38 in 1.4   Olivier
• [asterisk-users] Disabling Echo Cancellation on a per Call basis   Kevin P. Fleming
• [asterisk-users] Skype beta news ?   Tilghman Lesher
• [asterisk-users] CISCO 2950 -> 4 connections -> Cap of 512 Kbps -> How to bond ?   Vikas
• [asterisk-users] Multiple caller id ...   Benny Amorsen
• [asterisk-users] Asterisk Queue and URL Calling   Benny Amorsen
• [asterisk-users] Asterisk Queue and URL Calling   David Van Ginneken
• [asterisk-users] asterisk across a firewall   Erick Perez
• [asterisk-users] WiFi SIP phone w/VPN?   Frank Bulk - iName.com
• [asterisk-users] What t38pt_udptl is ? Explain T.38 in 1.4   Benny Amorsen
• [asterisk-users] Queue problem   Mike
• [asterisk-users] CISCO 2950 -> 4 connections -> Cap of 512 Kbps -> How to bond ?   Heath Roberts
• [asterisk-users] Queue problem   Danny Nicholas
• [asterisk-users] CISCO 2950 -> 4 connections -> Cap of 512 Kbps -> How to bond ?   Vikas
• [asterisk-users] CISCO 2950 -> 4 connections -> Cap of 512 Kbps -> How to bond ?   David Backeberg
• [asterisk-users] CISCO 2950 -> 4 connections -> Cap of 512 Kbps -> How to bond ?   Benny Amorsen
• [asterisk-users] Disabling Echo Cancellation on a per Call basis   Tobias Wolf
• [asterisk-users] Disabling Echo Cancellation on a per Call basis   Kevin P. Fleming
• [asterisk-users] CISCO 2950 -> 4 connections -> Cap of 512 Kbps -> How to bond ?   Jeff LaCoursiere
• [asterisk-users] CISCO 2950 -> 4 connections -> Cap of 512 Kbps -> How to bond ?   Michael
• [asterisk-users] call picking and transfers   Jeff LaCoursiere
• [asterisk-users] Skype beta news ?   Steve Totaro
• [asterisk-users] zaptel for asterisk   Talha Jalal
• [asterisk-users] Fwd: Manager Interface Originate (ASYNC) - How to get the Originate Status   Aloysius Thevarajah Lloyd (SunTel Technologies)
• [asterisk-users] MINNESOTA: TwinCities Asterisk Users Group Meeting, this Saturday. Valentine's Day Feb/14/2009 11:30am   asterisk_help at iwishi.nu
• [asterisk-users] MINNESOTA: TwinCities Asterisk Users Group Meeting, this Saturday. Valentine's Day Feb/14/2009 11:30am   Alex Balashov
• [asterisk-users] Asterisk Queue and URL Calling   Mohsen
• [asterisk-users] Re : Asterisk Queue and URL Calling   Mohsen
• [asterisk-users] ExitIf() convention?   Jack Bates
• [asterisk-users] Fwd: Manager Interface Originate (ASYNC) - How to get the Originate Status   Giedrius Augys
• [asterisk-users] ExitIf() convention?   Tilghman Lesher
• [asterisk-users] Fwd: Manager Interface Originate (ASYNC) - How to get the Originate Status   Aloysius Lloyd
• [asterisk-users] Fwd: Manager Interface Originate (ASYNC) - How to get the Originate Status   Aloysius Lloyd
• [asterisk-users] Asterisk Queue and URL Calling   Lenz Emilitri
• [asterisk-users] Hangup extensions via CLI?   Lenz Emilitri
• [asterisk-users] Fwd: Manager Interface Originate (ASYNC) - How to get the Originate Status   Giedrius Augys
• [asterisk-users] Fwd: Manager Interface Originate (ASYNC) - How to get the Originate Status   Aloysius Lloyd
• [asterisk-users] CISCO 2950 -> 4 connections -> Cap of 512 Kbps -> How to bond ?   Vikas
• [asterisk-users] WiFi SIP phone w/VPN?   Grygoriy Dobrovolskyy
• [asterisk-users] Broken Pipe error while using UpdateConfig command   Rilawich Ango
• [asterisk-users] WiFi SIP phone w/VPN?   k4rjj at bellsouth.net
• [asterisk-users] Hangup extensions via CLI?   Lenz Emilitri
• [asterisk-users] Multiple caller id ...   Julian Lyndon-Smith
• [asterisk-users] CISCO 2950 -> 4 connections -> Cap of 512 Kbps -> How to bond ?   Benny Amorsen
• [asterisk-users] OSLEC not being loaded on Ubuntu Intrepid   Alan Lord (News)
• [asterisk-users] WiFi SIP phone w/VPN?   Benny Amorsen
• [asterisk-users] CISCO 2950 -> 4 connections -> Cap of 512 Kbps -> How to bond ?   Alex Balashov
• [asterisk-users] PRI Test Lab   Lee Wilson
• [asterisk-users] Siemens Hipath PRI to Asterisk Call Routing?   joekane at gmail.com
• [asterisk-users] PRI Test Lab   Alex Balashov
• [asterisk-users] CISCO 2950 -> 4 connections -> Cap of 512 Kbps -> How to bond ?   Alex Balashov
• [asterisk-users] Cisco IP Phone 7940G.   Catalin S.
• [asterisk-users] PRI Test Lab   Lee Wilson
• [asterisk-users] Cisco IP Phone 7940G.   Alex Balashov
• [asterisk-users] PRI Test Lab   Alex Balashov
• [asterisk-users] Fwd: Manager Interface Originate (ASYNC) - How to get the Originate Status   Rayed Bs
• [asterisk-users] Cisco IP Phone 7940G.   Catalin S.
• [asterisk-users] Hangup extensions via CLI?   Dinesh Nair
• [asterisk-users] Cisco IP Phone 7940G.   Alex Balashov
• [asterisk-users] Cisco IP Phone 7940G.   k4rjj at bellsouth.net
• [asterisk-users] PRI Test Lab   Lee Wilson
• [asterisk-users] PRI Test Lab   Matt Florell
• [asterisk-users] PRI Test Lab   Alex Balashov
• [asterisk-users] PRI Test Lab   Olivier
• [asterisk-users] PRI Test Lab   Alex Balashov
• [asterisk-users] PRI Test Lab   Tom Moore
• [asterisk-users] CISCO 2950 -> 4 connections -> Cap of 512 Kbps -> How to bond ?   D Tucny
• [asterisk-users] Caller ID replacement   D Tucny
• [asterisk-users] PRI Test Lab   Matt Florell
• [asterisk-users] CISCO 2950 -> 4 connections -> Cap of 512 Kbps -> How to bond ?   Vikas
• [asterisk-users] PRI Test Lab   Olivier
• [asterisk-users] Cisco IP Phone 7940G.   Catalin S.
• [asterisk-users] Cisco IP Phone 7940G.   peder at networkoblivion.com
• [asterisk-users] PRI Test Lab   Kevin P. Fleming
• [asterisk-users] WiFi SIP phone w/VPN?   Olivier
• [asterisk-users] PRI Test Lab   Olivier
• [asterisk-users] PRI Test Lab   Kevin P. Fleming
• [asterisk-users] MINNESOTA: TwinCities Asterisk Users Group Meeting, this Saturday. Valentine's Day Feb/14/2009 11:30am   Jeff LaCoursiere
• [asterisk-users] PRI Test Lab   Dave Fullerton
• [asterisk-users] PRI Test Lab   Lee Wilson
• [asterisk-users] CISCO 2950 -> 4 connections -> Cap of 512 Kbps -> How to bond ?   Jeff LaCoursiere
• [asterisk-users] After Monitor() files disappear   Gunnar Schaller
• [asterisk-users] CISCO 2950 -> 4 connections -> Cap of 512 Kbps -> How to bond ?   D Tucny
• [asterisk-users] Cisco IP Phone 7940G.   Ronny Julian
• [asterisk-users] Cisco IP Phone 7940G.   Danny Nicholas
• [asterisk-users] Cisco IP Phone 7940G.   John Novack
• [asterisk-users] Cisco IP Phone 7940G.   John Novack
• [asterisk-users] CISCO 2950 -> 4 connections -> Cap of 512 Kbps -> How to bond ?   Vikas
• [asterisk-users] Cisco IP Phone 7940G.   Ronny Julian
• [asterisk-users] Hangup extensions via CLI?   Tim Nelson
• [asterisk-users] Cisco IP Phone 7940G.   Jeff LaCoursiere
• [asterisk-users] PRI Test Lab   Matt Florell
• [asterisk-users] CISCO 2950 -> 4 connections -> Cap of 512 Kbps -> How to bond ?   David Backeberg
• [asterisk-users] Cisco IP Phone 7940G.   John Novack
• [asterisk-users] Cisco IP Phone 7940G.   Ronny Julian
• [asterisk-users] PRI Test Lab   Tzafrir Cohen
• [asterisk-users] MINNESOTA: TwinCities Asterisk Users Group Meeting, this Saturday. Valentine's Day Feb/14/2009 11:30am   Anthony Messina
• [asterisk-users] Cisco IP Phone 7940G.   Tzafrir Cohen
• [asterisk-users] Cisco IP Phone 7940G.   Jeff LaCoursiere
• [asterisk-users] MINNESOTA: TwinCities Asterisk Users Group Meeting, this Saturday. Valentine's Day Feb/14/2009 11:30am   Tim Nelson
• [asterisk-users] Continue processing AGI script after hangup   cbbs70a at hotmail.com
• [asterisk-users] MINNESOTA: TwinCities Asterisk Users Group Meeting, this Saturday. Valentine's Day Feb/14/2009 11:30am   Shane Young
• [asterisk-users] Broken Pipe error while using UpdateConfig command   Örn Arnarson
• [asterisk-users] CISCO 2950 -> 4 connections -> Cap of 512 Kbps -> How to bond ?   peder at networkoblivion.com
• [asterisk-users] Asterisk and Amazon EC2 cloud service tutorial   Eric Chamberlain
• [asterisk-users] Continue processing AGI script after hangup   Tilghman Lesher
• [asterisk-users] Asterisk 1.6.0.5 and Aastra phones...   Carlos Chavez
• [asterisk-users] PRI Test Lab   Kevin P. Fleming
• [asterisk-users] PRI Test Lab   Kevin P. Fleming
• [asterisk-users] Asterisk on EC2 cloud computing - price assumptions - your brain needed   John Todd
• [asterisk-users] Cisco IP Phone 7940G.   David Gibbons
• [asterisk-users] Asterisk 1.6.0.5 and Aastra phones...   Anthony Messina
• [asterisk-users] Asterisk on EC2 cloud computing - price assumptions - your brain needed   Tzafrir Cohen
• [asterisk-users] Cisco IP Phone 7940G.   Anthony Francis
• [asterisk-users] OpenSky: Digium Skype gateway?   Philipp von Klitzing
• [asterisk-users] OpenSky: Digium Skype gateway?   randulo
• [asterisk-users] PRI Test Lab   Matt Florell
• [asterisk-users] Asterisk on EC2 cloud computing - price assumptions - your brain needed   John Todd
• [asterisk-users] OpenSky: Digium Skype gateway?   John Todd
• [asterisk-users] Cisco IP Phone 7940G.   David Gibbons
• [asterisk-users] PRI Test Lab   Kevin P. Fleming
• [asterisk-users] Asterisk on EC2 cloud computing - price assumptions - your brain needed   Tzafrir Cohen
• [asterisk-users] linksys PAP2t and asterisk   wassim Darwish
• [asterisk-users] PRI Test Lab   Matt Florell
• [asterisk-users] GUI interface to manage business edition   Miguel Martinez
• [asterisk-users] PRI Test Lab   Kevin P. Fleming
• [asterisk-users] Cisco IP Phone 7940G.   Catalin S.
• [asterisk-users] Asterisk 1.6.0.6 Release Candidate 1 Now Available   Asterisk Development Team
• [asterisk-users] Asterisk 1.6.0.5 and Aastra phones...   Carlos Chavez
• [asterisk-users] GUI interface to manage business edition   Bob Pierce
• [asterisk-users] Strange dialplan matching issue   Philipp Kempgen
• [asterisk-users] GUI interface to manage business edition   Kevin P. Fleming
• [asterisk-users] GUI interface to manage business edition   Jared Smith
• [asterisk-users] [OT] Gmail is broken (was: Re: WiFi SIP phone w/VPN?)   Philipp Kempgen
• [asterisk-users] Asterisk 1.6 CDR fields...   Carlos Chavez
• [asterisk-users] Strange dialplan matching issue   OCG Technical Support
• [asterisk-users] Slow hangup - Australia - analog - incoming calls   Paul Hales
• [asterisk-users] Asterisk 1.6.x timing API   Mike
• [asterisk-users] CISCO 2950 -> 4 connections -> Cap of 512 Kbps -> How to bond ?   Frank Bulk
• [asterisk-users] linksys PAP2t and asterisk   wassim Darwish
• [asterisk-users] Call Fowarding and Polycom Phone   Lee, John (Sydney)
• [asterisk-users] Progress() and Proceeding()   Philipp Kempgen
• [asterisk-users] OpenSky: Digium Skype gateway?   Steve Totaro
• [asterisk-users] Progress() and Proceeding()   Mike Trest
• [asterisk-users] Asterisk 1.6.x timing API   Kevin P. Fleming
• [asterisk-users] Asterisk CLI problem if run from /etc/inittab   Jim Boykin
• [asterisk-users] Siemens Hipath PRI to Asterisk Call Routing?   Philipp Kempgen
• [asterisk-users] Continue processing AGI script after hangup   Philipp Kempgen
• [asterisk-users] linksys PAP2t and asterisk   Jose Flores Galicia
• [asterisk-users] Call Fowarding and Polycom Phone   C F
• [asterisk-users] licensed g729   Nhadie
• [asterisk-users] Multiple caller id ...   Massimo Nuvoli
• [asterisk-users] Hangup extensions via CLI?   Tzafrir Cohen
• [asterisk-users] Asterisk CLI problem if run from /etc/inittab   Jim Boykin
• [asterisk-users] Asterisk CLI problem if run from /etc/inittab   Jim Boykin
• [asterisk-users] No such command 'core stop now'   Jim Boykin
• [asterisk-users] No such command 'core stop now'   Michiel van Baak
• [asterisk-users] No such command 'core stop now'   Jim Boykin
• [asterisk-users] Siemens Hipath PRI to Asterisk Call Routing?   Marco Mouta
• [asterisk-users] Asterisk on EC2 cloud computing - price assumptions - your brain needed   Olivier
• [asterisk-users] Asterisk on EC2 cloud computing - priceassumptions - your brain needed   Tom Moore
• [asterisk-users] licensed g729   Michael Graves
• [asterisk-users] Gizmo SIP / Skype gateway   Julian Lyndon-Smith
• [asterisk-users] linksys PAP2t and asterisk   wassim Darwish
• [asterisk-users] Multiple caller id ...   Philipp Kempgen
• [asterisk-users] MINNESOTA: TwinCities Asterisk Users Group Meeting, this Saturday. Valentine's Day Feb/14/2009 11:30am   Darrick Hartman
• [asterisk-users] call-limit   michel freiha
• [asterisk-users] Preferred Clock   Matt Riddell
• [asterisk-users] Broken Pipe error while using UpdateConfig command   Matt Riddell
• [asterisk-users] Autodialler query   Matt Riddell
• [asterisk-users] extensions ending with "#"...   Matt Riddell
• [asterisk-users] Credit Card processing machines   Matt Riddell
• [asterisk-users] VPN and Asterisk   Matt Riddell
• [asterisk-users] Michael Graves post   Matt Riddell
• [asterisk-users] Microsoft Recite   Dean Collins
• [asterisk-users] Hangup extensions via CLI?   Alexander Lopez
• [asterisk-users] Hangup extensions via CLI?   Tzafrir Cohen
• [asterisk-users] Gizmo SIP / Skype gateway   David Quinton
• [asterisk-users] Gizmo SIP / Skype gateway   Julian Lyndon-Smith
• [asterisk-users] Faxing with asterisk   Fabio Mosti
• [asterisk-users] Gizmo SIP / Skype gateway   Geoff Lane
• [asterisk-users] No such command 'core stop now'   Jim Boykin
• [asterisk-users] [OT] Gmail is broken (was: Re: WiFi SIP phone w/VPN?)   Grygoriy Dobrovolskyy
• [asterisk-users] Faxing with asterisk   Grygoriy Dobrovolskyy
• [asterisk-users] Asterisk on EC2 cloud computing - price assumptions - your brain needed   Grygoriy Dobrovolskyy
• [asterisk-users] Faxing with asterisk   Michael
• [asterisk-users] Faxing with asterisk   Michael
• [asterisk-users] Faxing with asterisk   Grygoriy Dobrovolskyy
• [asterisk-users] Faxing with asterisk   Grygoriy Dobrovolskyy
• [asterisk-users] Faxing with asterisk   Steve Underwood
• [asterisk-users] Faxing with asterisk   Steve Underwood
• [asterisk-users] SpanDSP question for Steve   Michael
• [asterisk-users] Faxing with asterisk   Fabio Mosti
• [asterisk-users] Faxing with asterisk   Doug Lytle
• [asterisk-users] OpenSky: Digium Skype gateway?   Olivier
• [asterisk-users] Faxing with asterisk   Rob Hillis
• [asterisk-users] Asterisk on EC2 cloud computing - price assumptions - your brain needed   SIP
• [asterisk-users] Faxing with asterisk   Marc STORCK
• [asterisk-users] [OT] Gmail is broken (was: Re: WiFi SIP phone w/VPN?)   Jerry Jones
• [asterisk-users] Michael Graves post   Michael Graves
• [asterisk-users] TeleKaam - Voice Portal for Students and Parents   Kashif Naeem
• [asterisk-users] Asterisk on EC2 cloud computing - price assumptions - your brain needed   Grygoriy Dobrovolskyy
• [asterisk-users] AstDB wildard searches   Geoff Lane
• [asterisk-users] AstDB wildard searches   Doug Lytle
• [asterisk-users] AstDB wildard searches   Jared Smith
• [asterisk-users] AstDB wildard searches   Geoff Lane
• [asterisk-users] command show channels concise   Jerry Geis
• [asterisk-users] Asterisk on EC2 cloud computing - price assumptions - your brain needed   Eric Chamberlain
• [asterisk-users] AstDB wildard searches   Jared Smith
• [asterisk-users] AstDB wildard searches   Geoff Lane
• [asterisk-users] command show channels concise   Mark Michelson
• [asterisk-users] Please help test the gender detection module at 575-613-4392   Asterisk Asterisk
• [asterisk-users] DTMF not completely muted   Michael Smith
• [asterisk-users] vmsecret question   Klaus Darilion
• [asterisk-users] DTMF not completely muted   Kevin P. Fleming
• [asterisk-users] Please help test the gender detection module at 575-613-4392   Asterisk Asterisk
• [asterisk-users] DTMF not completely muted   Michael Smith
• [asterisk-users] Michael Graves post   Matt Riddell
• [asterisk-users] [OT] Gmail is broken   Matt Riddell
• [asterisk-users] How to beep before transfer ...   Robert Augustyn
• [asterisk-users] WiFi SIP phone w/VPN?   Jay Milk
• [asterisk-users] DTMF not completely muted   Wilton Helm
• [asterisk-users] OpenSky: Digium Skype gateway?   Alejandro Lengua
• [asterisk-users] DTMF not completely muted   Michael Smith
• [asterisk-users] mp3player() to shuffle playlist   reza adinata
• [asterisk-users] Obtaining callerid on a PRI for billing purposes (with non toll-free numbers)   Alfred Monticello
• [asterisk-users] Obtaining callerid on a PRI for billing purposes (with non toll-free numbers)   Paul Hales
• [asterisk-users] Optimizing this script for calling Skype users from Asterisk   Michael Robertson
• [asterisk-users] Please help test the gender detection module at 575-613-4392   Gondar Monn
• [asterisk-users] TeleKaam - Voice Portal for Students and Parents   Kashif Naeem
• [asterisk-users] Credit Card processing machines   Andrew Joakimsen
• [asterisk-users] Stress Testing IVR   Rajkumar S
• [asterisk-users] unistim channel problem   Ralf Träskman
• [asterisk-users] zaptel compile kernel problem   reza adinata
• [asterisk-users] zaptel compile kernel problem   Dave Cotton
• [asterisk-users] zaptel compile kernel problem   reza adinata
• [asterisk-users] freemin managment for sim cards   Pezhman Lali
• [asterisk-users] zaptel compile kernel problem   Tzafrir Cohen
• [asterisk-users] zaptel compile kernel problem   reza adinata
• [asterisk-users] zaptel compile kernel problem   bails
• [asterisk-users] zaptel compile kernel problem   Tzafrir Cohen
• [asterisk-users] zaptel compile kernel problem   Tzafrir Cohen
• [asterisk-users] zaptel compile kernel problem   reza adinata
• [asterisk-users] zaptel compile kernel problem   Gordon Henderson
• [asterisk-users] zaptel compile kernel problem   Tzafrir Cohen
• [asterisk-users] zaptel compile kernel problem   Tzafrir Cohen
• [asterisk-users] What is the purpose of membermacro in queues.conf   Rajkumar S
• [asterisk-users] Pingable and Unreachable at the same time !   Olivier
• [asterisk-users] Pingable and Unreachable at the same time !   Marc STORCK
• [asterisk-users] Credit Card processing machines   Jeff LaCoursiere
• [asterisk-users] Pingable and Unreachable at the same time !   Olivier
• [asterisk-users] Pingable and Unreachable at the same time !   OCG Technical Support
• [asterisk-users] Packet Truncated - Choppy Audio   Matt King
• [asterisk-users] Hangup extensions via CLI?   Danny Nicholas
• [asterisk-users] Packet Truncated - Choppy Audio   Danny Nicholas
• [asterisk-users] Network architecture   michel freiha
• [asterisk-users] Network architecture   Jeff LaCoursiere
• [asterisk-users] Network architecture   Alex Balashov
• [asterisk-users] Network architecture   Danny Nicholas
• [asterisk-users] Network architecture   Grygoriy Dobrovolskyy
• [asterisk-users] Hangup extensions via CLI?   Tzafrir Cohen
• [asterisk-users] What is the purpose of membermacro in queues.conf   Mark Michelson
• [asterisk-users] Message   admin at viagra.com
• [asterisk-users] Obtaining callerid on a PRI for billing purposes (with non toll-free numbers)   C F
• [asterisk-users] DTMF not completely muted   Michael Smith
• [asterisk-users] Questions about OpenSky - Asterisk to Skype Gateway   Michael Robertson
• [asterisk-users] Asterisk 1.4.21.1 intermittent presence working with Polycom   JR Richardson
• [asterisk-users] Swift - detection of multiple digits unreliable on my system   Bob Hartwig
• [asterisk-users] Please help test the gender detection module at 575-613-4392   Asterisk Asterisk
• [asterisk-users] Credit Card processing machines   Don E. Wisdom
• [asterisk-users] Question regarding custom announcements in queues.conf   Christopher Aloi
• [asterisk-users] Stress Testing IVR   Philipp Kempgen
• [asterisk-users] Lost with Patton 5.3 web server. Registration ?   Olivier
• [asterisk-users] Question regarding custom announcements in queues.conf   Mark Michelson
• [asterisk-users] Question regarding custom announcements in queues.conf   Mark Michelson
• [asterisk-users] Asterisk supports SIP-T?   Daviramos Roussenq Fortunato
• [asterisk-users] Packet Truncated - Choppy Audio   Matt King
• [asterisk-users] Question regarding custom announcements in queues.conf   Christopher Aloi
• [asterisk-users] Network architecture   Asterisk Asterisk
• [asterisk-users] Credit Card processing machines   Jeff LaCoursiere
• [asterisk-users] Question regarding custom announcements in queues.conf   Christopher Aloi
• [asterisk-users] Question regarding custom announcements in queues.conf   Mark Michelson
• [asterisk-users] Credit Card processing machines   David Gibbons
• [asterisk-users] Question regarding custom announcements inqueues.conf   Danny Nicholas
• [asterisk-users] Please help test the gender detection module at 575-613-4392   Asterisk Asterisk
• [asterisk-users] Please help test the gender detection moduleat 575-613-4392   Jason Aarons (US)
• [asterisk-users] Question regarding custom announcements in queues.conf   Mark Michelson
• [asterisk-users] Credit Card processing machines   Jon Pounder
• [asterisk-users] Credit Card processing machines   Jeff LaCoursiere
• [asterisk-users] Credit Card processing machines   David Gibbons
• [asterisk-users] Network architecture   Wilton Helm
• [asterisk-users] Credit Card processing machines   Jerry Jones
• [asterisk-users] call file bug?   Ray Chen
• [asterisk-users] Credit Card processing machines   Jeff LaCoursiere
• [asterisk-users] Credit Card processing machines   Andrew Joakimsen
• [asterisk-users] Credit Card processing machines   Jonn Taylor
• [asterisk-users] Credit Card processing machines   Jeff LaCoursiere
• [asterisk-users] Credit Card processing machines   Jeff LaCoursiere
• [asterisk-users] call file bug?   Danny Nicholas
• [asterisk-users] Please help test the gender detection moduleat 575-613-4392   Asterisk Asterisk
• [asterisk-users] only ring phones that are not on a call   Jon Weisman
• [asterisk-users] Credit Card processing machines   Don E. Wisdom
• [asterisk-users] Updated modules to be released (FaxDetect, GenderDetect, MachineDetect, others)   Asterisk Asterisk
• [asterisk-users] Credit Card processing machines   Jon Pounder
• [asterisk-users] life safety system and VOIP   Jason Aarons (US)
• [asterisk-users] life safety system and VOIP   Jeff LaCoursiere
• [asterisk-users] life safety system and VOIP   Jon Pounder
• [asterisk-users] Caller Hangup detection   Mindaugas Kezys
• [asterisk-users] only ring phones that are not on a call   Gordon Henderson
• [asterisk-users] life safety system and VOIP   Jon Pounder
• [asterisk-users] life safety system and VOIP   Jeff LaCoursiere
• [asterisk-users] life safety system and VOIP   Don E. Wisdom
• [asterisk-users] only ring phones that are not on a call   Jon Weisman
• [asterisk-users] life safety system and VOIP   Jeff LaCoursiere
• [asterisk-users] life safety system and VOIP   Jon Pounder
• [asterisk-users] life safety system and VOIP   Jon Pounder
• [asterisk-users] only ring phones that are not on a call   Danny Nicholas
• [asterisk-users] ztdummy compile under 2.6.28 ?   Gordon Henderson
• [asterisk-users] SLA and Flashing BLF   Muiz Motani
• [asterisk-users] life safety system and VOIP   Jonn Taylor
• [asterisk-users] ztdummy compile under 2.6.28 ?   David Backeberg
• [asterisk-users] ztdummy compile under 2.6.28 ?   Shaun Ruffell
• [asterisk-users] Stress Testing IVR   David Backeberg
• [asterisk-users] life safety system and VOIP   Jason Aarons (US)
• [asterisk-users] ztdummy compile under 2.6.28 ?   Gordon Henderson
• [asterisk-users] life safety system and VOIP   John Novack
• [asterisk-users] SLA and Flashing BLF   Kevin P. Fleming
• [asterisk-users] life safety system and VOIP   Jonn Taylor
• [asterisk-users] life safety system and VOIP   Wilton Helm
• [asterisk-users] Asterisk supports SIP-T?   Raj Jain
• [asterisk-users] Open Source in an Economic Downturn: Asterisk stories needed   John Todd
• [asterisk-users] Open Source in an Economic Downturn: Asterisk stories needed   Michael
• [asterisk-users] Open Source in an Economic Downturn: Asterisk stories needed   Jim DeVito
• [asterisk-users] AgentCallbackLogin via dialaplan and device state   Jamin W. Collins
• [asterisk-users] life safety system and VOIP   Dave Platt
• [asterisk-users] call file bug?   Ex Vito
• [asterisk-users] xorcom hardware good idea?   David fire
• [asterisk-users] xorcom hardware good idea?   Ming Yong
• [asterisk-users] Open Source in an Economic Downturn: Asterisk stories needed   Gordon Henderson
• [asterisk-users] xorcom hardware good idea?   David fire
• [asterisk-users] Open Source in an Economic Downturn: Asterisk stories needed   Michael
• [asterisk-users] xorcom hardware good idea?   Ming Yong
• [asterisk-users] Stress Testing IVR   Rajkumar S
• [asterisk-users] Distributed presence in 1.6   Rajkumar S
• [asterisk-users] Credit Card processing machines   bilal ghayyad
• [asterisk-users] Network architecture   michel freiha
• [asterisk-users] Network architecture   michel freiha
• [asterisk-users] Setting SIP header on agent calls made by a queue   Lenz Emilitri
• [asterisk-users] Credit Card processing machines   Tim Panton
• [asterisk-users] Credit Card processing machines   Tzafrir Cohen
• [asterisk-users] AGI pdf book   michel freiha
• [asterisk-users] Asterisk supports SIP-T?   Daviramos Roussenq Fortunato
• [asterisk-users] Asterisk supports SIP-T?   Raj Jain
• [asterisk-users] AGI pdf book   David fire
• [asterisk-users] ztdummy compile under 2.6.28 ?   Gordon Henderson
• [asterisk-users] AGI pdf book   michel freiha
• [asterisk-users] Detecting which party initiates a hangup   Darren Murphy
• [asterisk-users] Ditech API   Dean Collins
• [asterisk-users] Accumulated call time   Geoff Lane
• [asterisk-users] What is the purpose of membermacro in queues.conf   Rajkumar S
• [asterisk-users] Network architecture   Benny Amorsen
• [asterisk-users] Credit Card processing machines   Jeff LaCoursiere
• [asterisk-users] US DID   Nhadie
• [asterisk-users] Network architecture   Steve Totaro
• [asterisk-users] asterisk-users Digest, Vol 55, Issue 52   bilal ghayyad
• [asterisk-users] Accumulated call time   David fire
• [asterisk-users] AGI pdf book   Danny Nicholas
• [asterisk-users] Setting SIP header on agent calls made by a queue   Danny Nicholas
• [asterisk-users] only ring phones that are not on a call   Benny Amorsen
• [asterisk-users] Setting SIP header on agent calls made by a queue   Benny Amorsen
• [asterisk-users] Need help on Forwarding   Max Alex
• [asterisk-users] Pharmacy Message 58005   asterisk-users at lists.digium.com
• [asterisk-users] US DID   Paul Chambers
• [asterisk-users] directrtpsetup=yes does not work in 1.4.23.1   Arturo Díaz Almagro
• [asterisk-users] Credit Card processing machines   Edwin Quijada
• [asterisk-users] Setting SIP header on agent calls made by a queue   Lenz Emilitri
• [asterisk-users] Setting SIP header on agent calls made by a queue   Lenz Emilitri
• [asterisk-users] US DID   Fred Posner
• [asterisk-users] asterisk-users Digest, Vol 55, Issue 52   Bill Michaelson
• [asterisk-users] Accumulated call time   Geoff Lane
• [asterisk-users] connection to siemens hipath   Jerry Geis
• [asterisk-users] Setting SIP header on agent calls made by aqueue   Danny Nicholas
• [asterisk-users] AGI pdf book   Jared Smith
• [asterisk-users] Network architecture   Jared Smith
• [asterisk-users] Open Source in an Economic Downturn: Asterisk stories needed   John Novack
• [asterisk-users] licensed g729   Thomas Kenyon
• [asterisk-users] Credit Card processing machines   bilal ghayyad
• [asterisk-users] Please help test the gender detection moduleat 575-613-4392   Asterisk Asterisk
• [asterisk-users] OT: Re: Credit Card processing machines   Richard Lyman
• [asterisk-users] Credit Card processing machines   Kinjal Dixit
• [asterisk-users] trunk to trunk   Leonja Cerebro
• [asterisk-users] Please help test the gender detection module at 575-613-4392   Steve Totaro
• [asterisk-users] OT: Re: Credit Card processing machines   Jeff LaCoursiere
• [asterisk-users] trunk to trunk   Robert Broyles
• [asterisk-users] OT: Re: Credit Card processing machines   Jon Pounder
• [asterisk-users] Please help test the gender detection module at 575-613-4392   Steve Totaro
• [asterisk-users] No Audio PlayBack Asterisk 1.6 Dahdi 2.1.0.3   Daviramos Roussenq Fortunato
• [asterisk-users] Credit Card processing machines   Andrew Joakimsen
• [asterisk-users] Open Source in an Economic Downturn: Asterisk stories needed   Chris Bagnall
• [asterisk-users] Open Source in an Economic Downturn: Asterisk stories needed   Michael
• [asterisk-users] benchg729 - no valid g729 license   Adam Robins
• [asterisk-users] Open Source in an Economic Downturn: Asterisk stories   JR Richardson
• [asterisk-users] call file FXO channel problem   Ray Chen
• [asterisk-users] benchg729 - no valid g729 license   Kevin P. Fleming
• [asterisk-users] Asterisk on the Cloud With a Click - pre-built Asterisk Amazon EC2 instance   Eric Chamberlain
• [asterisk-users] benchg729 - no valid g729 license   Adam Robins
• [asterisk-users] Asterisk on the Cloud With a Click - pre-built Asterisk Amazon EC2 instance   John Todd
• [asterisk-users] Understand SIP REFER   Klaus Darilion
• [asterisk-users] call file FXO channel problem   Eric Wieling, Asteria Solutions Group
• [asterisk-users] Please help test the gender detection module at 575-613-4392   Asterisk Asterisk
• [asterisk-users] Please help test the gender detection module at 575-613-4392   Jeff LaCoursiere
• [asterisk-users] Please help test the gender detection module at 575-613-4392   Darren Wiebe
• [asterisk-users] Please help test the gender detection moduleat 575-613-4392   Asterisk Asterisk
• [asterisk-users] AGI pdf book   Steve Edwards
• [asterisk-users] AGI pdf book   Steve Edwards
• [asterisk-users] AGI pdf book   Emmanuel Bruno
• [asterisk-users] AGI pdf book   Emmanuel Bruno
• [asterisk-users] AGI pdf book   Jose P. Espinal
• [asterisk-users] AGI pdf book   Michael
• [asterisk-users] AGI pdf book   Roderick A. Anderson
• [asterisk-users] SLA and Flashing BLF   Muiz Motani
• [asterisk-users] AGI pdf book   Michael
• [asterisk-users] DeadAgi Application in asterisk 1.6   Max Alex
• [asterisk-users] AOC-E pass through   Jean-Denis Girard
• [asterisk-users] Managing SIP hardphones call history   Olivier
• [asterisk-users] AOC-E pass through   Olivier
• [asterisk-users] Asterisk supports SIP-T?   Johansson Olle E
• [asterisk-users] Understand SIP REFER   Johansson Olle E
• [asterisk-users] Managing SIP hardphones call history   Johansson Olle E
• [asterisk-users] AGI pdf book   Hans Witvliet
• [asterisk-users] Friday Feb 20th 12 Noon EST: Jason Fischl from Counterpath on VUC   randulo
• [asterisk-users] AGI pdf book   Michael
• [asterisk-users] AGI pdf book   Grygoriy Dobrovolskyy
• [asterisk-users] Please help test the gender detection moduleat 575-613-4392   Grygoriy Dobrovolskyy
• [asterisk-users] Please help test the gender detection module at 575-613-4392   Thomas Kenyon
• [asterisk-users] Managing SIP hardphones call history   Olivier
• [asterisk-users] Network architecture   Grygoriy Dobrovolskyy
• [asterisk-users] sip phone cant hear the caller   Ralf Träskman
• [asterisk-users] About Hint Configuration in Asterisk.   Sunil Teli
• [asterisk-users] Asterisk BLF to Cisco CME   Enrico Pasqualotto
• [asterisk-users] AGI pdf book   SIP
• [asterisk-users] Not answering call when queue is full or has no members   Alex Hermann
• [asterisk-users] Understand SIP REFER   Klaus Darilion
• [asterisk-users] Busy status of a snom connected to two asterisk servers?   Rajkumar S
• [asterisk-users] AGI pdf book   Michael
• [asterisk-users] check if not human   Marcus Kvarsell
• [asterisk-users] Not answering call when queue is full or has no members   Lenz Emilitri
• [asterisk-users] Busy status of a snom connected to two asterisk servers?   Lenz Emilitri
• [asterisk-users] queue_variables() function   Philipp Kempgen
• [asterisk-users] Busy status of a snom connected to two asterisk servers?   Philipp Kempgen
• [asterisk-users] call picking and transfers   Jeff LaCoursiere
• [asterisk-users] Not answering call when queue is full or has no members   Philipp Kempgen
• [asterisk-users] Busy status of a snom connected to two asterisk servers?   Rajkumar S
• [asterisk-users] queue_variables() function   Jared Smith
• [asterisk-users] [PATCH] Auto answer channel after successfull queue entry (was Re: Not answering call when queue is full or has no members)   Alex Hermann
• [asterisk-users] Busy status of a snom connected to two asterisk servers?   Philipp Kempgen
• [asterisk-users] DeadAgi Application in asterisk 1.6   Philipp Kempgen
• [asterisk-users] check if not human   David fire
• [asterisk-users] check if not human   Jon Pounder
• [asterisk-users] AGI pdf book   Wilton Helm
• [asterisk-users] AOC-E pass through   Jean-Denis Girard
• [asterisk-users] check if not human   Miguel Molina
• [asterisk-users] Distributed presence in 1.6   Matt Riddell
• [asterisk-users] Need help on Forwarding   Philipp Kempgen
• [asterisk-users] check if not human   David Budny
• [asterisk-users] ztdummy compile under 2.6.28 ?   Philipp Kempgen
• [asterisk-users] What's the difference between the Jabber Client Mode And Component Mode?   Philippe Sultan
• [asterisk-users] Managing SIP hardphones call history   Eric Chamberlain
• [asterisk-users] Credit Card processing machines   Edwin Quijada
• [asterisk-users] Credit Card processing machines   bilal ghayyad
• [asterisk-users] Credit Card processing machines   Steve Edwards
• [asterisk-users] Annoying silence suppression effect on my digium E1 card with the VPMADT032 module   Loic Didelot
• [asterisk-users] Credit Card processing machines   Edwin Quijada
• [asterisk-users] Managing SIP hardphones call history   David Gibbons
• [asterisk-users] DTMF   David at ULC
• [asterisk-users] DTMF   Tim Nelson
• [asterisk-users] check if not human   Edwin Quijada
• [asterisk-users] DTMF   Steve Howes
• [asterisk-users] Annoying silence suppression effect on my digium E1 card with the VPMADT032 module   Kevin P. Fleming
• [asterisk-users] DTMF   David at ULC
• [asterisk-users] US DID   Asterisk Asterisk
• [asterisk-users] TDMOE Timing   Dave Fullerton
• [asterisk-users] AGI pdf book   Michael
• [asterisk-users] check if not human   Asterisk Asterisk
• [asterisk-users] call picking and transfers   Asterisk Asterisk
• [asterisk-users] directrtpsetup=yes does not work in 1.4.23.1   Jared Geiger
• [asterisk-users] Busy status of a snom connected to two asterisk servers?   Asterisk Asterisk
• [asterisk-users] Please help test the gender detection module at 575-613-4392   Asterisk Asterisk
• [asterisk-users] Busy status of a snom connected to two asterisk servers?   Philipp von Klitzing
• [asterisk-users] check if not human   David fire
• [asterisk-users] AGI script   michel freiha
• [asterisk-users] AGI script   Luis Morales
• [asterisk-users] AGI script   michel freiha
• [asterisk-users] AGI script   Steve Edwards
• [asterisk-users] Please help test the gender detection module at 575-613-4392   Steve Totaro
• [asterisk-users] AGI pdf book   Steve Totaro
• [asterisk-users] AGI pdf book   Michael
• [asterisk-users] TDMOE Timing   Steve Totaro
• [asterisk-users] AGI pdf book   Jeff LaCoursiere
• [asterisk-users] AGI pdf book   Wilton Helm
• [asterisk-users] AGI pdf book   Michael
• [asterisk-users] AGI pdf book   Steve Totaro
• [asterisk-users] AGI pdf book   Mike Diehl
• [asterisk-users] AGI pdf book   Steve Totaro
• [asterisk-users] AGI pdf book   Michael
• [asterisk-users] AGI pdf book   John Novack
• [asterisk-users] Please help test the gender detection module at 575-613-4392   Ira
• [asterisk-users] AGI pdf book   Michael
• [asterisk-users] AGI pdf book   Jeff LaCoursiere
• [asterisk-users] AGI pdf book   Emmanuel Bruno
• [asterisk-users] AGI pdf book   Emmanuel Bruno
• [asterisk-users] AGI pdf book   Michael
• [asterisk-users] US DID   Mik Cheez
• [asterisk-users] AGI script   Yawar Hadi
• [asterisk-users] AGI script   David fire
• [asterisk-users] SIP Proxy behind NAT talkinf to ASterisk with public IP   Jay Ray
• [asterisk-users] Busy status of a snom connected to two asterisk servers?   Rajkumar S
• [asterisk-users] Setting SIP header on agent calls made by a queue   Klaus Darilion
• [asterisk-users] SIP Proxy behind NAT talkinf to ASterisk with public IP   Stefan Schmidt
• [asterisk-users] AGI script   michel freiha
• [asterisk-users] DTMF   David at ULC
• [asterisk-users] Qualify sip users behind remote registrar   Philippe Sultan
• [asterisk-users] Setting SIP header on agent calls made by a queue   Lenz Emilitri
• [asterisk-users] check if not human   Lenz Emilitri
• [asterisk-users] Setting SIP header on agent calls made by a queue   Benny Amorsen
• [asterisk-users] SMS to generate a call back   Supa
• [asterisk-users] AGI script   Yawar Hadi
• [asterisk-users] check if not human   David fire
• [asterisk-users] SpanDSP question for Steve   Steve Underwood
• [asterisk-users] SMS Integration and SMS commands   Supa
• [asterisk-users] AGI pdf book   Andrew Thomas
• [asterisk-users] benchg729 - no valid g729 license   Leonardo Gomes Figueira
• [asterisk-users] real time connection failed   michel freiha
• [asterisk-users] real time connection failed   Philipp Kempgen
• [asterisk-users] Polycom Phones start to break up after being up a LONG time   Barry D. Hassler
• [asterisk-users] slip r errors nortel switch   Jerry Geis
• [asterisk-users] real time connection failed   Miguel Molina
• [asterisk-users] AGI script   Steve Edwards
• [asterisk-users] Polycom Phones start to break up after being up a LONG time   Asterisk Asterisk
• [asterisk-users] Table definitions for Realtime 1.6...   Carlos Chavez
• [asterisk-users] Please help test the gender detection module at 575-613-4392   Asterisk Asterisk
• [asterisk-users] check if not human   Asterisk Asterisk
• [asterisk-users] Please help test the gender detection module at 575-613-4392   Asterisk Asterisk
• [asterisk-users] zaptel telephone cards and asterisk in another pc   Ignacio
• [asterisk-users] Please help test the gender detection module at 575-613-4392   Asterisk Asterisk
• [asterisk-users] Polycom Phones start to break up after being up a LONG time   Jeff LaCoursiere
• [asterisk-users] zaptel telephone cards and asterisk in another pc   Eric Wieling, Asteria Solutions Group
• [asterisk-users] Please help test the gender detection module at 575-613-4392   Jeff LaCoursiere
• [asterisk-users] Polycom Phones start to break up after beingup a LONG time   Danny Nicholas
• [asterisk-users] Please help test the gender detection module at 575-613-4392   Tiago Durante
• [asterisk-users] Polycom Phones start to break up after beingupa LONG time   Alexander Lopez
• [asterisk-users] zaptel telephone cards and asterisk in another pc   Ignacio
• [asterisk-users] zaptel telephone cards and asterisk in another pc   Tim Nelson
• [asterisk-users] zaptel telephone cards and asterisk in another pc   Jeff LaCoursiere
• [asterisk-users] Polycom Phones start to break up after beingup a LONG time   Jeff LaCoursiere
• [asterisk-users] zaptel telephone cards and asterisk in another pc   Ignacio
• [asterisk-users] Polycom Phones start to break up after beingup a LONG time   Dave Fullerton
• [asterisk-users] Please help test the gender detection module at 575-613-4392   Tzafrir Cohen
• [asterisk-users] zaptel telephone cards and asterisk in another pc   Jeff LaCoursiere
• [asterisk-users] zaptel telephone cards and asterisk in another pc   Tzafrir Cohen
• [asterisk-users] zaptel telephone cards and asterisk in another pc   Matt Watson
• [asterisk-users] Please help test the gender detection module at 575-613-4392   Asterisk Asterisk
• [asterisk-users] zaptel telephone cards and asterisk in another pc   Ignacio
• [asterisk-users] Polycom Phones start to break up after beingup a LONG time   Danny Nicholas
• [asterisk-users] Please help test the gender detection module at 575-613-4392   Asterisk Asterisk
• [asterisk-users] SIP debug messages   Jeff LaCoursiere
• [asterisk-users] AGI script   michel freiha
• [asterisk-users] AGI script   Danny Nicholas
• [asterisk-users] asterisk-users Digest, Vol 55, Issue 64   Mark
• [asterisk-users] AGI script   Steve Edwards
• [asterisk-users] AGI script   Danny Nicholas
• [asterisk-users] AGI script   michel freiha
• [asterisk-users] AGI script   michel freiha
• [asterisk-users] AGI script   michel freiha
• [asterisk-users] AGI script   Steve Edwards
• [asterisk-users] AGI script   Steve Edwards
• [asterisk-users] zaptel telephone cards and asterisk in another pc   Jonn Taylor
• [asterisk-users] AGI script   Danny Nicholas
• [asterisk-users] AGI script   Steve Edwards
• [asterisk-users] AGI script   michel freiha
• [asterisk-users] AGI script   Danny Nicholas
• [asterisk-users] AGI script   michel freiha
• [asterisk-users] AGI script   Tzafrir Cohen
• [asterisk-users] CDR fields in 1.6   Carlos Chavez
• [asterisk-users] AGI script   Steve Edwards
• [asterisk-users] AGI script   Kai-Uwe Jensen
• [asterisk-users] AGI script   Tilghman Lesher
• [asterisk-users] AGI script   michel freiha
• [asterisk-users] CDR fields in 1.6   Tilghman Lesher
• [asterisk-users] AGI script   michel freiha
• [asterisk-users] AGI script   michel freiha
• [asterisk-users] hi   Vinicius Neves
• [asterisk-users] zaptel telephone cards and asterisk in another pc   Paul Hales
• [asterisk-users] zaptel telephone cards and asterisk in another pc   Paul Hales
• [asterisk-users] Vacation reply   vinicius.neves at live.com
• [asterisk-users] AGI script   Steve Edwards
• [asterisk-users] Vacation reply   RE Kushner List Account
• [asterisk-users] zaptel telephone cards and asterisk in another pc   Tom Lynn
• [asterisk-users] IAX2 - now known as RFC 5456   John Todd
• [asterisk-users] IAX2 - now known as RFC 5456   Gondar Monn
• [asterisk-users] Please help test the gender detection module at 575-613-4392   Steve Underwood
• [asterisk-users] help needed -- chanspy   Deepak
• [asterisk-users] Please help test the gender detection module at 575-613-4392   Darren Wiebe
• [asterisk-users] AGI script   Yawar Hadi
• [asterisk-users] AGI script   Yawar Hadi
• [asterisk-users] AGI script   Darren Murphy
• [asterisk-users] AGI script   Yawar Hadi
• [asterisk-users] Please help test the gender detection module at 575-613-4392   Tzafrir Cohen
• [asterisk-users] AGI script   michel freiha
• [asterisk-users] Cisco Phone losse regsitrations with Asterisk   amit salunkhe
• [asterisk-users] DIAL() application 'g' option   Geoff Lane
• [asterisk-users] Where to find db1_dump185 in debian packages ? [SOLVED]   Olivier
• [asterisk-users] AGI script   Luis Morales
• [asterisk-users] VoIP Information in CDRs   Costa Tsaousis
• [asterisk-users] Business Center Call Forward No Ans   Si Tai Fan
• [asterisk-users] Please help test the gender detection module at 575-613-4392   Asterisk Asterisk
• [asterisk-users] Please help test the gender detection module at 575-613-4392   Asterisk Asterisk
• [asterisk-users] Please help test the gender detection module at 575-613-4392   Asterisk Asterisk
• [asterisk-users] Please help test the gender detection module at 575-613-4392   Tzafrir Cohen
• [asterisk-users] DIAL() application 'g' option   michel freiha
• [asterisk-users] Please help test the gender detection module at 575-613-4392   Justin Newman
• [asterisk-users] Dial() application 'g' option   Philipp Kempgen
• [asterisk-users] Business Center Call Forward No Ans   Philipp Kempgen
• [asterisk-users] Business Center Call Forward No Ans   Don Kelly
• [asterisk-users] Dial() application 'g' option   Geoff Lane
• [asterisk-users] Dial() application 'g' option   Mindaugas Kezys
• [asterisk-users] VoIP Information in CDRs   Mindaugas Kezys
• [asterisk-users] Dial() application 'g' option   Geoff Lane
• [asterisk-users] Dial() application 'g' option   Jim Dickenson
• [asterisk-users] Network architecture   michel freiha
• [asterisk-users] Network architecture   Steve Edwards
• [asterisk-users] Intel Vs AMD   michel freiha
• [asterisk-users] Intel Vs AMD   Gordon Henderson
• [asterisk-users] I can`t send DTMFs through FXO lines - dahdi   Daniel - Asterisk
• [asterisk-users] Intel Vs AMD   Craig Van Ham
• [asterisk-users] Intel Vs AMD   David fire
• [asterisk-users] Intel Vs AMD   Wilton Helm
• [asterisk-users] Intel Vs AMD   Doug
• [asterisk-users] Intel Vs AMD   Darrick Hartman
• [asterisk-users] Asterisk 1.6.0.5 as non root and moh perm issue   Joseph L. Casale
• [asterisk-users] Business Center Call Forward No Ans   Si Tai Fan
• [asterisk-users] AGI script   Yawar Hadi
• [asterisk-users] AGI script   Steve Edwards
• [asterisk-users] AGI script   michel freiha
• [asterisk-users] AGI script   Yawar Hadi
• [asterisk-users] AGI script   Yawar Hadi
• [asterisk-users] Can't compile Dahdi on SuSE 10.1/Kernel 2.6.16.54-0.2.5-smp/GCC 4.1.2   Bruce Ferrell
• [asterisk-users] AGI script   Yawar Hadi
• [asterisk-users] Intel Vs AMD   Gordon Henderson
• [asterisk-users] Intel Vs AMD   Tzafrir Cohen
• [asterisk-users] AGI script   michel freiha
• [asterisk-users] receive fax problem   fateme fatah
• [asterisk-users] Intel Vs AMD   Gordon Henderson
• [asterisk-users] AGI script   Yawar Hadi
• [asterisk-users] Static extensions.conf|ael, best practice, hints (was: Re: What do you use? .conf or AEL?)   Philipp Kempgen
• [asterisk-users] AGI script   michel freiha
• [asterisk-users] Flash Operator Panel with Asterisk 1.6   Pablo Bernasconi
• [asterisk-users] Flash Operator Panel with Asterisk 1.6   Doug Lytle
• [asterisk-users] app_queue New Function ToggleQueueMemberUse()   Rajkumar S
• [asterisk-users] Asterisk 1.6.0.5 as non root and moh perm issue   Danny Nicholas
• [asterisk-users] Asterisk 1.6.0.5 as non root and moh perm issue   Joseph L. Casale
• [asterisk-users] Asterisk/Skype update   Dean Collins
• [asterisk-users] problem with nortel 2002 disconecting   Ralf Träskman
• [asterisk-users] Asterisk 1.6.0.5 as non root and moh perm issue   Danny Nicholas
• [asterisk-users] Asterisk 1.6.0.5 as non root and moh perm issue   David Backeberg
• [asterisk-users] Asterisk 1.6.0.5 as non root and moh perm issue   Philipp Kempgen
• [asterisk-users] 302 Redirect Handling   Arturo Díaz Almagro
• [asterisk-users] Please help test the gender detection module at 575-613-4392   Timothy Legge
• [asterisk-users] AGI script   Steve Edwards
• [asterisk-users] receive fax problem   David Backeberg
• [asterisk-users] AGI script   Steve Edwards
• [asterisk-users] Astlinux 0.6.3 Released   Darrick Hartman
• [asterisk-users] benchg729 - no valid g729 license   Kevin P. Fleming
• [asterisk-users] Static extensions.conf|ael, best practice, hints (was: Re: What do you use? .conf or AEL?)   Tilghman Lesher
• [asterisk-users] Delete all   David at ULC
• [asterisk-users] Delete all   David at ULC
• [asterisk-users] Delete all   Michael Iedema
• [asterisk-users] Delete all   Tzafrir Cohen
• [asterisk-users] Delete all   Steve Edwards
• [asterisk-users] Delete all   Matt Gibson
• [asterisk-users] [asterisk-dev] IAX2 - now known as RFC 5456   Kevin P. Fleming
• [asterisk-users] Delete all   Tzafrir Cohen
• [asterisk-users] FreePBX in a Cloud With a Click - FreePBX secured and optimized for Amazon EC2   Eric Chamberlain
• [asterisk-users] Asterisk 1.6.0.5 as non root and moh perm issue   Joseph L. Casale
• [asterisk-users] Delete all   David fire
• [asterisk-users] 5ess   Jerry Geis
• [asterisk-users] 5ess   Don Kelly
• [asterisk-users] check if not human   Edwin Quijada
• [asterisk-users] Voicemail and ADSI   Jeff LaCoursiere
• [asterisk-users] Delete all   Tzafrir Cohen
• [asterisk-users] Voicemail and ADSI   Eric Wieling, Asteria Solutions Group
• [asterisk-users] Voicemail and ADSI   Don Kelly
• [asterisk-users] Asterisk 1.6.0.6 Now Available!   Asterisk Development Team
• [asterisk-users] Compiling asterisk-addons-1.6.0 under Debian 2.6.18?   Bruce Komito
• [asterisk-users] Compiling asterisk-addons-1.6.0 under Debian 2.6.18?   Tzafrir Cohen
• [asterisk-users] don't get 2.0 gui to run on asterisk 1.6.0.5   Tamer Higazi
• [asterisk-users] Compiling asterisk-addons-1.6.0 under Debian 2.6.18?   Bruce Komito
• [asterisk-users] don't get 2.0 gui to run on asterisk 1.6.0.5   Danny Nicholas
• [asterisk-users] Inbound call to IVR drops after 21 seconds?   Doug
• [asterisk-users] don't get 2.0 gui to run on asterisk 1.6.0.5   Tamer Higazi
• [asterisk-users] 5ess   Jerry Geis
• [asterisk-users] IAX2 - now known as RFC 5456   Matt Florell
• [asterisk-users] don't get 2.0 gui to run on asterisk 1.6.0.5   Tzafrir Cohen
• [asterisk-users] don't get 2.0 gui to run on asterisk 1.6.0.5   Danny Nicholas
• [asterisk-users] don't get 2.0 gui to run on asterisk 1.6.0.5   Bob Pierce
• [asterisk-users] don't get 2.0 gui to run on asterisk 1.6.0.5   Danny Nicholas
• [asterisk-users] 5ess   Heath Roberts
• [asterisk-users] Windows Mobile MWI and asterisk   OCG Technical Support
• [asterisk-users] don't get 2.0 gui to run on asterisk 1.6.0.5   Tzafrir Cohen
• [asterisk-users] check if not human   Paul Hales
• [asterisk-users] strange text message:)   Catalin S.
• [asterisk-users] 5ess   Jared Smith
• [asterisk-users] strange text message:)   David fire
• [asterisk-users] GSM codec is a good choice ???   Alejandro Cabrera Obed
• [asterisk-users] AGI script   michel freiha
• [asterisk-users] building asterisk-1.6.0.6 failed!   Tamer Higazi
• [asterisk-users] GSM codec is a good choice ???   Steve Edwards
• [asterisk-users] strange text message:)   Catalin S.
• [asterisk-users] strange text message:)   OCG Technical Support
• [asterisk-users] weird problem   Michael
• [asterisk-users] HDD FULLL   David at ULC
• [asterisk-users] Intel Vs AMD   Andrew Joakimsen
• [asterisk-users] HDD FULLL   Darren Wiebe
• [asterisk-users] HDD FULLL   David fire
• [asterisk-users] HDD FULLL   David at ULC
• [asterisk-users] (no subject)   Lê Văn Hòa
• [asterisk-users] Polycom Spectralink 8002 Configuration   M Hulber
• [asterisk-users] HDD FULLL   Steve Edwards
• [asterisk-users] Managing the spiralling costs   Vikas
• [asterisk-users] HDD FULLL   David at ULC
• [asterisk-users] HDD FULLL   David at ULC
• [asterisk-users] HDD FULLL   Steve Edwards
• [asterisk-users] HDD FULLL   David fire
• [asterisk-users] HDD FULLL   Jeff LaCoursiere
• [asterisk-users] Managing the spiralling costs   David fire
• [asterisk-users] building asterisk-1.6.0.6 failed!   Matt Watson
• [asterisk-users] Managing the spiralling costs   Jeff LaCoursiere
• [asterisk-users] HDD FULLL   Matt Watson
• [asterisk-users] Managing the spiralling costs   Craig Van Ham
• [asterisk-users] Managing the spiralling costs   David Backeberg
• [asterisk-users] Managing the spiralling costs   Steve Edwards
• [asterisk-users] Legacy Cisco ATA186I1   drew einhorn
• [asterisk-users] Managing the spiralling costs   David Backeberg
• [asterisk-users] Inbound call to IVR drops after 21 seconds?   Doug
• [asterisk-users] HDD FULLL   Paul Hales
• [asterisk-users] HDD FULLL   Roderick A. Anderson
• [asterisk-users] Does audio have to flow through vitelity servers ?   Vikas
• [asterisk-users] AGI script   Yawar Hadi
• [asterisk-users] Free US DID   Plyschen
• [asterisk-users] multiple asterisks in a server   Rilawich Ango
• [asterisk-users] building asterisk-1.6.0.6 failed!   Tamer Higazi
• [asterisk-users] GSM codec is a good choice ???   Philipp Kempgen
• [asterisk-users] Swichted digits on received number from fax on an fxs port   Loic Didelot
• [asterisk-users] multiple asterisks in a server   Geraint Lee
• [asterisk-users] multiple asterisks in a server   Geraint Lee
• [asterisk-users] asterisk -f and restart now   Klaus Darilion
• [asterisk-users] chan_sip and database integration   Klaus Darilion
• [asterisk-users] chan_sip and database integration   Michiel van Baak
• [asterisk-users] building asterisk-1.6.0.6 failed!   Tzafrir Cohen
• [asterisk-users] HDD FULLL   Tzafrir Cohen
• [asterisk-users] asterisk -f and restart now   Tzafrir Cohen
• [asterisk-users] Swichted digits on received number from fax on an fxs port   Tzafrir Cohen
• [asterisk-users] Managing the spiralling costs   BJ Weschke
• [asterisk-users] asterisk -f and restart now   Klaus Darilion
• [asterisk-users] astdb and Debian : can't use db4.5_dump   Olivier
• [asterisk-users] multiple asterisks in a server   Olivier
• [asterisk-users] chan_sip and database integration   Johansson Olle E
• [asterisk-users] HDD FULLL   Barry L. Kline
• [asterisk-users] astdb and Debian : can't use db4.5_dump   Tzafrir Cohen
• [asterisk-users] db_dump185.c missing if Asterisk 1.4 source file   Olivier
• [asterisk-users] API hangup command   Jerry Geis
• [asterisk-users] GSM codec is a good choice ???   Tiago Durante
• [asterisk-users] COSTA RICA - E1   Luis Morales
• [asterisk-users] API hangup command   Danny Nicholas
• [asterisk-users] astdb and Debian : can't use db4.5_dump   Steve Howes
• [asterisk-users] GSM codec is a good choice ???   Alejandro Cabrera Obed
• [asterisk-users] astdb and Debian : can't use db4.5_dump   Olivier
• [asterisk-users] building asterisk-1.6.0.6 failed!   M Hulber
• [asterisk-users] astdb and Debian : can't use db4.5_dump   Philipp Kempgen
• [asterisk-users] GSM codec is a good choice ???   Olivier
• [asterisk-users] GSM codec is a good choice ???   Tiago Durante
• [asterisk-users] GSM codec is a good choice ???   Eric Wieling, Asteria Solutions Group
• [asterisk-users] GSM codec is a good choice ???   Alejandro Cabrera Obed
• [asterisk-users] astdb and Debian : can't use db4.5_dump   Olivier
• [asterisk-users] multiple asterisks in a server   David Backeberg
• [asterisk-users] multiple asterisks in a server   Danny Nicholas
• [asterisk-users] don't get 2.0 gui to run on asterisk 1.6.0.5   Leif Madsen
• [asterisk-users] Polycom Spectralink 8002 Configuration   M Hulber
• [asterisk-users] building asterisk-1.6.0.6 failed!   Leif Madsen
• [asterisk-users] multiple asterisks in a server   Geraint Lee
• [asterisk-users] Aastra phones   Mike
• [asterisk-users] building asterisk-1.6.0.6 failed!   Leif Madsen
• [asterisk-users] Aastra phones   John Millican
• [asterisk-users] Aastra phones   M Hulber
• [asterisk-users] building asterisk-1.6.0.6 failed!   M Hulber
• [asterisk-users] Aastra phones   Ira
• [asterisk-users] building asterisk-1.6.0.6 failed!   Tamer Higazi
• [asterisk-users] chan_sip and database integration   Klaus Darilion
• [asterisk-users] Aastra phones   Greg Kennedy
• [asterisk-users] multiple asterisks in a server   Klaus Darilion
• [asterisk-users] what is the correct character to separate application parameters: , or |   Klaus Darilion
• [asterisk-users] Swichted digits on received number from fax on an fxs port   Loic Didelot
• [asterisk-users] what is the correct character to separate application parameters: , or |   Danny Nicholas
• [asterisk-users] what is the correct character to separate application parameters: , or |   Eric Wieling, Asteria Solutions Group
• [asterisk-users] multiple asterisks in a server   Geraint Lee
• [asterisk-users] Gosub behavior change <=1.6.0.5 to 1.6.0.6   Barry L. Kline
• [asterisk-users] chan_sip and database integration   John Todd
• [asterisk-users] Aastra phones   Mike
• [asterisk-users] Aastra phones   Ira
• [asterisk-users] Polycom Phones start to break up after being up a LONG time   Barry D. Hassler
• [asterisk-users] Configuring chan_dahdi.conf for Sangoma A200/Remora FXO/FXS Analog AFT card   Ketema Harris
• [asterisk-users] Aastra phones   Mike
• [asterisk-users] receiving 1st digit from a variable   Tamer Higazi
• [asterisk-users] Polycom Phones start to break up after being up a LONG time   Mike
• [asterisk-users] Configuring chan_dahdi.conf for Sangoma A200/Remora FXO/FXS Analog AFT card   Ketema Harris
• [asterisk-users] receiving 1st digit from a variable   Danny Nicholas
• [asterisk-users] Polycom Phones start to break up after being up a LONG time   Danny Nicholas
• [asterisk-users] weird problem   David fire
• [asterisk-users] Configuring chan_dahdi.conf for Sangoma A200/Remora FXO/FXS Analog AFT card   Tzafrir Cohen
• [asterisk-users] receiving 1st digit from a variable   Tamer Higazi
• [asterisk-users] Polycom Phones start to break up after being up a LONG time   Mike
• [asterisk-users] Configuring chan_dahdi.conf for Sangoma A200/Remora FXO/FXS Analog AFT card   John Novack
• [asterisk-users] receiving 1st digit from a variable   Danny Nicholas
• [asterisk-users] multiple asterisks in a server   Klaus Darilion
• [asterisk-users] what is the correct character to separate application parameters: , or |   Klaus Darilion
• [asterisk-users] asterisk -f and restart now   Klaus Darilion
• [asterisk-users] Polycom Phones start to break up after being up a LONG time   Jeff LaCoursiere
• [asterisk-users] Multiple SIPGate accounts.   Razza
• [asterisk-users] receiving 1st digit from a variable   Klaus Darilion
• [asterisk-users] receiving 1st digit from a variable   Tamer Higazi
• [asterisk-users] Gosub behavior change <=1.6.0.5 to 1.6.0.6   Klaus Darilion
• [asterisk-users] asterisk -f and restart now   Klaus Darilion
• [asterisk-users] Incoming call   michel freiha
• [asterisk-users] multiple asterisks in a server   Senad Jordanovic
• [asterisk-users] asterisk -f and restart now   Ex Vito
• [asterisk-users] Polycom Spectralink 8002 Configuration   Jeff LaCoursiere
• [asterisk-users] Gosub behavior change <=1.6.0.5 to 1.6.0.6   Tilghman Lesher
• [asterisk-users] Gosub behavior change <=1.6.0.5 to 1.6.0.6   Tilghman Lesher
• [asterisk-users] Aastra phones   Greg Kennedy
• [asterisk-users] GSM codec is a good choice ???   Kristian Kielhofner
• [asterisk-users] GSM codec is a good choice ???   David fire
• [asterisk-users] GSM codec is a good choice ???   Alejandro Cabrera Obed
• [asterisk-users] multiple asterisks in a server   Rilawich Ango
• [asterisk-users] (no subject)   C F
• [asterisk-users] multiple asterisks in a server   Rilawich Ango
• [asterisk-users] GSM codec is a good choice ???   David fire
• [asterisk-users] Polycom Spectralink 8002 Configuration   Michael Graves
• [asterisk-users] weird problem   Michael
• [asterisk-users] GSM codec is a good choice ???   drew einhorn
• [asterisk-users] Problem redirecting user running a Dynamic feature   david
• [asterisk-users] Incoming call   David fire
• [asterisk-users] Asterisk security between two servers   arkda
• [asterisk-users] Dropping RTP packets   Jim Dickenson
• [asterisk-users] HDLC Errors   Daniel Johnson
• [asterisk-users] GSM codec is a good choice ???   Tzafrir Cohen
• [asterisk-users] Call files with extensions.ael : "One app must be specified"   Olivier
• [asterisk-users] Gosub behavior change <=1.6.0.5 to 1.6.0.6   Klaus Darilion
• [asterisk-users] Multiple SIPGate accounts.   Klaus Darilion
• [asterisk-users] switchtype QSIG and Asterisk implementation   Vieri
• [asterisk-users] switchtype QSIG and Asterisk implementation   Tzafrir Cohen
• [asterisk-users] switchtype QSIG and Asterisk implementation   Vieri
• [asterisk-users] bandwidth.com will not sell me a sip line since the address is in Citrus Heights CA   Vikas
• [asterisk-users] switchtype QSIG and Asterisk implementation   Olivier
• [asterisk-users] Asterisk with Internet connectivity   Klaus Darilion
• [asterisk-users] multiple asterisks in a server   Klaus Darilion
• [asterisk-users] bandwidth.com will not sell me a sip line since the address is in Citrus Heights CA   Gordon Henderson
• [asterisk-users] switchtype QSIG and Asterisk implementation   Artifex Maximus
• [asterisk-users] bandwidth.com will not sell me a sip line since the address is in Citrus Heights CA   Michael
• [asterisk-users] Asterisk with Internet connectivity   Administrator TOOTAI
• [asterisk-users] Asterisk with Internet connectivity   Grygoriy Dobrovolskyy
• [asterisk-users] Asterisk with Internet connectivity   Steve Howes
• [asterisk-users] trunk to trunk   Leonja Cerebro
• [asterisk-users] trunk to trunk   Robert Broyles
• [asterisk-users] switchtype QSIG and Asterisk implementation   Vieri
• [asterisk-users] strange text message:)   Catalin S.
• [asterisk-users] GSM codec is a good choice ???   Alejandro Cabrera Obed
• [asterisk-users] usegmtime=yes for cdr_custom   Klaus Darilion
• [asterisk-users] SIP_CODEC variable   Mike
• [asterisk-users] multiple asterisks in a server   Geraint Lee
• [asterisk-users] GSM codec is a good choice ???   Christian Victor
• [asterisk-users] GSM codec is a good choice ???   Michael Graves
• [asterisk-users] GSM codec is a good choice ???   Tzafrir Cohen
• [asterisk-users] GSM codec is a good choice ???   Michael Graves
• [asterisk-users] SIP_CODEC variable   Jared Smith
• [asterisk-users] Gosub behavior change <=1.6.0.5 to 1.6.0.6   sean darcy
• [asterisk-users] SIP_CODEC variable   Jeff LaCoursiere
• [asterisk-users] Stuck Parked Calls?   Jonathan C. Bailey
• [asterisk-users] Stuck Parked Calls?   Jonathan C. Bailey
• [asterisk-users] Gosub behavior change <=1.6.0.5 to 1.6.0.6   Jared Smith
• [asterisk-users] Polycom Spectralink 8002 Configuration   Bob Pierce
• [asterisk-users] TE121 on Asterisk   Oguzhan Kayhan
• [asterisk-users] TE121 on Asterisk   Eric Wieling, Asteria Solutions Group
• [asterisk-users] TE121 on Asterisk   Doug Lytle
• [asterisk-users] Polycom Spectralink 8002 Configuration   Jeff LaCoursiere
• [asterisk-users] Gosub behavior change <=1.6.0.5 to 1.6.0.6   Barry L. Kline
• [asterisk-users] Polycom Spectralink 8002 Configuration   Jeff LaCoursiere
• [asterisk-users] TE121 on Asterisk   Jared Smith
• [asterisk-users] dahdi wcb4xxp and fax   stoffell
• [asterisk-users] TE121 on Asterisk   Oguzhan Kayhan
• [asterisk-users] TE121 on Asterisk   Oguzhan Kayhan
• [asterisk-users] bandwidth.com will not sell me a sip line since the address is in Citrus Heights CA   Frank Bulk
• [asterisk-users] SIP_CODEC variable   Olivier
• [asterisk-users] bandwidth.com will not sell me a sip line since the address is in Citrus Heights CA   Jon Pounder
• [asterisk-users] Gosub behavior change <=1.6.0.5 to 1.6.0.6   Klaus Darilion
• [asterisk-users] dahdi wcb4xxp and fax   Lee Howard
• [asterisk-users] Polycom Spectralink 8002 Configuration   Bob Pierce
• [asterisk-users] Polycom Spectralink 8002 Configuration   M Hulber
• [asterisk-users] dahdi wcb4xxp and fax   Steve Underwood
• [asterisk-users] Polycom Spectralink 8002 Configuration   M Hulber
• [asterisk-users] dahdi wcb4xxp and fax   stoffell
• [asterisk-users] SIP_CODEC variable   Mike
• [asterisk-users] Aastra phones   Mike
• [asterisk-users] Gosub behavior change <=1.6.0.5 to 1.6.0.6   Tilghman Lesher
• [asterisk-users] Gosub behavior change <=1.6.0.5 to 1.6.0.6   sean darcy
• [asterisk-users] Polycom Spectralink 8002 Configuration   Bob Pierce
• [asterisk-users] Gosub behavior change <=1.6.0.5 to 1.6.0.6   Klaus Darilion
• [asterisk-users] Multiple SIPGate accounts.   Razza
• [asterisk-users] cannot allocate memory   wassim Darwish
• [asterisk-users] DID's in a specific rate center   Vikas
• [asterisk-users] DID's in a specific rate center   Danny Nicholas
• [asterisk-users] CDR - Asterisk-Stat and PHP5   Tiago Durante
• [asterisk-users] cannot allocate memory   Steve Edwards
• [asterisk-users] DID's in a specific rate center   Vikas
• [asterisk-users] DID's in a specific rate center   Jay Milk
• [asterisk-users] DID's in a specific rate center   Danny Nicholas
• [asterisk-users] DID's in a specific rate center   Danny Nicholas
• [asterisk-users] Patton 5.3. How to get incoming calls ?   Olivier
• [asterisk-users] Gosub behavior change <=1.6.0.5 to 1.6.0.6   Tilghman Lesher
• [asterisk-users] Gosub behavior change <=1.6.0.5 to 1.6.0.6   Eric Wieling, Asteria Solutions Group
• [asterisk-users] DID's in a specific rate center   Jonn Taylor
• [asterisk-users] Call from '6000' to extension rejected because extension not found   Chuck Coleman
• [asterisk-users] DID's in a specific rate center   Danny Nicholas
• [asterisk-users] Call from '6000' to extension rejected becauseextension not found   Danny Nicholas
• [asterisk-users] DID's in a specific rate center   Vikas
• [asterisk-users] DID's in a specific rate center   Danny Nicholas
• [asterisk-users] Realtime database function help   Elliot Murdock
• [asterisk-users] SheevaPlug Development Kit   Kristian Kielhofner
• [asterisk-users] DID's in a specific rate center   M Hulber
• [asterisk-users] SheevaPlug Development Kit   David fire
• [asterisk-users] Congestion Tone   Gustavo A Gonzalez
• [asterisk-users] SheevaPlug Development Kit   Brent Vrieze
• [asterisk-users] SheevaPlug Development Kit   Jon Pounder
• [asterisk-users] DID's in a specific rate center   Vikas
• [asterisk-users] DID's in a specific rate center   Jason Aarons (US)
• [asterisk-users] Call from '6000' to extension rejected because extension not found   David Gibbons
• [asterisk-users] SheevaPlug Development Kit   Tilghman Lesher
• [asterisk-users] DID's in a specific rate center   Jai Rangi
• [asterisk-users] AGI problem using mono (.Net)   Douglas Mortensen
• [asterisk-users] Gosub behavior change <=1.6.0.5 to 1.6.0.6   sean darcy
• [asterisk-users] AGI problem using mono (.Net)   Eric Wieling, Asteria Solutions Group
• [asterisk-users] AGI problem using mono (.Net)   Steve Edwards
• [asterisk-users] Call from '6000' to extension rejected because extension not found   Paul Hales
• [asterisk-users] Realtime database function help   Forrest Beck
• [asterisk-users] AGI problem using mono (.Net)   Luis Morales
• [asterisk-users] AGI problem using mono (.Net)   Steve Edwards
• [asterisk-users] DTMF Forwarking Problems.   Catalin S.
• [asterisk-users] Problems with Outbound Calls   Wye-khe Kwok
• [asterisk-users] DTMF Forwarking Problems.   Paul Hales
• [asterisk-users] asterisk 1.6.0.5 and IM   lord_fleg at iinet.net.au
• [asterisk-users] codec_dahdi and Asterisk 1.6.0.6   Brandon B.
• [asterisk-users] TE121 on Asterisk   Oguzhan Kayhan
• [asterisk-users] DID's in a specific rate center   Mik Cheez
• [asterisk-users] DID's in a specific rate center   Steve Edwards
• [asterisk-users] Gosub behavior change <=1.6.0.5 to 1.6.0.6   Klaus Darilion
• [asterisk-users] Gosub behavior change <=1.6.0.5 to 1.6.0.6   amit mehta
• [asterisk-users] TE121 on Asterisk   Michel Verbraak
• [asterisk-users] TE121 on Asterisk   Oguzhan Kayhan
• [asterisk-users] codec_dahdi and Asterisk 1.6.0.6   Olivier
• [asterisk-users] TE121 on Asterisk   Tzafrir Cohen
• [asterisk-users] Need US Dialing Account with Asterisk   Kashif Naeem
• [asterisk-users] TE121 on Asterisk   Oguzhan Kayhan
• [asterisk-users] CDR - Asterisk-Stat and PHP5   Paulo Santos
• [asterisk-users] Gosub behavior change <=1.6.0.5 to 1.6.0.6   Klaus Darilion
• [asterisk-users] Dictate   amit mehta
• [asterisk-users] HP DL380 G5 with TE420   Hans Konings
• [asterisk-users] Swichted digits on received number from fax on an fxs port   Loic Didelot
• [asterisk-users] Swichted digits on received number from fax on an fxs port   Tzafrir Cohen
• [asterisk-users] HP DL380 G5 with TE420   stoffell
• [asterisk-users] call-limit on a per destination basis   Jean-Michel Hiver
• [asterisk-users] HP DL380 G5 with TE420   Michel Verbraak
• [asterisk-users] call-limit on a per destination basis   Klaus Darilion
• [asterisk-users] Swichted digits on received number from fax on an fxs port   Loic Didelot
• [asterisk-users] DID's in a specific rate center   Frank Bulk
• [asterisk-users] DID's in a specific rate center   Alex Balashov
• [asterisk-users] HP DL380 G5 with TE420   Hans Konings
• [asterisk-users] Multiple SIPGate accounts.   Gordon Henderson
• [asterisk-users] DID's in a specific rate center   M Hulber
• [asterisk-users] DID's in a specific rate center   M Hulber
• [asterisk-users] Patton 5.3. How to get incoming calls ? [SOLVED]   Olivier
• [asterisk-users] HP DL380 G5 with TE420   Miguel Molina
• [asterisk-users] Getting SIP field P-Asserted-Identity from EAGI   equis software
• [asterisk-users] ATA recommendation (wih FTP provisioning)   Mike
• [asterisk-users] Problems with Outbound Calls   David fire
• [asterisk-users] ATA recommendation (wih FTP provisioning)   peder at networkoblivion.com
• [asterisk-users] Getting SIP field P-Asserted-Identity from EAGI   Danny Nicholas
• [asterisk-users] ATA recommendation (wih FTP provisioning)   Mike
• [asterisk-users] Odd Read App Issues   Robert Broyles
• [asterisk-users] Gosub behavior change <=1.6.0.5 to 1.6.0.6   Benny Amorsen
• [asterisk-users] Odd Read App Issues   Robert Broyles
• [asterisk-users] Congestion Tone   Gustavo A Gonzalez
• [asterisk-users] Getting SIP field P-Asserted-Identity from EAGI   Benny Amorsen
• [asterisk-users] ATA recommendation (wih FTP provisioning)   Benny Amorsen
• [asterisk-users] Odd Read App Issues   Eric Wieling, Asteria Solutions Group
• [asterisk-users] Problems with Outbound Calls   Tzafrir Cohen
• [asterisk-users] Odd Read App Issues   Robert Broyles
• [asterisk-users] [cdr_odbc] error: Cannot insert the value NULL into column 'calldate'   Rajkumar S
• [asterisk-users] Odd Read App Issues   Eric Wieling, Asteria Solutions Group
• [asterisk-users] ATA recommendation (wih FTP provisioning)   Jeff LaCoursiere
• [asterisk-users] Getting SIP field P-Asserted-Identity from EAGI   Tilghman Lesher
• [asterisk-users] Dictate   Brent Davidson
• [asterisk-users] DID's in a specific rate center   Wilton Helm
• [asterisk-users] Odd Read App Issues   Robert Broyles
• [asterisk-users] Getting SIP field P-Asserted-Identity from EAGI   equis software
• [asterisk-users] CDR - Asterisk-Stat and PHP5   Marco Signorini
• [asterisk-users] Odd Read App Issues   Brent Davidson
• [asterisk-users] DTMF tones mid conversation   Simon Dixey
• [asterisk-users] Odd Read App Issues   Robert Broyles
• [asterisk-users] Dictate   amit mehta
• [asterisk-users] ATA recommendation (wih FTP provisioning)   Wilton Helm
• [asterisk-users] ATA recommendation (wih FTP provisioning)   Jeff LaCoursiere
• [asterisk-users] Residential portals and real world scalability   J. Oquendo
• [asterisk-users] CDR - Asterisk-Stat and PHP5   Paulo Santos
• [asterisk-users] Can't build today's AGX Asterisk Addon with spandsp0.0.6pre3 or 4   Olivier
• [asterisk-users] Can't build today's AGX Asterisk Addon with spandsp0.0.6pre3 or 4   Olivier
• [asterisk-users] ATA recommendation (wih FTP provisioning)   Olivier
• [asterisk-users] ATA recommendation (wih FTP provisioning)   Olivier
• [asterisk-users] AGI problem using mono (.Net)   Douglas Mortensen
• [asterisk-users] incoming call problem   michel freiha
• [asterisk-users] AGI problem using mono (.Net)   Douglas Mortensen
• [asterisk-users] Problems with Outbound Calls   David fire
• [asterisk-users] DTMF tones mid conversation   stoffell
• [asterisk-users] CDR - Asterisk-Stat and PHP5   Tiago Durante
• [asterisk-users] asterisk 1.4.23.1 and mISDN 1.1.8 segfaults   stoffell
• [asterisk-users] Odd Read App Issues   Robert Broyles
• [asterisk-users] Friday Feb 27th at 12 Noon EST: Polycom Applications   randulo
• [asterisk-users] Odd Read App Issues   Brent Davidson
• [asterisk-users] Current state of Asterisk and Virtualization?   Gavin Henry
• [asterisk-users] ATA recommendation (wih FTP provisioning)   peder at networkoblivion.com
• [asterisk-users] AGI problem using mono (.Net)   Steve Edwards
• [asterisk-users] changing /etc/dahdi/system.conf   Brandon B.
• [asterisk-users] dahdi wcb4xxp and fax   Olivier
• [asterisk-users] changing /etc/dahdi/system.conf   Joseph L. Casale
• [asterisk-users] changing /etc/dahdi/system.conf   Tzafrir Cohen
• [asterisk-users] changing /etc/dahdi/system.conf   Barry L. Kline
• [asterisk-users] Dropping RTP packets   Brent Davidson
• [asterisk-users] ATA recommendation (wih FTP provisioning)   Wilton Helm
• [asterisk-users] Problems with Outbound Calls   Tzafrir Cohen
• [asterisk-users] ATA recommendation (wih FTP provisioning)   Jeff LaCoursiere
• [asterisk-users] Dropping RTP packets   Jim Dickenson
• [asterisk-users] SheevaPlug Development Kit   Hans Witvliet
• [asterisk-users] SheevaPlug Development Kit   Tzafrir Cohen
• [asterisk-users] ATA recommendation (wih FTP provisioning)   Ira
• [asterisk-users] ATA recommendation (wih FTP provisioning)   Steve Edwards
• [asterisk-users] Question about Do Not Disturb   Haim Dimer
• [asterisk-users] Problems with Outbound Calls   David fire
• [asterisk-users] using Cisco IP Communicator with SIP to Asterisk   Hugo Garcia Gomez
• [asterisk-users] Current state of Asterisk and Virtualization?   Senad Jordanovic
• [asterisk-users] call file concurrency   Bill Michaelson
• [asterisk-users] Question about Do Not Disturb   Alexander Lopez
• [asterisk-users] HP DL380 G5 with TE420   Hans Konings
• [asterisk-users] Question about Do Not Disturb   Gordon Henderson
• [asterisk-users] Problems with Outbound Calls   Wye-khe Kwok
• [asterisk-users] call file concurrency   Christian Victor
• [asterisk-users] Realtime mapping for 'queue_log' found to engine 'odbc', but the engine is not available   Rajkumar S
• [asterisk-users] incoming call problem   David fire
• [asterisk-users] incoming call problem   michel freiha
• [asterisk-users] Problems with Outbound Calls   Lee, John (Sydney)
• [asterisk-users] Odd Read App Issues - RESOLVED   Robert Broyles
• [asterisk-users] Problems with Outbound Calls   Tzafrir Cohen
• [asterisk-users] call-limit on a per destination basis   Jean-Michel Hiver
• [asterisk-users] change language and playback issue   Giedrius Augys
• [asterisk-users] [FIXED] Re: call-limit on a per destination basis   Jean-Michel Hiver
• [asterisk-users] [HOWTO] Priorize one destination over another on a link   Jean-Michel Hiver
• [asterisk-users] changing /etc/dahdi/system.conf   John Todd
• [asterisk-users] building a phone   Tzafrir Cohen
• [asterisk-users] Current state of Asterisk and Virtualization?   John Todd
• [asterisk-users] CDR - Asterisk-Stat and PHP5   Marco Signorini
• [asterisk-users] codec_dahdi and Asterisk 1.6.0.6   Moises Silva
• [asterisk-users] building a phone   Jon Pounder
• [asterisk-users] building a phone   SIP
• [asterisk-users] [FIXED] Re: call-limit on a per destination basis   Klaus Darilion
• [asterisk-users] building a phone   Marco Signorini
• [asterisk-users] building a phone   Jon Pounder
• [asterisk-users] ATA recommendation (wih FTP provisioning)   Wilton Helm
• [asterisk-users] building a phone   Marco Signorini
• [asterisk-users] building a phone   Tzafrir Cohen
• [asterisk-users] building a phone   Wilton Helm
• [asterisk-users] building a phone   Jon Pounder
• [asterisk-users] building a phone   Gordon Henderson
• [asterisk-users] building a phone   Grygoriy Dobrovolskyy
• [asterisk-users] building a phone   Tzafrir Cohen
• [asterisk-users] Problems with Outbound Calls   Danny Nicholas
• [asterisk-users] call file concurrency   Danny Nicholas
• [asterisk-users] call file concurrency   Steve Edwards
• [asterisk-users] API command monitor to record only the input channel   Jerry Geis
• [asterisk-users] building a phone   Gordon Henderson
• [asterisk-users] API command monitor to record only the input channel   Jim Dickenson
• [asterisk-users] building a phone   Tzafrir Cohen
• [asterisk-users] Switch Options for a service provider   Ignacio Ortega A.
• [asterisk-users] building a phone   Jon Pounder
• [asterisk-users] building a phone   Wilton Helm
• [asterisk-users] building a phone   Jon Pounder
• [asterisk-users] building a phone   Jon Pounder
• [asterisk-users] building a phone   Tzafrir Cohen
• [asterisk-users] building a phone   Tzafrir Cohen
• [asterisk-users] call file concurrency   Danny Nicholas
• [asterisk-users] building a phone   Wilton Helm
• [asterisk-users] call file concurrency   Eric Wieling, Asteria Solutions Group
• [asterisk-users] Continue in dialplan on hangup   Daniel Hazelbaker
• [asterisk-users] Continue in dialplan on hangup   Doug Lytle
• [asterisk-users] Continue in dialplan on hangup   Doug Lytle
• [asterisk-users] Current state of Asterisk and Virtualization?   Gavin Henry
• [asterisk-users] call file concurrency   Steve Edwards
• [asterisk-users] call file concurrency   Steve Edwards
• [asterisk-users] Echo on SIP to SIP calls?   Bruce Komito
• [asterisk-users] call file concurrency   Danny Nicholas
• [asterisk-users] Echo on SIP to SIP calls?   J. Oquendo
• [asterisk-users] Echo on SIP to SIP calls?   Stephen Davies
• [asterisk-users] call file concurrency   Steve Edwards
• [asterisk-users] TE121B server recommendation   Kevin DeGraaf
• [asterisk-users] changing /etc/dahdi/system.conf   Brandon B.
• [asterisk-users] TE121B server recommendation   Cory Andrews
• [asterisk-users] what is the effect of high LBO settings?   Brandon B.
• [asterisk-users] dialing timing problem?   Michael Higgins
• [asterisk-users] Continue in dialplan on hangup   Anthony Messina
• [asterisk-users] dialing timing problem?   Doug Lytle
• [asterisk-users] [asterisk-biz] Switch Options for a service provider   Alistair Cunningham
• [asterisk-users] dahdi wcb4xxp and fax   Matthew Fredrickson
• [asterisk-users] call file concurrency   James Sneeringer
• [asterisk-users] Continue in dialplan on hangup   Daniel Hazelbaker
• [asterisk-users] Continue in dialplan on hangup   Anthony Messina
• [asterisk-users] call file concurrency   Steve Edwards
• [asterisk-users] Question about Do Not Disturb   Haim Dimer
• [asterisk-users] call file concurrency   Mik Cheez
• [asterisk-users] rfc2833 vs. sipinfo and network weirdness   Michael
• [asterisk-users] what is the effect of high LBO settings?   Jared Smith
• [asterisk-users] building a phone   Michael Graves
• [asterisk-users] connecting 66 analog phones to asterisk - hardware suggestions   Erick Perez
• [asterisk-users] call file concurrency   Lenz Emilitri
• [asterisk-users] building a phone   Tzafrir Cohen
• [asterisk-users] [FIXED] Re: call-limit on a per destination basis   didier.cuffaut
• [asterisk-users] using an eicon diva server card with asterisk.   Andrea Borghi
• [asterisk-users] Continue in dialplan on hangup   David Backeberg
• [asterisk-users] Remote connection to an Asterisk server   Gary
• [asterisk-users] Remote connection to an Asterisk server   Steve Totaro
• [asterisk-users] Remote connection to an Asterisk server   Dale Noll
• [asterisk-users] Remote connection to an Asterisk server   Alex Balashov
• [asterisk-users] No rtp activity   michel freiha
• [asterisk-users] Temporaneamente assente   prandini at spe.net
• [asterisk-users] dialing timing problem?   Michael Higgins
• [asterisk-users] clone X100p+dahdi dial out works only after receiving call   Michael Higgins
• [asterisk-users] clone X100p+dahdi dial out works only after receiving call   Tzafrir Cohen
• [asterisk-users] building a phone   Paul Chambers
• [asterisk-users] building a phone   Michael Graves
• [asterisk-users] building a phone   Paul Chambers
• [asterisk-users] I can`t send DTMFs through FXO lines - dahdi   Daniel - Asterisk
• [asterisk-users] Called number as variable - how to?   Michael
• [asterisk-users] Called number as variable - how to?   Trevor Peirce

Last message date: Sat Feb 28 22:21:44 CST 2009
Archived on: Sat Feb 28 22:21:51 CST 2009

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