November 2008 Archives by date

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Starting: Sat Nov 1 01:25:59 CDT 2008
Ending: Sun Nov 30 23:25:03 CST 2008
Messages: 1503

• [asterisk-users] Wierd queue question   Dan Austin
• [asterisk-users] Wierd queue question   Julian Lyndon-Smith
• [asterisk-users] SPA3102 interdigit timers bug?   Rodolfo Alcazar Portillo
• [asterisk-users] Wierd queue question   Dan Austin
• [asterisk-users] 2 Or (971) Numbers for sale   Babcock, Michael Alex
• [asterisk-users] Call problems   Emmanuel Pascal Bruno
• [asterisk-users] Call problems   Rob Hillis
• [asterisk-users] asterisk-users Digest, Vol 52, Issue 1   Stefan Guenther
• [asterisk-users] Asterisk with SC440 Dell(Big Problem)   Jared Geiger
• [asterisk-users] Call problems   Emmanuel Pascal Bruno
• [asterisk-users] asterisk 1.2 and Dial with LIMIT_WARNING_FILE   Rafael Visser
• [asterisk-users] Wierd queue question   Julian Lyndon-Smith
• [asterisk-users] Gizmo ok but no audio when incoming   Rodolfo Alcazar Portillo
• [asterisk-users] giving a user asterisk CLI access: how bad could it get   Alexander Lopez
• [asterisk-users] giving a user asterisk CLI access: how bad could it get   Tilghman Lesher
• [asterisk-users] Ztdummy and Asterisk   Aldo D. Sudak
• [asterisk-users] Call terminates after 20 minutes   Jim Boykin
• [asterisk-users] Astribank loop current adjustment   Tzafrir Cohen
• [asterisk-users] Asterisk and Media gateway controller   Pedro G
• [asterisk-users] Question regarding keywords in sip.conf/users.conf   Rob Hillis
• [asterisk-users] Ztdummy and Asterisk   Dobry Dobrev
• [asterisk-users] Ztdummy and Asterisk   Tzafrir Cohen
• Returned mail: see transcript for details   Mail Delivery Subsystem
• [asterisk-users] Call problems   Emmanuel Pascal Bruno
• [asterisk-users] Call problems   Rob Hillis
• [asterisk-users] Call terminates after 20 minutes   Jim Boykin
• [asterisk-users] Ztdummy and Asterisk   Aldo D. Sudak
• [asterisk-users] Asterisk/Machine Hang after calling in/out ISDN   Lee, John (Sydney)
• [asterisk-users] VoIP traffic shaping   Kristian Kielhofner
• [asterisk-users] Question regarding keywords in sip.conf/users.conf   Paul Hales
• [asterisk-users] Call terminates after 20 minutes   Juan Rodríguez
• [asterisk-users] Polycom 430 no hangup after SIP BYE, Status 481 instead   Joel Pearson
• [asterisk-users] Call problems   Emmanuel Pascal Bruno
• [asterisk-users] asterisk and bigmem kernel   Edgar Guadamuz
• [asterisk-users] asterisk and bigmem kernel   Tzafrir Cohen
• [asterisk-users] Question regarding keywords in sip.conf/users.conf   Rob Hillis
• [asterisk-users] Call problems   Eberhard Roloff
• [asterisk-users] say load new   michael_t at gazeta.pl
• [asterisk-users] say load new   Igor Goncharovsky
• [asterisk-users] SPA3102 interdigit timers bug?   Steve Davies
• [asterisk-users] Hang up detection with TDM400P and Telewest/Virgin Media line   Ed W
• [asterisk-users] help with debugging phone call   Jerry Geis
• [asterisk-users] loading misdn.conf strange error regarding out of range   Julien Claassen
• [asterisk-users] help with debugging phone call   Jerry Geis
• [asterisk-users] Call quality issue across VPN-> POTS vs SIP   Lincoln King-Cliby
• [asterisk-users] Looking for a web video phone?   Robert Augustyn
• [asterisk-users] Call terminates after 20 minutes   John Todd
• [asterisk-users] Call quality issue across VPN-> POTS vs SIP   Bob Pierce
• [asterisk-users] Looking for a web video phone?   Fred Posner
• [asterisk-users] CDR Posting Delay   John Todd
• [asterisk-users] giving a user asterisk CLI access: how bad could it get   John Todd
• [asterisk-users] busylevel question   Jim Dickenson
• [asterisk-users] Looking for a web video phone?   Robert Augustyn
• [asterisk-users] CLI dial and echo recorder   Shaun Wingrin
• [asterisk-users] Call quality issue across VPN-> POTS vs SIP   Lincoln King-Cliby
• [asterisk-users] Blank Voicemail.Conf after Password Change   Leah Newmark
• [asterisk-users] Ztdummy and Asterisk   Stefan Tichy
• [asterisk-users] Call quality issue across VPN-> POTS vs SIP   Bob Pierce
• [asterisk-users] Call quality issue across VPN-> POTS vs SIP   Lincoln King-Cliby
• [asterisk-users] VoIP traffic shaping   Drew Gibson
• [asterisk-users] Ztdummy and Asterisk   Aldo D. Sudak
• [asterisk-users] asterisk src=dst   Ruddy Gbaguidi
• [asterisk-users] Blank Voicemail.Conf after Password Change   Tilghman Lesher
• [asterisk-users] Question regarding keywords in sip.conf/users.conf   Barry L. Kline
• [asterisk-users] Question regarding keywords in sip.conf/users.conf   Rob Hillis
• [asterisk-users] Question regarding keywords in sip.conf/users.conf   Eric "ManxPower" Wieling
• [asterisk-users] manager event privilege: call, all? what is?   Matanya Cohen
• [asterisk-users] Is SIPPEER curcalls working for you ?   Olivier
• [asterisk-users] SPA3102 interdigit timers bug?   Rodolfo Alcazar Portillo
• [asterisk-users] SPA-962 & Asterisk   Steve Anness
• [asterisk-users] SPA-962 Time on Asterisk   Steve Anness
• [asterisk-users] Ztdummy and Asterisk   Aldo D. Sudak
• [asterisk-users] Is SIPPEER curcalls working for you ?   Igor Zamocky
• [asterisk-users] 1.4.22 vs 1.4.21.2 - IAX2 regression ?   Igor Zamocky
• [asterisk-users] SPA-962 & Asterisk   David Gibbons
• [asterisk-users] VoIP Users Conference Call Friday Nov 7 On Wideband Voice & Conferencing   Michael Graves
• [asterisk-users] Is SIPPEER curcalls working for you ? [SOLVED]   Olivier
• [asterisk-users] VoIP Users Conference Call Friday Nov 7 On Wideband Voice & Conferencing   Tzafrir Cohen
• [asterisk-users] VoIP Users Conference Call Friday Nov 7 On Wideband Voice & Conferencing   Michael Graves
• [asterisk-users] SPA-962 Time on Asterisk   Steve Howes
• [asterisk-users] Call terminates after 20 minutes   Jim Boykin
• [asterisk-users] VoIP Users Conference Call Friday Nov 7 On Wideband Voice & Conferencing   randulo
• [asterisk-users] giving a user asterisk CLI access: how bad could it get   Dima
• [asterisk-users] VoIP Users Conference Call Friday Nov 7 On Wideband Voice & Conferencing   SIP
• [asterisk-users] VoIP Users Conference Call Friday Nov 7 On Wideband Voice & Conferencing   Michael Graves
• [asterisk-users] users.conf and hints   Jeremy Mann
• [asterisk-users] VoIP Users Conference Call Friday Nov 7 On Wideband Voice & Conferencing   Steve Edwards
• [asterisk-users] OT: Traffic Shaping   andrea
• [asterisk-users] VoIP Users Conference Call Friday Nov 7 On Wideband Voice & Conferencing   Michael Graves
• [asterisk-users] SPA-962 Time on Asterisk   Rob Hillis
• [asterisk-users] [Astlinux-users] VoIP Users Conference Call Friday Nov 7 On Wideband Voice & Conferencing   Kristian Kielhofner
• [asterisk-users] What is the best way to resale termination/origination?   Robert Augustyn
• [asterisk-users] [Astlinux-users] VoIP Users Conference Call Friday Nov 7 On Wideband Voice & Conferencing   randulo
• [asterisk-users] Some progress, anyway...   Philip Prindeville
• [asterisk-users] Some progress, anyway...   Darrick Hartman
• [asterisk-users] Blank Voicemail.Conf after Password Change   Leah Newmark
• [asterisk-users] Some progress, anyway...   Philip Prindeville
• [asterisk-users] dahdi trunk does not compile with kernel 2.6.27   John covici
• [asterisk-users] Blank Voicemail.Conf after Password Change   Tilghman Lesher
• [asterisk-users] dahdi trunk does not compile with kernel 2.6.27   Tzafrir Cohen
• [asterisk-users] dahdi trunk does not compile with kernel 2.6.27   John covici
• [asterisk-users] Call terminates after 20 minutes   John Todd
• [asterisk-users] SPA-962 & Asterisk   John Todd
• [asterisk-users] OT: Traffic Shaping   Matt Riddell
• [asterisk-users] shared voicemail box   Kelvin Chan
• [asterisk-users] 1.4.22 vs 1.4.21.2 - IAX2 regression ?   Ex Vito
• [asterisk-users] shared voicemail box   Carlos Chavez
• [asterisk-users] giving a user asterisk CLI access: how bad could it get   Ruddy Gbaguidi
• [asterisk-users] giving a user asterisk CLI access: how bad could it get   Jeff LaCoursiere
• [asterisk-users] Channels are increasing without limit - Please Help!   Daniel - Asterisk
• [asterisk-users] giving a user asterisk CLI access: how bad could it get   Tilghman Lesher
• [asterisk-users] giving a user asterisk CLI access: how bad could it get   Tilghman Lesher
• [asterisk-users] users.conf and hints   Tilghman Lesher
• [asterisk-users] WARNING message when calls get into a queue with realtime members (Local channel)   Daniel - Asterisk
• [asterisk-users] Sendmail using SMTP authorization   hugolivude
• [asterisk-users] Sendmail using SMTP authorization   Matt Gibson
• [asterisk-users] MS Exchange IMAP Voicemail   Andrew Joakimsen
• [asterisk-users] giving a user asterisk CLI access: how bad could it get   Tzafrir Cohen
• [asterisk-users] giving a user asterisk CLI access: how bad could it get   Tzafrir Cohen
• [asterisk-users] twice normal beep before busy tone ??   Stefan Guenther
• [asterisk-users] twice normal beep before busy tone ??   Matt Riddell
• [asterisk-users] How is it best to initialize specific SIP peer settings   Olivier
• [asterisk-users] AEL NoOp not working   Olivier
• [asterisk-users] giving a user asterisk CLI access: how bad could it get   Jeff LaCoursiere
• [asterisk-users] AEL NoOp not working   Atis Lezdins
• [asterisk-users] giving a user asterisk CLI access: how bad could it get   Tzafrir Cohen
• [asterisk-users] SIP Qualify is not working with Postgres   Marcelo Terres
• [asterisk-users] sas   sas sas
• [asterisk-users] Dundi Issues   Jeremy Mann
• [asterisk-users] 800 origination   Jeff LaCoursiere
• [asterisk-users] Dundi Issues   Jeremy Mann
• [asterisk-users] sas   Steve Howes
• [asterisk-users] Phishing attempt   Roderick A. Anderson
• [asterisk-users] AEL NoOp not working   Olivier
• [asterisk-users] Dundi Issues   Drew Gibson
• [asterisk-users] Dundi Issues   Jeremy Mann
• [asterisk-users] br.Doctor Ester   asterisk-users at lists.digium.com
• [asterisk-users] The skype channel...   Jared Smith
• [asterisk-users] Phishing attempt   Steve Totaro
• [asterisk-users] Phishing attempt   Alexandru Pirvulescu
• [asterisk-users] Phishing attempt   Atis Lezdins
• [asterisk-users] AEL NoOp not working   Atis Lezdins
• [asterisk-users] Phishing attempt   Roderick A. Anderson
• [asterisk-users] Phishing attempt   Roderick A. Anderson
• [asterisk-users] Phishing attempt   Tzafrir Cohen
• [asterisk-users] 800 origination   Babcock, Michael Alex
• [asterisk-users] The skype channel...   Julien Claassen
• [asterisk-users] bugs.digium.com Maintenance and Downtime   Kevin P. Fleming
• [asterisk-users] Dundi Issues   Drew Gibson
• [asterisk-users] SER/Asterisk interworking mailing list.   Alex Balashov
• [asterisk-users] Phishing attempt   John Todd
• [asterisk-users] ExtenSpy? am I doing it correctly?   Steve Gladden
• [asterisk-users] Phishing attempt   Roderick A. Anderson
• [asterisk-users] How is it best to initialize specific SIP peer settings   John Todd
• [asterisk-users] Phishing attempt   Igor Hernandez
• [asterisk-users] Phishing attempt   Philipp Kempgen
• [asterisk-users] Phishing attempt   Eric Fort
• [asterisk-users] Phishing attempt   Wilton Helm
• [asterisk-users] Phishing attempt   Eric Fort
• [asterisk-users] Inbound/Outbound undesired behavior   Ricardo Melendez
• [asterisk-users] a callerid question   Babcock, Michael Alex
• [asterisk-users] b410p mIDSN - RNIS signaling problems   Olivier Fauchon
• [asterisk-users] ExtenSpy? am I doing it correctly?   Steve Gladden
• [asterisk-users] Type 102 Millwatt Test Line   John Meksavan
• [asterisk-users] Phishing attempt   Thomas Kenyon
• [asterisk-users] Type 102 Millwatt Test Line   Alex Balashov
• [asterisk-users] Phishing attempt   Mark Michelson
• [asterisk-users] SER/Asterisk interworking mailing list.   Jai Rangi
• [asterisk-users] Phishing attempt   Jeff LaCoursiere
• [asterisk-users] Cisco 776M Any good for connection to local Asterisk server?   Ronny Julian
• [asterisk-users] Phishing attempt   Matt Riddell
• [asterisk-users] TDM400 with FXS some handsets not ringing   Mike
• [asterisk-users] TDM400 with FXS some handsets not ringing   Brent Davidson
• [asterisk-users] TDM400 with FXS some handsets not ringing   Matt Riddell
• [asterisk-users] Asterisk Realtime Configuration   Pedram M
• [asterisk-users] Polycom's lose BLF after Asterisk restart   Thermal Wetland
• [asterisk-users] Asterisk Realtime Configuration   Matt Riddell
• [asterisk-users] Asterisk Realtime Configuration   Pedram M
• [asterisk-users] Agent Question   Joseph L. Casale
• [asterisk-users] Asterisk Realtime Configuration   Matt Riddell
• [asterisk-users] Call Files   David Klaverstyn
• [asterisk-users] Call Files   Matt Riddell
• [asterisk-users] Missing callerid   Yelson Vivas
• [asterisk-users] Asterisk Realtime Configuration   Pedram M
• [asterisk-users] Recommend Wireless IP Phone   Pedram M
• [asterisk-users] ISDN Cause Code 100, Bosch Integral Management Connection   Wolfgang Pichler
• [asterisk-users] Recommend Wireless IP Phone   Emmanuel Pascal Bruno
• [asterisk-users] TDM400 with FXS some handsets not ringing   Gordon Henderson
• [asterisk-users] Recommend Wireless IP Phone   Gordon Henderson
• [asterisk-users] Phishing attempt   Thomas Kenyon
• [asterisk-users] Phishing attempt   Matt Riddell
• [asterisk-users] twice normal beep before busy tone ??   Stefan Guenther
• [asterisk-users] Phishing attempt   Thomas Kenyon
• [asterisk-users] Phishing attempt   Tzafrir Cohen
• [asterisk-users] asterisk.conf ====> maxload   andrea
• [asterisk-users] Phishing attempt   Thomas Kenyon
• [asterisk-users] Asterisk and rawplayer   Ade Vickers
• [asterisk-users] HD Voice conference Friday Nov 7th @ 12 Noon EST   randulo
• [asterisk-users] Spam from DIDForSale <contact-sales at didforsale.com>   Gordon Henderson
• [asterisk-users] TDM400 with FXS some handsets not ringing   Mike
• [asterisk-users] Spam from DIDForSale <contact-sales at didforsale.com>   bails
• [asterisk-users] ExtenSpy? am I doing it correctly?   Marco Signorini
• [asterisk-users] Errors in console for zap   Jon Weisman
• [asterisk-users] crashes after upgrade from 1.2.16 to 1.4.21.2   Louis-David Mitterrand
• [asterisk-users] RFC: multiple packages editing asterisk config files   Tzafrir Cohen
• [asterisk-users] Polycom's lose BLF after Asterisk restart   Mr Shunz
• [asterisk-users] Polycom's lose BLF after Asterisk restart   Ed W
• [asterisk-users] Recommend Wireless IP Phone   Grygoriy Dobrovolskyy
• [asterisk-users] crashes after upgrade from 1.2.16 to 1.4.21.2   Doug Lytle
• [asterisk-users] tired of "midget packet received" warnings   Louis-David Mitterrand
• [asterisk-users] Phishing attempt   Jeff LaCoursiere
• [asterisk-users] AEL NoOp not working   Olivier
• [asterisk-users] Phishing attempt   Steve Totaro
• [asterisk-users] Recommend Wireless IP Phone   Fred Posner
• [asterisk-users] Recommend Wireless IP Phone   randulo
• [asterisk-users] ODBCExec from Dialplan   Sebastian Gutierrez
• [asterisk-users] Recommend Wireless IP Phone   Michael Graves
• [asterisk-users] RFC: multiple packages editing asterisk config files   Tzafrir Cohen
• [asterisk-users] Recommend Wireless IP Phone   randulo
• [asterisk-users] tired of "midget packet received" warnings   Kevin P. Fleming
• [asterisk-users] ODBCExec from Dialplan   Jared Smith
• [asterisk-users] tired of "midget packet received" warnings   Louis-David Mitterrand
• [asterisk-users] RFC: multiple packages editing asterisk config files   Tzafrir Cohen
• [asterisk-users] tired of "midget packet received" warnings   Tilghman Lesher
• [asterisk-users] ODBCExec from Dialplan   Sebastian Gutierrez
• [asterisk-users] Call Files   Ruddy Gbaguidi
• [asterisk-users] Asterisk trunking   Jan Prunk
• [asterisk-users] twice normal beep before busy tone ??   Eric "ManxPower" Wieling
• [asterisk-users] Spam from DIDForSale <contact-sales at didforsale.com>   Steve Edwards
• [asterisk-users] Spam from DIDForSale <contact-sales at didforsale.com>   David Gibbons
• [asterisk-users] Variable Scope Question   Brent Davidson
• [asterisk-users] Asking again about busylevel   Jim Dickenson
• [asterisk-users] Spam from DIDForSale <contact-sales at didforsale.com>   Gonzalo Servat
• [asterisk-users] Variable Scope Question   Atis Lezdins
• [asterisk-users] Spam from DIDForSale <contact-sales at didforsale.com>   Anthony Francis
• [asterisk-users] Variable Scope Question   Tilghman Lesher
• [asterisk-users] Recommend Wireless IP Phone   Wilton Helm
• [asterisk-users] Spam from DIDForSale <contact-sales at didforsale.com>   Greg Woods
• [asterisk-users] Variable Scope Question   Brent Davidson
• [asterisk-users] Spam from DIDForSale <contact-sales at didforsale.com>   Anthony Francis
• [asterisk-users] twice normal beep before busy tone ??   Wilton Helm
• [asterisk-users] [OT] Capitalism (was: Spam from DIDForSale <contact-sales at didforsale.com>)   Atis Lezdins
• [asterisk-users] Variable Scope Question   Tilghman Lesher
• [asterisk-users] Recommend Wireless IP Phone   Michael Graves
• [asterisk-users] twice normal beep before busy tone ??   Stefan Guenther
• [asterisk-users] Phishing attempt   Philipp Kempgen
• [asterisk-users] Spam from DIDForSale <contact-sales at didforsale.com>   David Gibbons
• [asterisk-users] Spam from DIDForSale <contact-sales at didforsale.com>   Singer X.J. Wang
• [asterisk-users] ODBCExec and Asterisk 1.6 New Thread   Sebastian Gutierrez
• [asterisk-users] [OT] Re: Phishing attempt   Philipp Kempgen
• [asterisk-users] ODBCExec and Asterisk 1.6 New Thread   Tilghman Lesher
• [asterisk-users] Spam from DIDForSale <contact-sales at didforsale.com>   SIP
• [asterisk-users] AEL NoOp not working   Steve Murphy
• [asterisk-users] Spam from DIDForSale <contact-sales at didforsale.com>   Brent Davidson
• [asterisk-users] RFC: multiple packages editing asterisk config files   Steve Murphy
• [asterisk-users] twice normal beep before busy tone ??   Eric "ManxPower" Wieling
• [asterisk-users] Spam from DIDForSale <contact-sales at didforsale.com>   Anthony Plack
• [asterisk-users] ExtenSpy? am I doing it correctly?   Jim Dickenson
• [asterisk-users] SPA-962 & Asterisk   Paul Hales
• [asterisk-users] SPA-962 & Asterisk   Wilton Helm
• [asterisk-users] Polycom 430 no hangup after SIP BYE, Status 481 instead   Joel Pearson
• [asterisk-users] ODBCExec and Asterisk 1.6 New Thread   Sebastian Gutierrez
• [asterisk-users] asterisk and bigmem kernel   Edgar Guadamuz
• [asterisk-users] ODBCExec and Asterisk 1.6 New Thread   Tilghman Lesher
• [asterisk-users] Help on g729 CODEC   vivek rastogi
• [asterisk-users] Help on g729 CODEC   Tilghman Lesher
• [asterisk-users] Use of optional new number in ISDN release 22   Cristian Dimache
• [asterisk-users] [OT] Reporting Spam   randulo
• [asterisk-users] SPA-962 & Asterisk   Tzafrir Cohen
• [asterisk-users] tired of "midget packet received" warnings   Louis-David Mitterrand
• [asterisk-users] tired of "midget packet received" warnings   Rob Hillis
• [asterisk-users] asterisk - avaya ip office SIP trunking   Krishna Sumanth Chava
• [asterisk-users] tired of "midget packet received" warnings   Tim Panton
• [asterisk-users] tired of "midget packet received" warnings   Louis-David Mitterrand
• [asterisk-users] RFC: multiple packages editing asterisk config files   Tzafrir Cohen
• [asterisk-users] ISDN Cause Code 100, Bosch Integral Management Connection   Johann Steinwendtner
• [asterisk-users] AEL NoOp not working [SOLVED]   Olivier
• [asterisk-users] ODBCExec and Asterisk 1.6 New Thread   Sebastian Gutierrez
• [asterisk-users] is it possible to deactivate RTCP?   Klaus Darilion
• [asterisk-users] Help with asterisk and avaya SIP trunking   Krishna Sumanth Chava
• [asterisk-users] T.38 without port changes   Klaus Darilion
• [asterisk-users] Cisco 776M Any good for connection to local Asterisk server?   peder at networkoblivion.com
• [asterisk-users] 1.6 Production ready??   Sebastian Gutierrez
• [asterisk-users] TE121B Doesn't Fit PCI-E Slot   David Budny
• [asterisk-users] help with dialplan   Jerry Geis
• [asterisk-users] tired of "midget packet received" warnings   Tim Panton
• [asterisk-users] help with dialplan   Brent Davidson
• [asterisk-users] help with dialplan   Jerry Geis
• [asterisk-users] help with dialplan   Brent Davidson
• [asterisk-users] help with dialplan   Doug Lytle
• [asterisk-users] DNS A queries for channel   samuel
• [asterisk-users] help with dialplan   Steve Totaro
• [asterisk-users] help with dialplan   Steve Totaro
• [asterisk-users] 1.6 Production ready??   Matthew Fredrickson
• [asterisk-users] help with dialplan   Doug Lytle
• [asterisk-users] is it possible to deactivate RTCP?   Anthony Francis
• [asterisk-users] 1.6 Production ready??   Steve Totaro
• [asterisk-users] Providing Ringback   Igor Hernandez
• [asterisk-users] 1.6 Production ready??   Matthew Fredrickson
• [asterisk-users] REFER problems with Asterisk and OpenSER   Michał Ostrowski
• [asterisk-users] 1.6 Production ready??   Sebastian Gutierrez
• [asterisk-users] DNS A queries for channel   John Todd
• [asterisk-users] is it possible to deactivate RTCP?   John Todd
• [asterisk-users] 1.6 Production ready??   Jonn R Taylor
• [asterisk-users] 1.6 Production ready??   Freddi Hansen
• [asterisk-users] Help with asterisk and avaya SIP trunking   Robert Boardman
• [asterisk-users] tired of "midget packet received" warnings   Benny Amorsen
• [asterisk-users] Outgoing SIP calls dropped after 30 seconds.   Kurt Knudsen
• [asterisk-users] SPA-962 & Asterisk   Wilton Helm
• [asterisk-users] tired of "midget packet received" warnings   Tzafrir Cohen
• [asterisk-users] Outgoing SIP calls dropped after 30 seconds.   SIP
• [asterisk-users] SPA-962 & Asterisk   Tzafrir Cohen
• [asterisk-users] Outgoing SIP calls dropped after 30 seconds.   Doug
• [asterisk-users] Outgoing SIP calls dropped after 30 seconds.   Kurt Knudsen
• [asterisk-users] Outgoing SIP calls dropped after 30 seconds.   Kurt Knudsen
• [asterisk-users] 1.6 Production ready??   Matthew Fredrickson
• [asterisk-users] 1.6 Production ready??   Matthew Fredrickson
• [asterisk-users] tired of "midget packet received" warnings   Paul Hales
• [asterisk-users] TE121B Doesn't Fit PCI-E Slot   Joseph L. Casale
• [asterisk-users] tired of "midget packet received" warnings   Rob Hillis
• [asterisk-users] Outgoing SIP calls dropped after 30 seconds.   Grey Man
• [asterisk-users] Providing Ringback   Grey Man
• [asterisk-users] Providing Ringback   Igor Hernandez
• [asterisk-users] ISDN for Dahdi get hangs   Clive.Chan(ATN)
• [asterisk-users] tired of "midget packet received" warnings   Louis-David Mitterrand
• [asterisk-users] Outgoing SIP calls dropped after 30 seconds.   Steve Totaro
• [asterisk-users] tired of "midget packet received" warnings   Tzafrir Cohen
• [asterisk-users] tired of "midget packet received" warnings   Rob Hillis
• [asterisk-users] TE121B Doesn't Fit PCI-E Slot   David Budny
• [asterisk-users] Outgoing SIP calls dropped after 30 seconds.   Kurt Knudsen
• [asterisk-users] TE121B Doesn't Fit PCI-E Slot   Steve Totaro
• [asterisk-users] Help with asterisk and avaya SIP trunking   Robert Boardman
• [asterisk-users] TE121B Doesn't Fit PCI-E Slot   Eberhard Roloff
• [asterisk-users] tired of "midget packet received" warnings   Atis Lezdins
• [asterisk-users] tired of "midget packet received" warnings   Tzafrir Cohen
• [asterisk-users] Help with asterisk and avaya SIP trunking   Krishna Sumanth Chava
• [asterisk-users] Rolled Distro?   Jason Lixfeld
• [asterisk-users] Help with asterisk and avaya SIP trunking   Shaun Ewing
• [asterisk-users] Sendmail using SMTP authorization   hugolivude
• [asterisk-users] Rolled Distro?   Joseph L. Casale
• [asterisk-users] TE121B Doesn't Fit PCI-E Slot   Joseph L. Casale
• [asterisk-users] Sendmail using SMTP authorization   OCG Technical Support
• [asterisk-users] set(CALLERID(name) not working   sean darcy
• [asterisk-users] set(CALLERID(name) not working   Trevor Peirce
• [asterisk-users] set(CALLERID(name) not working   OCG Technical Support
• [asterisk-users] tired of "midget packet received" warnings   Russell Bryant
• [asterisk-users] tired of "midget packet received" warnings   Rob Hillis
• [asterisk-users] Codec problems when using G.723   Thomas Winter
• [asterisk-users] Codec problems when using G.723   Eric "ManxPower" Wieling
• [asterisk-users] TE121B Doesn't Fit PCI-E Slot   Eberhard Roloff
• [asterisk-users] Recommend Wireless IP Phone   Ed W
• [asterisk-users] Rolled Distro?   Kevin P. Fleming
• [asterisk-users] Recommend Wireless IP Phone   Wilton Helm
• [asterisk-users] Rolled Distro?   Matt Riddell
• [asterisk-users] set(CALLERID(name) not working   sean darcy
• [asterisk-users] set(CALLERID(name) not working   sean darcy
• [asterisk-users] Help with asterisk and avaya SIP trunking   Krishna Sumanth Chava
• [asterisk-users] Help with asterisk and avaya SIP trunking   Shaun Ewing
• [asterisk-users] Rolled Distro?   Tzafrir Cohen
• [asterisk-users] crashes after upgrade from 1.2.16 to 1.4.21.2   Louis-David Mitterrand
• [asterisk-users] changing the size of voice packets   Pezhman Lali
• [asterisk-users] changing the size of voice packets   Igor Goncharovsky
• [asterisk-users] DNS A queries for channel   samuel
• [asterisk-users] changing the size of voice packets   Alex Balashov
• [asterisk-users] GEN-GEN and Manual Ring-Down (MRD)?   Tony Mountifield
• [asterisk-users] analog issues using xen virtualization   Jesus Lee
• [asterisk-users] set(CALLERID(name) not working   Doug Lytle
• [asterisk-users] DNS A queries for channel   samuel
• [asterisk-users] changing the size of voice packets   Pezhman Lali
• [asterisk-users] directrtpsetup without reinvite   regs at kinetix.gr
• [asterisk-users] Using AMI to determine PRI Channels Used   David Budny
• [asterisk-users] TE121B Doesn't Fit PCI-E Slot   David Budny
• [asterisk-users] Using AMI to determine PRI Channels Used   Godson Gera
• [asterisk-users] SRTP support in asterisk 1.6   Max Alex
• [asterisk-users] Codec problems when using G.723   Thomas Winter
• [asterisk-users] Codec problems when using G.723   Eric "ManxPower" Wieling
• [asterisk-users] Sendmail using SMTP authorization   hugolivude
• [asterisk-users] Recommend Wireless IP Phone   Drew Gibson
• [asterisk-users] Codec problems when using G.723   Thomas Winter
• [asterisk-users] Recommend Wireless IP Phone   Eric "ManxPower" Wieling
• [asterisk-users] Using AMI to determine PRI Channels Used   Richard Lyman
• [asterisk-users] Using AMI to determine PRI Channels Used   Anthony Francis
• [asterisk-users] Recommend Wireless IP Phone   Michael Graves
• [asterisk-users] Asterisk daemon dies about once per day   Douglas Mortensen
• [asterisk-users] directrtpsetup without reinvite   Kristian Kielhofner
• [asterisk-users] directrtpsetup without reinvite   Alex Balashov
• [asterisk-users] directrtpsetup without reinvite   Kevin P. Fleming
• [asterisk-users] console/dsp asterisk seg fault   Jerry Geis
• [asterisk-users] console/dsp asterisk seg fault   Steve Totaro
• [asterisk-users] Recommend Wireless IP Phone   Wilton Helm
• [asterisk-users] Asterisk daemon dies about once per day   Steve Murphy
• [asterisk-users] Asterisk daemon dies about once per day   Tim Nelson
• [asterisk-users] Help with asterisk and avaya SIP trunking   Krishna Sumanth Chava
• [asterisk-users] console/dsp asterisk seg fault   Jerry Geis
• [asterisk-users] console/dsp asterisk seg fault   Jerry Geis
• [asterisk-users] changing the size of voice packets   John Todd
• [asterisk-users] Fwd: console/dsp asterisk seg fault   Steve Totaro
• [asterisk-users] changing the size of voice packets   Alex Balashov
• [asterisk-users] changing the size of voice packets   Kristian Kielhofner
• [asterisk-users] changing the size of voice packets   Kristian Kielhofner
• [asterisk-users] Asterisk daemon dies about once per day   Steve Murphy
• [asterisk-users] Voicemail IMAP ./configure error   c james
• [asterisk-users] directrtpsetup without reinvite   regs at kinetix.gr
• [asterisk-users] changing the size of voice packets   Alex Balashov
• [asterisk-users] Voicemail IMAP ./configure error   Mark Michelson
• [asterisk-users] DNS A queries for channel   Matt Riddell
• [asterisk-users] changing the size of voice packets   Kevin P. Fleming
• [asterisk-users] changing the size of voice packets   Alex Balashov
• [asterisk-users] changing the size of voice packets   Kristian Kielhofner
• [asterisk-users] Voicemail IMAP ./configure error   c james
• [asterisk-users] GEN-GEN and Manual Ring-Down (MRD)?   Tilghman Lesher
• [asterisk-users] changing the size of voice packets   Kevin P. Fleming
• [asterisk-users] changing the size of voice packets   Kristian Kielhofner
• [asterisk-users] Voicemail IMAP ./configure error   Mark Michelson
• [asterisk-users] Help with asterisk and avaya SIP trunking   Shaun Ewing
• [asterisk-users] What makes TDM400 FXS Connection to TELCO go into Off Hook State?   Jim Duda
• [asterisk-users] Server for 25-30 phones, sip trunks over the net   nb
• [asterisk-users] set(CALLERID(name) not working   C F
• [asterisk-users] dial a number while play the sound   邱磊
• [asterisk-users] Server for 25-30 phones, sip trunks over the net   Gordon Henderson
• [asterisk-users] music on hold   邱磊
• [asterisk-users] music on hold   Lee, John (Sydney)
• [asterisk-users] GEN-GEN and Manual Ring-Down (MRD)?   Raj Jain
• [asterisk-users] OT: Disable Polycom 650 Forward Softkey   Andrew Joakimsen
• [asterisk-users] music on hold   Tzafrir Cohen
• [asterisk-users] TE410P alarms stay RED with 1.4.22   Louis-David Mitterrand
• [asterisk-users] DNS A queries for channel   samuel
• [asterisk-users] What makes TDM400 FXS Connection to TELCO go into Off Hook State?   Tzafrir Cohen
• [asterisk-users] music on hold   邱磊
• [asterisk-users] music on hold   邱磊
• [asterisk-users] DNS A queries for channel   Matt Riddell
• [asterisk-users] Forcing repacketization on SIP to SIP call   Richard Brady
• [asterisk-users] TE410P alarms stay RED with 1.4.22   Laurent Caron
• [asterisk-users] Dial outside number using the E1 Link   fateme fatah
• [asterisk-users] TE410P alarms stay RED with 1.4.22   Laurent Caron
• [asterisk-users] Inbound/Outbound undesired behavior   Lenz Emilitri
• [asterisk-users] DNS A queries for channel   Philipp Kempgen
• [asterisk-users] music on hold   Jeff LaCoursiere
• [asterisk-users] Voicemail IMAP ./configure error   c james
• [asterisk-users] changing the size of voice packets   Pezhman Lali
• [asterisk-users] help with call with no sound via PSTN   César García
• [asterisk-users] TE410P alarms stay RED with 1.4.22   Louis-David Mitterrand
• [asterisk-users] view the current calls and their codec   nik600
• [asterisk-users] Dial outside number using the E1 Link   Pezhman Lali
• [asterisk-users] view the current calls and their codec   Tim Nelson
• [asterisk-users] OT: Polycom Firmware available (by accident?)   Dave Fullerton
• [asterisk-users] DNS A queries for channel   samuel
• [asterisk-users] music on hold   Uros Djokic
• [asterisk-users] Asterisk CDR Error ??   Ruddy Gbaguidi
• [asterisk-users] OT: Polycom Firmware available (by accident?)   Vinícius Fontes
• [asterisk-users] OT: Polycom Firmware available (by accident?)   Doug Lytle
• [asterisk-users] OT: Polycom Firmware available (by accident?)   Jared Smith
• [asterisk-users] OT: Polycom Firmware available (by accident?)   Dave Fullerton
• [asterisk-users] view the current calls and their codec   nik600
• [asterisk-users] Server for 25-30 phones, sip trunks over the net   Ben Hauger
• [asterisk-users] view the current calls and their codec   Tim Nelson
• [asterisk-users] GEN-GEN and Manual Ring-Down (MRD)?   Jorge Mendoza
• [asterisk-users] DNS A queries for channel   Tilghman Lesher
• [asterisk-users] OT: Polycom Firmware available (by accident?)   Tilghman Lesher
• [asterisk-users] ztdummy: rtc: lost some interrupts at 1024Hz.   Giorgio Incantalupo
• [asterisk-users] OT: Polycom Firmware available (by accident?)   Dave Fullerton
• [asterisk-users] view the current calls and their codec   nik600
• [asterisk-users] ztdummy: rtc: lost some interrupts at 1024Hz.   Remi Quezada
• [asterisk-users] OT: Polycom Firmware available (by accident?)   Doug
• [asterisk-users] ztdummy: rtc: lost some interrupts at 1024Hz   JR Richardson
• [asterisk-users] view the current calls and their codec   Anthony Francis
• [asterisk-users] Use the NEW ulaw/alaw codecs (slower, but cleaner)   Karl Fife
• [asterisk-users] Use the NEW ulaw/alaw codecs (slower, but cleaner)   Steve Totaro
• [asterisk-users] play file from url   Mike Clark
• [asterisk-users] play file from url   Singer X.J. Wang
• [asterisk-users] Use the NEW ulaw/alaw codecs (slower, but cleaner)   Steve Murphy
• [asterisk-users] Request for testing of new driver for B410P Quad-Port BRI   Shaun Ruffell
• [asterisk-users] Use the NEW ulaw/alaw codecs (slower, but cleaner)   Wilton Helm
• [asterisk-users] AsteriskNOW 1.5 - app_voicemail_imapstorage.so won't talk to IMAP server   Jason Lixfeld
• [asterisk-users] AsteriskNOW 1.5 - app_voicemail_imapstorage.so won't talk to IMAP server   Jason Parker
• [asterisk-users] Use the NEW ulaw/alaw codecs (slower, but cleaner)   Steve Murphy
• [asterisk-users] What makes TDM400 FXS Connection to TELCO go into Off Hook State?   Jim Duda
• [asterisk-users] Use the NEW ulaw/alaw codecs (slower, but cleaner)   Steve Underwood
• [asterisk-users] Grandstream and pickup   Julian Lyndon-Smith
• [asterisk-users] OT: Polycom Firmware available (by accident?)   Darryl Dunkin
• [asterisk-users] Grandstream and pickup   Julian Lyndon-Smith
• [asterisk-users] AS5200 <-> T100P - No alarms but no calls either...   Don Fanning
• [asterisk-users] DNS A queries for channel   Matt Riddell
• [asterisk-users] Use DECT GAP handsets with Snom M3 base?   Paul Chambers
• [asterisk-users] Use DECT GAP handsets with Snom M3 base?   Michael Graves
• [asterisk-users] set(CALLERID(name) not working   sean darcy
• [asterisk-users] set(CALLERID(name) not working   sean darcy
• [asterisk-users] set(CALLERID(name) not working   Daniel Lynes
• [asterisk-users] Use the NEW ulaw/alaw codecs (slower, but cleaner)   Wilton Helm
• [asterisk-users] What makes TDM400 FXS Connection to TELCO go into Off Hook State?   Tzafrir Cohen
• [asterisk-users] AS5200 <-> T100P - No alarms but no calls either...   Tzafrir Cohen
• [asterisk-users] play file from url   Pezhman Lali
• [asterisk-users] Grandstream and pickup   Julian Lyndon-Smith
• [asterisk-users] Grandstream and pickup   Gordon Henderson
• [asterisk-users] Grandstream and pickup   Julian Lyndon-Smith
• [asterisk-users] The sound is played but I did not hear   jhon digital21
• [asterisk-users] test OpenVox B400P and junghans card for dahdi BRI wcb4xxp   lizhong zhu
• [asterisk-users] ztdummy: rtc: lost some interrupts at 1024Hz   Giorgio Incantalupo
• [asterisk-users] AS5200 <-> T100P - No alarms but no calls either...   Steve Howes
• [asterisk-users] test OpenVox B400P and junghans card for dahdi BRI wcb4xxp   Tzafrir Cohen
• [asterisk-users] Query about Call Recording with Asterisk / Freeswitch in Cisco IPCC deployment   Kashif Naeem
• [asterisk-users] QueueLog from AMI   Sebastian Gutierrez
• [asterisk-users] What are the minimum realtime fields for sipusers?   Eric Chamberlain
• [asterisk-users] QueueLog from AMI   Atis Lezdins
• [asterisk-users] E1 PRI to and from SIP screeching   Peter Lindquist
• [asterisk-users] QueueLog from AMI   Sebastian Gutierrez
• [asterisk-users] PSTN Channels merging with SIP channels!!!   R_O_L_A_N_D at hotmail.com
• [asterisk-users] How to get correct dial result for outgoing calls thru ISDN?   Daniel - Asterisk
• [asterisk-users] E1 PRI to and from SIP screeching   Steve Totaro
• [asterisk-users] PSTN Channels merging with SIP channels!!!   Dave Fullerton
• [asterisk-users] QueueLog from AMI   Atis Lezdins
• [asterisk-users] QueueLog from AMI   Sebastian Gutierrez
• [asterisk-users] AsteriskNOW 1.5 - app_voicemail_imapstorage.so won't talk to IMAP server   Jason Parker
• [asterisk-users] PSTN Channels merging with SIP channels!!!   R_O_L_A_N_D at hotmail.com
• [asterisk-users] Request for testing of new driver for B410P Quad-Port BRI   stoffell
• [asterisk-users] SIP provider and NAT   Jim Dickenson
• [asterisk-users] 1.4.22 vs 1.4.21.2 - IAX2 regression ?   Igor Zamocky
• [asterisk-users] SIP provider and NAT   Alex Balashov
• [asterisk-users] SIP provider and NAT   Steve Totaro
• [asterisk-users] set(CALLERID(name) not working   sean darcy
• [asterisk-users] Voicemail IMAP ./configure error   Mark Michelson
• [asterisk-users] SIP provider and NAT   Sebastian
• [asterisk-users] Why Nat=yes Nat=no Option?   Steve Totaro
• [asterisk-users] Why Nat=yes Nat=no Option?   Alex Balashov
• [asterisk-users] set(CALLERID(name) not working   Doug Lytle
• [asterisk-users] Why Nat=yes Nat=no Option?   Steve Totaro
• [asterisk-users] Why Nat=yes Nat=no Option?   Alex Balashov
• [asterisk-users] set(CALLERID(name) not working   Matt Riddell
• [asterisk-users] Why Nat=yes Nat=no Option?   Steve Totaro
• [asterisk-users] List eating mail again?   Doug Lytle
• [asterisk-users] QueueLog from AMI   Atis Lezdins
• [asterisk-users] set(CALLERID(name) not working   sean darcy
• [asterisk-users] Use DECT GAP handsets with Snom M3 base?   Paul Chambers
• [asterisk-users] List eating mail again?   Doug Lytle
• [asterisk-users] List eating mail again?   Atis Lezdins
• [asterisk-users] CANCEL FORWAR   Alex Balashov
• [asterisk-users] CANCEL FORWAR   Alex Balashov
• [asterisk-users] Why Nat=yes Nat=no Option?   Alex Balashov
• [asterisk-users] 1.4.22 CALLERID(num)   James Fromm
• [asterisk-users] What makes TDM400 FXS Connection to TELCO go into Off Hook State?   Jim Duda
• [asterisk-users] Why Nat=yes Nat=no Option?   Steve Totaro
• [asterisk-users] Why Nat=yes Nat=no Option?   Tilghman Lesher
• [asterisk-users] What are the minimum realtime fields for sipusers?   Eric Chamberlain
• [asterisk-users] CANCEL FORWAR   Bruno Rodrigues
• [asterisk-users] Why Nat=yes Nat=no Option?   Steve Totaro
• [asterisk-users] set(CALLERID(name) not working   sean darcy
• [asterisk-users] What makes TDM400 FXS Connection to TELCO go into Off Hook State?   Tzafrir Cohen
• [asterisk-users] test OpenVox B400P and junghans card for dahdi BRI wcb4xxp   lizhong zhu
• [asterisk-users] E1 PRI to and from SIP screeching   Peter Lindquist
• [asterisk-users] E1 PRI to and from SIP screeching   Eric "ManxPower" Wieling
• [asterisk-users] Why Nat=yes Nat=no Option?   Eric "ManxPower" Wieling
• [asterisk-users] How to get correct dial result for outgoing calls thru ISDN?   Eric "ManxPower" Wieling
• [asterisk-users] How to get correct dial result for outgoing calls thru ISDN?   Tilghman Lesher
• [asterisk-users] AsteriskNOW 1.5 - app_voicemail_imapstorage.so won't talk to IMAP server   Jason Lixfeld
• [asterisk-users] E1 PRI to and from SIP screeching   Peter Lindquist
• [asterisk-users] How to get correct dial result for outgoing calls thru ISDN?   Eric "ManxPower" Wieling
• [asterisk-users] One little pill and you are on your way to sex!   asterisk-users at lists.digium.com
• [asterisk-users] asterisk setup w/ voIP phones   Mike
• [asterisk-users] How to get correct dial result for outgoing calls thru ISDN?   Tilghman Lesher
• [asterisk-users] asterisk setup w/ voIP phones   Gordon Henderson
• [asterisk-users] asterisk setup w/ voIP phones   Matt Gibson
• [asterisk-users] Parking help - causing Asterisk crash   Daniel Johnson
• [asterisk-users] cisco voice gw / cisco call manager /asterisk for voice record, ivr   Litzler Mihály
• [asterisk-users] Parking help - causing Asterisk crash   Doug Lytle
• [asterisk-users] Problems with Licensed g729a codec from Digium   Murray Blakeman
• [asterisk-users] 1.6 Production ready??   Sherwood McGowan
• [asterisk-users] Tos_sip   Anael DIAZ
• [asterisk-users] [Fwd: [OpenSIPS-Devel] RFC: new opensips design]   Bogdan-Andrei Iancu
• [asterisk-users] database queries from extensions.conf   Klaus Darilion
• [asterisk-users] How long will Asterisk 1.4.x supported/maintained   Klaus Darilion
• [asterisk-users] Why Nat=yes Nat=no Option?   Klaus Darilion
• [asterisk-users] asterisk setup w/ voIP phones   Justin Case
• [asterisk-users] Why Nat=yes Nat=no Option?   Alex Balashov
• [asterisk-users] Why Nat=yes Nat=no Option?   Alex Balashov
• [asterisk-users] database queries from extensions.conf   Jared Smith
• [asterisk-users] database queries from extensions.conf   Wolfgang Pichler
• [asterisk-users] 1.6 Production ready??   César García
• [asterisk-users] Why Nat=yes Nat=no Option?   Steve Totaro
• [asterisk-users] Why Nat=yes Nat=no Option?   Alex Balashov
• [asterisk-users] Why Nat=yes Nat=no Option?   Steve Totaro
• [asterisk-users] Why Nat=yes Nat=no Option?   Alex Balashov
• [asterisk-users] Why Nat=yes Nat=no Option?   Klaus Darilion
• [asterisk-users] Why Nat=yes Nat=no Option?   Klaus Darilion
• [asterisk-users] Why Nat=yes Nat=no Option?   Alex Balashov
• [asterisk-users] Why Nat=yes Nat=no Option?   Alex Balashov
• [asterisk-users] How long will Asterisk 1.4.x supported/maintained   Tilghman Lesher
• [asterisk-users] database queries from extensions.conf   Klaus Darilion
• [asterisk-users] database queries from extensions.conf   Klaus Darilion
• [asterisk-users] Why Nat=yes Nat=no Option?   Klaus Darilion
• [asterisk-users] Why Nat=yes Nat=no Option?   Klaus Darilion
• [asterisk-users] database queries from extensions.conf   Wolfgang Pichler
• [asterisk-users] Voicemail IMAP ./configure error   c james
• [asterisk-users] Why Nat=yes Nat=no Option?   Alex Balashov
• [asterisk-users] Asterisk and Zaptel version numbers -- how close is close enough?   Steve Edwards
• [asterisk-users] Asterisk and Zaptel version numbers -- how close is close enough?   Steve Edwards
• [asterisk-users] Voicemail IMAP ./configure error   Mark Michelson
• [asterisk-users] Preserving DID numbers on PRI pass through   Mikel Lindsaar
• [asterisk-users] ParkandAnnounce?   Positively Optimistic
• [asterisk-users] Virtual Question   Babcock, Michael Alex
• [asterisk-users] openLDAP   jude NOA
• [asterisk-users] Dedicated Servers   Sahil Gupta
• [asterisk-users] RTP LOG   Max Alex
• [asterisk-users] Dedicated Servers   Tom Moore
• [asterisk-users] ParkandAnnounce?   Doug Lytle
• [asterisk-users] Preserving DID numbers on PRI pass through   Tony Mountifield
• [asterisk-users] asterisk/E1   Khaled Chehab
• [asterisk-users] kick from conference message on 1.2.23   Jerry Geis
• [asterisk-users] asterisk/E1   Steve Totaro
• [asterisk-users] asterisk/E1   Godson Gera
• [asterisk-users] asterisk/E1   Khaled Chehab
• [asterisk-users] Looking for a good lightweight Linux softPhone   Gordon Henderson
• [asterisk-users] Looking for a good lightweight Linux softPhone   Alex Balashov
• [asterisk-users] asterisk/E1   Steve Totaro
• [asterisk-users] Looking for a good lightweight Linux softPhone   Cory Andrews
• [asterisk-users] asterisk/E1   Khaled Chehab
• [asterisk-users] kick from conference message on 1.2.23   Tony Mountifield
• [asterisk-users] asterisk/E1   Tony Mountifield
• [asterisk-users] RTP LOG   Positively Optimistic
• [asterisk-users] Looking for a good lightweight Linux softPhone   Alan Lord
• [asterisk-users] asterisk/E1   Steve Totaro
• [asterisk-users] Looking for a good lightweight Linux softPhone   Tzafrir Cohen
• [asterisk-users] Queue App - Set monitoring dynamically   equis software
• [asterisk-users] Looking for a good lightweight Linux softPhone   Gordon Henderson
• [asterisk-users] Looking for a good lightweight Linux softPhone   Tzafrir Cohen
• [asterisk-users] no dial to busy sip line   Christophorus Laube
• [asterisk-users] no dial to busy sip line   David Gibbons
• [asterisk-users] Looking for a good lightweight Linux softPhone   Tilghman Lesher
• [asterisk-users] Looking for a good lightweight Linux softPhone   Gordon Henderson
• [asterisk-users] Looking for a good lightweight Linux softPhone   Jeff LaCoursiere
• [asterisk-users] Looking for a good lightweight Linux softPhone   Gordon Henderson
• [asterisk-users] Looking for a good lightweight Linux softPhone   zoachien at securax.org
• [asterisk-users] PRI users, please read   Tilghman Lesher
• [asterisk-users] Looking for a good lightweight Linux softPhone   Atis Lezdins
• [asterisk-users] Manilla inbound DID   Fred Posner
• [asterisk-users] Linksys SPA 400, 901 and 921 with asterisk   Valentin Bud
• [asterisk-users] Looking for a good lightweight Linux softPhone   Tzafrir Cohen
• [asterisk-users] asterisk/E1   Edwin Lam
• [asterisk-users] asterisk/E1   Valentin Bud
• [asterisk-users] Originate on AMI   Marco Eduardo Cordeiro
• [asterisk-users] RTP LOG   Benny Amorsen
• [asterisk-users] installation   DesRae Mason
• [asterisk-users] installation   Steve Totaro
• [asterisk-users] Best way to handle include files?   Doug
• [asterisk-users] AS5200 <-> T100P - No alarms but no calls either...   Don Fanning
• [asterisk-users] Preserving DID numbers on PRI pass through   Mikel Lindsaar
• [asterisk-users] PBX -> PRI -> * -> Telco not working   Mikel Lindsaar
• [asterisk-users] RTP LOG   Max Alex
• [asterisk-users] music on hold   fateme fatah
• [asterisk-users] RTP LOG   Atis Lezdins
• [asterisk-users] Looking for a good lightweight Linux softPhone   Atis Lezdins
• [asterisk-users] Polycom low volume   hin lee
• [asterisk-users] PBX -> PRI -> * -> Telco not working   Tony Mountifield
• [asterisk-users] PBX -> PRI -> * -> Telco not working   Mikel Lindsaar
• [asterisk-users] PBX -> PRI -> * -> Telco not working   Mikel Lindsaar
• [asterisk-users] PBX -> PRI -> * -> Telco not working   Mikel Lindsaar
• [asterisk-users] Polycom low volume   Michael Graves
• [asterisk-users] MixMonitor and Queues   Sebastian
• [asterisk-users] PBX -> PRI -> * -> Telco not working   Mikel Lindsaar
• [asterisk-users] Polycom low volume   Darrick Hartman
• [asterisk-users] Asterisk GUI and SIP registration   Joseph L. Casale
• [asterisk-users] PBX -> PRI -> * -> Telco not working   Tony Mountifield
• [asterisk-users] IAX2 client for "eee pc 1000"   Joseph
• [asterisk-users] RV: MixMonitor and Queues   Sebastian
• [asterisk-users] IAX2 client for "eee pc 1000"   David
• [asterisk-users] Asterisk GUI and SIP registration   Brandon Kruse
• [asterisk-users] IAX2 client for "eee pc 1000"   Joseph
• [asterisk-users] IAX2 client for "eee pc 1000"   David
• [asterisk-users] IAX2 client for "eee pc 1000"   Alex Robar
• [asterisk-users] IAX2 client for "eee pc 1000"   Joseph
• [asterisk-users] PBX -> PRI -> * -> Telco not working   Mikel Lindsaar
• [asterisk-users] IAX2 client for "eee pc 1000"   Alex Balashov
• [asterisk-users] IAX2 client for "eee pc 1000"   Alex Robar
• [asterisk-users] Polycom low volume   hin lee
• [asterisk-users] Polycom low volume   Doug
• [asterisk-users] IAX2 client for "eee pc 1000"   Rob Hillis
• [asterisk-users] IAX2 client for "eee pc 1000"   Joseph
• [asterisk-users] Polycom low volume   hin lee
• [asterisk-users] IAX2 client for "eee pc 1000"   Tzafrir Cohen
• [asterisk-users] * + Legacy PBX works but strange problem   Sriram
• [asterisk-users] * + Legacy PBX works but strange problem   Robert Boardman
• [asterisk-users] Info about dstchannel   Chris Maciejewski
• [asterisk-users] * + Legacy PBX works but strange problem   Steve Totaro
• [asterisk-users] Polycom low volume   Michael Graves
• [asterisk-users] * + Legacy PBX works but strange problem   Sriram
• [asterisk-users] IAX2 client for "eee pc 1000"   henry at henrythebig.com
• [asterisk-users] * + Legacy PBX works but strange problem   ram
• [asterisk-users] What kind of IAX2 client will install/run on EEE PC 1000 (stock Linux software)?   Bill Michaelson
• [asterisk-users] * + Legacy PBX works but strange problem   Sriram
• [asterisk-users] iPhone SIP or IAX client (without proxy)?   Lincoln King-Cliby
• [asterisk-users] IAX2 client for "eee pc 1000"   Joseph
• [asterisk-users] IAX2 client for "eee pc 1000"   Tzafrir Cohen
• [asterisk-users] Caching Asterisk SIP useragent info?   veselin at campbell-lange.net
• [asterisk-users] DialPlan   michel freiha
• [asterisk-users] dahdi compile error on svn   Jerry Geis
• [asterisk-users] PBX -> PRI -> * -> Telco not working   Tony Mountifield
• [asterisk-users] iPhone SIP or IAX client (without proxy)?   Eric Chamberlain
• [asterisk-users] Queue App - Set monitoring dynamically   Matt Riddell
• [asterisk-users] Record Application   Kannan Murali
• [asterisk-users] dahdi compile error on svn   Tzafrir Cohen
• [asterisk-users] Caching Asterisk SIP useragent info?   Paul Hales
• [asterisk-users] dahdi compile error on svn   Jerry Geis
• [asterisk-users] dahdi compile error on svn   Jerry Geis
• [asterisk-users] upgrade to 1.6   Jerry Geis
• [asterisk-users] IAX2 client for 'eee pc 1000'   Roger Marquis
• [asterisk-users] Debugging Asterisk   Mikel Lindsaar
• [asterisk-users] Debugging Asterisk   Valentin Bud
• [asterisk-users] Caching Asterisk SIP useragent info?   Veselin K
• [asterisk-users] Caching Asterisk SIP useragent info?   andrea
• [asterisk-users] upgrade to 1.6   Hakan C
• [asterisk-users] Queue App - Set monitoring dynamically   equis software
• [asterisk-users] E1/channels   Khaled Chehab
• [asterisk-users] upgrade to 1.6   Jerry Geis
• [asterisk-users] ALL of DIDx Down?   Andrew Joakimsen
• [asterisk-users] asterisk conference   Giedrius Augys
• [asterisk-users] Debugging Asterisk   Terry Wilson
• [asterisk-users] Full Duplex   Ken Williams
• [asterisk-users] Debugging Asterisk   Atis Lezdins
• [asterisk-users] no dial to busy sip line   Christophorus Laube
• [asterisk-users] Hints and realtime   Julian Lyndon-Smith
• [asterisk-users] Digium Card Noice issue   Bipin
• [asterisk-users] Debugging Asterisk   Tilghman Lesher
• [asterisk-users] dahdi and asterisk 1.4.22   Jerry Geis
• [asterisk-users] Caching Asterisk SIP useragent info?   Tilghman Lesher
• [asterisk-users] Digium Card Noice issue   Steve Howes
• [asterisk-users] Hints and realtime   Tilghman Lesher
• [asterisk-users] AMI Events disabling.   Simith Nambiar
• [asterisk-users] Full Duplex   Gordon Henderson
• [asterisk-users] Full Duplex   Doug Lytle
• [asterisk-users] Full Duplex   Ken Williams
• [asterisk-users] Deny FOP originated calls   Rodolfo Alcazar Portillo
• [asterisk-users] Full Duplex   Ken Williams
• [asterisk-users] Digium Card Noice issue   Andrew Joakimsen
• [asterisk-users] Full Duplex   Doug Lytle
• [asterisk-users] Use the NEW ulaw/alaw codecs (slower, but cleaner)   John Todd
• [asterisk-users] Full Duplex   Singer X.J. Wang
• [asterisk-users] How long will Asterisk 1.4.x supported/maintained   Philipp Kempgen
• [asterisk-users] Use the NEW ulaw/alaw codecs (slower, but cleaner)   Wilton Helm
• [asterisk-users] AMI Events disabling.   Simith Nambiar
• [asterisk-users] Digium Card Noice issue   Matthew Fredrickson
• [asterisk-users] Use the NEW ulaw/alaw codecs (slower, but cleaner)   Matthew Fredrickson
• [asterisk-users] MixMonitor Problem   Sebastian
• [asterisk-users] Full Duplex   Matthew Fredrickson
• [asterisk-users] Digium Card Noice issue   Benny Amorsen
• [asterisk-users] Polycom low volume   hin lee
• [asterisk-users] Use the NEW ulaw/alaw codecs (slower, but cleaner)   Benny Amorsen
• [asterisk-users] Use the NEW ulaw/alaw codecs (slower, but cleaner)   Matthew Fredrickson
• [asterisk-users] How long will Asterisk 1.4.x supported/maintained   Tilghman Lesher
• [asterisk-users] Use the NEW ulaw/alaw codecs (slower, but cleaner)   Wilton Helm
• [asterisk-users] How long will Asterisk 1.4.x supported/maintained   Anthony Francis
• [asterisk-users] How long will Asterisk 1.4.x supported/maintained   Tilghman Lesher
• [asterisk-users] Picked up calls die in exactly 20 seconds   Juan Carlos Castro y Castro
• [asterisk-users] Caching Asterisk SIP useragent info?   Paul Hales
• [asterisk-users] How long will Asterisk 1.4.x supported/maintained   Philipp Kempgen
• [asterisk-users] Use the NEW ulaw/alaw codecs (slower, but cleaner)   Steve Underwood
• [asterisk-users] upgrade to 1.6   Matt Riddell
• [asterisk-users] ALL of DIDx Down?   Matt Riddell
• [asterisk-users] Use the NEW ulaw/alaw codecs (slower, but cleaner)   Matthew Fredrickson
• [asterisk-users] two sip listening ports for single asterisk   Rizwan Hisham
• [asterisk-users] Use the NEW ulaw/alaw codecs (slower, but cleaner)   Matthew Fredrickson
• [asterisk-users] Message 12216   asterisk-users at lists.digium.com
• [asterisk-users] Deny FOP originated calls   Tilghman Lesher
• [asterisk-users] two sip listening ports for single asterisk   Robert McNaught
• [asterisk-users] How long will Asterisk 1.4.x supported/maintained   Atis Lezdins
• [asterisk-users] How long will Asterisk 1.4.x supported/maintained   Tilghman Lesher
• [asterisk-users] HPEC performance   Joseph L. Casale
• [asterisk-users] Polycom low volume   Doug
• [asterisk-users] MixMonitor Problem   Trevor Peirce
• [asterisk-users] How to Barge specific extensions   amit salunkhe
• [asterisk-users] test OpenVox B400P and junghans card for dahdi BRI wcb4xxp   lizhong zhu
• [asterisk-users] PoE switch recommendations?   Jesse Molina
• [asterisk-users] How long will Asterisk 1.4.x supported/maintained   Tzafrir Cohen
• [asterisk-users] test OpenVox B400P and junghans card for dahdi BRI wcb4xxp   Tzafrir Cohen
• [asterisk-users] HPEC performance   Tzafrir Cohen
• [asterisk-users] Asterisk Realtime and device contexts   Sawan Vithlani
• [asterisk-users] Caching Asterisk SIP useragent info?   Veselin K
• [asterisk-users] HPEC performance   Gordon Henderson
• [asterisk-users] busy-level / busy-limit Asterisk 1.4.22   nik600
• [asterisk-users] busy-level / busy-limit Asterisk 1.4.22   Steve Howes
• [asterisk-users] Use the NEW ulaw/alaw codecs (slower, but cleaner)   Benny Amorsen
• [asterisk-users] busy-level / busy-limit Asterisk 1.4.22   nik600
• [asterisk-users] FOP with Asterisk 1.6. No call Information.   David Klaverstyn
• [asterisk-users] Use the NEW ulaw/alaw codecs (slower, but cleaner)   Tony Mountifield
• [asterisk-users] Use the NEW ulaw/alaw codecs (slower, but cleaner)   Steve Underwood
• [asterisk-users] Asterisk with or without OpenSER   Yehavi Bourvine
• [asterisk-users] Asterisk not reading fast DTMFs, was: PBX -> PRI -> * -> Telco not working   Mikel Lindsaar
• [asterisk-users] Picked up calls die in exactly 20 seconds   Steve Totaro
• [asterisk-users] Asterisk not reading fast DTMFs, was: PBX -> PRI -> * -> Telco not working   Mikel Lindsaar
• [asterisk-users] meetme command from 1.4 to 1.6   Jerry Geis
• [asterisk-users] Asterisk with or without OpenSER   Steve Totaro
• [asterisk-users] FOP with Asterisk 1.6. No call Information.   Doug Lytle
• [asterisk-users] Asterisk 1.4.21.2 and gtalk2voip   Administrator TOOTAI
• [asterisk-users] Crash when rebooting or unload xorcom modules   Loic Didelot
• [asterisk-users] How to Barge specific extensions   troxlinux
• [asterisk-users] Asterisk not reading fast DTMFs, was: PBX -> PRI -> * -> Telco not working   Tony Mountifield
• [asterisk-users] Configuring Sangoma BRI with zaptel?   Claus Herwig
• [asterisk-users] Asterisk not reading fast DTMFs, was: PBX -> PRI -> * -> Telco not working   Mikel Lindsaar
• [asterisk-users] Incoming Transfer   Joseph L. Casale
• [asterisk-users] FOP with Asterisk 1.6. No call Information.   Carlos Chavez
• [asterisk-users] meetme command from 1.4 to 1.6   Jeff Peeler
• [asterisk-users] sound quality between two back-to-back asterisk   mark morreny
• [asterisk-users] Asterisk 1.4.21.2 and gtalk2voip   Administrator TOOTAI
• [asterisk-users] diax debian package   Joseph
• [asterisk-users] diax debian package   Tzafrir Cohen
• [asterisk-users] Do Digium Digital Cards Handle Remote Loopback Command?   George Pajari
• [asterisk-users] Do Digium Digital Cards Handle Remote Loopback Command?   Alex Balashov
• [asterisk-users] setting up callback   Михаил (Плюс Плюс)
• [asterisk-users] Asterisk not reading fast DTMFs, was: PBX -> PRI -> * -> Telco not working   Mikel Lindsaar
• [asterisk-users] Asterisk with or without OpenSER   John Todd
• [asterisk-users] sound quality between two back-to-back asterisk   Godson Gera
• [asterisk-users] Asterisk with or without OpenSER   Yehavi Bourvine
• [asterisk-users] Use the NEW ulaw/alaw codecs (slower, but cleaner)   Benny Amorsen
• [asterisk-users] Incoming Transfer   Godson Gera
• [asterisk-users] Realtime MOH   Sebastian
• [asterisk-users] Do Digium Digital Cards Handle Remote Loopback Command?   Kevin P. Fleming
• [asterisk-users] Fwd: Polycom phone time behind one hour.   Doug Smith
• [asterisk-users] How to Barge specific extensions   Hakan C
• [asterisk-users] Fwd: Polycom phone time behind one hour.   Brent Davidson
• [asterisk-users] Use the NEW ulaw/alaw codecs (slower, but cleaner)   Steve Underwood
• [asterisk-users] Caching Asterisk SIP useragent info?   Paul Hales
• [asterisk-users] Asterisk not reading fast DTMFs, was: PBX -> PRI -> * -> Telco not working   Brent Davidson
• [asterisk-users] Asterisk 1.6 call files Disposition=NO ANSWER   David Klaverstyn
• [asterisk-users] changing the size of voice packets   Dan Austin
• [asterisk-users] help with dahdi   Jerry Geis
• [asterisk-users] Fwd: Polycom phone time behind one hour.   Bryan M. Johns
• [asterisk-users] help with dahdi   Joseph L. Casale
• [asterisk-users] help with dahdi   Jerry Geis
• [asterisk-users] help with dahdi   Jerry Geis
• [asterisk-users] question about connecting with Mobile Base Station   mark morreny
• [asterisk-users] question about connecting with Mobile Base Station   Andrew Joakimsen
• [asterisk-users] help with dahdi   Joseph L. Casale
• [asterisk-users] two sip listening ports for single asterisk   Matthew J. Roth
• [asterisk-users] Aeterisk NOW 1.5beta1 - CDR problem....   Bipin
• [asterisk-users] question about connecting with Mobile Base Station   mark morreny
• [asterisk-users] presence with polycom DND   cfh
• [asterisk-users] help with dahdi   Tzafrir Cohen
• [asterisk-users] Asterisk not reading fast DTMFs, was: PBX -> PRI -> * -> Telco not working   Mikel Lindsaar
• [asterisk-users] help with dahdi   Hakan C
• [asterisk-users] Asterisk not reading fast DTMFs, was: PBX -> PRI -> * -> Telco not working   Hakan C
• [asterisk-users] Asterisk not reading fast DTMFs, was: PBX -> PRI -> * -> Telco not working   Mikel Lindsaar
• [asterisk-users] Role of asterisk   Valentin Bud
• [asterisk-users] Forcing repacketization on SIP to SIP call   Richard Brady
• [asterisk-users] P2P   ml at bortal.de
• [asterisk-users] PoE switch recommendations?   Mike Jagdis
• [asterisk-users] P2P   Hakan C
• [asterisk-users] question about connecting with Mobile Base Station   Hakan C
• [asterisk-users] P2P   Valentin Bud
• [asterisk-users] Monitoring   Jon Weisman
• [asterisk-users] Monitoring   Hakan C
• [asterisk-users] Aeterisk NOW 1.5beta1 - CDR problem....   Hakan C
• [asterisk-users] Configuring Sangoma BRI with zaptel?   Hakan C
• [asterisk-users] Picked up calls die in exactly 20 seconds   Hakan C
• [asterisk-users] asterisk conference   Hakan C
• [asterisk-users] Monitoring   Jon Weisman
• [asterisk-users] P2P   ml at bortal.de
• [asterisk-users] help with dahdi   Tzafrir Cohen
• [asterisk-users] IF else   michel freiha
• [asterisk-users] P2P   Valentin Bud
• [asterisk-users] help with dahdi   Jerry Geis
• [asterisk-users] Monitoring   Hakan C
• [asterisk-users] Monitoring   Jon Weisman
• [asterisk-users] Monitoring   Giorgio Ciccarelli
• [asterisk-users] Asterisk NOW - Where to start   Ferguson, Michael
• [asterisk-users] Monitoring   federico fetto
• [asterisk-users] IF else   Gordon Henderson
• [asterisk-users] Role of asterisk   Jared Smith
• [asterisk-users] Asterisk NOW - Where to start - FOUND, Thanks   Ferguson, Michael
• [asterisk-users] Role of asterisk   Jared Smith
• [asterisk-users] help with dahdi   Hakan C
• [asterisk-users] help with dahdi   Tzafrir Cohen
• [asterisk-users] Monitoring   Jon Weisman
• [asterisk-users] Monitoring   Jon Weisman
• [asterisk-users] IF else   Atis Lezdins
• [asterisk-users] Monitoring   Giorgio Ciccarelli
• [asterisk-users] help with dahdi   Jerry Geis
• [asterisk-users] Asterisk with or without OpenSER   mail-lists
• [asterisk-users] Monitoring   federico fetto
• [asterisk-users] Asterisk with or without OpenSER   Steve Totaro
• [asterisk-users] Asterisk with or without OpenSER   Alex Balashov
• [asterisk-users] IF else   Steve Edwards
• [asterisk-users] P2P   Jim Dickenson
• [asterisk-users] IF else   Atis Lezdins
• [asterisk-users] Asterisk not reading fast DTMFs, was: PBX -> PRI -> * -> Telco not working   Brent Davidson
• [asterisk-users] Howto grab back call transfered from SIP phone   Russell Brown
• [asterisk-users] Asterisk with or without OpenSER   Gordon Henderson
• [asterisk-users] Asterisk with or without OpenSER   Alex Balashov
• [asterisk-users] Asterisk with or without OpenSER   Alex Balashov
• [asterisk-users] Asterisk with or without OpenSER   Steve Totaro
• [asterisk-users] VoiceMail - audio problem   Shaun Wingrin
• [asterisk-users] question about connecting with Mobile Base Station   John Todd
• [asterisk-users] Asterisk with or without OpenSER   Gordon Henderson
• [asterisk-users] Howto grab back call transfered from SIP phone   Chris Bagnall
• [asterisk-users] TDM400 (?) zap hangup   Roderick A. Anderson
• [asterisk-users] Best way to handle include files?   Doug
• [asterisk-users] Upgrading Asterisk and FreePBX from 1.2 to 1.4   Carlos Chavez
• [asterisk-users] TDM400 (?) zap hangup   Tzafrir Cohen
• [asterisk-users] Best way to handle include files?   Tzafrir Cohen
• [asterisk-users] Role of asterisk   Valentin Bud
• [asterisk-users] puzzle   Jeff LaCoursiere
• [asterisk-users] HPEC performance   Joseph L. Casale
• [asterisk-users] TDM400 (?) zap hangup   Roderick A. Anderson
• [asterisk-users] TDM400 (?) zap hangup   Jared Smith
• [asterisk-users] puzzle   Brent Davidson
• [asterisk-users] puzzle   Danny Nicholas
• [asterisk-users] puzzle   Jeff LaCoursiere
• [asterisk-users] Asterisk with or without OpenSER   Alex Balashov
• [asterisk-users] puzzle   Danny Nicholas
• [asterisk-users] Asterisk with or without OpenSER   mail-lists
• [asterisk-users] Howto grab back call transfered from SIP phone   Steve Totaro
• [asterisk-users] TDM400 (?) zap hangup   Roderick A. Anderson
• [asterisk-users] Howto grab back call transfered from SIP phone   Steve Totaro
• [asterisk-users] HPEC performance   Tzafrir Cohen
• [asterisk-users] Asterisk 1.6 call files Disposition=NO ANSWER   Steve Murphy
• [asterisk-users] VoiceMail - audio problem   David A. Bandel
• [asterisk-users] puzzle   Tzafrir Cohen
• [asterisk-users] puzzle   Jeff LaCoursiere
• [asterisk-users] Best way to handle include files?   Doug
• [asterisk-users] VoiceMail - audio problem   Shaun Wingrin
• [asterisk-users] puzzle   Jeff LaCoursiere
• [asterisk-users] puzzle   Tzafrir Cohen
• [asterisk-users] puzzle   Danny Nicholas
• [asterisk-users] Asterisk 1.6 call files Disposition=NO ANSWER   Steve Murphy
• [asterisk-users] HPEC performance   Joseph L. Casale
• [asterisk-users] puzzle   Jeff LaCoursiere
• [asterisk-users] HPEC performance   Joseph L. Casale
• [asterisk-users] dahdi_test drops after restarting Sangoma driver   Andres
• [asterisk-users] puzzle   Steve Totaro
• [asterisk-users] puzzle   Jeff LaCoursiere
• [asterisk-users] puzzle   Steve Totaro
• [asterisk-users] Meetme "talker optimization" always on even when no "o" option present.   Dan Austin
• [asterisk-users] puzzle   Jeff LaCoursiere
• [asterisk-users] puzzle   Alex Balashov
• [asterisk-users] puzzle   Steve Totaro
• [asterisk-users] Upgrading Asterisk and FreePBX from 1.2 to 1.4   Rob Hillis
• [asterisk-users] puzzle   Jeff LaCoursiere
• [asterisk-users] puzzle   Steve Totaro
• [asterisk-users] puzzle   Alex Balashov
• [asterisk-users] puzzle   Jeff LaCoursiere
• [asterisk-users] Any other "free" toll free SIP providers out there?   Tom Browning
• [asterisk-users] puzzle   Steve Totaro
• [asterisk-users] two sip listening ports for single asterisk   Mike
• [asterisk-users] dahdi_test drops after restarting Sangoma driver   Hakan C
• [asterisk-users] How long will Asterisk 1.4.x supported/maintained   Olivier
• [asterisk-users] HPEC performance   Gordon Henderson
• [asterisk-users] echo cancellation for sip phones   Pezhman Lali
• [asterisk-users] echo cancellation for sip phones   Alan Lord
• [asterisk-users] How long will Asterisk 1.4.x supported/maintained   Tzafrir Cohen
• [asterisk-users] Any other "free" toll free SIP providers out there?   Tzafrir Cohen
• [asterisk-users] echo cancellation for sip phones   Pezhman Lali
• [asterisk-users] jitterbuffer   farah.auf at epfl.ch
• [asterisk-users] Configuring Sangoma BRI with zaptel?   Claus Herwig
• [asterisk-users] Voice Mail   michel freiha
• [asterisk-users] Voice Mail   Doug Lytle
• [asterisk-users] puzzle   Philipp Kempgen
• [asterisk-users] DTMF payload   michel freiha
• [asterisk-users] Collect digits from the Callee after the Call is connected.   Doug Lytle
• [asterisk-users] Any other "free" toll free SIP providers out there?   SIP
• [asterisk-users] Collect digits from the Callee after the Call is connected.   Darrin Henshaw
• [asterisk-users] Load balancing Asterisk.   Nitzan Kon
• [asterisk-users] Voicemail in Real Time   Ali Jawad
• [asterisk-users] Load balancing Asterisk.   Grygoriy Dobrovolskyy
• [asterisk-users] Load balancing Asterisk.   Nitzan Kon
• [asterisk-users] Load balancing Asterisk.   Alex Balashov
• [asterisk-users] Voicemail in Real Time   Philipp Kempgen
• [asterisk-users] Collect digits from the Callee after the Call is connected.   Doug Lytle
• [asterisk-users] Collect digits from the Callee after the Call is connected.   Simith Nambiar
• [asterisk-users] Load balancing Asterisk.   Jeff LaCoursiere
• [asterisk-users] puzzle   Steve Totaro
• [asterisk-users] Load balancing Asterisk.   Nitzan Kon
• [asterisk-users] Load balancing Asterisk.   Alex Balashov
• [asterisk-users] Load balancing Asterisk.   SIP
• [asterisk-users] Load balancing Asterisk.   Alex Balashov
• [asterisk-users] Load balancing Asterisk.   Grey Man
• [asterisk-users] Load balancing Asterisk.   Nitzan Kon
• [asterisk-users] Collect digits from the Callee after the Call is connected.   Simith Nambiar
• [asterisk-users] Load balancing Asterisk.   Alex Balashov
• [asterisk-users] jitterbuffer   farah.auf at epfl.ch
• [asterisk-users] Load balancing Asterisk.   Alex Balashov
• [asterisk-users] Role of asterisk   Bill Andersen
• [asterisk-users] Load balancing Asterisk.   SIP
• [asterisk-users] Load balancing Asterisk.   Grygoriy Dobrovolskyy
• [asterisk-users] Role of asterisk   Valentin Bud
• [asterisk-users] Load balancing Asterisk.   Nitzan Kon
• [asterisk-users] Macro conversion in 1.6   Joseph L. Casale
• [asterisk-users] Load balancing Asterisk.   Alex Balashov
• [asterisk-users] Using MAC or extension number as SIP identifier   Olivier
• [asterisk-users] Any other "free" toll free SIP providers out there?   Atis Lezdins
• [asterisk-users] Load balancing Asterisk.   Grey Man
• [asterisk-users] Macro conversion in 1.6   Atis Lezdins
• [asterisk-users] Using MAC or extension number as SIP identifier   Daniel Hazelbaker
• [asterisk-users] Using MAC or extension number as SIP identifier   Steve Howes
• [asterisk-users] Using MAC or extension number as SIP identifier   Philipp Kempgen
• [asterisk-users] Using MAC or extension number as SIP identifier   Olivier
• [asterisk-users] Macro conversion in 1.6   Joseph L. Casale
• [asterisk-users] Using MAC or extension number as SIP identifier   Olivier
• [asterisk-users] SIP to IAX2 with delayed echo   c james
• [asterisk-users] Using MAC or extension number as SIP identifier   Eric "ManxPower" Wieling
• [asterisk-users] two sip listening ports for single asterisk   Matthew J. Roth
• [asterisk-users] SIP to IAX2 with delayed echo   Tim Nelson
• [asterisk-users] setting up callback   Mikhail (Plus Plus)
• [asterisk-users] Load balancing Asterisk.   Alex Balashov
• [asterisk-users] Using MAC or extension number as SIP identifier   Daniel Hazelbaker
• [asterisk-users] SIP to IAX2 with delayed echo   Steve Totaro
• [asterisk-users] Any other "free" toll free SIP providers out there?   Tom Browning
• [asterisk-users] Sending / Receiving sms messages with Portech 370   Julian Lyndon-Smith
• [asterisk-users] Low RX volume and half duplex/"walkie-talkie" on AEX-804E   Lincoln King-Cliby
• [asterisk-users] Disable native bridge?   Tod Fitch
• [asterisk-users] Using MAC or extension number as SIP identifier   Olivier
• [asterisk-users] Subversion Mirror Down for Maintenance   Russell Bryant
• [asterisk-users] SIP to IAX2 with delayed echo   c james
• [asterisk-users] SIP to IAX2 with delayed echo   c james
• [asterisk-users] SIP to IAX2 with delayed echo   Steve Totaro
• [asterisk-users] How long will Asterisk 1.4.x supported/maintained   Steve Totaro
• [asterisk-users] SIP to IAX2 with delayed echo   Dave Fullerton
• [asterisk-users] SIP to IAX2 with delayed echo   Steve Totaro
• [asterisk-users] Using MAC or extension number as SIP identifier   Eric "ManxPower" Wieling
• [asterisk-users] VB 6 developer needed   Gregory Malsack
• [asterisk-users] Rv: sas   sas sas
• [asterisk-users] SIP to IAX2 with delayed echo   Drew Gibson
• [asterisk-users] Using MAC or extension number as SIP identifier   Philipp Kempgen
• [asterisk-users] A question about how much an Asterisk Dcap consultant and a Sipmaster make   Scott Berry
• [asterisk-users] A question about how much an Asterisk Dcap consultant and a Sipmaster make   Steve Totaro
• [asterisk-users] Elastix workshop in Toronto; Wed Nov 26th, 2008   Simon P. Ditner
• [asterisk-users] dial console/dsp hear crackling in headset   Jerry Geis
• [asterisk-users] SIP to IAX2 with delayed echo   Tim Panton
• [asterisk-users] DTMF issue   michel freiha
• [asterisk-users] SIP to IAX2 with delayed echo   Benny Amorsen
• [asterisk-users] SIP to IAX2 with delayed echo   c james
• [asterisk-users] SIP to IAX2 with delayed echo   c james
• [asterisk-users] Limit the number of users in a meetme conference?   Noah Miller
• [asterisk-users] Limit the number of users in a meetme conference?   Danny Nicholas
• [asterisk-users] OT: ATA causes random DTMF in stream   OCG Technical Support
• [asterisk-users] Playback using AMI   Jim Dickenson
• [asterisk-users] Limit the number of users in a meetme conference?   Singer X.J. Wang
• [asterisk-users] Limit the number of users in a meetme conference?   Dan Austin
• [asterisk-users] Playback using AMI   Danny Nicholas
• [asterisk-users] ISDN Cause codes   Robert Boardman
• [asterisk-users] A way to run extenrnotify when IMAP events take place...   Jeffrey Phelps
• [asterisk-users] ISDN Cause codes   Don Kelly
• [asterisk-users] SIP to IAX2 with delayed echo   Dave Platt
• [asterisk-users] SVN - DIGIUM   Luis Morales
• [asterisk-users] A way to run extenrnotify when IMAP events take place...   Philipp Kempgen
• [asterisk-users] SVN - Digium   Philipp Kempgen
• [asterisk-users] Playback using AMI   Jim Dickenson
• [asterisk-users] SVN - Digium   Luis Morales
• [asterisk-users] SIP to IAX2 with delayed echo   Steve Totaro
• [asterisk-users] Full Duplex   Matt Riddell
• [asterisk-users] A way to run extenrnotify when IMAP events take place...   Jeffrey Phelps
• [asterisk-users] Full Duplex   Adam Lovegrove
• [asterisk-users] Group count not being preserved when transferring a call into a conference   Daniel Johnson
• [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities   Yehavi Bourvine
• [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities   Dan Austin
• [asterisk-users] ISDN Cause codes   Robert Boardman
• [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities   Al Baker
• [asterisk-users] sip trunking and call transfer   nik600
• [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities   Gordon Henderson
• [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities   EdPimentl
• [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities   Grygoriy Dobrovolskyy
• [asterisk-users] PSTN Gateway setup   Valentin Bud
• [asterisk-users] SVN - DIGIUM   Grygoriy Dobrovolskyy
• [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities   Yehavi Bourvine
• [asterisk-users] A way to run extenrnotify when IMAP events take place...   Philipp Kempgen
• [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities   RE Kushner List Account
• [asterisk-users] SPA2100 transfer to ASTERISK CID   Sebastian Milioto
• [asterisk-users] ISDN Cause codes   Martin Smith
• [asterisk-users] A way to run extenrnotify when IMAP events take place...   Danny Nicholas
• [asterisk-users] Large Asterisk installations (~10, 000 extensions), preferably at universities   Jason Aarons (US)
• [asterisk-users] Large Asterisk installations (~10, 000 extensions), preferably at universities   Alex Balashov
• [asterisk-users] A way to run extenrnotify when IMAP events take place...   Philipp Kempgen
• [asterisk-users] Large Asterisk installations (~10, 000 extensions), preferably at universities   Alex Balashov
• [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities   Alex Balashov
• [asterisk-users] Ping   Sebastian Milioto
• [asterisk-users] Limit the number of users in a meetme conference?   Noah Miller
• [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities   Noah Miller
• [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities   Alex Balashov
• [asterisk-users] [SOLVED] TDM400 (?) zap hangup   Roderick A. Anderson
• [asterisk-users] How long will Asterisk 1.4.x supported/maintained   Matt Florell
• [asterisk-users] Ping   Atis Lezdins
• [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities   Gonzalo Servat
• [asterisk-users] Limit the number of users in a meetmeconference?   Danny Nicholas
• [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities   Alex Balashov
• [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities   RE Kushner List Account
• [asterisk-users] upgrade from 1.2 to 1.4 and now half channel audio   Jerry Geis
• [asterisk-users] OT - SIP message encoding to enhance text display   Olivier
• [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities   Gonzalo Servat
• [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities   Noah Miller
• [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities   Alex Balashov
• [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities   Alex Balashov
• [asterisk-users] upgrade from 1.2 to 1.4 and now half channel audio   Danny Nicholas
• [asterisk-users] upgrade from 1.2 to 1.4 and now half channel audio   Jerry Geis
• [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities   Gonzalo Servat
• [asterisk-users] upgrade from 1.2 to 1.4 and now half channelaudio   Danny Nicholas
• [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities   Yehavi Bourvine
• [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities   Alex Balashov
• [asterisk-users] Log level of 500 Server Internal Error.   Atis Lezdins
• [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities   Alex Balashov
• [asterisk-users] Log level of 500 Server Internal Error.   Alex Balashov
• [asterisk-users] Log level of 500 Server Internal Error.   Atis Lezdins
• [asterisk-users] Log level of 500 Server Internal Error.   Alex Balashov
• [asterisk-users] Log level of 500 Server Internal Error.   Tom Moore
• [asterisk-users] hint priority with 50 channels   Loic Didelot
• [asterisk-users] Limit the number of users in a meetmeconference?   Atis Lezdins
• [asterisk-users] How long will Asterisk 1.4.x supported/maintained   Tilghman Lesher
• [asterisk-users] TrixBox problem...   Gregory Malsack
• [asterisk-users] Full Duplex   Matthew Fredrickson
• [asterisk-users] TrixBox problem...   Tim Nelson
• [asterisk-users] TrixBox problem...   Andres
• [asterisk-users] How long will Asterisk 1.4.x supported/maintained   Matt Florell
• [asterisk-users] Best way to handle include files?   Doug
• [asterisk-users] hint priority with 50 channels   Danny Nicholas
• [asterisk-users] hint priority with 50 channels   Philipp Kempgen
• [asterisk-users] upgrade from 1.2 to 1.4 and now half channel audio   Jerry Geis
• [asterisk-users] How long will Asterisk 1.4.x supported/maintained   Steve Totaro
• [asterisk-users] How long will Asterisk 1.4.x supported/maintained   Alex Balashov
• [asterisk-users] How long will Asterisk 1.4.x supported/maintained   Matt Florell
• [asterisk-users] ISDN Cause codes   Robert Boardman
• [asterisk-users] hint priority with 50 channels   Anthony Francis
• [asterisk-users] ISDN Cause codes   Martin Smith
• [asterisk-users] Setting up to reveive faxes.   Ken D'Ambrosio
• [asterisk-users] MozIAX - Mozilla IAX2 soft-phone 3sec delay   Joseph
• [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities   Terry Wilson
• [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities   Terry Wilson
• [asterisk-users] MOH Realtime Problem   Sebastian
• [asterisk-users] MoH in a loop   Robert Augustyn
• [asterisk-users] Large Asterisk installations (~10, 000 extensions), preferably at universities   Wilton Helm
• [asterisk-users] MOH Realtime Problem   Sebastian
• [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities   Michael Collins
• [asterisk-users] CDR Desgin   Grey Man
• [asterisk-users] Need Recording Solution in Asterisk   Kashif Naeem
• [asterisk-users] Asterisk Instant message passing with eyebeam   Max Alex
• [asterisk-users] Upgrade 1.4.19 to 1.6 => segementation fault   Ronald Wiplinger (Lists)
• [asterisk-users] hint priority with 50 channels   Loic Didelot
• [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities   Tim Panton
• [asterisk-users] MozIAX - Mozilla IAX2 soft-phone 3sec delay   Tim Panton
• [asterisk-users] IMAP voicemail with Exchange (was: A way to run extenrnotify when IMAP events take place...)   Noah Miller
• [asterisk-users] [SOLVED] Re: Upgrade 1.4.19 to 1.6 => segementation fault   Ronald Wiplinger (Lists)
• [asterisk-users] Setting up to reveive faxes.   Noah Miller
• [asterisk-users] Setting up to reveive faxes.   Gordon Henderson
• [asterisk-users] hint priority with 50 channels   Anthony Francis
• [asterisk-users] Setting up to reveive faxes.   Anthony Messina
• [asterisk-users] hint priority with 50 channels   Loic Didelot
• [asterisk-users] ISDN Cause codes   Robert Boardman
• [asterisk-users] MozIAX - Mozilla IAX2 soft-phone 3sec delay   Joseph
• [asterisk-users] hint priority with 50 channels   Tilghman Lesher
• [asterisk-users] Setting up to reveive faxes.   Tilghman Lesher
• [asterisk-users] Setting up to reveive faxes.   Gordon Henderson
• [asterisk-users] hint priority with 50 channels   Loic Didelot
• [asterisk-users] Need Recording Solution in Asterisk   David Cook
• [asterisk-users] MozIAX - Mozilla IAX2 soft-phone 3sec delay   Jean-Denis Girard
• [asterisk-users] CDR Desgin   Grey Man
• [asterisk-users] presence with polycom DND   Paul Hales
• [asterisk-users] IMAP voicemail with Exchange (was: A way to run extenrnotify when IMAP events take place...)   Sigma Networks
• [asterisk-users] Asterisk Instant message passing with eyebeam   Alex Balashov
• [asterisk-users] Large Asterisk installations (~10, 000 extensions), preferably at universities   William Muriithi
• [asterisk-users] SendImage()   Philipp Kempgen
• [asterisk-users] MOH Realtime Problem   Sebastian
• [asterisk-users] How does IMAP notify Asterisk that I've read a message?   Barry L. Kline
• [asterisk-users] SendImage()   Rob Hillis
• [asterisk-users] database queries from extensions.conf   Al Baker
• [asterisk-users] database queries from extensions.conf   Al Baker
• [asterisk-users] Asterisk 1.6 mysql cdr log problem   Artifex Maximus
• [asterisk-users] hint priority with 50 channels   Loic Didelot
• [asterisk-users] hint priority with 50 channels   Loic Didelot
• [asterisk-users] How does IMAP notify Asterisk that I've read amessage?   joea at j4computers.com
• [asterisk-users] CDR Design   Grey Man
• [asterisk-users] How does IMAP notify Asterisk that I've read amessage?   Barry L. Kline
• [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities   Eric Chamberlain
• [asterisk-users] Need Recording Solution in Asterisk   Daniel Varella
• [asterisk-users] Collect digits from the Callee after the Call is connected.   Doug Lytle
• [asterisk-users] Collect digits from the Callee after the Call is connected.   Simith Nambiar
• [asterisk-users] SendImage()   Tilghman Lesher
• [asterisk-users] How does IMAP notify Asterisk that I've read a message?   Tilghman Lesher
• [asterisk-users] Asterisk 1.6, IMAP Voicemail and externnotify   Barry L. Kline
• [asterisk-users] How does IMAP notify Asterisk that I've read a message?   Barry L. Kline
• [asterisk-users] sip trunking and call transfer   nik600
• [asterisk-users] Problem with DAHDI and OSLEC integration.   Marco Signorini
• [asterisk-users] asterisk and vlc   Jerry Geis
• [asterisk-users] sip trunking and call transfer   Eric "ManxPower" Wieling
• [asterisk-users] Problem with DAHDI and OSLEC integration.   Joseph L. Casale
• [asterisk-users] sip trunking and call transfer   Raj Jain
• [asterisk-users] Problem with DAHDI and OSLEC integration.   Tzafrir Cohen
• [asterisk-users] Problem with DAHDI and OSLEC integration.   Joseph L. Casale
• [asterisk-users] IMAP voicemail with Exchange (was: A way to run extenrnotify when IMAP events take place...)   Noah Miller
• [asterisk-users] asterisk and v4l   Jerry Geis
• [asterisk-users] pick up IAX2 calls   Bruno Castelo Branco
• [asterisk-users] pick up IAX2 calls   Luis Morales
• [asterisk-users] Problem with DAHDI and OSLEC integration.   Marco Signorini
• [asterisk-users] SendText and non-ASCII characters   Olivier
• [asterisk-users] Asterisk Server Fine Tuning for Best Performance   Fernando Berretta
• [asterisk-users] asterisk and v4l   Tzafrir Cohen
• [asterisk-users] Fwd: SPA2100 transfer to ASTERISK CID   Sebastian Milioto
• [asterisk-users] Fwd: SPA2100 transfer to ASTERISK CID   vision_admin at vision21networks.com
• [asterisk-users] play sound while executing agi script   Giedrius Augys
• [asterisk-users] CDR Design   Grey Man
• [asterisk-users] play sound while executing agi script   Tilghman Lesher
• [asterisk-users] CDR Design   regs at kinetix.gr
• [asterisk-users] ISDN Cause codes   Martin Smith
• [asterisk-users] database queries from extensions.conf   Jared Smith
• [asterisk-users] IMAP voicemail with Exchange (was: A way to run extenrnotify when IMAP events take place...)   Jeffrey Phelps
• [asterisk-users] SendText and non-ASCII characters   Philipp Kempgen
• [asterisk-users] IMAP voicemail with Exchange (was: A way torun extenrnotify when IMAP events take place...)   Jeffrey Phelps
• [asterisk-users] SendText and non-ASCII characters   Philipp Kempgen
• [asterisk-users] SendText and non-ASCII characters   Philipp Kempgen
• [asterisk-users] Asterisk 1.6, IMAP Voicemail and externnotify   Jeffrey Phelps
• [asterisk-users] database queries from extensions.conf   Jared Smith
• [asterisk-users] sip trunking and call transfer   nik600
• [asterisk-users] Asterisk 1.6, IMAP Voicemail and externnotify   Steve Totaro
• [asterisk-users] Asterisk 1.6, IMAP Voicemail and externnotify   Barry L. Kline
• [asterisk-users] play sound while executing agi script   Steve Edwards
• [asterisk-users] CDR Design   Anthony Francis
• [asterisk-users] SendText and non-ASCII characters   Olivier
• [asterisk-users] RFC 3428 (was: Re: SendText and non-ASCII characters)   Philipp Kempgen
• [asterisk-users] MOH Realtime Problem   dinhtrung
• [asterisk-users] RFC 3428 (was: Re: SendText and non-ASCII characters)   Olivier
• [asterisk-users] SendText and non-ASCII characters   Philipp Kempgen
• [asterisk-users] RFC 3428   Philipp Kempgen
• [asterisk-users] pick up IAX2 calls   Eric "ManxPower" Wieling
• [asterisk-users] pick up IAX2 calls   Steve Totaro
• [asterisk-users] CDR Design   regs at kinetix.gr
• [asterisk-users] HPEC performance   Tzafrir Cohen
• [asterisk-users] SendImage()   Atis Lezdins
• [asterisk-users] Asterisk 1.6 mysql cdr log problem   Atis Lezdins
• [asterisk-users] reducing iax packet size   Pezhman Lali
• [asterisk-users] Log level of 500 Server Internal Error.   Atis Lezdins
• [asterisk-users] SendText and non-ASCII characters   Olivier
• [asterisk-users] database queries from extensions.conf   Julian Lyndon-Smith
• [asterisk-users] database queries from extensions.conf   Atis Lezdins
• [asterisk-users] database queries from extensions.conf   Julian Lyndon-Smith
• [asterisk-users] CDR Desgin   Steve Murphy
• [asterisk-users] SendText and non-ASCII characters   Philipp Kempgen
• [asterisk-users] Large Asterisk installations (~10, 000 extensions), preferably at universities   Steve Totaro
• [asterisk-users] Need Recording Solution in Asterisk   Stephen Wingfield
• [asterisk-users] Asterisk 1.6 mysql cdr log problem   Tilghman Lesher
• [asterisk-users] pick up IAX2 calls   Bruno Castelo Branco
• [asterisk-users] Asterisk 1.6 mysql cdr log problem   Tilghman Lesher
• [asterisk-users] cdr mysql error   Nhadie
• [asterisk-users] cdr mysql error   Jai Rangi
• [asterisk-users] CDR Desgin   regs at kinetix.gr
• [asterisk-users] CDR Desgin   Grey Man
• [asterisk-users] OSLEC build errors on DAHDI [was: Re: HPEC performance]   Tzafrir Cohen
• [asterisk-users] Collect digits from the Callee after the Call is connected.   Simith Nambiar
• [asterisk-users] Asterisk 1.6 mysql cdr log problem   Atis Lezdins
• [asterisk-users] Set a specific registration expiry value to a given peer without touching defaultexpiry in sip.conf ?   Olivier
• [asterisk-users] Disabling Call-Waiting   Elliot Murdock
• [asterisk-users] Disabling Call-Waiting   Sebastian
• [asterisk-users] Asterisk 1.6 mysql cdr log problem   Artifex Maximus
• [asterisk-users] Disabling Call-Waiting   Elliot Murdock
• [asterisk-users] Asterisk 1.6 mysql cdr log problem   Sebastian
• [asterisk-users] Disabling Call-Waiting   Gordon Henderson
• [asterisk-users] AsteriskNOW 1.5 upgrade from 1.4 to 1.6   Jason Lixfeld
• [asterisk-users] The sound is played but I did not hear   jhon digital21
• [asterisk-users] The sound is played but I did not hear   Doug Lytle
• [asterisk-users] Disabling Call-Waiting   Mikhail (Plus Plus)
• [asterisk-users] AsteriskNOW 1.5 upgrade from 1.4 to 1.6   Jason Parker
• [asterisk-users] Disabling Call-Waiting   Gordon Henderson
• [asterisk-users] Disabling Call-Waiting   Sebastian
• [asterisk-users] Disabling Call-Waiting   Elliot Murdock
• [asterisk-users] Disabling Call-Waiting   Tilghman Lesher
• [asterisk-users] CDR Design   Anthony Francis
• [asterisk-users] Disabling Call-Waiting   Tilghman Lesher
• [asterisk-users] CDR Design   Steve Murphy
• [asterisk-users] CDR Desgin   Steve Murphy
• [asterisk-users] Disabling Call-Waiting   Sebastian
• [asterisk-users] MOH Realtime   Sebastian
• [asterisk-users] Disabling Call-Waiting   Yehavi Bourvine
• [asterisk-users] CDR Design   regs at kinetix.gr
• [asterisk-users] Disabling Call-Waiting   Sebastian
• [asterisk-users] pick up IAX2 calls   Tim Panton
• [asterisk-users] half channel audio after upgrade to 1.4.18   Jerry Geis
• [asterisk-users] Disabling Call-Waiting   Elliot Murdock
• [asterisk-users] two sip listening ports for single asterisk   Rizwan Hisham
• [asterisk-users] CDR Desgin   Benny Amorsen
• [asterisk-users] Disabling Call-Waiting   Benny Amorsen
• [asterisk-users] Connecting AS5350XM with Asterisk   A T I F
• [asterisk-users] Connecting AS5350XM with Asterisk   A T I F
• [asterisk-users] Connecting AS5350XM with Asterisk   A T I F
• [asterisk-users] Connecting AS5350XM with Asterisk   Alex Balashov
• [asterisk-users] Disabling Call-Waiting   Paul Hales
• [asterisk-users] half channel audio after upgrade to 1.4.18   Paul Hales
• [asterisk-users] Connecting AS5350XM with Asterisk   A T I F
• [asterisk-users] Connecting AS5350XM with Asterisk   A T I F
• [asterisk-users] Connecting AS5350XM with Asterisk   Alex Balashov
• [asterisk-users] Connecting AS5350XM with Asterisk   Alex Balashov
• [asterisk-users] Connecting AS5350XM with Asterisk   A T I F
• [asterisk-users] Connecting AS5350XM with Asterisk   Alex Balashov
• [asterisk-users] Connecting AS5350XM with Asterisk   A T I F
• [asterisk-users] CDR Desgin   Freddi Hansen
• [asterisk-users] Connecting AS5350XM with Asterisk   Alex Balashov
• [asterisk-users] Connecting AS5350XM with Asterisk   A T I F
• [asterisk-users] Connecting AS5350XM with Asterisk   Alex Balashov
• [asterisk-users] pick up IAX2 calls   Bruno Castelo Branco
• [asterisk-users] bridging - Didn't get a frame from channel   Tony Gaspar
• [asterisk-users] Ring/Off-hook in strange state 6 channel X   research at businesstz.com
• [asterisk-users] The sound is played but I did not hear   jhon digital21
• [asterisk-users] CDR Desgin   regs at kinetix.gr
• [asterisk-users] Ring/Off-hook in strange state 6 channel X   Tzafrir Cohen
• [asterisk-users] Asterisk voicemail and Lotus Notes   Olivier
• [asterisk-users] The sound is played but I did not hear   Doug Lytle
• [asterisk-users] SVN   Alex Montoanelli
• [asterisk-users] SVN   Michiel van Baak
• [asterisk-users] Channel variable to identify the calling SIP peer   Richard Brady
• [asterisk-users] Channel variable to identify the calling SIP peer   Grey Man
• [asterisk-users] half channel audio after upgrade to 1.4.18   Jerry Geis
• [asterisk-users] Hints stopped working suddently   Mike
• [asterisk-users] Hints stopped working suddently   Danny Nicholas
• [asterisk-users] Mobile as FXO   Irfan Malik
• [asterisk-users] Mobile as FXO   Steve Totaro
• [asterisk-users] language and meetme issue   Giedrius Augys
• [asterisk-users] SVN   Atis Lezdins
• [asterisk-users] Dahdi, b410p and looping from 1 port to another   Olivier
• [asterisk-users] Mobile as FXO   Danny Nicholas
• [asterisk-users] Mobile as FXO   Irfan Malik
• [asterisk-users] language and meetme issue   Danny Nicholas
• [asterisk-users] Mobile as FXO   Danny Nicholas
• [asterisk-users] Mobile as FXO   Steve Totaro
• [asterisk-users] Mobile as FXO   Irfan Malik
• [asterisk-users] Mobile as FXO   Irfan Malik
• [asterisk-users] Mobile as FXO   Irfan Malik
• [asterisk-users] MS Exchange IMAP Voicemail   Jeffrey Phelps
• [asterisk-users] The sound is played but I did not hear   Doug Lytle
• [asterisk-users] Mobile as FXO   Danny Nicholas
• [asterisk-users] Dahdi, b410p and looping from 1 port to another   Kevin P. Fleming
• [asterisk-users] The sound is played but I did not hear   jhon digital21
• [asterisk-users] language and meetme issue   Giedrius Augys
• [asterisk-users] Dahdi, b410p and looping from 1 port to another   Olivier
• [asterisk-users] language and meetme issue   Danny Nicholas
• [asterisk-users] SVN   Alex Montoanelli
• [asterisk-users] Hints stopped working suddently   Kai-Uwe Jensen
• [asterisk-users] 1.4.x Strange Vocemail delay   Sven Geggus
• [asterisk-users] Hints stopped working suddently   Danny Nicholas
• [asterisk-users] language and meetme issue   Giedrius Augys
• [asterisk-users] Customized CDR Records   David Budny
• [asterisk-users] Dahdi, b410p and looping from 1 port to another   Tzafrir Cohen
• [asterisk-users] spandsp not recognized by menuselect on Lenny   Olivier
• [asterisk-users] Hints stopped working suddently   Mike
• [asterisk-users] Hints stopped working suddently   Mike
• [asterisk-users] Hints stopped working suddently   Danny Nicholas
• [asterisk-users] sip MWI Messages-Waiting: always reports no messages   Mark G. Thomas
• [asterisk-users] Hints stopped working suddently   Mike
• [asterisk-users] Dahdi, b410p and looping from 1 port to another   Olivier
• [asterisk-users] Customized CDR Records   Tilghman Lesher
• [asterisk-users] Dahdi, b410p and looping from 1 port to another   Kevin P. Fleming
• [asterisk-users] The sound is played but I did not hear   Doug Lytle
• [asterisk-users] Problems with Rhino Channelbank...   Carlos Chavez
• [asterisk-users] spandsp not recognized by menuselect on Lenny   Tzafrir Cohen
• [asterisk-users] pick up IAX2 calls   Eric "ManxPower" Wieling
• [asterisk-users] spandsp not recognized by menuselect on Lenny   Philipp Kempgen
• [asterisk-users] spandsp not recognized by menuselect on Lenny   Tzafrir Cohen
• [asterisk-users] Dahdi, b410p and looping from 1 port to another   Olivier
• [asterisk-users] spandsp not recognized by menuselect on Lenny   Olivier
• [asterisk-users] spandsp not recognized by menuselect on Lenny [SOLVED]   Olivier
• [asterisk-users] spandsp not recognized by menuselect on Lenny   Philipp Kempgen
• [asterisk-users] spandsp not recognized by menuselect on Lenny   Philipp Kempgen
• [asterisk-users] resample not recognized by menuselect on Lenny   Philipp Kempgen
• [asterisk-users] sip MWI Messages-Waiting: always reports no messages   Lincoln King-Cliby
• [asterisk-users] Asterisk daemon dies about once per day   Douglas Mortensen
• [asterisk-users] resample not recognized by menuselect on Lenny   Tzafrir Cohen
• [asterisk-users] spandsp not recognized by menuselect on Lenny   Tzafrir Cohen
• [asterisk-users] Asterisk daemon dies about once per day   Tzafrir Cohen
• [asterisk-users] 2 Asterisks to one PBX - E1 conection   dubravko caric
• [asterisk-users] 2 Asterisks to one PBX - E1 conection   Steve Totaro
• [asterisk-users] Softphone IP publico x privado   Luis Antonio Prata Barbosa
• [asterisk-users] Dahdi, b410p and looping from 1 port to another   Kevin P. Fleming
• [asterisk-users] resample not recognized by menuselect on Lenny   Kevin P. Fleming
• [asterisk-users] CDR Hangupcause   Sebastian
• [asterisk-users] resample not recognized by menuselect on Lenny   Philipp Kempgen
• [asterisk-users] Dahdi, b410p and looping from 1 port to another   Olivier
• [asterisk-users] Dahdi, b410p and looping from 1 port to another   Olivier
• [asterisk-users] Dahdi, b410p and looping from 1 port to another   Jon Pounder
• [asterisk-users] Dahdi, b410p and looping from 1 port to another   Kevin P. Fleming
• [asterisk-users] Asterisk daemon dies about once per day   Tilghman Lesher
• [asterisk-users] Asterisk daemon dies about once per day   Tzafrir Cohen
• [asterisk-users] 2 Asterisks to one PBX - E1 conection   Alejandro Kauffmann
• [asterisk-users] 2 Asterisks to one PBX - E1 conection   David Backeberg
• [asterisk-users] bridging - Didn't get a frame from channel   Tony Gaspar
• [asterisk-users] pick up IAX2 calls   Bruno Castelo Branco
• [asterisk-users] Ring/Off-hook in strange state 6 channel X   research at businesstz.com
• [asterisk-users] Softphones with RPID and BLF   Yehavi Bourvine
• [asterisk-users] pick up IAX2 calls   coco
• [asterisk-users] 2 Asterisks to one PBX - E1 conection   dubravko caric
• [asterisk-users] 2 Asterisks to one PBX - E1 conection   dubravko caric
• [asterisk-users] Unknown signalling method 'bri_cpe   Olivier
• [asterisk-users] Unknown signalling method 'bri_cpe   Tzafrir Cohen
• [asterisk-users] Unknown signalling method 'bri_cpe   Olivier
• [asterisk-users] Disable Transfer   Max Alex
• [asterisk-users] Any 1.6 SendFAX example ?   Olivier
• [asterisk-users] trunk peer not registering after migrating installation   John Taylor
• [asterisk-users] Mobile as FXO   Irfan Malik
• [asterisk-users] Unknown signalling method 'bri_cpe   Tzafrir Cohen
• [asterisk-users] Mobile as FXO   Tzafrir Cohen
• [asterisk-users] Unknown signalling method 'bri_cpe   Olivier
• [asterisk-users] Any 1.6 SendFAX example ?   Atis Lezdins
• [asterisk-users] Unknown signalling method 'bri_cpe   Tzafrir Cohen
• [asterisk-users] Wellgate & Asterisk   Ronald Wiplinger (Lists)
• [asterisk-users] Wellgate & Asterisk   Eberhard Roloff
• [asterisk-users] Unknown signalling method 'bri_cpe   Olivier
• [asterisk-users] Any 1.6 SendFAX example ?   Olivier
• [asterisk-users] Wellgate & Asterisk   Guillermo Salas M.
• [asterisk-users] Any 1.6 SendFAX example ?   Steve Underwood
• [asterisk-users] Hints stopped working suddently   Mike
• [asterisk-users] originate problem   Johan Sandgren
• [asterisk-users] FW: cdr_addon_mysql.so did not register itselfduringload   Matthias Urlichs
• [asterisk-users] originate problem   Tzafrir Cohen
• [asterisk-users] Hints stopped working suddently   Julian Lyndon-Smith
• [asterisk-users] pick up IAX2 calls   Eric "ManxPower" Wieling
• [asterisk-users] Any 1.6 SendFAX example ?   Atis Lezdins
• [asterisk-users] MOH Realtime   Sebastian
• [asterisk-users] force channel hangup   Kelvin Chan
• [asterisk-users] Windows Mobile 6 SIP client: Remote host can't match request NOTIFY to call   OCG Technical Support
• [asterisk-users] MOH Realtime   dinhtrung
• [asterisk-users] Large Asterisk installations (~10, 000 extensions), preferably at universities   Yehavi Bourvine
• [asterisk-users] [Solved] Wellgate & Asterisk   Ronald Wiplinger (Lists)
• [asterisk-users] Windows Mobile 6 SIP client: Remote host can't match request NOTIFY to call   Sebastian
• [asterisk-users] MOH Realtime   Sebastian
• [asterisk-users] Anonymous callerid   Max Alex
• [asterisk-users] Large Asterisk installations (~10, 000 extensions), preferably at universities   Grygoriy Dobrovolskyy
• [asterisk-users] force channel hangup   Mr Shunz
• [asterisk-users] MixMonitor with non-20ms packets   Grigoriy Puzankin
• [asterisk-users] [SPAM] - Re: FW: cdr_addon_mysql.so did notregister itselfduringload - Email found in subject   Andrew Thomas
• [asterisk-users] originate problem   Johan Sandgren
• [asterisk-users] Priority between calls from different queues   equis software
• [asterisk-users] Asterisk and S-Bus   Steven.Moughan at lakecommunications.com
• [asterisk-users] [SPAM] - Asterisk and S-Bus - Email found in subject   Andrew Thomas
• [asterisk-users] length of field names   Jerry Geis
• [asterisk-users] Priority between calls from different queues   Atis Lezdins
• [asterisk-users] Priority between calls from different queues   equis software
• [asterisk-users] Priority between calls from different queues   Darrin Henshaw
• [asterisk-users] Anonymous callerid   Philipp Kempgen
• [asterisk-users] Priority between calls from different queues   Atis Lezdins
• [asterisk-users] Priority between calls from different queues   equis software
• [asterisk-users] Priority between calls from different queues   equis software
• [asterisk-users] RTCP too short   michel freiha
• [asterisk-users] Calls drop after a couple of minutes.   Simon Tennant
• [asterisk-users] Asterisk SIP security   Mike
• [asterisk-users] Hints stopped working suddently   Mike
• [asterisk-users] Hints stopped working suddently   Danny Nicholas
• [asterisk-users] Friday at 12 Noon ET, the VoIP Users Conference reminder   randulo
• [asterisk-users] Priority between calls from different queues   Atis Lezdins
• [asterisk-users] Asterisk and multicast RTP   Cesc Santa
• [asterisk-users] Priority between calls from different queues   equis software
• [asterisk-users] RTCP too short   Jon Weisman
• [asterisk-users] RTCP too short   Danny Nicholas
• [asterisk-users] RTCP too short   Jon Weisman
• [asterisk-users] RTCP too short   Danny Nicholas
• [asterisk-users] RTCP too short   Drew Gibson
• [asterisk-users] How to disable trunk from the cli?   Robert Augustyn
• [asterisk-users] Help with asterisk and avaya SIP trunking   Krishna Sumanth Chava
• [asterisk-users] How to disable trunk from the cli?   Danny Nicholas
• [asterisk-users] Asterisk SIP security   Steve Totaro
• [asterisk-users] RTCP too short   Jon Weisman
• [asterisk-users] RTCP too short   Jon Weisman
• [asterisk-users] RTCP too short   Danny Nicholas
• [asterisk-users] Any 1.6 SendFAX example ?   Anthony Messina
• [asterisk-users] force channel hangup   Anthony Messina
• [asterisk-users] force channel hangup   Danny Nicholas
• [asterisk-users] force channel hangup   Alex Balashov
• [asterisk-users] force channel hangup   Danny Nicholas
• [asterisk-users] force channel hangup   Tzafrir Cohen
• [asterisk-users] force channel hangup   Tzafrir Cohen
• [asterisk-users] received wrong state events for originate command   Sun xiaoshuang
• [asterisk-users] Trixbox 2.6.1.13 OpenR2   Yuri
• [asterisk-users] Trixbox 2.6.1.13 OpenR2   Yuri
• [asterisk-users] libspandsp.so.0: cannot open shared object file: No such file or directory   Doug
• [asterisk-users] Trixbox 2.6.1.13 OpenR2   Peter Lindquist
• [asterisk-users] libspandsp.so.0: cannot open shared object file: No such file or directory   Alex Balashov
• [asterisk-users] Anonymous callerid   Max Alex
• [asterisk-users] GSM gateways - which one ?   Julian Lyndon-Smith
• [asterisk-users] GSM gateways - which one ?   Julian Lyndon-Smith
• [asterisk-users] GSM gateways - which one ?   Michael Graves
• [asterisk-users] Anonymous callerid   Tilghman Lesher
• [asterisk-users] Trixbox 2.6.1.13 OpenR2   Moises Silva
• [asterisk-users] asterisk-users Digest, Vol 52, Issue 81   Dallas Carter
• [asterisk-users] Anonymous callerid   Philipp Kempgen
• [asterisk-users] asterisk / sipura call breaking up on silence?!   R_O_L_A_N_D at hotmail.com
• [asterisk-users] asterisk / sipura call breaking up on silence?!   Philipp Kempgen
• [asterisk-users] libspandsp.so.0: cannot open shared object file: No such file or directory   Doug
• [asterisk-users] libspandsp.so.0: cannot open shared object file: No such file or directory   Doug Lytle
• [asterisk-users] libspandsp.so.0: cannot open shared object file: No such file or directory   Tzafrir Cohen
• [asterisk-users] pp_each_user(), pp_each_extension()   Philipp Kempgen
• [asterisk-users] DAHDI issue in dialplan   Joseph L. Casale
• [asterisk-users] DAHDI issue in dialplan   Joseph L. Casale
• [asterisk-users] libspandsp.so.0: cannot open shared object file: No such file or directory   Doug
• [asterisk-users] libspandsp.so.0: cannot open shared object file: No such file or directory   Doug
• [asterisk-users] libspandsp.so.0: cannot open shared object file: No such file or directory   Tzafrir Cohen
• [asterisk-users] libspandsp.so.0: cannot open shared object file: No such file or directory   Doug Lytle
• [asterisk-users] Anonymous callerid   Conrad Wood
• [asterisk-users] DTMF Tones   Robert Boardman
• [asterisk-users] DTMF Tones   Alex Balashov
• [asterisk-users] DTMF Tones   Robert Boardman
• [asterisk-users] DTMF Tones   Philipp Kempgen
• [asterisk-users] DTMF Tones   Robert Boardman
• [asterisk-users] libspandsp.so.0: cannot open shared object file: No such file or directory   Doug
• [asterisk-users] libspandsp.so.0: cannot open shared object file: No such file or directory   Doug
• [asterisk-users] Channel variable to identify the calling SIP peer   Philipp Kempgen
• [asterisk-users] asterisk / sipura call breaking up on silence?!   R_O_L_A_N_D at hotmail.com
• [asterisk-users] asterisk / sipura call breaking up on silence?!   Philipp Kempgen
• [asterisk-users] libspandsp.so.0: cannot open shared object file: No such file or directory   Tzafrir Cohen
• [asterisk-users] Oslec issue   Joseph L. Casale
• [asterisk-users] Unknown signalling method 'bri_cpe   Olivier
• [asterisk-users] asterisk / sipura call breaking up on silence?!   Kai-Uwe Jensen
• [asterisk-users] libspandsp.so.0: cannot open shared object file: No such file or directory   Doug
• [asterisk-users] Howto analyze concurrent ISDN channel usage   Jay Taylor
• [asterisk-users] libspandsp.so.0: cannot open shared object file: No such file or directory   Tzafrir Cohen
• [asterisk-users] libspandsp.so.0: cannot open shared object file: No such file or directory   Doug

Last message date: Sun Nov 30 23:25:03 CST 2008
Archived on: Tue Dec 2 14:45:14 CST 2008

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