October 2008 Archives by date

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Starting: Wed Oct 1 00:14:38 CDT 2008
Ending: Fri Oct 31 23:36:12 CDT 2008
Messages: 1767

• [asterisk-users] is DNS SRV enough for failover?   Nhadie
• [asterisk-users] is DNS SRV enough for failover?   Andres
• [asterisk-users] Software patents (was G723 on asterisk 1.4.1)   Atis Lezdins
• [asterisk-users] Software patents (was G723 on asterisk 1.4.1)   Steve Underwood
• [asterisk-users] Asterisk Documentation now on voip-info.org Wiki   Olivier
• [asterisk-users] Asterisk Documentation now on voip-info.org Wiki   Rob Hillis
• [asterisk-users] How can Block a pri channel   Giorgio Incantalupo
• [asterisk-users] Software patents (was G723 on asterisk 1.4.1)   Atis Lezdins
• [asterisk-users] GSM / 3g channel bank   Julian Lyndon-Smith
• [asterisk-users] Asterisk Documentation now on voip-info.org Wiki   Josiah Bryan
• [asterisk-users] GSM / 3g channel bank   Jeff Johnson
• [asterisk-users] ATA for large networks   Steve Totaro
• [asterisk-users] GSM / 3g channel bank   Gordon Henderson
• [asterisk-users] Debugging   Wilton Helm
• [asterisk-users] Call Center   broadband Voice
• [asterisk-users] Asterisk queue not play muscinhold or hangup   broadband Voice
• [asterisk-users] New VICIDIAL astGUIclient Release: 2.0.4   broadband Voice
• [asterisk-users] RTP sent before the INVITE ACK (for voicemail app)   tic tac
• [asterisk-users] Asterisk in VM.   Joseph L. Casale
• [asterisk-users] Software patents (was G723 on asterisk 1.4.1)   Steve Underwood
• [asterisk-users] GSM / 3g channel bank   Steve Kennedy
• [asterisk-users] How can Block a pri channel   Rob Hillis
• [asterisk-users] GSM / 3g channel bank   Julian Lyndon-Smith
• [asterisk-users] Sip Header Help   voip crazy
• [asterisk-users] GSM / 3g channel bank   Jeff LaCoursiere
• [asterisk-users] RTP sent before the INVITE ACK (for voicemail app)   Grey Man
• [asterisk-users] Ongoing calls with SIPPEER, curcalls   Olivier
• [asterisk-users] Ongoing calls with SIPPEER, curcalls   Doug Lytle
• [asterisk-users] G723 on asterisk 1.4.1   Tilghman Lesher
• [asterisk-users] SIP users limit   Alejandro Facultad
• [asterisk-users] Asterisk in VM.   Bill Michaelson
• [asterisk-users] Software patents (was G723 on asterisk 1.4.1)   Atis Lezdins
• [asterisk-users] [Fwd: asterisk-users Digest, Vol 51, Issue 2]   Bill Michaelson
• [asterisk-users] New VICIDIAL astGUIclient Release: 2.0.4   Matt Florell
• [asterisk-users] Asterisk in VM.   Alexander Benaguev
• [asterisk-users] Ongoing calls with SIPPEER, curcalls   Olivier
• [asterisk-users] RTP sent before the INVITE ACK (for voicemail app)   tic tac
• [asterisk-users] Ongoing calls with SIPPEER, curcalls   Doug Lytle
• [asterisk-users] ietf-sipping-config-framework   Olivier
• [asterisk-users] Ongoing calls with SIPPEER, curcalls   Olivier
• [asterisk-users] Ongoing calls with SIPPEER, curcalls   Doug Lytle
• [asterisk-users] zap destroy   Jeff Peeler
• [asterisk-users] GSM / 3g channel bank   Sam Tam
• [asterisk-users] How can Block a pri channel   Sean Bright
• [asterisk-users] How can Block a pri channel   Dwayne Hubbard
• [asterisk-users] Asterisk - Failover System   Nelson Granados
• [asterisk-users] Asterisk - Failover System   Alex Balashov
• [asterisk-users] Asterisk - Failover System   Jai Rangi
• [asterisk-users] Asterisk in VM.   Hans Witvliet
• [asterisk-users] Asterisk in VM.   Jeff Johnson
• [asterisk-users] Software patents (was G723 on asterisk 1.4.1)   Michael Collins
• [asterisk-users] Asterisk - Failover System   Steve Totaro
• [asterisk-users] Software patents (was G723 on asterisk 1.4.1)   Steve Totaro
• [asterisk-users] Asterisk - Failover System   Alex Balashov
• [asterisk-users] Software patents (was G723 on asterisk 1.4.1)   Alex Balashov
• [asterisk-users] zap destroy   Daniel Hazelbaker
• [asterisk-users] Software patents (was G723 on asterisk 1.4.1)   Steve Totaro
• [asterisk-users] Software patents (was G723 on asterisk 1.4.1)   Michael Collins
• [asterisk-users] zap destroy   Jeff LaCoursiere
• [asterisk-users] Asterisk - Failover System   Jai Rangi
• [asterisk-users] no audio, firewall problem?   tic tac
• [asterisk-users] rebooting snoms in 1.6   Dr. Michael J. Chudobiak
• [asterisk-users] Asterisk - Failover System   Steve Edwards
• [asterisk-users] ATA for large networks   C F
• [asterisk-users] Aheeva With Asterisk   broadband Voice
• [asterisk-users] cisco VAD and Asterisk recordings   Gabriel Ortiz Lour
• [asterisk-users] Asterisk - Failover System   Alex Balashov
• [asterisk-users] Asterisk - Failover System   Alex Balashov
• [asterisk-users] Aheeva With Asterisk   Matt Florell
• [asterisk-users] Asterisk - Failover System   Darren Sessions
• [asterisk-users] Aheeva With Asterisk   broadband Voice
• [asterisk-users] ATA for large networks   nav_asr_1 at hotmail.com
• [asterisk-users] Asterisk custom functions   Max Alex
• [asterisk-users] zap destroy   Tzafrir Cohen
• [asterisk-users] Software patents (was G723 on asterisk 1.4.1)   Tzafrir Cohen
• [asterisk-users] Software patents (was G723 on asterisk 1.4.1)   Steve Totaro
• [asterisk-users] B410p question   voip crazy
• [asterisk-users] Software patents (was G723 on asterisk 1.4.1)   Tzafrir Cohen
• [asterisk-users] DTMF Problem   michel freiha
• [asterisk-users] How to find the CDR call start time value   Steve Hanselman
• [asterisk-users] RTP sent before the INVITE ACK (for voicemail app)   Grey Man
• [asterisk-users] How to find the CDR call start time value   Krunal Patel
• [asterisk-users] ATA for large networks   Nicolas Ross
• [asterisk-users] How to find the CDR call start time value   Steve Hanselman
• [asterisk-users] rebooting snoms in 1.6   Dr. Michael J. Chudobiak
• [asterisk-users] Ultramonkey LVS + asterisk   Nhadie
• [asterisk-users] IP address on mysql cdr   ronald ramos
• [asterisk-users] rebooting snoms in 1.6   Benny Amorsen
• [asterisk-users] DTMF issue   michel freiha
• [asterisk-users] Cisco Dropping SIP support?   Stefan Gofferje
• [asterisk-users] How can Block a pri channel   Sean Bright
• [asterisk-users] How can Block a pri channel   Steve Totaro
• [asterisk-users] Asterisk - Failover System   Steve Totaro
• [asterisk-users] Zaptel-1.4.1 error cross compile   satish patel
• [asterisk-users] Zaptel-1.4.1 error cross compile   Tzafrir Cohen
• [asterisk-users] Is SIPPEER curcalls working for you ? (was: Ongoing calls with SIPPEER, curcalls)   Olivier
• [asterisk-users] Zaptel-1.4.1 error cross compile   Satish Patel
• [asterisk-users] Is SIPPEER curcalls working for you ? (was: Ongoing calls with SIPPEER, curcalls)   Doug Lytle
• [asterisk-users] DTMF Problem   Fred Posner
• [asterisk-users] Zaptel-1.4.1 error cross compile   Tzafrir Cohen
• [asterisk-users] Zaptel-1.4.1 error cross compile   Satish Patel
• [asterisk-users] Asterisk custom functions   Tilghman Lesher
• [asterisk-users] zap destroy   Jeff Peeler
• [asterisk-users] zap destroy   Daniel Hazelbaker
• [asterisk-users] OT - Is sip.instance useful ?   Olivier
• [asterisk-users] Asterisk Queue question   voip crazy
• [asterisk-users] OT - Is sip.instance useful ?   SIP
• [asterisk-users] Asterisk Queue question   Atis Lezdins
• [asterisk-users] Zaptel-1.4.1 error cross compile   Satish Patel
• [asterisk-users] OT - Is sip.instance useful ?   Olivier
• [asterisk-users] DTMF Problem   bilal ghayyad
• [asterisk-users] dahdi-linux 2.0.0 and dahdi-tools 2.0.0 released   Asterisk Development Team
• [asterisk-users] Asterisk 1.4.22 and 1.6.0 Released   Asterisk Development Team
• [asterisk-users] Channels crossing...   Carlos Chavez
• [asterisk-users] Channels crossing...   Doug Lytle
• [asterisk-users] VOIP Provider   Gregory Malsack
• [asterisk-users] VOIP Provider   Steve Totaro
• [asterisk-users] DTMF   Barton Fisher
• [asterisk-users] Asterisk Queue question   Daniel - Asterisk
• [asterisk-users] VOIP Provider   Rafael Canchola
• [asterisk-users] Channels crossing...   Steve Totaro
• [asterisk-users] dahdi service start   Jerry Geis
• [asterisk-users] Cisco Dropping SIP support?   peder at networkoblivion.com
• [asterisk-users] uninstalling zaptel   Jerry Geis
• [asterisk-users] Channels crossing...   Nicolás Gudiño
• [asterisk-users] dahdi service start   Jason Parker
• [asterisk-users] t1 cards   Eric Fort
• [asterisk-users] t1 cards   Gordon Henderson
• [asterisk-users] t1 cards   Eric Fort
• [asterisk-users] 2 stage dialing and 484 address incomplete   Olivier
• [asterisk-users] Improving the voice Quality,   Jai Rangi
• [asterisk-users] sip clients for smart phones?   Babcock, Michael Alex
• [asterisk-users] sip clients for smart phones?   Alex Balashov
• [asterisk-users] Improving the voice Quality,   Alex Balashov
• [asterisk-users] sip clients for smart phones?   Babcock, Michael Alex
• [asterisk-users] How to add Callee's name into Dial command ?   Olivier
• [asterisk-users] Asterisk 1.4.22 and 1.6.0 Released   Stefan Gofferje
• [asterisk-users] How to add Callee's name into Dial command ?   Olivier
• [asterisk-users] sip clients for smart phones?   randulo
• [asterisk-users] t1 cards   Gordon Henderson
• [asterisk-users] How to add Callee's name into Dial command ?   Rob Hillis
• [asterisk-users] sip clients for smart phones?   Gordon Henderson
• [asterisk-users] sip clients for smart phones?   Rob Hillis
• [asterisk-users] DTMF   michel freiha
• [asterisk-users] t1 cards   Eric Fort
• [asterisk-users] DTMF   Tzafrir Cohen
• [asterisk-users] t1 cards   Jeff LaCoursiere
• [asterisk-users] How to add Callee's name into Dial command ?   Mr Shunz
• [asterisk-users] Improving the voice Quality,   Al Baker
• [asterisk-users] How to add Callee's name into Dial command ?   Olivier
• [asterisk-users] How to add Callee's name into Dial command ?   Mr Shunz
• [asterisk-users] How to add Callee's name into Dial command ?   Josiah Bryan
• [asterisk-users] How to add Callee's name into Dial command ?   Mr Shunz
• [asterisk-users] How to add Callee's name into Dial command ?   Mr Shunz
• [asterisk-users] MWI with Siemens Gigaset S450IP   Olivier
• [asterisk-users] Ok message   michel freiha
• [asterisk-users] How to add Callee's name into Dial command ?   Joe Pukepail
• [asterisk-users] How to add Callee's name into Dial command ?   Olivier
• [asterisk-users] t1 cards   Lyle Giese
• [asterisk-users] How to add Callee's name into Dial command ?   Olivier
• [asterisk-users] How to add Callee's name into Dial command ?   Olivier
• [asterisk-users] Ok message   Philipp Kempgen
• [asterisk-users] MWI with Siemens Gigaset S450IP   Kevin P. Fleming
• [asterisk-users] sip clients for smart phones?   Matt Gibson
• [asterisk-users] Ok message   Alex Balashov
• [asterisk-users] How to add Callee's name into Dial command ?   satish patel
• [asterisk-users] sip clients for smart phones?   Babcock, Michael Alex
• [asterisk-users] How to add Callee's name into Dial command ?   Mark Hamilton
• [asterisk-users] Improving the voice Quality,   Jai Rangi
• [asterisk-users] sip clients for smart phones?   Mark Hamilton
• [asterisk-users] Improving the voice Quality,   Gordon Henderson
• [asterisk-users] sip clients for smart phones?   Babcock, Michael Alex
• [asterisk-users] catch the use of h option in dial   Max Börebäck
• [asterisk-users] sip clients for smart phones?   Matt Gibson
• [asterisk-users] sip clients for smart phones?   Tariq ..
• [asterisk-users] sip clients for smart phones?   Babcock, Michael Alex
• [asterisk-users] sip clients for smart phones?   Mark Hamilton
• [asterisk-users] network monitoring - triggering a phone call in asterisk   Eric Fort
• [asterisk-users] sip clients for smart phones?   Babcock, Michael Alex
• [asterisk-users] Improving the voice Quality,   Alex Balashov
• [asterisk-users] network monitoring - triggering a phone call in asterisk   Alex Balashov
• [asterisk-users] sip clients for smart phones?   Tariq ..
• [asterisk-users] OT: Re: sip clients for smart phones?   Philipp Kempgen
• [asterisk-users] Improving the voice Quality,   Jai Rangi
• [asterisk-users] OT: Re: sip clients for smart phones?   Tariq ..
• [asterisk-users] sip clients for smart phones?   Babcock, Michael Alex
• [asterisk-users] sip clients for smart phones?   Matt Gibson
• [asterisk-users] network monitoring - triggering a phone call in asterisk   Matt Gibson
• [asterisk-users] Asterisk 1.6.1-beta1 Now Available   Asterisk Development Team
• [asterisk-users] DTMF issues...   Carlos Chavez
• [asterisk-users] DTMF Talk Off   Sam
• [asterisk-users] DTMF issues...   Moises Silva
• [asterisk-users] OT: Re: sip clients for smart phones?   Philipp Kempgen
• [asterisk-users] Improving the voice Quality,   Jai Rangi
• [asterisk-users] Improving the voice Quality,   Steve Totaro
• [asterisk-users] is DNS SRV enough for failover?   CunningPike
• [asterisk-users] Ultramonkey LVS + asterisk   Igor Hernandez
• [asterisk-users] Aastra phones and dns srv records   Tom Moore
• [asterisk-users] How to add Callee's name into Dial command ?   Olivier
• [asterisk-users] uninstalling zaptel   Hakan C
• [asterisk-users] 2 stage dialing and 484 address incomplete [SOLVED]   Olivier
• [asterisk-users] Zaptel-1.4.1 error cross compile   Hakan C
• [asterisk-users] Mimic SIP Events framework in Asterisk without coding ...   Olivier
• [asterisk-users] OT: Re: sip clients for smart phones?   Tarek Sawah
• [asterisk-users] Mimic SIP Events framework in Asterisk without coding ...   Tzafrir Cohen
• [asterisk-users] sip clients for smart phones?   Mark Hamilton
• [asterisk-users] IAX denial of service   Guillermo V. Salas
• [asterisk-users] Vitelity Asterisk configuration help   Stephen Reese
• [asterisk-users] voicemail quota   tic tac
• [asterisk-users] Improving the voice Quality,   Alex Balashov
• [asterisk-users] Music on hold for sub tenants   carl Lougher
• [asterisk-users] Asterisk Load Balancing   John D
• [asterisk-users] Asterisk Load Balancing   Darren Sessions
• [asterisk-users] Asterisk Load Balancing   Alex Balashov
• [asterisk-users] Asterisk Load Balancing   Darren Sessions
• [asterisk-users] How to add Callee's name into Dial command ?   Mr Shunz
• [asterisk-users] IAX denial of service   Tilghman Lesher
• [asterisk-users] How to add Callee's name into Dial command ?   Olivier
• [asterisk-users] sip clients for smart phones?   Andrew Kohlsmith (lists)
• [asterisk-users] OT: Re: sip clients for smart phones?   Andrew Kohlsmith (lists)
• [asterisk-users] Mimic SIP Events framework in Asterisk without coding ...   Olivier
• [asterisk-users] Zaptel-1.4.1 error cross compile   satish patel
• [asterisk-users] Zaptel-1.4.1 error cross compile   Tzafrir Cohen
• [asterisk-users] Zaptel-1.4.1 error cross compile   Tzafrir Cohen
• [asterisk-users] OT: text/plain (was: Re: Re: sip clients for smart phones?)   Philipp Kempgen
• [asterisk-users] sip clients for smart phones?   Grygoriy Dobrovolskyy
• [asterisk-users] OT: text/plain (was: Re: Re: sip clients for smart phones?)   Andrew Kohlsmith (lists)
• [asterisk-users] OT: text/plain   Philipp Kempgen
• [asterisk-users] Zaptel-1.4.1 error cross compile   satish patel
• [asterisk-users] asterisk, phpagi and singleton   Giedrius Augys
• [asterisk-users] asterisk, phpagi and singleton   Philipp Kempgen
• [asterisk-users] asterisk, phpagi and singleton   Steve Edwards
• [asterisk-users] OT: text/plain   Mark Hamilton
• [asterisk-users] OT: text/plain   SIP
• [asterisk-users] OT: headsets   Bill Michaelson
• [asterisk-users] Is SIPPEER curcalls working for you ? (was: Ongoing calls with SIPPEER, curcalls)   Doug Lytle
• [asterisk-users] MWI with Siemens Gigaset S450IP   robert.boardman at gmail.com
• [asterisk-users] Zaptel-1.4.1 error cross compile   Satish Patel
• [asterisk-users] no per mailbox imapfolder override? wow.   Brian J. Murrell
• [asterisk-users] Music on hold for sub tenants   Andrew Joakimsen
• [asterisk-users] t1 cards   Andrew Joakimsen
• [asterisk-users] cisco VAD and Asterisk recordings   Andrew Joakimsen
• [asterisk-users] MS Exchange IMAP Voicemail   Andrew Joakimsen
• [asterisk-users] MS Exchange IMAP Voicemail   David Backeberg
• [asterisk-users] MS Exchange IMAP Voicemail   Andrew Joakimsen
• [asterisk-users] Dial out DAHDI Channel?   Jim Duda
• [asterisk-users] t1 cards   Eric Fort
• [asterisk-users] Dial out DAHDI Channel?   Jim Duda
• [asterisk-users] Dial out DAHDI Channel?   Jim Duda
• [asterisk-users] asteriskt38.com   Andrew Joakimsen
• [asterisk-users] asteriskt38.com   John Faubion
• [asterisk-users] asterisk, phpagi and singleton   Alex Balashov
• [asterisk-users] sip clients for smart phones?   Dinesh Nair
• [asterisk-users] Zaptel-1.4.1 error cross compile   Alan Lord
• [asterisk-users] Fwd: Fonolo: Visually Navigate & Dial IVR Phone Menus in Web/Mobile Browser   Babcock, Michael Alex
• [asterisk-users] Alarm events + asterisk dies   Roberts Klotins
• [asterisk-users] R key with Siemens Gigaset IP (was MWI with Siemens Gigaset S450IP)   Olivier
• [asterisk-users] MS Exchange IMAP Voicemail   Olivier
• [asterisk-users] MWI with Siemens Gigaset S450IP   Olivier
• [asterisk-users] asterisk, phpagi and singleton   Giedrius Augys
• [asterisk-users] asterisk, phpagi and singleton   Alex Balashov
• [asterisk-users] Tribox   broadband Voice
• [asterisk-users] queue members randomly become paused after upgrade to Asterisk 1.4   Giorgio Incantalupo
• [asterisk-users] Ok message   michel freiha
• [asterisk-users] Tribox   Steven Howes
• [asterisk-users] asteriskt38.com   Atis Lezdins
• [asterisk-users] asteriskt38.com   Daniel Ferenci
• [asterisk-users] Zaptel-1.4.1 error cross compile   satish patel
• [asterisk-users] Tribox   Tarek Sawah
• [asterisk-users] Zaptel-1.4.1 error cross compile   Alan Lord
• [asterisk-users] OT: headsets   Jay R. Ashworth
• [asterisk-users] How to implement Ringing through a sound card for overhead paging   Robert Augustyn
• [asterisk-users] Conneting Asterisk to Swyx pri   Geraint Lee
• [asterisk-users] AEL and swap from macros to contexts   Pavel Jezek
• [asterisk-users] Conneting Asterisk to Swyx pri   Gordon Henderson
• [asterisk-users] "No route to destination" error   Martin Seebach
• [asterisk-users] How to implement Ringing through a sound card for overhead paging   Doug Lytle
• [asterisk-users] Conneting Asterisk to Swyx pri   Geraint Lee
• [asterisk-users] Hook Flash   Lucas Alvarez
• [asterisk-users] PoE switch recommendations?   Ken D'Ambrosio
• [asterisk-users] PoE switch recommendations?   Geraint Lee
• [asterisk-users] PoE switch recommendations?   Singer Wang
• [asterisk-users] PoE switch recommendations?   Chris Bagnall
• [asterisk-users] Semi OT: Global Crossing   Ken Williams
• [asterisk-users] PoE switch recommendations?   Gordon Henderson
• [asterisk-users] PoE switch recommendations?   Geraint Lee
• [asterisk-users] Hook Flash   Jeff Peeler
• [asterisk-users] PoE switch recommendations?   David Gibbons
• [asterisk-users] Tribox   Guillermo Salas M.
• [asterisk-users] Tribox   Guillermo Salas M.
• [asterisk-users] PoE switch recommendations?   Alexander Lopez
• [asterisk-users] PoE switch recommendations?   David Gibbons
• [asterisk-users] PoE switch recommendations?   Robert Augustyn
• [asterisk-users] How to implement Ringing through a sound card for overhead paging   Robert Augustyn
• [asterisk-users] AEL and swap from macros to contexts   Atis Lezdins
• [asterisk-users] AEL and swap from macros to contexts   Pavel Jezek
• [asterisk-users] How to implement Ringing through a sound card for overhead paging   Doug Lytle
• [asterisk-users] OT: headsets   Bill Michaelson
• [asterisk-users] AEL and swap from macros to contexts   Kevin P. Fleming
• [asterisk-users] asterisk, phpagi and singleton   Steve Edwards
• [asterisk-users] OT: headsets   Michael Graves
• [asterisk-users] asteriskt38.com   Andrew Joakimsen
• [asterisk-users] PoE switch recommendations?   David Gibbons
• [asterisk-users] PoE switch recommendations?   Jonathan C. Bailey
• [asterisk-users] PoE switch recommendations?   Norman Franke
• [asterisk-users] Nice recording interfaces   Chris Bagnall
• [asterisk-users] PoE switch recommendations?   Jerry Jones
• [asterisk-users] cdr,gsm file format   Sriram
• [asterisk-users] Missing 'Queue' Application in 1.4.21.2   Josiah Bryan
• [asterisk-users] asteriskt38.com   Atis Lezdins
• [asterisk-users] How to implement Ringing through a sound card for overhead paging   Robert Augustyn
• [asterisk-users] Missing 'Queue' Application in 1.4.21.2   Andres
• [asterisk-users] OT: headsets   Jay R. Ashworth
• [asterisk-users] PoE switch recommendations?   Jay R. Ashworth
• [asterisk-users] Missing 'Queue' Application in 1.4.21.2   Josiah Bryan
• [asterisk-users] Help with remote users   Steve Anness
• [asterisk-users] Asterisk 1.4 or 1.6 ???   Alejandro Facultad
• [asterisk-users] PoE switch recommendations?   Karl Fife
• [asterisk-users] Help with remote users   Jerry Jones
• [asterisk-users] PoE switch recommendations?   Dave Walker
• [asterisk-users] asteriskt38.com   Andrew Joakimsen
• [asterisk-users] cdr,gsm file format   Dave Walker
• [asterisk-users] Help with remote users   Steve Anness
• [asterisk-users] Asterisk 1.4 or 1.6 ???   Gordon Henderson
• [asterisk-users] Matching *, + and # in the dialplan   Karl Fife
• [asterisk-users] How to implement Ringing through a sound card for overhead paging   Doug Lytle
• [asterisk-users] Asterisk 1.4 or 1.6 ???   Brendan Martens
• [asterisk-users] Tribox   Tarek Sawah
• [asterisk-users] Tribox   Ron Stephan
• [asterisk-users] Asterisk 1.4 or 1.6 ???   Jason Aarons (US)
• [asterisk-users] ldap usage in 1.6.0   Brendan Martens
• [asterisk-users] Tribox   Tarek Sawah
• [asterisk-users] No reply to our critical packet   Andrew Joakimsen
• [asterisk-users] t1 cards   Nick B.
• [asterisk-users] Tribox   Guillermo Salas M.
• [asterisk-users] Vitelity Asterisk configuration help   Darren Severino
• [asterisk-users] AEL and swap from macros to contexts   Steve Murphy
• [asterisk-users] No reply to our critical packet   SIP
• [asterisk-users] ldap usage in 1.6.0   Olivier
• [asterisk-users] ldap usage in 1.6.0   Brendan Martens
• [asterisk-users] No reply to our critical packet   Andrew Joakimsen
• [asterisk-users] PoE switch recommendations?   Andrew Joakimsen
• [asterisk-users] PoE switch recommendations?   Karl Fife
• [asterisk-users] No reply to our critical packet   Atis Lezdins
• [asterisk-users] PoE switch recommendations?   Daniel Hazelbaker
• [asterisk-users] Asterisk/AJAM Console   Forrest Beck
• [asterisk-users] Asterisk 1.4 or 1.6 ???   Al Baker
• [asterisk-users] regcontext   Nhadie
• [asterisk-users] Vitelity Asterisk configuration help   Stephen Reese
• [asterisk-users] AEL and swap from macros to contexts   Pavel Jezek
• [asterisk-users] can't find mysqlclient : asterisk-addons-1.6.0   Klaverstyn, David C
• [asterisk-users] asteriskt38.com   Daniel Ferenci
• [asterisk-users] Help with remote users   Andrew Joakimsen
• [asterisk-users] can't find mysqlclient : asterisk-addons-1.6.0   Stefan Schmidt
• [asterisk-users] can't find mysqlclient : asterisk-addons-1.6.0   Lee, John (Sydney)
• [asterisk-users] Asterisk 1.4 or 1.6 ???   Giorgio Incantalupo
• [asterisk-users] How to implement Ringing through a sound card for overhead paging   Julien Claassen
• [asterisk-users] asterisk, phpagi and singleton   Giedrius Augys
• [asterisk-users] automatic call pickup   Vieri
• [asterisk-users] automatic call pickup   Pavel Jezek
• [asterisk-users] How to implement Ringing through a sound card for overhead paging   Robert Augustyn
• [asterisk-users] automatic call pickup   Anselm Martin Hoffmeister
• [asterisk-users] AEL and swap from macros to contexts   Atis Lezdins
• [asterisk-users] Cisco 7906g & SIP   Sasa
• [asterisk-users] Creating Asterisk Binary Package   Dobry Dobrev
• [asterisk-users] call center   ould ould
• [asterisk-users] Cisco 7906g & SIP   Duncan Turnbull
• [asterisk-users] PoE switch recommendations?   Chris Bagnall
• [asterisk-users] AEL and swap from macros to contexts   Pavel Jezek
• [asterisk-users] AEL and swap from macros to contexts   Atis Lezdins
• [asterisk-users] regcontext   Jared Smith
• [asterisk-users] PoE switch recommendations?   Christian Victor
• [asterisk-users] ldap usage in 1.6.0   Josiah Bryan
• [asterisk-users] OT: text/plain   Benny Amorsen
• [asterisk-users] ldap usage in 1.6.0   Brendan Martens
• [asterisk-users] Asterisk 1.4 or 1.6 ???   Benny Amorsen
• [asterisk-users] Cisco 7906g & SIP   Sasa
• [asterisk-users] Creating Asterisk Binary Package   Brendan Martens
• [asterisk-users] Efax from Agi script   Riccardo Cupardo
• [asterisk-users] Vitelity Asterisk configuration help   Stephen Reese
• [asterisk-users] include in the DAHDI system.conf file and chan_dahdi.conf   Jerry Geis
• [asterisk-users] include in the DAHDI system.conf file and chan_dahdi.conf   Tzafrir Cohen
• [asterisk-users] Help with remote users   Steve Anness
• [asterisk-users] OT: headsets   James Sneeringer
• [asterisk-users] Vitelity Asterisk configuration help   Stephen Reese
• [asterisk-users] Creating Asterisk Binary Package   Dobry Dobrev
• [asterisk-users] Help with remote users   Andres
• [asterisk-users] Vitelity Asterisk configuration help   Darren Severino
• [asterisk-users] Asterisk Callerid Help Needed   Max Alex
• [asterisk-users] PoE switch recommendations?   Daniel Hazelbaker
• [asterisk-users] automatic call pickup   Vieri
• [asterisk-users] Bad Destinations   Mr surfit
• [asterisk-users] Vitelity Asterisk configuration help   Roderick A. Anderson
• [asterisk-users] Vitelity Asterisk configuration help   Darren Severino
• [asterisk-users] How to implement Ringing through a sound card for overhead paging   Julien Claassen
• [asterisk-users] Conneting Asterisk to Swyx pri   Geraint Lee
• [asterisk-users] Setting up Asterisk   Wilton Helm
• [asterisk-users] Setting up Asterisk   satish patel
• [asterisk-users] Setting up Asterisk   satish patel
• [asterisk-users] Matching *, + and # in the dialplan   Tilghman Lesher
• [asterisk-users] cisco phones getting SIP 401 unauthorized   Jerry Geis
• [asterisk-users] cisco phones getting SIP 401 unauthorized   Jerry Geis
• [asterisk-users] How to implement Ringing through a sound card for overhead paging   Robert Augustyn
• [asterisk-users] cisco phones getting SIP 401 unauthorized   Matt Gibson
• [asterisk-users] How to implement Ringing through a sound card for overhead paging   Julien Claassen
• [asterisk-users] asteriskt38.com   Andrew Joakimsen
• [asterisk-users] How to implement Ringing through a sound card for overhead paging   Robert Augustyn
• [asterisk-users] asteriskt38.com   Daniel Ferenci
• [asterisk-users] How to implement Ringing through a sound card for overhead paging   Julien Claassen
• [asterisk-users] Question on screening calls / Question about the Dial g option   Mike
• [asterisk-users] Question on screening calls / Question about the Dial g option   Doug Lytle
• [asterisk-users] How to implement Ringing through a sound card for overhead paging   Robert Augustyn
• [asterisk-users] Question on screening calls / Question about the Dial g option   Mike
• [asterisk-users] cisco phones getting SIP 401 unauthorized   Jerry Geis
• [asterisk-users] cisco phones getting SIP 401 unauthorized   Matt Gibson
• [asterisk-users] Help with remote users   Steve Anness
• [asterisk-users] can't find mysqlclient : asterisk-addons-1.6.0   Klaverstyn, David C
• [asterisk-users] Question on screening calls / Question about the Dial g option   Doug Lytle
• [asterisk-users] help no ring on caller side   Nhadie
• [asterisk-users] "requested special control 20" ??   sean darcy
• [asterisk-users] changing passwords   Ken Zarifes
• [asterisk-users] can't find mysqlclient : asterisk-addons-1.6.0   Steve Totaro
• [asterisk-users] can't find mysqlclient : asterisk-addons-1.6.0   Philipp Kempgen
• [asterisk-users] can't find mysqlclient : asterisk-addons-1.6.0   Philipp Kempgen
• [asterisk-users] changing passwords   Philipp Kempgen
• [asterisk-users] changing passwords   Steve Totaro
• [asterisk-users] help no ring on caller side   Steve Totaro
• [asterisk-users] Question on screening calls / Question about the Dial g option   Steve Totaro
• [asterisk-users] Efax from Agi script   Andrew Joakimsen
• [asterisk-users] changing passwords   Ken Zarifes
• [asterisk-users] changing passwords   Ken Zarifes
• [asterisk-users] changing passwords   Andrew Joakimsen
• [asterisk-users] Help with remote users   Andrew Joakimsen
• [asterisk-users] can't find mysqlclient : asterisk-addons-1.6.0   Andrew Joakimsen
• [asterisk-users] help no ring on caller side   Andrew Joakimsen
• [asterisk-users] Matching *, + and # in the dialplan   Karl Fife
• [asterisk-users] automatic call pickup   Andrew Joakimsen
• [asterisk-users] help no ring on caller side   Nhadie
• [asterisk-users] registration limit   Nhadie
• [asterisk-users] Bad Destinations   Andrew Joakimsen
• [asterisk-users] Can't find the path to Phoneprov directory   hin lee
• [asterisk-users] registration limit   Andrew Joakimsen
• [asterisk-users] Matching *, + and # in the dialplan   Rob Hillis
• [asterisk-users] regcontext   Nhadie
• [asterisk-users] Zaptel -> DAHDI for dummies?   Remco Barendse
• [asterisk-users] automatic call pickup   Vieri
• [asterisk-users] Call-limit bug in 1.2 ?   Chris Bagnall
• [asterisk-users] automatic call pickup   Andrew Joakimsen
• [asterisk-users] How is automatic redial/callback when available implemented ?   Olivier
• [asterisk-users] SRV DNS failover - dial to proxy list   Matteo Piazza
• [asterisk-users] PoE switch recommendations?   Sigma Networks
• [asterisk-users] debugging hints in 1.6   Dr. Michael J. Chudobiak
• [asterisk-users] PoE switch recommendations?   Michael Graves
• [asterisk-users] cisco phones getting SIP 401 unauthorized   Jerry Geis
• [asterisk-users] Call-limit bug in 1.2 ?   Philipp Kempgen
• [asterisk-users] debugging hints in 1.6   Philipp Kempgen
• [asterisk-users] debugging hints in 1.6   Dr. Michael J. Chudobiak
• [asterisk-users] debugging hints in 1.6   Philipp Kempgen
• [asterisk-users] Matching *, + and # in the dialplan   Tilghman Lesher
• [asterisk-users] debugging hints in 1.6   Dr. Michael J. Chudobiak
• [asterisk-users] Matching *, + and # in the dialplan   Tilghman Lesher
• [asterisk-users] fastagi example   Giedrius Augys
• [asterisk-users] Mobile (cell) phone in a queue (but act like an agent   Gareth Llewellyn
• [asterisk-users] cisco phones getting SIP 401 unauthorized   Matt Gibson
• [asterisk-users] cisco phones getting SIP 401 unauthorized   Jerry Geis
• [asterisk-users] make func_realtime work like app_realtime (1.6)   Wesley Haut
• [asterisk-users] cisco phones getting SIP 401 unauthorized   David Gibbons
• [asterisk-users] Sample code fragement for subscribing to hints wanted.   Russell Brown
• [asterisk-users] cisco phones getting SIP 401 unauthorized   Jerry Geis
• [asterisk-users] Sample code fragement for subscribing to hints wanted.   Jared Smith
• [asterisk-users] make func_realtime work like app_realtime (1.6)   Rob Hillis
• [asterisk-users] PoE switch recommendations?   Drew Gibson
• [asterisk-users] make func_realtime work like app_realtime (1.6)   Tilghman Lesher
• [asterisk-users] Sample code fragement for subscribing to hints wanted.   Philipp Kempgen
• [asterisk-users] Help with IAX Trunking   Steve Anness
• [asterisk-users] make func_realtime work like app_realtime (1.6)   BJ Weschke
• [asterisk-users] Vitelity Asterisk configuration help   Stephen Reese
• [asterisk-users] No reply to our critical packet   Andrew Joakimsen
• [asterisk-users] Sip Trunking   Brent Davidson
• [asterisk-users] Update (IAX Trunking Help)   Steve Anness
• [asterisk-users] Tribox   Grygoriy Dobrovolskyy
• [asterisk-users] Update (IAX Trunking Help)   Alejandro Kauffmann
• [asterisk-users] conntrack_sip, iptables, and asterisk   OCG Technical Support
• [asterisk-users] conntrack_sip, iptables, and asterisk   Alex Balashov
• [asterisk-users] make func_realtime work like app_realtime (1.6)   Rob Hillis
• [asterisk-users] retransmitting NAT   Nhadie
• [asterisk-users] Menu for call forwarding or voicemail   Stephen Reese
• [asterisk-users] Question on using DMZ   C. Savinovich
• [asterisk-users] Record name for conference...   Carlos Chavez
• [asterisk-users] Record name for conference...   Dean Collins
• [asterisk-users] Record name for conference...   Ming Yong
• [asterisk-users] Record name for conference...   Alejandro Kauffmann
• [asterisk-users] Interpreting Asterisk Logs   Darren Murphy
• [asterisk-users] Friday Oct 10 @12 Noon EDT VoIP Users Conference   randulo
• [asterisk-users] Howto analyze concurrent ISDN channel usage   Patrick
• [asterisk-users] H323   michel freiha
• [asterisk-users] H323   broadband Voice
• [asterisk-users] Interrupt Asterisk's SayDigits()   broadband Voice
• [asterisk-users] Ringtones for the console   Julien Claassen
• [asterisk-users] Howto analyze concurrent ISDN channel usage   Stefan Schmidt
• [asterisk-users] Ringtones for the console   Julien Claassen
• [asterisk-users] Hang up detection with TDM400P and Telewest/Virgin Media line   Mike
• [asterisk-users] Cisco 7906g & SIP   Sasa
• [asterisk-users] Asterisk restarts on call parking   David Harty
• [asterisk-users] Cisco 7906g & SIP   Stefan Gofferje
• [asterisk-users] Cisco 7906g & SIP   Sasa
• [asterisk-users] Cisco 7906g & SIP   David Gibbons
• [asterisk-users] Ringtones for the console   Robert Augustyn
• [asterisk-users] Hang up detection with TDM400P and Telewest/Virgin Media line   Luis Morales
• [asterisk-users] Howto analyze concurrent ISDN channel usage   Luis Morales
• [asterisk-users] Cisco 7906g & SIP   Sasa
• [asterisk-users] Question on using DMZ   Atis Lezdins
• [asterisk-users] Cisco 7906g & SIP   David Gibbons
• [asterisk-users] Inbound Calls From AS5300 Rejected with 488 Code   Ketema Harris
• [asterisk-users] cisco phones getting SIP 401 unauthorized   peder at networkoblivion.com
• [asterisk-users] cisco phones getting SIP 401 unauthorized   peder at networkoblivion.com
• [asterisk-users] Inbound Calls From AS5300 Rejected with 488 Code   Alex Balashov
• [asterisk-users] make func_realtime work like app_realtime (1.6)   Tilghman Lesher
• [asterisk-users] Inbound Calls From AS5300 Rejected with 488 Code   Ketema Harris
• [asterisk-users] Inbound Calls From AS5300 Rejected with 488 Code   Alex Balashov
• [asterisk-users] Record name for conference...   Fred Posner
• [asterisk-users] Asterisk-Panasonic TDA 600 error   Javier Prieto Gomez
• [asterisk-users] Cisco 7906g & SIP   Sasa
• [asterisk-users] Cisco 7906g & SIP   David Gibbons
• [asterisk-users] Tribox   Tarek Sawah
• [asterisk-users] Update (IAX Trunking Help)   Steve Anness
• [asterisk-users] cisco phones getting SIP 401 unauthorized - solved   Jerry Geis
• [asterisk-users] Inbound Calls From AS5300 Rejected with 488 Code   Kristian Kielhofner
• [asterisk-users] Inbound Calls From AS5300 Rejected with 488 Code   Norman Franke
• [asterisk-users] Hang up detection with TDM400P and Telewest/Virgin Media line   Gordon Henderson
• [asterisk-users] Update (IAX Trunking Help)   Michiel van Baak
• [asterisk-users] Hang up detection with TDM400P and Telewest/Virgin Media line   Luis Morales
• [asterisk-users] ATA hangs up with fax detection on...   Carlos Chavez
• [asterisk-users] asterisk-users Digest, Vol 51, Issue 26   Russell Brown
• [asterisk-users] Hang up detection with TDM400P and Telewest/Virgin Media line   Gordon Henderson
• [asterisk-users] Hang up detection with TDM400P and Telewest/Virgin Media line   robb
• [asterisk-users] Update (IAX Trunking Help)   Steve Anness
• [asterisk-users] Sample code fragement for subscribing to hints wanted (was: Re: asterisk-users Digest, Vol 51, Issue 26)   Philipp Kempgen
• [asterisk-users] Asterisk 1.6.0 CDR billsec and duration not working from h extension   Eric Chamberlain
• [asterisk-users] SIP problems?   Kristian Kielhofner
• [asterisk-users] SIP problems?   Alex Balashov
• [asterisk-users] Asterisk 1.6.0 CDR billsec and duration not working from h extension   Senad Jordanovic
• [asterisk-users] Hang up detection with TDM400P and Telewest/Virgin Media line   Mike
• [asterisk-users] Transfer/Park Question.   Brent Davidson
• [asterisk-users] Cisco 7960 sccp, Skinny and 1.4   Wayne
• [asterisk-users] Asterisk 1.6.0 CDR billsec and duration not working from h extension   Steve Murphy
• [asterisk-users] DAHDI aaaaaaaaaaaaaaarrrrrrrrrghhhhhhhhh :((((   Remco Barendse
• [asterisk-users] DAHDI aaaaaaaaaaaaaaarrrrrrrrrghhhhhhhhh :((((   Steve Totaro
• [asterisk-users] DAHDI aaaaaaaaaaaaaaarrrrrrrrrghhhhhhhhh :((((   Alex Balashov
• [asterisk-users] Transfer/Park Question.   Daniel Hazelbaker
• [asterisk-users] Hang up detection with TDM400P and Telewest/Virgin Media line   Luis Morales
• [asterisk-users] Polycom 330 not dialing 4 digit extensions beginning with 11xx   Ed DeHart
• [asterisk-users] Polycom 330 not dialing 4 digit extensions beginning with 11xx   Michael Graves
• [asterisk-users] Polycom 330 not dialing 4 digit extensions beginning with 11xx   Philipp Kempgen
• [asterisk-users] Polycom 330 not dialing 4 digit extensions beginning with 11xx   hin lee
• [asterisk-users] Asterisk CDR Analyser   Klaverstyn, David C
• [asterisk-users] Asterisk-Panasonic TDA 600 error   C F
• [asterisk-users] DAHDI aaaaaaaaaaaaaaarrrrrrrrrghhhhhhhhh :((((   Sean Bright
• [asterisk-users] DAHDI aaaaaaaaaaaaaaarrrrrrrrrghhhhhhhhh :((((   sean darcy
• [asterisk-users] Polycom 330 not dialing 4 digit extensions beginning with 11xx   Luis Morales
• [asterisk-users] Asterisk CDR Analyser   Sean Bright
• [asterisk-users] Update (IAX Trunking Help)   Alejandro Kauffmann
• [asterisk-users] DAHDI aaaaaaaaaaaaaaarrrrrrrrrghhhhhhhhh :((((   Steve Totaro
• [asterisk-users] DAHDI aaaaaaaaaaaaaaarrrrrrrrrghhhhhhhhh :((((   Tzafrir Cohen
• [asterisk-users] Asterisk CDR Analyser   Klaverstyn, David C
• [asterisk-users] 1.6.0.1 ??   sean darcy
• [asterisk-users] 1.6.0.1 ??   Brendan Martens
• [asterisk-users] DAHDI aaaaaaaaaaaaaaarrrrrrrrrghhhhhhhhh :((((   Brendan Martens
• [asterisk-users] DAHDI aaaaaaaaaaaaaaarrrrrrrrrghhhhhhhhh :((((   Anthony Messina
• [asterisk-users] Multicore process   Setta Punpeng
• [asterisk-users] DAHDI aaaaaaaaaaaaaaarrrrrrrrrghhhhhhhhh :((((   Tzafrir Cohen
• [asterisk-users] How to barge Inbound calls   amit salunkhe
• [asterisk-users] Compile logger-mysql.c with UNDEFINED REF to `mysql_error'   Lee, John (Sydney)
• [asterisk-users] DAHDI aaaaaaaaaaaaaaarrrrrrrrrghhhhhhhhh :((((   Remco Barendse
• [asterisk-users] DAHDI aaaaaaaaaaaaaaarrrrrrrrrghhhhhhhhh :((((   Remco Barendse
• [asterisk-users] Compile logger-mysql.c with UNDEFINED REF to `mysql_error'   Atis Lezdins
• [asterisk-users] Be aware of callcheap.com and Mike Low - It is scam   Zeeshan Zakaria
• [asterisk-users] How to enable inbound CLI for X-Lite/Asterisk phone.   Phil Reynolds
• [asterisk-users] Compile logger-mysql.c with UNDEFINED REF to`mysql_error'   Lee, John (Sydney)
• [asterisk-users] How to enable inbound CLI for X-Lite/Asterisk phone.   Syed Nasruddin
• [asterisk-users] DAHDI aaaaaaaaaaaaaaarrrrrrrrrghhhhhhhhh :((((   David A. Bandel
• [asterisk-users] Hang up detection with TDM400P and Telewest/Virgin Media line   Gordon Henderson
• [asterisk-users] Cisco 7960 sccp, Skinny and 1.4   David Gibbons
• [asterisk-users] Unable to create channel of type 'DAHDI' (cause 0 - Unknown)   Jim Duda
• [asterisk-users] Unable to create channel of type 'DAHDI' (cause 0 - Unknown)   Brendan Martens
• [asterisk-users] Unable to create channel of type 'DAHDI' (cause 0 - Unknown)   Kevin P. Fleming
• [asterisk-users] Unable to create channel of type 'DAHDI' (cause 0 - Unknown)   Brendan Martens
• [asterisk-users] Unable to create channel of type 'DAHDI' (cause 0 - Unknown)   Jim Duda
• [asterisk-users] Asterisk SIP calls stop working having more than 300 calls (more than 600 channels)   Juan Rodríguez
• [asterisk-users] DAHDI aaaaaaaaaaaaaaarrrrrrrrrghhhhhhhhh :((((   Shaun Ruffell
• [asterisk-users] DAHDI aaaaaaaaaaaaaaarrrrrrrrrghhhhhhhhh :((((   Matthew Fredrickson
• [asterisk-users] Howto analyze concurrent ISDN channel usage   Jay Taylor
• [asterisk-users] Asterisk SIP calls stop working having more than 300 calls (more than 600 channels)   Kristian Kielhofner
• [asterisk-users] Asterisk SIP calls stop working having more than 300 calls (more than 600 channels)   Juan Rodríguez
• [asterisk-users] Question about echo cancelation   Olivier
• [asterisk-users] softclient for customized apps like a call center?   JD
• [asterisk-users] Asterisk SIP calls stop working having more than 300 calls (more than 600 channels)   Kristian Kielhofner
• [asterisk-users] Block Caller ID   Sriram
• [asterisk-users] Got event 17 (Polarity Reversal)...   Jim Duda
• [asterisk-users] Transfer/Park Question.   Brent Davidson
• [asterisk-users] Got event 17 (Polarity Reversal)...   Eric "ManxPower" Wieling
• [asterisk-users] Be aware of callcheap.com and Mike Low - It is scam   Steve Totaro
• [asterisk-users] Question about echo cancelation   Eric "ManxPower" Wieling
• [asterisk-users] Howto analyze concurrent ISDN channel usage   Patrick
• [asterisk-users] Howto analyze concurrent ISDN channel usage   Patrick
• [asterisk-users] Got event 17 (Polarity Reversal)...   Tzafrir Cohen
• [asterisk-users] Asterisk SIP calls stop working having more than 300 calls (more than 600 channels)   Tzafrir Cohen
• [asterisk-users] Howto analyze concurrent ISDN channel usage   Patrick
• [asterisk-users] Transfer/Park Question.   Doug Lytle
• [asterisk-users] Cisco 7960 sccp, Skinny and 1.4   Michiel van Baak
• [asterisk-users] Got event 17 (Polarity Reversal)...   Wilton Helm
• [asterisk-users] Asterisk SIP calls stop working having more than 300 calls (more than 600 channels)   Juan Rodríguez
• [asterisk-users] Got event 17 (Polarity Reversal)...   Dan Peters
• [asterisk-users] Got event 17 (Polarity Reversal)...   Doug Lytle
• [asterisk-users] Menu for call forwarding or voicemail   Stephen Reese
• [asterisk-users] Transfer/Park Question.   Brent Davidson
• [asterisk-users] Transfer/Park Question.   Doug Lytle
• [asterisk-users] Budge Tones pick up wrong calls   Paul Douglas Franklin
• [asterisk-users] Transfer/Park Question.   Daniel Hazelbaker
• [asterisk-users] Cisco 7960 sccp, Skinny and 1.4   Wayne
• [asterisk-users] Transfer/Park Question.   Brent Davidson
• [asterisk-users] Transfer/Park Question.   Brent Davidson
• [asterisk-users] Help need for debuging the core file.   gary
• [asterisk-users] is there a way   Babcock, Michael Alex
• [asterisk-users] Configuring Bandwidth.com SIP trunks to prevent one-way audio   Kurt Knudsen
• [asterisk-users] Cisco 7960 sccp, Skinny and 1.4   Michiel van Baak
• [asterisk-users] Help need for debuging the core file.   Tilghman Lesher
• [asterisk-users] Configuring Bandwidth.com SIP trunks to prevent one-way audio   Steve Totaro
• [asterisk-users] Be aware of callcheap.com and Mike Low - It is scam   Zeeshan Zakaria
• [asterisk-users] Caller ID service and the ethernet stucking   bilal ghayyad
• [asterisk-users] Got event 17 (Polarity Reversal)...   Jim Duda
• [asterisk-users] is there a way   Brent Davidson
• [asterisk-users] Be aware of callcheap.com and Mike Low - It is scam   Steve Totaro
• [asterisk-users] is there a way   Steve Totaro
• [asterisk-users] Configuring Bandwidth.com SIP trunks to prevent one-way audio   Kurt Knudsen
• [asterisk-users] Configuring Bandwidth.com SIP trunks to prevent one-way audio   Steve Totaro
• [asterisk-users] Menu for call forwarding or voicemail   Stephen Reese
• [asterisk-users] Budge Tones pick up wrong calls   Alex Balashov
• [asterisk-users] is there a way   Babcock, Michael Alex
• [asterisk-users] is there a way   Babcock, Michael Alex
• [asterisk-users] Configuring Bandwidth.com SIP trunks to prevent one-way audio   Kurt Knudsen
• [asterisk-users] Configuring Bandwidth.com SIP trunks to prevent one-way audio   Steve Totaro
• [asterisk-users] Mitel 5220 firmware   Bro
• [asterisk-users] is there a way   Eric Fort
• [asterisk-users] Budge Tones pick up wrong calls   Andrew Joakimsen
• [asterisk-users] is there a way   Steve Totaro
• [asterisk-users] Got event 17 (Polarity Reversal)...   Tzafrir Cohen
• [asterisk-users] is there a way   Tzafrir Cohen
• [asterisk-users] Matching *, + and # in the dialplan   Benny Amorsen
• [asterisk-users] Sip Trunking   Benny Amorsen
• [asterisk-users] Question about echo cancelation   Olivier
• [asterisk-users] Got event 17 (Polarity Reversal)...   Jim Duda
• [asterisk-users] is there a way   Bill Michaelson
• [asterisk-users] Got event 17 (Polarity Reversal)...   Jorge Mendoza
• [asterisk-users] Asterisk For Windows ?   Meftah Tayeb
• [asterisk-users] Is there a way to test SIP credentials without making a call?   Eric Chamberlain
• [asterisk-users] Is there a way to test SIP credentials withoutmaking a call?   Meftah Tayeb
• [asterisk-users] Asterisk For Windows ?   Chris Rowson
• [asterisk-users] Question about echo cancelation   Eric "ManxPower" Wieling
• [asterisk-users] Is there a way to test SIP credentials withoutmaking a call?   Eric Chamberlain
• [asterisk-users] Got event 17 (Polarity Reversal)...   Gordon Henderson
• [asterisk-users] Asterisk For Windows ?   Tzafrir Cohen
• [asterisk-users] Is there a way to test SIP credentialswithoutmaking a call?   Meftah Tayeb
• [asterisk-users] Asterisk For Windows ?   Meftah Tayeb
• [asterisk-users] Asterisk For Windows ?   Meftah Tayeb
• [asterisk-users] Asterisk For Windows ?   Tzafrir Cohen
• [asterisk-users] 1 second delay when connecting calls   nrbwpi at gmail.com
• [asterisk-users] Asterisk For Windows ?   Roderick A. Anderson
• [asterisk-users] Asterisk For Windows ?   Babcock, Michael Alex
• [asterisk-users] Asterisk 1.6.1 + openais   Edgar Guadamuz
• [asterisk-users] 1 second delay when connecting calls   Eric "ManxPower" Wieling
• [asterisk-users] Question about echo cancelation   Olivier
• [asterisk-users] Configuring Bandwidth.com SIP trunks to prevent one-way audio   Kurt Knudsen
• [asterisk-users] Question about echo cancelation   Eric "ManxPower" Wieling
• [asterisk-users] Asterisk For Windows ?   Guillermo Salas M.
• [asterisk-users] Is there a way to test SIP credentials withoutmaking a call?   Rob Hillis
• [asterisk-users] Is there a way to test SIP credentials withoutmaking a call?   Eric Chamberlain
• [asterisk-users] cli commands missing   Eric Fort
• [asterisk-users] Is there a way to test SIP credentials withoutmaking a call?   Rob Hillis
• [asterisk-users] cli commands missing   Tzafrir Cohen
• [asterisk-users] cli commands missing   Eric Fort
• [asterisk-users] cli commands missing   Andrew Joakimsen
• [asterisk-users] cli commands missing   Michiel van Baak
• [asterisk-users] Asterisk For Windows ?   Meftah Tayeb
• [asterisk-users] Asterisk For Windows ?   Meftah Tayeb
• [asterisk-users] Asterisk For Windows ?   Meftah Tayeb
• [asterisk-users] Asterisk For Windows ?   Tzafrir Cohen
• [asterisk-users] Question about echo cancelation   Olivier
• [asterisk-users] Asterisk For Windows ?   Meftah Tayeb
• [asterisk-users] 1 second delay when connecting calls   Vieri
• [asterisk-users] Hang up detection with TDM400P and Telewest/Virgin Media line   Gordon Henderson
• [asterisk-users] Asterisk For Windows ?   Chris Rowson
• [asterisk-users] Asterisk For Windows ?   Meftah Tayeb
• [asterisk-users] Conneting Asterisk to Swyx pri   Geraint Lee
• [asterisk-users] One Way Audio Problem   GNUbie
• [asterisk-users] One Way Audio Problem   Tzafrir Cohen
• [asterisk-users] One Way Audio Problem   Gordon Henderson
• [asterisk-users] Caller ID sip trunk   robb
• [asterisk-users] setup for fax machine   sean darcy
• [asterisk-users] setup for fax machine   Doug Lytle
• [asterisk-users] setup for fax machine   Andrew Joakimsen
• [asterisk-users] Menu for call forwarding or voicemail   Stephen Reese
• [asterisk-users] cli commands missing   Eric Fort
• [asterisk-users] One Way Audio Problem   GNUbie
• [asterisk-users] One Way Audio Problem   GNUbie
• [asterisk-users] Interpreting Asterisk Logs   Roberts Klotins
• [asterisk-users] setup for fax machine   Gordon Henderson
• [asterisk-users] realtime queue_log to mySQL backport to 1.4   Lee, John (Sydney)
• [asterisk-users] Question about echo cancelation   Olivier
• [asterisk-users] One Way Audio Problem   Tzafrir Cohen
• [asterisk-users] Question about echo cancelation   Olivier
• [asterisk-users] Question about echo cancelation   Olivier
• [asterisk-users] realtime queue_log to mySQL backport to 1.4   Atis Lezdins
• [asterisk-users] ERROR:Failed to create H323 listener   Ali Jawad
• [asterisk-users] echo over digital line   Vieri
• [asterisk-users] Installing CdrTool on free bsd   Mohit Saxena
• [asterisk-users] One Way Audio Problem   GNUbie
• [asterisk-users] Cisco 7906g & SIP   Sasa
• [asterisk-users] Cisco 7906g & SIP   David Gibbons
• [asterisk-users] One Way Audio Problem   Steve Totaro
• [asterisk-users] Cisco 7906g & SIP   Sasa
• [asterisk-users] Unknown call every 30 minutes on the dot.   Kurt Knudsen
• [asterisk-users] Cisco 7906g & SIP   David Gibbons
• [asterisk-users] How to enable inbound CLI forX-Lite/Asterisk phone.   Phil Reynolds
• [asterisk-users] One Way Audio Problem   GNUbie
• [asterisk-users] MOH Bad   peder at networkoblivion.com
• [asterisk-users] One Way Audio Problem   Steve Totaro
• [asterisk-users] One Way Audio Problem   Norman Franke
• [asterisk-users] IP 650 Sidecar   Jeremy Mann
• [asterisk-users] Need help for debuging   gary
• [asterisk-users] How to enable inboundCLI forX-Lite/Asterisk phone.   Phil Reynolds
• [asterisk-users] How to enable inbound CLI forX-Lite/Asterisk phone.   Syed Nasruddin
• [asterisk-users] IP 650 Sidecar   James Sneeringer
• [asterisk-users] Need help for debuging   Tilghman Lesher
• [asterisk-users] One Way Audio Problem   GNUbie
• [asterisk-users] One Way Audio Problem   GNUbie
• [asterisk-users] setup for fax machine   Anthony Messina
• [asterisk-users] 1 second delay when connecting calls   nrbwpi at gmail.com
• [asterisk-users] One Way Audio Problem   Jeff LaCoursiere
• [asterisk-users] How to enable inboundCLI forX-Lite/Asterisk phone.   Syed Nasruddin
• [asterisk-users] One Way Audio Problem   Daniel Hazelbaker
• [asterisk-users] Need help for debuging   gary
• [asterisk-users] One Way Audio Problem   Karsten Wemheuer
• [asterisk-users] Need help for debuging   Philipp Kempgen
• [asterisk-users] Need help for debuging   Tilghman Lesher
• [asterisk-users] Is there a way to test SIP credentials withoutmaking a call?   Eric Chamberlain
• [asterisk-users] setup for fax machine   Gordon Henderson
• [asterisk-users] cli commands missing   Karsten Wemheuer
• [asterisk-users] Is there a way to test SIP credentials withoutmaking a call?   Kristian Kielhofner
• [asterisk-users] AGI Hangup   Gnu Devel
• [asterisk-users] Softphone Framework or Libraries   Ricardo Melendez
• [asterisk-users] Text messaging and Asterisk   C. Savinovich
• [asterisk-users] setup for fax machine   Steve Totaro
• [asterisk-users] Softphone Framework or Libraries   Dean Collins
• [asterisk-users] Text messaging and Asterisk   Pavel Jezek
• [asterisk-users] ifbyphone/google analytics   Dean Collins
• [asterisk-users] Text messaging and Asterisk   C. Savinovich
• [asterisk-users] Softphone Framework or Libraries   Michael Graves
• [asterisk-users] Cisco 7906g & SIP   Peter Mueller
• [asterisk-users] ifbyphone/google analytics   Eric Chamberlain
• [asterisk-users] Text messaging and Asterisk   Eric Chamberlain
• [asterisk-users] Softphone Framework or Libraries   Senad Jordanovic
• [asterisk-users] Text messaging and Asterisk   Kurt Knudsen
• [asterisk-users] Text messaging and Asterisk   C. Savinovich
• [asterisk-users] Text messaging and Asterisk   Matt Gibson
• [asterisk-users] Got event 17 (Polarity Reversal)...   Jorge Mendoza
• [asterisk-users] ISDN   Wilton Helm
• [asterisk-users] ISDN   Michael Graves
• [asterisk-users] Got event 17 (Polarity Reversal)...   Gordon Henderson
• [asterisk-users] Matching *, + and # in the dialplan   Steve Murphy
• [asterisk-users] Question about echo cancelation   Tilghman Lesher
• [asterisk-users] Tracking T1/PRI channel status - inbound vs. outbound   Kevin DeGraaf
• [asterisk-users] voicemail issues with 1.6.0   Brendan Martens
• [asterisk-users] ISDN   Steve Totaro
• [asterisk-users] ISDN   Hans Witvliet
• [asterisk-users] Tracking T1/PRI channel status - inbound vs. outbound   Steve Totaro
• [asterisk-users] ISDN   Michael Graves
• [asterisk-users] Tracking T1/PRI channel status - inbound vs. outbound   Steve Totaro
• [asterisk-users] Asterisk help please   Ramkumar Subramanian
• [asterisk-users] ISDN   Joe Greco
• [asterisk-users] ISDN   Steve Totaro
• [asterisk-users] Hang up detection with TDM400P and Telewest/Virgin Media line   Mike
• [asterisk-users] Matching *, + and # in the dialplan   Karl Fife
• [asterisk-users] ISDN   Tzafrir Cohen
• [asterisk-users] realtime queue_log to mySQL backport to 1.4   Lee, John (Sydney)
• [asterisk-users] Matching *, + and # in the dialplan   Eric "ManxPower" Wieling
• [asterisk-users] cli commands missing   Tzafrir Cohen
• [asterisk-users] realtime queue_log to mySQL backport to 1.4   Lee, John (Sydney)
• [asterisk-users] Panasonic x Asterisk if I can emulate Panasonic fast!   Rodolfo Alcazar Portillo
• [asterisk-users] ISDN   Wilton Helm
• [asterisk-users] ISDN   Wilton Helm
• [asterisk-users] ISDN   Wilton Helm
• [asterisk-users] Panasonic x Asterisk if I can emulate Panasonic fast!   Jorge Mendoza
• [asterisk-users] Text messaging and Asterisk   John Todd
• [asterisk-users] Is there a way to test SIP credentials without making a call?   John Todd
• [asterisk-users] Budge Tones pick up wrong calls   Trevor Peirce
• [asterisk-users] Aastra phones and dns srv records   Trevor Peirce
• [asterisk-users] Alarm events + asterisk dies   Roberts Klotins
• [asterisk-users] Question about echo cancelation   Olivier
• [asterisk-users] echo over digital line   Olivier
• [asterisk-users] echo over digital line   Kevin P. Fleming
• [asterisk-users] Softphone Framework or Libraries   Tim Panton
• [asterisk-users] echo over digital line   Vieri
• [asterisk-users] realtime queue_log to mySQL backport to 1.4   Atis Lezdins
• [asterisk-users] Cisco 7906g & SIP   Sasa
• [asterisk-users] Alarm events + asterisk dies   broadband Voice
• [asterisk-users] Matching *, + and # in the dialplan   Benny Amorsen
• [asterisk-users] ISDN   Tzafrir Cohen
• [asterisk-users] Cisco 7906g & SIP   Duncan Turnbull
• [asterisk-users] Asterisk voicemail   Chris Rowson
• [asterisk-users] ISDN   Tzafrir Cohen
• [asterisk-users] Alarm events + asterisk dies   Tzafrir Cohen
• [asterisk-users] Asterisk voicemail   Chris Rowson
• [asterisk-users] Cisco 7906g & SIP   Sasa
• [asterisk-users] Asterisk voicemail   Steven Howes
• [asterisk-users] Asterisk voicemail   Atis Lezdins
• [asterisk-users] Asterisk voicemail   Chris Rowson
• [asterisk-users] Asterisk voicemail   Chris Rowson
• [asterisk-users] Hang up detection with TDM400P and Telewest/Virgin Media line   Gordon Henderson
• [asterisk-users] ISDN   Gordon Henderson
• [asterisk-users] Aastra phones and dns srv records   Tom Moore
• [asterisk-users] ISDN   Joe Greco
• [asterisk-users] Panasonic x Asterisk if I can emulate Panasonic fast!   Rodolfo Alcazar Portillo
• [asterisk-users] realtime queue_log to mySQL backport to 1.4   Tilghman Lesher
• [asterisk-users] Panasonic x Asterisk if I can emulate Panasonic fast!   Tilghman Lesher
• [asterisk-users] ISDN   Joe Greco
• [asterisk-users] Asterisk voicemail   Tilghman Lesher
• [asterisk-users] Call files   Steven Howes
• [asterisk-users] is there a way   Drew Gibson
• [asterisk-users] Alarm events + asterisk dies   Roberts Klotins
• [asterisk-users] Asterisk 1.4.10.1 : PRI congestion warnings   veselin at campbell-lange.net
• [asterisk-users] is there a way   Brent Davidson
• [asterisk-users] Asterisk 1.6.1 + openais   Russell Bryant
• [asterisk-users] asterisk+heartbeat   Nhadie
• [asterisk-users] ISDN   Wilton Helm
• [asterisk-users] Asterisk voicemail   Chris Rowson
• [asterisk-users] Call files   Christian Victor
• [asterisk-users] Speex Problem   Brent Davidson
• [asterisk-users] Speex Problem   Brent Davidson
• [asterisk-users] Text messaging and Asterisk   Drew Gibson
• [asterisk-users] Matching *, + and # in the dialplan   Karl Fife
• [asterisk-users] Cisco 7906g & SIP   Duncan Turnbull
• [asterisk-users] Question about echo cancelation   Tilghman Lesher
• [asterisk-users] ISDN   Wilton Helm
• [asterisk-users] Looking for a mentor   Cliff Sifton
• [asterisk-users] Looking for a mentor   Martin Smith
• [asterisk-users] ISDN   Steve Totaro
• [asterisk-users] Looking for a mentor   Tzafrir Cohen
• [asterisk-users] ISDN   Steve Totaro
• [asterisk-users] BRI or PRI callerid   Steve Totaro
• [asterisk-users] Budge Tones pick up wrong calls   Paul Douglas Franklin
• [asterisk-users] SIP channels seem not to close after call is finished   Daniel - Asterisk
• [asterisk-users] Budge Tones pick up wrong calls   Darryl Dunkin
• [asterisk-users] Help With AMI   Jim Dickenson
• [asterisk-users] Looking for a mentor   Cliff Sifton
• [asterisk-users] Matching *, + and # in the dialplan   Benny Amorsen
• [asterisk-users] Matching *, + and # in the dialplan   Karl Fife
• [asterisk-users] Text messaging and Asterisk   C. Savinovich
• [asterisk-users] Text messaging and Asterisk   Steve Totaro
• [asterisk-users] realtime queue_log to mySQL backport to 1.4   Lee, John (Sydney)
• [asterisk-users] realtime queue_log to mySQL backport to 1.4   Lee, John (Sydney)
• [asterisk-users] FW: asterisk-users Digest, Vol 51, Issue 48   rinor bytyqi
• [asterisk-users] Asterisk 1.4.21.2 - Issues with call parking   Muthukrishnan Venkatakrishnan
• [asterisk-users] Asterisk 1.4.21.2 - Issues with call parking   Muthukrishnan Venkatakrishnan
• [asterisk-users] Asterisk 1.4.21.2 - Issues with call parking   Doug Lytle
• [asterisk-users] realtime queue_log to mySQL backport to 1.4   Atis Lezdins
• [asterisk-users] Matching *, + and # in the dialplan   Jared Smith
• [asterisk-users] Help With AMI   Martin Smith
• [asterisk-users] LDAP authentication   Adrien Couet
• [asterisk-users] mismatched callerid on phone and CDR ?   Louis-David Mitterrand
• [asterisk-users] mismatched callerid on phone and CDR ?   Tilghman Lesher
• [asterisk-users] Zaptel compile error after make update.   Freddi Hansen
• [asterisk-users] asterisk+heartbeat   Gleim, Jason
• [asterisk-users] phoniceq e400p driver for DAHDI   Josh Edwards
• [asterisk-users] Matching *, + and # in the dialplan   Steve Murphy
• [asterisk-users] SIP channels seem not to close after call is finished   Steve Murphy
• [asterisk-users] Zaptel compile error after make update.   Tzafrir Cohen
• [asterisk-users] voicemail.conf   jonathan augenstine
• [asterisk-users] voicemail.conf   Tilghman Lesher
• [asterisk-users] Help With AMI   Jim Dickenson
• [asterisk-users] Configuring SIP TLS   Rafael Puga
• [asterisk-users] asterisk+heartbeat   Steve Totaro
• [asterisk-users] asterisk+heartbeat   Edgar Guadamuz
• [asterisk-users] Cisco 7960 not always receiving incoming calls   Stephen Reese
• [asterisk-users] Queue problem   Ing. Jorge S Alanís Garza
• [asterisk-users] Panasonic x Asterisk if I can emulate Panasonic fast!   C F
• [asterisk-users] Panasonic x Asterisk if I can emulate Panasonic fast!   C F
• [asterisk-users] asterisk+heartbeat   Nhadie
• [asterisk-users] asterisk+heartbeat   Nhadie
• [asterisk-users] One Way Audio Problem   GNUbie
• [asterisk-users] One Way Audio Problem   GNUbie
• [asterisk-users] One Way Audio Problem   GNUbie
• [asterisk-users] Matching *, + and # in the dialplan   Karl Fife
• [asterisk-users] asterisk+heartbeat   Edgar Guadamuz
• [asterisk-users] asterisk+heartbeat   Steve Totaro
• [asterisk-users] Telrad Analog CID   Andrew Joakimsen
• [asterisk-users] How to invoke an external C program and output an integer to the program?   johnny_xing
• [asterisk-users] One Way Audio Problem   Steve Totaro
• [asterisk-users] One Way Audio Problem   Steve Totaro
• [asterisk-users] asterisk+heartbeat   Nhadie
• [asterisk-users] One Way Audio Problem   Steve Totaro
• [asterisk-users] One Way Audio Problem   GNUbie
• [asterisk-users] One Way Audio Problem   Steve Totaro
• [asterisk-users] One Way Audio Problem   Steve Totaro
• [asterisk-users] One Way Audio Problem   GNUbie
• [asterisk-users] One Way Audio Problem   Tzafrir Cohen
• [asterisk-users] mismatched callerid on phone and CDR ?   Louis-David Mitterrand
• [asterisk-users] Configuring SIP TLS   Hans Witvliet
• [asterisk-users] How to launch batch whenever Asterisk (re)start ?   Olivier
• [asterisk-users] How to launch batch whenever Asterisk (re)start ?   Steve Totaro
• [asterisk-users] How to launch batch whenever Asterisk (re)start ?   Torbjörn Abrahamsson
• [asterisk-users] realtime queue_log to mySQL backport to 1.4   Lee, John (Sydney)
• [asterisk-users] Zaptel compile error after make update.   Tzafrir Cohen
• [asterisk-users] How to launch batch whenever Asterisk (re)start ?   Tzafrir Cohen
• [asterisk-users] app transfer problem   Enrico Pasqualotto
• [asterisk-users] DTMF issue   michel freiha
• [asterisk-users] How to launch batch whenever Asterisk (re)start ?   Olivier
• [asterisk-users] How to launch batch whenever Asterisk (re)start ?   Olivier
• [asterisk-users] Call files   Steven Howes
• [asterisk-users] How to launch batch whenever Asterisk (re)start ?   Olivier
• [asterisk-users] prective dialer   yavuz yildirim
• [asterisk-users] Panasonic x Asterisk ... NO PROBLEM!   Rodolfo Alcazar Portillo
• [asterisk-users] asterisk cmd mysql and stored procedures   Giedrius Augys
• [asterisk-users] Sharing my Asterisk + SPA3102/PAP2 setup: What I've learned in 1 week.   Rodolfo Alcazar Portillo
• [asterisk-users] Panasonic x Asterisk if I can emulate Panasonic fast!   Olivier
• [asterisk-users] Difference between followme and substitution ?   Olivier
• [asterisk-users] Panasonic x Asterisk if I can emulate Panasonic fast!   Olivier
• [asterisk-users] Panasonic x Asterisk if I can emulate Panasonic fast!   Rodolfo Alcazar Portillo
• [asterisk-users] Cli with COLORS   Lucas Alvarez
• [asterisk-users] Cli with COLORS   César García
• [asterisk-users] Panasonic x Asterisk if I can emulate Panasonic fast!   Steve Totaro
• [asterisk-users] app_confcall build issues   jonathan augenstine
• [asterisk-users] Cli with COLORS   Mr Shunz
• [asterisk-users] app_confcall build issues   Steven Howes
• [asterisk-users] asterisk-users Digest, Vol 51, Issue 51   Norman Franke
• [asterisk-users] Triggering a call from bash   Rodolfo Alcazar Portillo
• [asterisk-users] Matching *, + and # in the dialplan   Olivier
• [asterisk-users] SIP: difference between Grandstream and Cisco when behind NAT   Tony Mountifield
• [asterisk-users] RELEASE message in q931.c   Julian Lyndon-Smith
• [asterisk-users] Triggering a call from bash   Barry L. Kline
• [asterisk-users] Triggering a call from bash   David A. Bandel
• [asterisk-users] DTMF issue   John Meksavan
• [asterisk-users] Multiple Repeated tones with TDM02B   Manolet Gmail
• [asterisk-users] SIP: difference between Grandstream and Cisco when behind NAT   Gordon Henderson
• [asterisk-users] Matching *, + and # in the dialplan   Tilghman Lesher
• [asterisk-users] Triggering a call from bash   Rodolfo Alcazar Portillo
• [asterisk-users] Triggering a call from bash   Rodolfo Alcazar Portillo
• [asterisk-users] [Asterisk-users] asterisk +heartbeat   Wilton Helm
• [asterisk-users] SIP: difference between Grandstream and Cisco when behind NAT   Tony Mountifield
• [asterisk-users] One Way Audio Problem   Karsten Wemheuer
• [asterisk-users] How to launch batch whenever Asterisk (re)start ?   Torbjörn Abrahamsson
• [asterisk-users] RELEASE message in q931.c   Tzafrir Cohen
• [asterisk-users] International calls/pridialplan from a legacy PBX.   Ken D'Ambrosio
• [asterisk-users] DAHDI and wait 'w'   Jerry Geis
• [asterisk-users] Matching *, + and # in the dialplan   Karl Fife
• [asterisk-users] SIP: difference between Grandstream and Cisco when behind NAT   Gordon Henderson
• [asterisk-users] [Asterisk-users] asterisk +heartbeat   Steve Totaro
• [asterisk-users] DAHDI and wait 'w'   John Novack
• [asterisk-users] International calls/pridialplan from a legacy PBX.   Tilghman Lesher
• [asterisk-users] Matching *, + and # in the dialplan   Steve Murphy
• [asterisk-users] SIP: difference between Grandstream and Cisco when behind NAT   Benny Amorsen
• [asterisk-users] Matching *, + and # in the dialplan   Tilghman Lesher
• [asterisk-users] 1 second delay when connecting calls   "Juan E. Rodríguez"
• [asterisk-users] One way voice after call transfer (bugs 9305, 13120)   broadband Voice
• [asterisk-users] SIP: difference between Grandstream and Cisco when behind NAT   peder at networkoblivion.com
• [asterisk-users] Meetme "talker optimization" always on even when no "o" option present.   William F. Acker WB2FLW +1-303-722-7209
• [asterisk-users] Matching *, + and # in the dialplan   John Todd
• [asterisk-users] Panasonic x Asterisk if I can emulate Panasonic fast!   C F
• [asterisk-users] Panasonic x Asterisk if I can emulate Panasonic fast!   C F
• [asterisk-users] Panasonic x Asterisk ... NO PROBLEM!   C F
• [asterisk-users] One way voice after call transfer (bugs 9305, 13120)   Yehavi Bourvine
• [asterisk-users] Asterisk SIP calls stop working having more than 300 calls (more than 600 channels)   Juan Rodríguez
• [asterisk-users] prective dialer   ram
• [asterisk-users] Send me an SMS   Sunkara RaviPrakash
• [asterisk-users] realtime queue_log to mySQL backport to 1.4   Lee, John (Sydney)
• [asterisk-users] How to add contexts in asterisk realtime?   Zeeshan Zakaria
• [asterisk-users] Alarm events + asterisk dies   Roberts Klotins
• [asterisk-users] prective dialer   Steve Totaro
• [asterisk-users] Panasonic x Asterisk if I can emulate Panasonic fast!   Olivier
• [asterisk-users] RELEASE message in q931.c   Kevin P. Fleming
• [asterisk-users] [Asterisk-users] asterisk +heartbeat (Wilton Helm)   Freddi Hansen
• [asterisk-users] How to launch batch whenever Asterisk (re)start ?   Olivier
• [asterisk-users] One way voice after call transfer (bugs 9305, 13120)   broadband Voice
• [asterisk-users] Cisco 7906g & SIP   Sasa
• [asterisk-users] Cisco 7906g & SIP   Duncan Turnbull
• [asterisk-users] Asterisk SIP calls stop working having more than 300 calls (more than 600 channels)   Tzafrir Cohen
• [asterisk-users] How to launch batch whenever Asterisk (re)start ? [SOLVED]   Olivier
• [asterisk-users] Panasonic x Asterisk ... NO PROBLEM!   Rodolfo Alcazar Portillo
• [asterisk-users] How to launch batch whenever Asterisk (re)start ? [SOLVED]   Tzafrir Cohen
• [asterisk-users] on livemsn to cindy   Rodolfo Alcazar Portillo
• [asterisk-users] [Asterisk-users] asterisk +heartbeat (Wilton Helm)   Steve Totaro
• [asterisk-users] srv records not being honoured properly   Brian J. Murrell
• [asterisk-users] realtime queue_log to mySQL backport to 1.4   Tilghman Lesher
• [asterisk-users] GET DATA Returning only a single digit   Akintayo Olusegun
• [asterisk-users] How to launch batch whenever Asterisk (re)start ? [SOLVED]   Olivier
• [asterisk-users] Whenever Asterisk restarts, what should happen to ongoing subscriptions ?   Olivier
• [asterisk-users] Strip prefix   michel freiha
• [asterisk-users] Asterisk SIP and SRTP   Artem Makhutov
• [asterisk-users] Matching *, + and # in the dialplan   Olivier
• [asterisk-users] Strip prefix   Eric "ManxPower" Wieling
• [asterisk-users] Asterisk SIP and SRTP   Russell Bryant
• [asterisk-users] Strip prefix   michel freiha
• [asterisk-users] srv records not being honoured properly   Andres
• [asterisk-users] srv records not being honoured properly   Brian J. Murrell
• [asterisk-users] [Asterisk-users] asterisk +heartbeat (Wilton Helm)   Kristian Kielhofner
• [asterisk-users] prective dialer   Richard Lyman
• [asterisk-users] Snom M3 firmware Update   Joseph L. Casale
• [asterisk-users] Strip prefix   Tilghman Lesher
• [asterisk-users] FW: Under the Radar Blog   Dean Collins
• [asterisk-users] srv records not being honoured properly   Eric "ManxPower" Wieling
• [asterisk-users] srv records not being honoured properly   Brian J. Murrell
• [asterisk-users] Panasonic x Asterisk ... NO PROBLEM!   C F
• [asterisk-users] srv records not being honoured properly   Eric "ManxPower" Wieling
• [asterisk-users] Matching *, + and # in the dialplan   Tilghman Lesher
• [asterisk-users] srv records not being honoured properly   Brian J. Murrell
• [asterisk-users] [Asterisk-users] +heartbeat   Wilton Helm
• [asterisk-users] Snom M3 firmware Update   Tim Litwiller
• [asterisk-users] One way voice after call transfer (bugs 9305, 13120)   Yehavi Bourvine
• [asterisk-users] srv records not being honoured properly   Tilghman Lesher
• [asterisk-users] Snom M3 firmware Update   Torbjörn Abrahamsson
• [asterisk-users] Snom M3 firmware Update   Joseph L. Casale
• [asterisk-users] srv records not being honoured properly   Brian J. Murrell
• [asterisk-users] Asterisk SIP calls stop working having more than 300 calls (more than 600 channels)   "Juan E. Rodríguez"
• [asterisk-users] Asterisk SIP calls stop working having more than 300 calls (more than 600 channels)   Tzafrir Cohen
• [asterisk-users] One Way Audio Problem   Jeff LaCoursiere
• [asterisk-users] Transfering Calls back on the same PRI   Ron Joffe
• [asterisk-users] anoyingly answers already in use pstn line   Jack Bates
• [asterisk-users] anoyingly answers already in use pstn line   Gleim, Jason
• [asterisk-users] One Way Audio Problem   Brent Davidson
• [asterisk-users] Transfering Calls back on the same PRI   Ron Joffe
• [asterisk-users] Transfering Calls back on the same PRI   Steve Totaro
• [asterisk-users] anoyingly answers already in use pstn line   Tzafrir Cohen
• [asterisk-users] Phones lose contact   Paul Douglas Franklin
• [asterisk-users] Phones lose contact   Jerry Jones
• [asterisk-users] Cisco 7960 not always receiving incoming calls   Stephen Reese
• [asterisk-users] Cisco 7960 not always receiving incoming calls   Darryl Dunkin
• [asterisk-users] Cisco 7960 not always receiving incoming calls   Darryl Dunkin
• [asterisk-users] SER + Asterisk   Joseph
• [asterisk-users] Transfering Calls back on the same PRI   Ron Joffe
• [asterisk-users] SER + Asterisk   Alex Balashov
• [asterisk-users] SER + Asterisk   Kristian Kielhofner
• [asterisk-users] SER + Asterisk   Joseph
• [asterisk-users] SER + Asterisk   Joseph
• [asterisk-users] SER + Asterisk   Grey Man
• [asterisk-users] SER + Asterisk   ram
• [asterisk-users] SER + Asterisk   Joseph
• [asterisk-users] SER + Asterisk   Joseph
• [asterisk-users] Asterisk 1.4 and openLDAP   Anael DIAZ
• [asterisk-users] Asterisk 1.4 and openLDAP   Ming Yong
• [asterisk-users] Asterisk 1.4 and openLDAP   Tilghman Lesher
• [asterisk-users] SER + Asterisk   Alex Balashov
• [asterisk-users] SER + Asterisk   Alex Balashov
• [asterisk-users] SER + Asterisk   Alex Balashov
• [asterisk-users] Looking to replicate OnSIP ........SER + Asterisk   EdPimentl
• [asterisk-users] strange h323 delay issue   Giedrius Augys
• [asterisk-users] OT: Polycom IP330 user problem   Bill Michaelson
• [asterisk-users] SER + Asterisk   Joseph
• [asterisk-users] SER + Asterisk   Alex Balashov
• [asterisk-users] SER + Asterisk   Joseph
• [asterisk-users] SER + Asterisk   Alex Balashov
• [asterisk-users] OT: Polycom IP330 user problem   Doug Lytle
• [asterisk-users] SER + Asterisk   Joseph
• [asterisk-users] SER + Asterisk   Alex Balashov
• [asterisk-users] SER + Asterisk   Joseph
• [asterisk-users] SER + Asterisk   Steve Totaro
• [asterisk-users] SER + Asterisk   Alex Balashov
• [asterisk-users] SER + Asterisk   Steve Totaro
• [asterisk-users] Cisco 7960 not always receiving incoming calls   Stephen Reese
• [asterisk-users] SER + Asterisk   Alex Balashov
• [asterisk-users] Cisco 7960 not always receiving incoming calls   Darryl Dunkin
• [asterisk-users] Is there a way to specify the fromdomain from the dialplan?   Eric Chamberlain
• [asterisk-users] Does asterisk 1.6 support an authname with a domain?   Eric Chamberlain
• [asterisk-users] Cisco 7960 not always receiving incoming calls   Stephen Reese
• [asterisk-users] IP Address on CDR   Nhadie
• [asterisk-users] app_confcall on Asterisk 1.6 update   jonathan augenstine
• [asterisk-users] Latency woes, qos the fix?   Stephen Reese
• [asterisk-users] IP Address on CDR   Juan Rodríguez
• [asterisk-users] IP Address on CDR   Alex Balashov
• [asterisk-users] Latency woes, qos the fix?   Alex Balashov
• [asterisk-users] Latency woes, qos the fix?   Stephen Reese
• [asterisk-users] Latency woes, qos the fix?   Alex Balashov
• [asterisk-users] srv records not being honoured properly   Kevin P. Fleming
• [asterisk-users] strange h323 delay issue   Mindaugas Kezys
• [asterisk-users] strange h323 delay issue   Giedrius Augys
• [asterisk-users] IP Address on CDR   Nhadie
• [asterisk-users] srv records not being honoured properly   Brian J. Murrell
• [asterisk-users] strange h323 delay issue   Giedrius Augys
• [asterisk-users] Latency woes, qos the fix?   Jerry Jones
• [asterisk-users] DAHDI and wait 'w'   Jerry Geis
• [asterisk-users] srv records not being honoured properly   Kevin P. Fleming
• [asterisk-users] srv records not being honoured properly   Kevin P. Fleming
• [asterisk-users] IP Address on CDR   Stefan Schmidt
• [asterisk-users] Latency woes, qos the fix?   Stephen Reese
• [asterisk-users] IP Address on CDR   Nhadie
• [asterisk-users] adding a second extension   Stephen Reese
• [asterisk-users] srv records not being honoured properly   Brian J. Murrell
• [asterisk-users] Asterisk Problem   Ahmed torinto
• [asterisk-users] adding a second extension   Juan Rodríguez
• [asterisk-users] Asterisk Problem   Kristian Kielhofner
• [asterisk-users] adding a second extension   Stephen Reese
• [asterisk-users] Asterisk Problem   Juan Rodríguez
• [asterisk-users] Asterisk Problem   Ahmed Torintino
• [asterisk-users] Asterisk Problem   Ahmed Torintino
• [asterisk-users] adding a second extension   Stephen Reese
• [asterisk-users] Got SIP response 603 "Declined" back from 81.15.xx.xx   Joseph
• [asterisk-users] Asterisk 1.4 and openLDAP   Gavin Henry
• [asterisk-users] Asterisk 1.4 and openLDAP   Gavin Henry
• [asterisk-users] Asterisk 1.6.1 + openais   Edgar Guadamuz
• [asterisk-users] OT: Polycom IP330 user problem   Andrew Joakimsen
• [asterisk-users] Latency woes, qos the fix?   Andrew Joakimsen
• [asterisk-users] Asterisk Problem   Jai Rangi
• [asterisk-users] Panama DID   Dean Collins
• [asterisk-users] adding a second extension   Stephen Reese
• [asterisk-users] adding a second extension   Juan Rodríguez
• [asterisk-users] Zaptel FXO offhook when connected to PSTN   CSB
• [asterisk-users] adding a second extension   Stephen Reese
• [asterisk-users] adding a second extension   Juan Rodríguez
• [asterisk-users] adding a second extension   Eric "ManxPower" Wieling
• [asterisk-users] QoS VoIP   Anael DIAZ
• [asterisk-users] asterisk setup   Mike
• [asterisk-users] asterisk setup   Gordon Henderson
• [asterisk-users] asterisk setup   randulo
• [asterisk-users] asterisk setup   Mike
• [asterisk-users] Zaptel FXO offhook when connected to PSTN   Tzafrir Cohen
• [asterisk-users] B410P and asterisk 1.6   Enrico Maistro
• [asterisk-users] Asterisk 1.6.1 + openais   Russell Bryant
• [asterisk-users] Asterisk Problem   Antoine Megalla
• [asterisk-users] Asterisk Problem   Tzafrir Cohen
• [asterisk-users] ISDN PRI Caller ID problem   A.R. Nasir Qureshi
• [asterisk-users] QoS VoIP   Alex Balashov
• [asterisk-users] anoyingly answers already in use pstn line   Drew Gibson
• [asterisk-users] Problem in extensions.conf Configuration ${CALLINGPRES}   Hiren Mistry
• [asterisk-users] adding a second extension   Stephen Reese
• [asterisk-users] adding a second extension   Stephen Reese
• [asterisk-users] adding a second extension   Juan Rodríguez
• [asterisk-users] adding a second extension   Stephen Reese
• [asterisk-users] Configuring Bandwidth.com SIP trunks to prevent one-way audio   Kurt Knudsen
• [asterisk-users] Configuring Bandwidth.com SIP trunks to prevent one-way audio   Jeremy Mann
• [asterisk-users] I have probleme with asterisk   diop cheikhtacko
• [asterisk-users] Configuring Bandwidth.com SIP trunks to prevent one-way audio   Kurt Knudsen
• [asterisk-users] Configuring Bandwidth.com SIP trunks to prevent one-way audio   Jeremy Mann
• [asterisk-users] Is there a way to specify the fromdomain from the dialplan?   Philipp Kempgen
• [asterisk-users] Transferring Outbound Calls   Joseph L. Casale
• [asterisk-users] OT: Polycom IP330 user problem   Mark Hamilton
• [asterisk-users] OPENR2 in Thailand   Peter Lindquist
• [asterisk-users] a little regex help needed   sean darcy
• [asterisk-users] Configuring Bandwidth.com SIP trunks to prevent one-way audio   Kurt Knudsen
• [asterisk-users] a little regex help needed   Jared Smith
• [asterisk-users] Configuring Bandwidth.com SIP trunks to prevent one-way audio   Jeremy Mann
• [asterisk-users] How Secure Is Asterisk   Steve Anness
• [asterisk-users] How Secure Is Asterisk   Sam Tam
• [asterisk-users] How Secure Is Asterisk   Alex Balashov
• [asterisk-users] Configuring Bandwidth.com SIP trunks to prevent one-way audio   Kurt Knudsen
• [asterisk-users] SERVICE CODES   Robert Boardman
• [asterisk-users] How Secure Is Asterisk   broadband Voice
• [asterisk-users] a little regex help needed   Atis Lezdins
• [asterisk-users] ISDN PRI Caller ID problem   Matthew Fredrickson
• [asterisk-users] QoS VoIP   sean darcy
• [asterisk-users] a little regex help needed   sean darcy
• [asterisk-users] a little regex help needed   Edwin Lam
• [asterisk-users] a little regex help needed   sean darcy
• [asterisk-users] TDM410P with EC doesn't detect DTMF after being on for ~1 hour   Kurt Knudsen
• [asterisk-users] a little regex help needed   Steve Murphy
• [asterisk-users] a little regex help needed   sean darcy
• [asterisk-users] prepaid approach   Nhadie
• [asterisk-users] SER + Asterisk   Tobias Wolf
• [asterisk-users] SER + Asterisk   Alex Balashov
• [asterisk-users] prepaid approach   Rafael Cedeño
• [asterisk-users] come back ring   jordan pan
• [asterisk-users] SER + Asterisk   Andres
• [asterisk-users] Zaptel FXO offhook when connected to PSTN   CSB
• [asterisk-users] OPENR2 in Thailand   Moises Silva
• [asterisk-users] OPENR2 in Thailand   Chris Ziomkowski
• [asterisk-users] OPENR2 in Thailand   Peter Lindquist
• [asterisk-users] OPENR2 in Thailand   Peter Lindquist
• [asterisk-users] ISDN PRI Caller ID problem   A.R. Nasir Qureshi
• [asterisk-users] OPENR2 in Thailand   Chris Ziomkowski
• [asterisk-users] How Secure Is Asterisk   Sam Tam
• [asterisk-users] Asterisk 1.4: ISDN congestion warnings   veselin at campbell-lange.net
• [asterisk-users] SER + Asterisk   SIP
• [asterisk-users] Realtime : switch any context dynamically   morteza kashani
• [asterisk-users] [help] Realtime Swich any context dinamically   morteza kashani
• [asterisk-users] [help] Realtime Swich any context dinamically   Atis Lezdins
• [asterisk-users] OPENR2 in Thailand   Steve Underwood
• [asterisk-users] SER + Asterisk   Alex Balashov
• [asterisk-users] Panasonic x Asterisk ... NO PROBLEM!   César García
• [asterisk-users] SER + Asterisk   SIP
• [asterisk-users] adding a second extension   Stephen Reese
• [asterisk-users] SER + Asterisk   Alex Balashov
• [asterisk-users] adding a second extension   Stephen Reese
• [asterisk-users] come back ring   Tilghman Lesher
• [asterisk-users] Problem with Portech   Sasa
• [asterisk-users] Cisco 7906g & SIP   Sasa
• [asterisk-users] Cisco 7906g & SIP   David Gibbons
• [asterisk-users] [help] Realtime Swich any context dinamically   Terry Wilson
• [asterisk-users] Generating 484 "Address Incomplete"   Vahan Yerkanian
• [asterisk-users] Panasonic x Asterisk ... NO PROBLEM!   Rodolfo Alcazar Portillo
• [asterisk-users] Generating 484 "Address Incomplete"   Terry Wilson
• [asterisk-users] Asterisk Console color   Armand Fumal
• [asterisk-users] Asterisk Console color   Dwayne Hubbard
• [asterisk-users] For Dial(), when calling party hangs up, redirect called party to another location in the dialplan?   Martin Smith
• [asterisk-users] How to add contexts in asterisk realtime?   Zeeshan Zakaria
• [asterisk-users] Asterisk Console color   Tilghman Lesher
• [asterisk-users] For Dial(), when calling party hangs up, redirect called party to another location in the dialplan?   Josiah Bryan
• [asterisk-users] How to add contexts in asterisk realtime?   Terry Wilson
• [asterisk-users] How Secure Is Asterisk   Jonn R Taylor
• [asterisk-users] Asterisk Console color   Armand Fumal
• [asterisk-users] Panasonic x Asterisk ... NO PROBLEM!   Jonn R Taylor
• [asterisk-users] How Secure Is Asterisk   SIP
• [asterisk-users] hex b1 in CallerID sent by Asterisk On PRI   Bob Pierce
• [asterisk-users] Asterisk Console color   Tilghman Lesher
• [asterisk-users] How Secure Is Asterisk   Singer Wang
• [asterisk-users] How Secure Is Asterisk   Eric Chamberlain
• [asterisk-users] Does Asterisk support SIP Join Headers   Bruce Reeves
• [asterisk-users] Does Asterisk support SIP Join Headers   Alex Balashov
• [asterisk-users] Does Asterisk support SIP Join Headers   Bruce Reeves
• [asterisk-users] How Secure Is Asterisk   Kristian Kielhofner
• [asterisk-users] How Secure Is Asterisk   RE Kushner List Account
• [asterisk-users] adding a second extension   Stephen Reese
• [asterisk-users] How Secure Is Asterisk   Kristian Kielhofner
• [asterisk-users] Causes of auto-congestion on SIP?   James Lamanna
• [asterisk-users] need asterisk tech to relocate to riyadh   A_ Navone
• [asterisk-users] How Secure Is Asterisk   Tzafrir Cohen
• [asterisk-users] a question about linux/asterisk/commands   Babcock, Michael Alex
• [asterisk-users] come back ring   jordan pan
• [asterisk-users] How to add contexts in asterisk realtime?   morteza kashani
• [asterisk-users] come back ring   Eric Fort
• [asterisk-users] How to add contexts in asterisk realtime?   morteza kashani
• [asterisk-users] How Secure Is Asterisk   Nikolai Lusan
• [asterisk-users] sip and nat   Johanna NIRINA
• [asterisk-users] sip and nat   Johanna NIRINA
• [asterisk-users] sip and nat   Jai Rangi
• [asterisk-users] sip and nat   Johanna NIRINA
• [asterisk-users] WebCall application   voip crazy
• [asterisk-users] How Secure Is Asterisk   Tim Panton
• [asterisk-users] How Secure Is Asterisk   Tim Panton
• [asterisk-users] WebCall application   Tim Panton
• [asterisk-users] 2 asterisk boxes   Jerry Geis
• [asterisk-users] Asterisk Console color   Ivar Dahl
• [asterisk-users] WebCall application   Rob Hillis
• [asterisk-users] WebCall application   Tim Panton
• [asterisk-users] Asterisk Console color   Tzafrir Cohen
• [asterisk-users] 2 asterisk boxes   Robin Rodriguez
• [asterisk-users] sip and nat   Robin Rodriguez
• [asterisk-users] Parking Issue   Craig Van Ham
• [asterisk-users] hex b1 in CallerID sent by Asterisk On PRI   Bob Pierce
• [asterisk-users] fax / t38 gateway   Brendan Martens
• [asterisk-users] asterisk video   Nhadie
• [asterisk-users] Sonicwall potentially causing long ping times to SIP phones   James Lamanna
• [asterisk-users] How to add contexts in asterisk realtime?   JR Richardson
• [asterisk-users] fax / t38 gateway   Jonn R Taylor
• [asterisk-users] Sonicwall potentially causing long ping times toSIP phones   Jeff Johnson
• [asterisk-users] Sonicwall potentially causing long ping times toSIP phones   Craig Van Ham
• [asterisk-users] changing from default codec   Max McGraw
• [asterisk-users] : Parking Issue   Craig Van Ham
• [asterisk-users] asterisk video   Tilghman Lesher
• [asterisk-users] How to add contexts in asterisk realtime?   Terry Wilson
• [asterisk-users] Sonicwall potentially causing long ping timestoSIP phones   Jeff Johnson
• [asterisk-users] hex b1 in CallerID sent by Asterisk On PRI   Bob Pierce
• [asterisk-users] : Parking Issue   Steven Howes
• [asterisk-users] adding a second extension   Stephen Reese
• [asterisk-users] adding a second extension   Juan Rodríguez
• [asterisk-users] asterisk video   Nhadie
• [asterisk-users] adding a second extension   Stephen Reese
• [asterisk-users] FW: [wwwac] Thursday 23 October 2008 NYLUG: Paul Charles Leddy on Asterisk, the Free Software Telephone System   Dean Collins
• [asterisk-users] Panasonic x Asterisk ... NO PROBLEM!   C F
• [asterisk-users] Panasonic x Asterisk ... NO PROBLEM!   C F
• [asterisk-users] fax / t38 gateway   Brendan Martens
• [asterisk-users] fax / t38 gateway   Andrew Joakimsen
• [asterisk-users] fax / t38 gateway   Olivier
• [asterisk-users] command - set sip_codec- does not work with asterisk-1.4.21   lizhong zhu
• [asterisk-users] Sonicwall potentially causing long ping times toSIP phones   Dr. Michael J. Chudobiak
• [asterisk-users] asterisk video   Gordon Henderson
• [asterisk-users] asterisk video   Nhadie
• [asterisk-users] problems with some incoming/outgoing calls   Fernando Serto
• [asterisk-users] Astribank loop current adjustment   Udo Schacht-Wiegand
• [asterisk-users] asterisk video   Gordon Henderson
• [asterisk-users] asterisk video   Robert Augustyn
• [asterisk-users] asterisk video   Nhadie
• [asterisk-users] switching from 1.6.0-beta9 to 1.6.0.1 problems   Julien Claassen
• [asterisk-users] problems with some incoming/outgoing calls   Lucas Alvarez
• [asterisk-users] users.conf and sip call-limit   Jeremy Mann
• [asterisk-users] Atxfer Command   David Monteagudo Sanz
• [asterisk-users] Atxfer Command   Mark Michelson
• [asterisk-users] Hylafax asterisk iaxmodem problem   voip crazy
• [asterisk-users] Devstate and Voicemail   Marc Hudson
• [asterisk-users] Channels are increasing without limit - Please Help!   Daniel - Asterisk
• [asterisk-users] adding a second extension   Juan Rodríguez
• [asterisk-users] Sonicwall potentially causing long ping times to SIP phones   Bill Michaelson
• [asterisk-users] Hylafax asterisk iaxmodem problem   Lee Howard
• [asterisk-users] Channels are increasing without limit - Please Help!   Daniel - Asterisk
• [asterisk-users] Sonicwall potentially causing long ping times to SIP phones   Bruce Komito
• [asterisk-users] Hylafax asterisk iaxmodem problem   Alex Balashov
• [asterisk-users] Sonicwall potentially causing long ping times to SIP phones   James Lamanna
• [asterisk-users] Channels are increasing without limit - Please Help!   Daniel - Asterisk
• [asterisk-users] Sonicwall potentially causing long ping times to SIP phones   Craig Van Ham
• [asterisk-users] adding a second extension   Stephen Reese
• [asterisk-users] Sonicwall potentially causing long ping times to SIP phones   Kristian Kielhofner
• [asterisk-users] fax / t38 gateway   Brendan Martens
• [asterisk-users] Sonicwall potentially causing long ping times to SIP phones   Bruce Komito
• [asterisk-users] Devstate and Voicemail   Jared Smith
• [asterisk-users] Devstate and Voicemail   Philipp Kempgen
• [asterisk-users] is there a reference guide to "pri debug span" messages?   John Cheng
• [asterisk-users] is there a reference guide to "pri debug span" messages?   bkruse
• [asterisk-users] Returning to Voicemail after returning call   Mark Wiater
• [asterisk-users] is there a reference guide to "pri debug span" messages?   Jerry Jones
• [asterisk-users] Inbound DID + voice ports needed for vote monitoring project   David Troy
• [asterisk-users] Inbound DID + voice ports needed for vote monitoring project   Jai Rangi
• [asterisk-users] adding a second extension   Stephen Reese
• [asterisk-users] Inbound DID + voice ports needed for vote monitoring project   michel freiha
• [asterisk-users] Emerging dilema? DID forwarding meets SMS   Karl Fife
• [asterisk-users] changing from default codec   Max McGraw
• [asterisk-users] Asterisk and Cisco Call Manager Express (CME)   Stephen Reese
• [asterisk-users] problems with some incoming/outgoing calls   Fernando Serto
• [asterisk-users] Asterisk and Cisco Call Manager Express (CME)   David Gibbons
• [asterisk-users] Asterisk and Cisco Call Manager Express (CME)   Stephen Reese
• [asterisk-users] How to add contexts in asterisk realtime?   Steve Murphy
• [asterisk-users] Devstate and Voicemail   Marc Hudson
• [asterisk-users] How to add contexts in asterisk realtime?   Olivier
• [asterisk-users] fax / t38 gateway   Olivier
• [asterisk-users] problems with some incoming/outgoing calls   Tzafrir Cohen
• [asterisk-users] Asterisk and Cisco Call Manager Express (CME)   Steven Howes
• [asterisk-users] switching from 1.6.0-beta9 to 1.6.0.1 problems   Enrico Maistro
• [asterisk-users] Emerging dilema? DID forwarding meets SMS   Gordon Henderson
• [asterisk-users] fax / t38 gateway   Steve Underwood
• [asterisk-users] Asterisk and Cisco Call Manager Express (CME)   David Gibbons
• [asterisk-users] Problems with zaptel/ztdummy/asterisk.   Richard Horton
• [asterisk-users] Asterisk and Cisco Call Manager Express (CME)   Stephen Reese
• [asterisk-users] prective dialer   Alex Balashov
• [asterisk-users] fax / t38 gateway   Brendan Martens
• [asterisk-users] fax / t38 gateway   Brendan Martens
• [asterisk-users] Emerging dilema? DID forwarding meets SMS   Drew Gibson
• [asterisk-users] fax / t38 gateway   Olivier
• [asterisk-users] fax / t38 gateway   Olivier
• [asterisk-users] Freepbx or Trixbox Presentation   Torintino T
• [asterisk-users] Problems with zaptel/ztdummy/asterisk.   Tzafrir Cohen
• [asterisk-users] SIP channels seem not to close after call is finished   Daniel - Asterisk
• [asterisk-users] fax / t38 gateway   Senad Jordanovic
• [asterisk-users] Advice on ISDN and Asterisk in the UK   Phil Knighton
• [asterisk-users] Emerging dilema? DID forwarding meets SMS   Gordon Henderson
• [asterisk-users] Advice on ISDN and Asterisk in the UK   Andres
• [asterisk-users] Advice on ISDN and Asterisk in the UK   Gordon Henderson
• [asterisk-users] fax / t38 gateway   Wilton Helm
• [asterisk-users] Advice on ISDN and Asterisk in the UK   Dave Cotton
• [asterisk-users] Advice on ISDN and Asterisk in the UK   Alan Lord
• [asterisk-users] fax / t38 gateway   Daniel Hazelbaker
• [asterisk-users] Problems with zaptel/ztdummy/asterisk.   Richard Horton
• [asterisk-users] Agents log in afterhours   Ing. Jorge S Alanís Garza
• [asterisk-users] Panasonic x Asterisk ... NO PROBLEM!   Richard Scobie
• [asterisk-users] fax / t38 gateway   Wilton Helm
• [asterisk-users] Advice on ISDN and Asterisk in the UK   Gordon Henderson
• [asterisk-users] OT: Disable Polycom 650 Forward Softkey   Bill Andersen
• [asterisk-users] OT: Disable Polycom 650 Forward Softkey   Darryl Dunkin
• [asterisk-users] Sonicwall potentially causing long ping times to SIP phones   Bill Michaelson
• [asterisk-users] Sonicwall potentially causing long ping timesto SIP phones   Darryl Dunkin
• [asterisk-users] OT: Disable Polycom 650 Forward Softkey   Bill Andersen
• [asterisk-users] OT: Disable Polycom 650 Forward Softkey   John Todd
• [asterisk-users] Panasonic x Asterisk ... NO PROBLEM!   Torintino T
• [asterisk-users] Emerging dilema? DID forwarding meets SMS   John Todd
• [asterisk-users] OT: Disable Polycom 650 Forward Softkey   Bill Andersen
• [asterisk-users] Fresh installed box   Torintino T
• [asterisk-users] Sporadic One Way Audio   Brent Davidson
• [asterisk-users] Fresh installed box   Matt Gibson
• [asterisk-users] Asterisk iaxy adapter annoying beep during conversation   Joseph
• [asterisk-users] Sporadic One Way Audio   OCG Technical Support
• [asterisk-users] Sporadic One Way Audio   Brent Davidson
• [asterisk-users] Sporadic One Way Audio   OCG Technical Support
• [asterisk-users] Sporadic One Way Audio   Christian Stredicke
• [asterisk-users] Fresh installed box   Mark Hamilton
• [asterisk-users] Fresh installed box   Matt Gibson
• [asterisk-users] OT: Disable Polycom 650 Forward Softkey   Chris Walton
• [asterisk-users] Sonicwall potentially causing long ping times to SIP phones   Stephen Davies
• [asterisk-users] Advice on ISDN and Asterisk in the UK   Alan Lord
• [asterisk-users] Astribank loop current adjustment   Stephen Davies
• [asterisk-users] Fresh installed box   Torintino T
• [asterisk-users] Astribank loop current adjustment   Tzafrir Cohen
• [asterisk-users] Advice on ISDN and Asterisk in the UK   Gordon Henderson
• [asterisk-users] Advice on ISDN and Asterisk in the UK   Tim Panton
• [asterisk-users] The skype channel...   Julien Claassen
• [asterisk-users] fax / t38 gateway   Anthony Messina
• [asterisk-users] The skype channel...   Dean Collins
• [asterisk-users] The skype channel...   Thomas Kenyon
• [asterisk-users] The skype channel...   Julien Claassen
• [asterisk-users] The skype channel...   randulo
• [asterisk-users] The skype channel...   Thomas Kenyon
• [asterisk-users] Returning to Voicemail after returning call   Andrew Joakimsen
• [asterisk-users] Cheapest 4 port FXO   Joseph L. Casale
• [asterisk-users] someone to test gtalk with me?   Julien Claassen
• [asterisk-users] Fresh installed box   Matt Gibson
• [asterisk-users] gtalk dialstring?   Julien Claassen
• [asterisk-users] gtalk dialstring?   Philippe Sultan
• [asterisk-users] gtalk dialstring?   Julien Claassen
• [asterisk-users] Cheapest 4 port FXO   Alex Balashov
• [asterisk-users] Cheapest 4 port FXO   Joseph L. Casale
• [asterisk-users] Agents log in afterhours   Atis Lezdins
• [asterisk-users] Strange ring tone: Long-Short-Short   Joseph
• [asterisk-users] Queue Warning messages   César García
• [asterisk-users] hammering imap vmail storage   Brian J. Murrell
• [asterisk-users] bug in Asterisk 1.4.22?   Abel Monzon
• [asterisk-users] bug in Asterisk 1.4.22?   Vahan Yerkanian
• [asterisk-users] bug in Asterisk 1.4.22?   Tzafrir Cohen
• [asterisk-users] bug in Asterisk 1.4.22?   Juan Rodríguez
• [asterisk-users] Cheapest 4 port FXO   Gordon Henderson
• [asterisk-users] fax / t38 gateway   Olivier
• [asterisk-users] Strange ring tone: Long-Short-Short   SIP
• [asterisk-users] Cheapest 4 port FXO   Hans Witvliet
• [asterisk-users] fax / t38 gateway   Steve Underwood
• [asterisk-users] jingle/gtalk still very troubling   Julien Claassen
• [asterisk-users] Strange ring tone: Long-Short-Short   Eric Moniz
• [asterisk-users] jingle/gtalk still very troubling   Philippe Sultan
• [asterisk-users] jingle/gtalk still very troubling   Julien Claassen
• [asterisk-users] jingle/gtalk still very troubling   Philippe Sultan
• [asterisk-users] jingle/gtalk still very troubling   Julien Claassen
• [asterisk-users] Cheapest 4 port FXO   Joseph L. Casale
• [asterisk-users] fax / t38 gateway   Benny Amorsen
• [asterisk-users] fax / t38 gateway   Benny Amorsen
• [asterisk-users] jingle/gtalk still very troubling   Philippe Sultan
• [asterisk-users] fax / t38 gateway   Benny Amorsen
• [asterisk-users] Cheapest 4 port FXO   Eric Fort
• [asterisk-users] jingle/gtalk still very troubling   Julien Claassen
• [asterisk-users] Strange ring tone: Long-Short-Short   Joseph
• [asterisk-users] jingle/gtalk still very troubling   Philippe Sultan
• [asterisk-users] Cheapest 4 port FXO   Robert Augustyn
• [asterisk-users] No incoming audio on Dahdi channels (TDM410P)   Kurt Knudsen
• [asterisk-users] jingle/gtalk still very troubling   Julien Claassen
• [asterisk-users] Cheapest 4 port FXO   Gergo Csibra
• [asterisk-users] Asterisk on Freebsd 7.0 Release.   Abel Monzon
• [asterisk-users] Fresh installed box   Torintino T
• [asterisk-users] Cheapest 4 port FXO   Gordon Henderson
• [asterisk-users] Cheapest 4 port FXO   Andrew Joakimsen
• [asterisk-users] OT: Disable Polycom 650 Forward Softkey   Andrew Joakimsen
• [asterisk-users] Emerging dilema? DID forwarding meets SMS   Andrew Joakimsen
• [asterisk-users] fax / t38 gateway   Steve Underwood
• [asterisk-users] fax / t38 gateway   Steve Underwood
• [asterisk-users] OT: Disable Polycom 650 Forward Softkey   Darrick Hartman
• [asterisk-users] Panasonic x Asterisk ... NO PROBLEM!   C F
• [asterisk-users] CDR Records are not working   Pedram M
• [asterisk-users] CDR Records are not working   Klaverstyn, David C
• [asterisk-users] Asterisk and voice recognition   Christian
• [asterisk-users] OT: Disable Polycom 650 Forward Softkey   Andrew Joakimsen
• [asterisk-users] Asterisk and voice recognition   Darren Sessions
• [asterisk-users] Asterisk and voice recognition   Christian
• [asterisk-users] Asterisk 1.6 CDR no Clid information   David Klaverstyn
• [asterisk-users] Asterisk and voice recognition   Darren Sessions
• [asterisk-users] OT: Disable Polycom 650 Forward Softkey   Tilghman Lesher
• [asterisk-users] Panasonic x Asterisk ... NO PROBLEM!   Richard Scobie
• [asterisk-users] autodialed call forwarding via meetme or queue (was predictive dialer)   Roi Stork
• [asterisk-users] openser+asterisk   jordan pan
• [asterisk-users] Asterisk 1.6 pbx_lua not creating any contexts   Charles Duffy
• [asterisk-users] jingle/gtalk still very troubling   Philippe Sultan
• [asterisk-users] openser+asterisk   Alex Balashov
• [asterisk-users] Asterisk and voice recognition   Christian
• [asterisk-users] Fring: Open VPN client to be installed on the mobile, which mobile?   bilal ghayyad
• [asterisk-users] Fring: Open VPN client to be installed on the mobile, which mobile?   Alex Balashov
• [asterisk-users] Fring: Open VPN client to be installed on the mobile, which mobile?   bilal ghayyad
• [asterisk-users] Fring: Open VPN client to be installed on the mobile, which mobile?   Alex Balashov
• [asterisk-users] Fring: Open VPN client to be installed on the mobile, which mobile?   bilal ghayyad
• [asterisk-users] Fring: Open VPN client to be installed on the mobile, which mobile?   Atis Lezdins
• [asterisk-users] KEY SYSTEM, Intercom   hbk
• [asterisk-users] Secure WAN connection too * extention   hbk
• [asterisk-users] Door phone   hbk
• [asterisk-users] OT: Disable Polycom 650 Forward Softkey   Dave Fullerton
• [asterisk-users] KEY SYSTEM, Intercom   Alex Balashov
• [asterisk-users] Secure WAN connection too * extention   Alex Balashov
• [asterisk-users] Secure WAN connection too * extention   Steve Totaro
• [asterisk-users] Door phone   Steve Totaro
• [asterisk-users] Door phone   Darren Severino
• [asterisk-users] KEY SYSTEM, Intercom   Gordon Henderson
• [asterisk-users] Fring: Open VPN client to be installed on the mobile, which mobile?   bilal ghayyad
• [asterisk-users] Fring: Open VPN client to be installed on the mobile, which mobile?   Tzafrir Cohen
• [asterisk-users] OT: Disable Polycom 650 Forward Softkey   Michael Graves
• [asterisk-users] How to bind a SIP channel to an IP   srinivas Antarvedi
• [asterisk-users] fax / t38 gateway   Brendan Martens
• [asterisk-users] hammering imap vmail storage   Brendan Martens
• [asterisk-users] hammering imap vmail storage   Brian J. Murrell
• [asterisk-users] Sporadic One Way Audio   Brent Davidson
• [asterisk-users] Forcing repacketization on SIP to SIP call   Richard Brady
• [asterisk-users] change codec mid-call   Roger Casaponsa
• [asterisk-users] Fring: Open VPN client to be installed on the mobile, which mobile?   Mark Hamilton
• [asterisk-users] fax / t38 gateway   hh174
• [asterisk-users] Fring: Open VPN client to be installed on the mobile, which mobile?   bilal ghayyad
• [asterisk-users] hammering imap vmail storage   Mark Michelson
• [asterisk-users] make config update-rc.d on Debian   Philipp Kempgen
• [asterisk-users] Door phone   C F
• [asterisk-users] Panasonic x Asterisk ... NO PROBLEM!   C F
• [asterisk-users] Fring: Open VPN client to be installed on the mobile, which mobile?   Eric Chamberlain
• [asterisk-users] Asterisk and rawplayer   Ade Vickers
• [asterisk-users] Fring: Open VPN client to be installed on the mobile, which mobile?   Administrator TOOTAI
• [asterisk-users] Fring: Open VPN client to be installed on the mobile, which mobile?   Jeff LaCoursiere
• [asterisk-users] fax / t38 gateway   Wilton Helm
• [asterisk-users] fax / t38 gateway   Kristian Kielhofner
• [asterisk-users] hammering imap vmail storage   Brian J. Murrell
• [asterisk-users] Fring: Open VPN client to be installed on the mobile, which mobile?   Andrew Kohlsmith (lists)
• [asterisk-users] Fring: Open VPN client to be installed on the mobile, which mobile?   Jerry Jones
• [asterisk-users] fax / t38 gateway   Benny Amorsen
• [asterisk-users] fax / t38 gateway   Brendan Martens
• [asterisk-users] Fring: Open VPN client to be installed on the mobile, which mobile?   Jeff LaCoursiere
• [asterisk-users] Sending a text-message to JABBER via CLI   Julien Claassen
• [asterisk-users] asterisk on freebsd   Abel Monzon
• [asterisk-users] Fring: Open VPN client to be installed on the mobile, which mobile?   Alex Balashov
• [asterisk-users] whisper time remaining   Victor Alvarez
• [asterisk-users] whisper time remaining   Alex Balashov
• [asterisk-users] Fring: Open VPN client to be installed on the mobile, which mobile?   Kristian Kielhofner
• [asterisk-users] whisper time remaining   Victor Alvarez
• [asterisk-users] gtalk/jingle full report   Julien Claassen
• [asterisk-users] Asterisk 1.6 CDR no Clid information   David Klaverstyn
• [asterisk-users] whisper time remaining   Fred Posner
• [asterisk-users] [SOLVED] Strange ring tone: Long-Short-Short   Joseph
• [asterisk-users] whisper time remaining   Wilton Helm
• [asterisk-users] whisper time remaining   Alexander Lopez
• [asterisk-users] Best Sales 2008!   asterisk-users at lists.digium.com
• [asterisk-users] OT - Choose PCI or mini-PCI for appliance ?   Olivier
• [asterisk-users] whisper time remaining   Victor Alvarez
• [asterisk-users] jingle/gtalk still very troubling   Julien Claassen
• [asterisk-users] gtalk/jingle full report   Philippe Sultan
• [asterisk-users] gtalk/jingle full report   Julien Claassen
• [asterisk-users] jingle/gtalk still very troubling   Philippe Sultan
• [asterisk-users] Re Fring: Open VPN client to be installed on the mobile, which mobile?   bilal ghayyad
• [asterisk-users] Cheapest 4 port FXO   Thomas Kenyon
• [asterisk-users] AMI - Status Event.   Diego Aguirre (DagMoller)
• [asterisk-users] AMI - Status Event.   Simith Nambiar
• [asterisk-users] Anyone using an Intel Atom ?   Gordon Henderson
• [asterisk-users] Anyone using an Intel Atom ?   Steve Totaro
• [asterisk-users] Dealing with progress codes   arkda
• [asterisk-users] fax / t38 gateway   JD
• [asterisk-users] fax / t38 gateway   Andrew Kohlsmith (lists)
• [asterisk-users] whisper time remaining   Jose P. Espinal
• [asterisk-users] Anyone using an Intel Atom ?   Gordon Henderson
• [asterisk-users] fax / t38 gateway   Gordon Henderson
• [asterisk-users] Anyone using an Intel Atom ?   Steve Totaro
• [asterisk-users] Anyone using an Intel Atom ?   Jared Geiger
• [asterisk-users] Anyone using an Intel Atom ?   Gordon Henderson
• [asterisk-users] fax / t38 gateway   Wilton Helm
• [asterisk-users] Anyone using an Intel Atom ?   Alan Lord
• [asterisk-users] Dropped Calls / Maximum Retries Exceeded / No Reply to Our Critical Packet   Lincoln King-Cliby
• [asterisk-users] fax / t38 gateway   JD
• [asterisk-users] Anyone using an Intel Atom ?   Jeff LaCoursiere
• [asterisk-users] Anyone using an Intel Atom ?   Julian Lyndon-Smith
• [asterisk-users] Dropped Calls / Maximum Retries Exceeded / No Reply to Our Critical Packet   Mark Michelson
• [asterisk-users] fax / t38 gateway   JR Richardson
• [asterisk-users] fax / t38 gateway   Jonn R Taylor
• [asterisk-users] MWI with Siemens Gigaset S450IP   Robert Boardman
• [asterisk-users] R key with Siemens Gigaset IP (was MWI with Siemens Gigaset S450IP)   Robert Boardman
• [asterisk-users] Forcing repacketization on SIP to SIP call   John Todd
• [asterisk-users] Dropped Calls / Maximum Retries Exceeded / No Reply to Our Critical Packet   Benny Amorsen
• [asterisk-users] Multiline Analog Setup   Joseph L. Casale
• [asterisk-users] fax / t38 gateway   Bill Andersen
• [asterisk-users] XML Cisco config file   César García
• [asterisk-users] Dealing with progress codes   arkda
• [asterisk-users] XML Cisco config file   Lincoln King-Cliby
• [asterisk-users] Sendmail for Voicemail   asterisk-users at kfife.mailworks.org
• [asterisk-users] Sendmail for Voicemail   Lyle Giese
• [asterisk-users] CHANUNAVAIL with a TDM800 card   hin lee
• [asterisk-users] Decent Voip Phones for enterprise   Kev Szaszvari
• [asterisk-users] fax / t38 gateway   Steve Underwood
• [asterisk-users] fax / t38 gateway   Steve Underwood
• [asterisk-users] Decent Voip Phones for enterprise   Alex Balashov
• [asterisk-users] Decent Voip Phones for enterprise   Paul Hales
• [asterisk-users] Decent Voip Phones for enterprise   Rob Hillis
• [asterisk-users] fax / t38 gateway   Jonn R Taylor
• [asterisk-users] Decent Voip Phones for enterprise   Ex Vito
• [asterisk-users] Sendmail for Voicemail   David
• [asterisk-users] Snom - we are puzzled   Ronald Wiplinger (Lists)
• [asterisk-users] Decent Voip Phones for enterprise   David Gibbons
• [asterisk-users] any dialplan action on received jabber msgs?   Brian J. Murrell
• [asterisk-users] Dealing with progress codes   Juan Rodríguez
• [asterisk-users] Decent Voip Phones for enterprise   Daniel Hazelbaker
• [asterisk-users] R key with Siemens Gigaset IP (was MWI with Siemens Gigaset S450IP)   Olivier
• [asterisk-users] MWI with Siemens Gigaset S450IP   Olivier
• [asterisk-users] any dialplan action on received jabber msgs?   Olivier
• [asterisk-users] Anyone using an Intel Atom ?   Gordon Henderson
• [asterisk-users] Decent Voip Phones for enterprise   Gordon Henderson
• [asterisk-users] Decent Voip Phones for enterprise   Bruno Castelo Branco
• [asterisk-users] R key with Siemens Gigaset IP (was MWI with Siemens Gigaset S450IP)   Alan Lord
• [asterisk-users] Dial() - any way to limit waiting for a "RINGING" state?   Anton
• [asterisk-users] Anyone using an Intel Atom ?   Steve Totaro
• [asterisk-users] Decent Voip Phones for enterprise   David Gibbons
• [asterisk-users] SIP ACCOUNT CODE not included in CDR when SIP Status is "Unknown"   Shaun Wingrin
• [asterisk-users] Dial() - any way to limit waiting for a "RINGING" state?   Vinícius Fontes
• [asterisk-users] Complete OS/Asterisk disk   Julian Lyndon-Smith
• [asterisk-users] codec not in channel variables   Stanisław Pitucha
• [asterisk-users] Anyone using an Intel Atom ?   Gordon Henderson
• [asterisk-users] network design philosophy and practice   Bill Michaelson
• [asterisk-users] Complete OS/Asterisk disk   Tzafrir Cohen
• [asterisk-users] network design philosophy and practice   Andrew Kohlsmith (lists)
• [asterisk-users] Complete OS/Asterisk disk   Steve Totaro
• [asterisk-users] network design philosophy and practice   Jerry Jones
• [asterisk-users] network design philosophy and practice   Steve Totaro
• [asterisk-users] network design philosophy and practice   Drew Gibson
• [asterisk-users] network design philosophy and practice   Kristian Kielhofner
• [asterisk-users] network design philosophy and practice   Gordon Henderson
• [asterisk-users] network design philosophy and practice   Alex Balashov
• [asterisk-users] What syntax to send user:pass in SIP Dial string?   JR Richardson
• [asterisk-users] network design philosophy and practice   David Gibbons
• [asterisk-users] codec not in channel variables   michel freiha
• [asterisk-users] network design philosophy and practice   Alex Balashov
• [asterisk-users] Dropped Calls / Maximum Retries Exceeded / No Reply to Our Critical Packet   michel freiha
• [asterisk-users] How to bind a SIP channel to an IP   ram
• [asterisk-users] openser+asterisk   ram
• [asterisk-users] Headset Recommendation   Jeremy Mann
• [asterisk-users] Dropped Calls / Maximum Retries Exceeded / No Reply to Our Critical Packet   Lincoln King-Cliby
• [asterisk-users] Sendmail for Voicemail   Todd
• [asterisk-users] Intergrating vicidial with trixbox   James Mutuku
• [asterisk-users] codec not in channel variables   Stanisław Pitucha
• [asterisk-users] Intergrating vicidial with trixbox   Alex Balashov
• [asterisk-users] network design philosophy and practice   Daniel Hazelbaker
• [asterisk-users] network design philosophy and practice   Tilghman Lesher
• [asterisk-users] Intergrating vicidial with trixbox   Ron Byer Jr.
• [asterisk-users] Sendmail for Voicemail   Gordon Henderson
• [asterisk-users] network design philosophy and practice   Alex Balashov
• [asterisk-users] network design philosophy and practice   Jeff LaCoursiere
• [asterisk-users] [OT] Flash player for call recordings - 8khz   Atis Lezdins
• [asterisk-users] network design philosophy and practice   Darrick Hartman
• [asterisk-users] network design philosophy and practice   Drew Gibson
• [asterisk-users] Blank Voicemail.Conf after Password Change   Leah Newmark
• [asterisk-users] Best Sales 2008!   asterisk-users at lists.digium.com
• [asterisk-users] network design philosophy and practice   Daniel Hazelbaker
• [asterisk-users] network design philosophy and practice   Drew Gibson
• [asterisk-users] network design philosophy and practice   David Gibbons
• [asterisk-users] Current Open Source Billing Package   Jerry Jones
• [asterisk-users] Intergrating vicidial with trixbox   Matt Florell
• [asterisk-users] Current Open Source Billing Package   Guillermo V. Salas
• [asterisk-users] Snom - we are puzzled   Christian Stredicke
• [asterisk-users] Current Open Source Billing Package   Jeff LaCoursiere
• [asterisk-users] network design philosophy and practice   Bill Michaelson
• [asterisk-users] network design philosophy and practice   Jerry Jones
• [asterisk-users] Is anyone using * for 2 way video conferencing?   Robert Augustyn
• [asterisk-users] FW: Thecus N7700   Dean Collins
• [asterisk-users] network design philosophy and practice   Jerry Jones
• [asterisk-users] CDP (was Re: network design philosophy and practice)   Kristian Kielhofner
• [asterisk-users] SIP ACCOUNT CODE not included in CDR when SIP Status is "Unknown"   Shaun Wingrin
• [asterisk-users] Is anyone using * for 2 way video conferencing?   Gordon Henderson
• [asterisk-users] app_swift installation problems   dlynam at gmail.com
• [asterisk-users] Is anyone using * for 2 way video conferencing?   Robert Augustyn
• [asterisk-users] Dealing with progress codes   arkda
• [asterisk-users] Is anyone using * for 2 way video conferencing?   Jeff LaCoursiere
• [asterisk-users] Dealing with progress codes   Eric "ManxPower" Wieling
• [asterisk-users] Dealing with progress codes   arkda
• [asterisk-users] Dimensioning a telephony system based on openser!   Nuno Marques
• [asterisk-users] [OpenSIPS-Users] Dimensioning a telephony system based on openser!   Alex Balashov
• [asterisk-users] [OpenSIPS-Users] Dimensioning a telephony system based on openser!   Nuno Marques
• [asterisk-users] XML Cisco config file   César García
• [asterisk-users] [OpenSIPS-Users] Dimensioning a telephony system based on openser!   Alex Balashov
• [asterisk-users] [OpenSIPS-Users] Dimensioning a telephony system based on openser!   Nuno Marques
• [asterisk-users] [OpenSIPS-Users] Dimensioning a telephony system based on openser!   Jai Rangi
• [asterisk-users] XML Cisco config file   OCG Technical Support
• [asterisk-users] [OpenSIPS-Users] Dimensioning a telephony system based on openser!   Alex Balashov
• [asterisk-users] app_swift installation problems   Darren Sessions
• [asterisk-users] 'Asterisk is not thread safe' message   joe mcguckin
• [asterisk-users] [OpenSIPS-Users] Dimensioning a telephony system based on openser!   Jai Rangi
• [asterisk-users] [OpenSIPS-Users] Dimensioning a telephony system based on openser!   Alex Balashov
• [asterisk-users] Current Open Source Billing Package   Darren Wiebe
• [asterisk-users] 'Asterisk is not thread safe' message   Tilghman Lesher
• [asterisk-users] Dealing with progress codes   Juan Rodríguez
• [asterisk-users] Dial() - any way to limit waiting for a "RINGING" state?   Anton
• [asterisk-users] R key with Siemens Gigaset IP (was MWI with Siemens Gigaset S450IP)   Olivier
• [asterisk-users] list-testing   Johan Sandgren
• [asterisk-users] list-testing2   Johan Sandgren
• [asterisk-users] [Kamailio-Users] [OpenSIPS-Users] Dimensioning a telephony system based on openser!   LetMeKnow
• [asterisk-users] R key with Siemens Gigaset IP (was MWI with Siemens Gigaset S450IP)   Alan Lord
• [asterisk-users] Current Open Source Billing Package   Stephen Wingfield
• [asterisk-users] Asterisk 1.6 pbx_lua not creating any contexts   Charles Duffy
• [asterisk-users] up to 3000 lines capacity asterisk Deployment   Dumpolid Exeplish
• [asterisk-users] SIP REGISTER   michel freiha
• [asterisk-users] Is anyone using * for 2 way video conferencing?   Gordon Henderson
• [asterisk-users] mp3player and shoutcast   ACL
• [asterisk-users] Asterisk and rawplayer   Ade Vickers
• [asterisk-users] Linux Kernel >=2.6.25 Realtime issues   Sven Geggus
• [asterisk-users] Linux Kernel >=2.6.25 Realtime issues   Tzafrir Cohen
• [asterisk-users] SIP REGISTER   Alex Balashov
• [asterisk-users] Linux Kernel >=2.6.25 Realtime issues   Sven Geggus
• [asterisk-users] SIP REGISTER   michel freiha
• [asterisk-users] Trouble with SIP/NAT   Noro Hasina
• [asterisk-users] SIP REGISTER   Alex Balashov
• [asterisk-users] Linux Kernel >=2.6.25 Realtime issues   Tzafrir Cohen
• [asterisk-users] Current Open Source Billing Package   Outback Dingo
• [asterisk-users] Sangoma Question   Jeremy Mann
• [asterisk-users] Linux Kernel >=2.6.25 Realtime issues   Sven Geggus
• [asterisk-users] XML Cisco config file   César García
• [asterisk-users] Dealing with progress codes   arkda
• [asterisk-users] Asterisk and rawplayer   BJ Weschke
• [asterisk-users] Sangoma Question   BJ Weschke
• [asterisk-users] network design philosophy and practice   Andrew Latham
• [asterisk-users] up to 3000 lines capacity asterisk Deployment   Andrew Latham
• [asterisk-users] Asterisk settings   michel freiha
• [asterisk-users] R key with Siemens Gigaset IP (was MWI with Siemens Gigaset S450IP)   Olivier
• [asterisk-users] Connection two asterisk via SIP (call forward)   Frank Becker
• [asterisk-users] Asterisk SVN bug segfaulting   Julien Claassen
• [asterisk-users] Sendmail for Voicemail   Olivier
• [asterisk-users] SIP # DTMF   Rodolfo Alcazar Portillo
• [asterisk-users] SIP # DTMF   Eric "ManxPower" Wieling
• [asterisk-users] SIP # DTMF   Anthony Francis
• [asterisk-users] Old mantis e-mails   Daniel Hazelbaker
• [asterisk-users] Asterisk settings   Steve Howes
• [asterisk-users] Old mantis e-mails   Mark Michelson
• [asterisk-users] Old mantis e-mails   Anthony Messina
• [asterisk-users] Other lists   Adam Moffett
• [asterisk-users] Asterisk Legacy PBX   Sriram
• [asterisk-users] [SOLVED] SIP # DTMF   Rodolfo Alcazar Portillo
• [asterisk-users] CHANUNAVAIL with a TDM800 card   hin lee
• [asterisk-users] CHANUNAVAIL with a TDM800 card   Tzafrir Cohen
• [asterisk-users] Dealing with progress codes   Juan Rodríguez
• [asterisk-users] Music On Hold (from a Sound card) Help   Timothy Smith
• [asterisk-users] ISDN - BRI   Wilton Helm
• [asterisk-users] Asterisk Legacy PBX   Mark Best
• [asterisk-users] Dealing with progress codes   Nathan Bowyer
• [asterisk-users] General development funding: discussion and survey   John Todd
• [asterisk-users] ISDN - BRI   Tzafrir Cohen
• [asterisk-users] CHANUNAVAIL with a TDM800 card   hin lee
• [asterisk-users] CHANUNAVAIL with a TDM800 card   Tzafrir Cohen
• [asterisk-users] CHANUNAVAIL with a TDM800 card   hin lee
• [asterisk-users] Message 245058   asterisk-users at lists.digium.com
• [asterisk-users] 1.4.22 vs 1.4.21.2 - IAX2 regression ?   Ex Vito
• [asterisk-users] Linux Kernel >=2.6.25 Realtime issues   Sven Geggus
• [asterisk-users] fax / t38 gateway   Steve Underwood
• [asterisk-users] network design philosophy and practice   Paul Hales
• [asterisk-users] Decent Voip Phones for enterprise   Paul Hales
• [asterisk-users] Asterisk with SC440 Dell(Big Problem)   Edwin Quijada
• [asterisk-users] autodialed call forwarding via meetme or queue (was predictive dialer)   Roi Stork
• [asterisk-users] fax / t38 gateway   Jonn R Taylor
• [asterisk-users] Enter Value and continue dialplan   David Klaverstyn
• [asterisk-users] Enter Value and continue dialplan   Stuart Sheldon
• [asterisk-users] fax / t38 gateway   Kristian Kielhofner
• [asterisk-users] twice normal beep before busy tone ??   Stefan Guenther
• [asterisk-users] dtmf have not callerid   jordan pan
• [asterisk-users] MusicOnHold from a Sound card   Timothy Smith
• [asterisk-users] Asterisk with SC440 Dell(Big Problem)   Tzafrir Cohen
• [asterisk-users] Friday Halloween Edition Oct 31 12 Noon EDT   randulo
• [asterisk-users] giving a user asterisk CLI access: how bad could it get   fadey
• [asterisk-users] Friday Halloween Edition Oct 31 12 Noon EDT   Hans Witvliet
• [asterisk-users] giving a user asterisk CLI access: how bad could it get   Julien Claassen
• [asterisk-users] Call terminates after 20 minutes   Jim Boykin
• [asterisk-users] giving a user asterisk CLI access: how bad could it get   Tzafrir Cohen
• [asterisk-users] Asterisk/Machine Hang after calling in/out ISDN   Lee, John (Sydney)
• [asterisk-users] Call terminates after 20 minutes   Marcin J. Kowalczyk
• [asterisk-users] Asterisk/Machine Hang after calling in/out ISDN   Tzafrir Cohen
• [asterisk-users] Copy protection issues with G.729 codec in Solaris   Peter Galiovsky
• [asterisk-users] Decent Voip Phones for enterprise   Olivier
• [asterisk-users] Friday Halloween Edition Oct 31 12 Noon EDT   randulo
• [asterisk-users] Asterisk/Machine Hang after calling in/out ISDN   Tarek Sawah
• [asterisk-users] Copy protection issues with G.729 codec in Solaris   Rob Hillis
• [asterisk-users] Monitor group calls (recording calls)   Paolo Carpo
• [asterisk-users] Monitor group calls (recording calls)   Martin Smith
• [asterisk-users] No audio after transferring to voicemail   Jeremy Phillips
• [asterisk-users] Asterisk Legacy pBx help   Sriram
• [asterisk-users] VoIP phones supporting speex   Rajkumar S
• [asterisk-users] Asterisk with SC440 Dell(Big Problem)   Edwin Quijada
• [asterisk-users] up to 3000 lines capacity asterisk Deployment   Alexander Lopez
• [asterisk-users] CDR Posting Delay   Douglas Garstang
• [asterisk-users] Sendmail for Voicemail   Brent Davidson
• [asterisk-users] Sendmail for Voicemail   Tzafrir Cohen
• [asterisk-users] Call problems   Emmanuel Pascal Bruno
• [asterisk-users] Asterisk installation   Christian
• [asterisk-users] Asterisk installation   Darrick Hartman
• [asterisk-users] Asterisk installation   Christian
• [asterisk-users] Asterisk installation   Christian
• [asterisk-users] Asterisk installation   David Klaverstyn
• [asterisk-users] giving a user asterisk CLI access: how bad could it get   Dima
• [asterisk-users] twice normal beep before busy tone ??   Robert Lister
• [asterisk-users] Asterisk installation   Eric "ManxPower" Wieling
• [asterisk-users] giving a user asterisk CLI access: how bad could it get   Tzafrir Cohen
• [asterisk-users] Blank Voicemail.Conf after Password Change   Robert Lister
• [asterisk-users] Asterisk installation   Christian
• [asterisk-users] fax / t38 gateway   Jonn R Taylor
• [asterisk-users] giving a user asterisk CLI access: how bad could it get   Jeff LaCoursiere
• [asterisk-users] twice normal beep before busy tone ??   Robert Lister
• [asterisk-users] VoIP traffic shaping   OCG Technical Support

Last message date: Fri Oct 31 23:36:12 CDT 2008
Archived on: Fri Oct 31 23:36:18 CDT 2008

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