March 2008 Archives by date

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Starting: Sat Mar 1 00:17:21 CST 2008
Ending: Mon Mar 31 21:34:07 CDT 2008
Messages: 1944

• [asterisk-users] Help asterisk connectivity with MS SQL   Rahul Yadav
• [asterisk-users] which phones to use ??   Gordon Henderson
• [asterisk-users] Help asterisk connectivity with MS SQL   Tilghman Lesher
• [asterisk-users] iax2 reload does not update channel config for removed key from config file   Hans Feringa
• [asterisk-users] which phones to use ??   Rob Hillis
• [asterisk-users] which phones to use ??   Thomas Kenyon
• [asterisk-users] ASTCC installation error install: invalid user `apache'   admin at ekoonet.com
• [asterisk-users] which phones to use ??   Michael Graves
• [asterisk-users] "callpark" feature in ABE?   Kevin P. Fleming
• [asterisk-users] Cisco 79xx users/consultants, 7970G color in particular share information   Sigma Networks
• [asterisk-users] Mediatrix 1124 and Audiocodes MP Series gateways   Mark Adams
• [asterisk-users] Cisco 79xx users/consultants, 7970G color in particular share information   OCG Technical Support
• [asterisk-users] Cisco 79xx users/consultants, 7970G color in particular share information   Lacy Moore
• [asterisk-users] Cisco 79xx users/consultants, 7970G color in particular share information   Sigma Networks
• [asterisk-users] Cisco 79xx users/consultants, 7970G color in particular share information   Andres Jimenez
• [asterisk-users] Cisco 79xx users/consultants, 7970G color in particular share information   Sigma Networks
• [asterisk-users] Cisco 79xx users/consultants, 7970G color in particular share information   Eric Wieling
• [asterisk-users] Page app, Polycom IP 601, 60 SIP peers, Interesting Issue WORKING NOW   JR Richardson
• [asterisk-users] I need the least expensive way to do this   Timothy C Litwiller
• [asterisk-users] which phones to use ??   shadowym
• [asterisk-users] I need the least expensive way to do this   Tom Moore
• [asterisk-users] ASTCC installation error install: invalid user `apache'   admin at ekoonet.com
• [asterisk-users] I need the least expensive way to do this   Steve Totaro
• [asterisk-users] ASTCC installation error install: invalid user `apache'   admin at ekoonet.com
• [asterisk-users] real zaptel call durations   aymen warfalli
• [asterisk-users] I need the least expensive way to do this   John Novack
• [asterisk-users] ASTCC installation error install: invalid user `apache'   Steve Totaro
• [asterisk-users] real zaptel call durations   Eric Wieling
• [asterisk-users] I need the least expensive way to do this   Sigma Networks
• [asterisk-users] which phones to use ??   Rob Hillis
• [asterisk-users] Page app, Polycom IP 601, 60 SIP peers, Interesting Issue WORKING NOW   Rob Hillis
• [asterisk-users] I need the least expensive way to do this   Andrew Ladanowski
• [asterisk-users] scripts to convert .conf files to SQL for realtime   Vieri
• [asterisk-users] real zaptel call durations   Tzafrir Cohen
• [asterisk-users] I need the least expensive way to do this   John Novack
• [asterisk-users] Cisco 7910 and Asterisk   Mr Gabriel Ogunleye
• [asterisk-users] I need the least expensive way to do this   Senad Jordanovic
• [asterisk-users] I need the least expensive way to do this   Timothy C Litwiller
• [asterisk-users] I need the least expensive way to do this   Steve Totaro
• [asterisk-users] I need the least expensive way to do this   C F
• [asterisk-users] I need the least expensive way to do this   C F
• [asterisk-users] which phones to use ??   C F
• [asterisk-users] Cisco 79xx users/consultants, 7970G color in particular share information   Sean Dennis
• [asterisk-users] which phones to use ??   randulo
• [asterisk-users] which phones to use ??   randulo
• [asterisk-users] DID number   Mike
• [asterisk-users] DID number   Gordon Henderson
• [asterisk-users] Cisco 79xx users/consultants, 7970G color in particular share information   Alberto Pastore
• [asterisk-users] DID number   Mark Edwards
• [asterisk-users] Cisco 7965g and asterisk   npf-mlists at eurotux.com
• [asterisk-users] manager ignore my settings   Tzafrir Cohen
• [asterisk-users] override/redefine asterisk DB function   Vieri
• [asterisk-users] override/redefine asterisk DB function   Tilghman Lesher
• [asterisk-users] real zaptel call durations   aymen warfalli
• [asterisk-users] OT - CEBIT next week!   Zoa
• [asterisk-users] real zaptel call durations   Tzafrir Cohen
• [asterisk-users] I need the least expensive way to do this   Michael Graves
• [asterisk-users] Cisco 7970 - register with NAT phone   Sigma Networks
• [asterisk-users] which phones to use ??   Michael Graves
• [asterisk-users] Cisco 79xx users/consultants, 7970G color in particular share information (was: Re: asterisk-users Digest, Vol 44, Issue 3)   Matthew Rubenstein
• [asterisk-users] TDM400P dialout problem   Martin
• [asterisk-users] which phones to use ??   Rob Hillis
• [asterisk-users] DID number   Erik Anderson
• [asterisk-users] DID number   randulo
• [asterisk-users] Page app, Polycom IP 601, 60 SIP peers, Interesting Issue WORKING NOW   JR Richardson
• [asterisk-users] TDM400P dialout problem   sean darcy
• [asterisk-users] TDM400P dialout problem   Tzafrir Cohen
• [asterisk-users] TDM400P dialout problem   sean darcy
• [asterisk-users] TDM400P dialout problem   Kevin P. Fleming
• [asterisk-users] TDM400P dialout problem   sean darcy
• [asterisk-users] TDM400P dialout problem   sean darcy
• [asterisk-users] override/redefine asterisk DB function   Benny Amorsen
• [asterisk-users] override/redefine asterisk DB function   Tzafrir Cohen
• [asterisk-users] chan_ss7 0.10   marek cervenka
• [asterisk-users] TDM400P dialout problem   sean darcy
• [asterisk-users] Speex: complexity, VBR, ABR, CBR, quality   bilal ghayyad
• [asterisk-users] Speex: complexity, VBR, ABR, CBR, quality   Steve Totaro
• [asterisk-users] TDM400P dialout problem   Martin
• [asterisk-users] Newbie on VoIP   NOC Ph
• [asterisk-users] Newbie on VoIP   markgreene at gmail.com
• [asterisk-users] TDM400P dialout problem   sean darcy
• [asterisk-users] TDM400P dialout problem   Martin
• [asterisk-users] [Asterisk-bsd] Newbie on VoIP   NOC Ph
• [asterisk-users] Newbie on VoIP   Vincent
• [asterisk-users] How to get a clean, basic configuration?   Vincent
• [asterisk-users] chan_ss7 0.10   Joel at Gmail
• [asterisk-users] Newbie on VoIP   Tzafrir Cohen
• [asterisk-users] when we try to add CURL code to file channel.c we get an error - undefined reference to curl_easy_init   Prashant Sharma
• [asterisk-users] Newbie on VoIP   NOC Ph
• [asterisk-users] Newbie on VoIP   Erik Anderson
• [asterisk-users] Newbie on VoIP   Tzafrir Cohen
• [asterisk-users] Newbie on VoIP   NOC Ph
• [asterisk-users] when we try to add CURL code to file channel.c we get an error - undefined reference to curl_easy_init   Grey Man
• [asterisk-users] Newbie on VoIP   Hans Witvliet
• [asterisk-users] Newbie on VoIP   Steve Totaro
• [asterisk-users] when we try to add CURL code to file channel.c we get an error - undefined reference to curl_easy_init   Tilghman Lesher
• [asterisk-users] when we try to add CURL code to file channel.c we get an error - undefined reference to curl_easy_init   Prashant Sharma
• [asterisk-users] when we try to add CURL code to file channel.c we get an error - undefined reference to curl_easy_init   Prashant Sharma
• [asterisk-users] SPA3102 registration problem   Jaap Winius
• [asterisk-users] when we try to add CURL code to file channel.c we get an error - undefined reference to curl_easy_init   Tilghman Lesher
• [asterisk-users] Page app, Polycom IP 601, 60 SIP peers, Interesting Issue WORKING NOW   Bill Andersen
• [asterisk-users] when we try to add CURL code to file channel.c we get an error - undefined reference to curl_easy_init   Tzafrir Cohen
• [asterisk-users] DID number   Corey Potts
• [asterisk-users] problem transferring calls some of the times   Raúl Gómez C.
• [asterisk-users] Polycom VSX 7000e Series & Asterisk   Ken Williams
• [asterisk-users] problem transferring calls some of the times   Raúl Gómez C.
• [asterisk-users] DID number   Mojo with Horan & Company, LLC
• [asterisk-users] T1, Rhino, & Nortel   Gleim, Jason
• [asterisk-users] Aastra phones and park/pickup feature   OCG Technical Support
• [asterisk-users] Aastra phones and park/pickup feature   Darren Wright
• [asterisk-users] DID number   Andrew Ladanowski
• [asterisk-users] OT - Mime-construct, exim4 and job numbers   Olivier
• [asterisk-users] Switchvox feedback   C F
• [asterisk-users] Switchvox feedback   Steve Totaro
• [asterisk-users] ekiga sip registration fails; externip no help   sean darcy
• [asterisk-users] T1, Rhino, & Nortel   Steve Totaro
• [asterisk-users] DID number   Andrew Ladanowski
• [asterisk-users] Polycom IP600 + PC share same switch port withVLAN   Lee, John (Sydney)
• [asterisk-users] Had it with Dell Garbage   Ex Vito
• [asterisk-users] Had it with Dell Garbage   Darren Wright
• [asterisk-users] Had it with Dell Garbage   Tilghman Lesher
• [asterisk-users] when we try to add CURL code to file channel.c we get an error - undefined reference to curl_easy_init   Prashant Sharma
• [asterisk-users] problem transferring calls some of the times   Ian
• [asterisk-users] Had it with Dell Garbage   Tzafrir Cohen
• [asterisk-users] Aastra Park Softkey   Russell Brown
• [asterisk-users] [Fwd: OT - CEBIT next week!] - updated list   zoa
• [asterisk-users] TDM400P dialout problem   Martin
• [asterisk-users] when we try to add CURL code to file channel.c we get an error - undefined reference to curl_easy_init   Bruce Reeves
• [asterisk-users] Problems with removing zaptel   Martin
• [asterisk-users] Switchvox feedback   Bruce Reeves
• [asterisk-users] problem transferring calls some of the times   Raúl Gómez C.
• [asterisk-users] incoming call popup   marek cervenka
• [asterisk-users] astmanproxy and core dump   Julian Lyndon-Smith
• [asterisk-users] astmanproxy and core dump   Tzafrir Cohen
• [asterisk-users] Clustering Meetme over multiple boxes?   Tony Mountifield
• [asterisk-users] Coppercom and Asterisk   Mike Hammett
• [asterisk-users] Clustering Meetme over multiple boxes?   Matt Florell
• [asterisk-users] Page() command   Jerry Geis
• [asterisk-users] Coppercom and Asterisk   Tilghman Lesher
• [asterisk-users] TDM400P dialout problem   Shaun Ruffell
• [asterisk-users] Clustering Meetme over multiple boxes?   Tony Mountifield
• [asterisk-users] PPP dialout via * server   Greg Woods
• [asterisk-users] Clustering Meetme over multiple boxes?   Matt Florell
• [asterisk-users] PPP dialout via * server   Kevin P. Fleming
• [asterisk-users] Cisco 7960 SIP Upgrade   Mike Hammett
• [asterisk-users] Coppercom and Asterisk   Mike Hammett
• [asterisk-users] Cisco 7960 SIP Upgrade   Sigma Networks
• [asterisk-users] PPP dialout via * server   Adam Moffett
• [asterisk-users] Coppercom and Asterisk   Shane Burrell
• [asterisk-users] Problems configuring Astribank   Andres Jimenez
• [asterisk-users] Cisco 7960 SIP Upgrade   Mike Hammett
• [asterisk-users] Asterisk and Avaya...   Carlos Chavez
• [asterisk-users] Had it with Dell Garbage   Ex Vito
• [asterisk-users] PPP dialout via * server   Greg Woods
• [asterisk-users] Problems configuring Astribank   Andres Jimenez
• [asterisk-users] Asterisk and Avaya...   Doug Lytle
• [asterisk-users] Asterisk and Avaya...   Doug Lytle
• [asterisk-users] speaker volume on Polycom SIP phones   Peter Hessler
• [asterisk-users] Mitel SX-200 + *   Mark Best
• [asterisk-users] Problems configuring Astribank   Tzafrir Cohen
• [asterisk-users] Mitel SX-200 + *   Eric Wieling
• [asterisk-users] console dsp   Jerry Geis
• [asterisk-users] Mitel SX-200 + *   Mark Best
• [asterisk-users] Mitel SX-200 + *   Jared Smith
• [asterisk-users] Mitel SX-200 + *   Jared Smith
• [asterisk-users] Mitel SX-200 + *   Mark Best
• [asterisk-users] speaker volume on Polycom SIP phones   Eric Wieling
• [asterisk-users] missing ${DIALSTATUS} in hangup extension?   Brooks Bridges
• [asterisk-users] Mitel SX-200 + *   Eric Wieling
• [asterisk-users] T1, Rhino, & Nortel   Gleim, Jason
• [asterisk-users] console dsp   Jerry Geis
• [asterisk-users] ekiga sip registration fails; externip no help   sean darcy
• [asterisk-users] Asterisk and Cisco Unity?   Facundo Ameal
• [asterisk-users] Newbie dialplan: dial 0 for outside line   Lee, John (Sydney)
• [asterisk-users] Newbie dialplan: dial 0 for outside line   Bruce Reeves
• [asterisk-users] problem transferring calls some of the times   Ian
• [asterisk-users] Newbie dialplan: dial 0 for outside line   Lee, John (Sydney)
• [asterisk-users] Asterisk as useragent registered using 2 accounts   Rizwan Hisham
• [asterisk-users] {s} - extension   Daniel Suleyman
• [asterisk-users] {s} - extension   Andres Jimenez
• [asterisk-users] {s} - extension   Tzafrir Cohen
• [asterisk-users] {s} - extension   Daniel Suleyman
• [asterisk-users] Digium TDMXXB and Electronic Noises   Fons van der Beek
• [asterisk-users] {s} - extension   Tzafrir Cohen
• [asterisk-users] {s} - extension   Andres Jimenez
• [asterisk-users] Linksys SPA devices and CID   Jaap Winius
• [asterisk-users] Passing variables between two DUNDi/IAX2 peers   Vieri
• [asterisk-users] {s} - extension   Tzafrir Cohen
• [asterisk-users] Linksys SPA devices and CID   Tim Johnson
• [asterisk-users] incoming call popup   Rajkumar S
• [asterisk-users] How to restrict a Polycom from receiving unauthorized calls   Yehavi Bourvine +972-8-9489444
• [asterisk-users] {s} - extension   Andres Jimenez
• [asterisk-users] Passing variables between two DUNDi/IAX2 peers   Vieri
• [asterisk-users] no audio between two asterisk servers   Jerry Geis
• [asterisk-users] Passing variables between two DUNDi/IAX2 peers   Richard Lyman
• [asterisk-users] Digium TDMXXB and Electronic Noises   Jared Smith
• [asterisk-users] SIP REFER Message, over NAT   Marc Fargas
• [asterisk-users] fonality new version   Dean Collins
• [asterisk-users] DNS Changes never picked up with Asterisk 1.4.18 chan_sip?   Joe Schmid
• [asterisk-users] incoming call popup   Scott Wolfe
• [asterisk-users] Linksys SPA devices and CID   Jaap Winius
• [asterisk-users] Linksys SPA devices and CID   Tim Johnson
• [asterisk-users] NIN Ghosts music (free download) safe for MOH?   Justin Newman
• [asterisk-users] Aastra Park Softkey   shadowym
• [asterisk-users] NIN Ghosts music (free download) safe for MOH?   Andrew Latham
• [asterisk-users] problem transferring calls some of the times   Raúl Gómez C.
• [asterisk-users] Transferring Unanswered Calls   Raúl Gómez C.
• [asterisk-users] C compiler cannot create executables when building zaptel   CSB
• [asterisk-users] Passing variables between two DUNDi/IAX2 peers   Vieri
• [asterisk-users] NIN Ghosts music (free download) safe for MOH?   Tzafrir Cohen
• [asterisk-users] NIN Ghosts music (free download) safe for MOH?   Joshua Kinard
• [asterisk-users] Problem between Asterisk and an Aastra 57i   Stefan Guenther
• [asterisk-users] codec_g729-v34 Builds Now Available   The Asterisk Development Team
• [asterisk-users] C compiler cannot create executables when building zaptel   Stefan Guenther
• [asterisk-users] Aastra Park Softkey   Ira
• [asterisk-users] Problem between Asterisk and an Aastra 57i   Gareth Owen
• [asterisk-users] g729 to GSM translator is needed for voicemail to work fine, how?   bilal ghayyad
• [asterisk-users] Codec Preferences   bilal ghayyad
• [asterisk-users] codec_g729-v34 Builds Now Available   Bruce McAlister
• [asterisk-users] g729 to GSM translator is needed for voicemail to work fine, how?   Tilghman Lesher
• [asterisk-users] Linksys SPA devices and CID   Jaap Winius
• [asterisk-users] Problem between Asterisk and an Aastra 57i   Gleim, Jason
• [asterisk-users] C compiler cannot create executables when building zaptel   Michiel van Baak
• [asterisk-users] C compiler cannot create executables when building zaptel   Tzafrir Cohen
• [asterisk-users] C compiler cannot create executables when building zaptel   Tzafrir Cohen
• [asterisk-users] OT How to Change Polycom Web Admin User:Pass via Web   Steve Totaro
• [asterisk-users] Queue calls drop to voicemail intermittantly   Chris Earle
• [asterisk-users] Aastra-Asterisk: 6 beeps then voice quality degrades   OCG Technical Support
• [asterisk-users] Asterisk in the call center - how do you do it?   Ade Vickers
• [asterisk-users] Asterisk 1.4.19-rc1 Now Available   The Asterisk Development Team
• [asterisk-users] Asterisk 1.6.0-beta5 Now Available   The Asterisk Development Team
• [asterisk-users] Asterisk in the call center - how do you do it?   Kev S
• [asterisk-users] OT How to Change Polycom Web Admin User:Pass via Web   Noah Miller
• [asterisk-users] Codec Preferences   Noah Miller
• [asterisk-users] How to restrict a Polycom from receiving unauthorized calls   Kevin P. Fleming
• [asterisk-users] Asterisk in the call center - how do you do it?   Matt Florell
• [asterisk-users] Voice quality is bad from one side and good from another side   bilal ghayyad
• [asterisk-users] Asterisk based UNIX   bilal ghayyad
• [asterisk-users] Asterisk based UNIX   Alex Balashov
• [asterisk-users] Voice quality is bad from one side and good from another side   Alex Balashov
• [asterisk-users] Asterisk based UNIX   Bill Andersen
• [asterisk-users] Asterisk based UNIX   Alex Balashov
• [asterisk-users] Asterisk based UNIX   Hans Witvliet
• [asterisk-users] Asterisk based UNIX   Tzafrir Cohen
• [asterisk-users] Voice quality is bad from one side and good from another side   Steve Totaro
• [asterisk-users] Voice quality is bad from one side and good from another side   Steve Totaro
• [asterisk-users] Asterisk in the call center - how do you do it?   Paul Hales
• [asterisk-users] Asterisk in the call center - how do you do it?   Matt Florell
• [asterisk-users] Call recording problems from queue   Ex Vito
• [asterisk-users] OT How to Change Polycom Web Admin User:Pass via Web   Steve Totaro
• [asterisk-users] Voice quality is bad from one side and good from another side   Alex Balashov
• [asterisk-users] Transferring Unanswered Calls   Ex Vito
• [asterisk-users] Asterisk in the call center - how do you do it?   Paul Hales
• [asterisk-users] Transferring Unanswered Calls   Steve Totaro
• [asterisk-users] Newbie Polycom: how to effect change in sip.cfg?   Lee, John (Sydney)
• [asterisk-users] LDAP   Gonzalo Servat
• [asterisk-users] Asterisk in the call center - how do you do it?   Rob Hillis
• [asterisk-users] Passing variables between two DUNDi/IAX2 peers   Craig Guy
• [asterisk-users] Asterisk in the call center - how do you do it?   Matt Florell
• [asterisk-users] OT How to Change Polycom Web Admin User:Pass via Web   Yehavi Bourvine +972-8-9489444
• [asterisk-users] OT How to Change Polycom Web Admin User:Pass via Web   Lee, John (Sydney)
• [asterisk-users] Newbie Polycom: how to effect change in sip.cfg?   Rob Hillis
• [asterisk-users] Asterisk in the call center - how do you do it?   Rob Hillis
• [asterisk-users] Asterisk in the call center - how do you do it?   Matt Florell
• [asterisk-users] Newbie Polycom: how to effect change insip.cfg?   Lee, John (Sydney)
• [asterisk-users] Asterisk based UNIX   Carole Migden
• [asterisk-users] Polycom IP600 + PC share same switch port withVLAN   James Sneeringer
• [asterisk-users] Polycom IP600 + PC share same switch portwithVLAN   Lee, John (Sydney)
• [asterisk-users] Newbie Polycom: how to effect change insip.cfg?   Rob Hillis
• [asterisk-users] Asterisk in the call center - how do you do it?   Rob Hillis
• [asterisk-users] LDAP   Gonzalo Servat
• [asterisk-users] Newbie Polycom: how to effect change insip.cfg?   Lee, John (Sydney)
• [asterisk-users] problem transferring calls some of the times   Ian
• [asterisk-users] Message sequence of a conference   preethy varghese
• [asterisk-users] Asterisk based UNIX   Carole Migden
• [asterisk-users] Problem between Asterisk and an Aastra 57i   Stefan Guenther
• [asterisk-users] Problem between Asterisk and an Aastra 57i   Stefan Guenther
• [asterisk-users] RFC for conference   preethy varghese
• [asterisk-users] Asterisk based UNIX   randulo
• [asterisk-users] Newbie Polycom: IP600 Headset Problem   Lee, John (Sydney)
• [asterisk-users] LDAP   Faraz Khan
• [asterisk-users] Call recording problems from queue   Scott Gifford
• [asterisk-users] [Invitation] VoIP Users Conference @ Fri Mar 7 12:00 - 13:00 ()   Randy R
• [asterisk-users] VoIP Users Conference for Friday March 7th @ 12 Noon EST   randulo
• [asterisk-users] Fax in AGX Extra Addons for Asterisk causes Asterisk to die   A.R. Nasir Qureshi
• [asterisk-users] Accepted: VoIP Users Conference   Steven C. Blair
• [asterisk-users] Accepted: [Invitation] VoIP Users Conference @ Fri Mar 712:00 - 13:00 ()   Morgan Gilroy
• [asterisk-users] Asterisk based UNIX   Hans Witvliet
• [asterisk-users] New Time Proposed: [Invitation] VoIP Users Conference @ Fri Mar 712:00 - 13:00 ()   Giles Coochey
• [asterisk-users] Asterisk based UNIX   randulo
• [asterisk-users] Declined: VoIP Users Conference   Jacobson
• [asterisk-users] Tentative: VoIP Users Conference   Jacobson
• [asterisk-users] {s} - extension   Daniel Suleyman
• [asterisk-users] Asterisk authentication by SIP Proxy   Mayur
• [asterisk-users] Newbie Polycom: IP600 Headset Problem   Doug Lytle
• [asterisk-users] VoIP Users Conference for Friday March 7th @ 12 Noon EST   Tzafrir Cohen
• [asterisk-users] codec_g729-v34 Builds Now Available   Bruce McAlister
• [asterisk-users] VoIP Users Conference for Friday March 7th @ 12 Noon EST   randulo
• [asterisk-users] AEL - SQL and TIMEDIFF()   Tobias Ahlander
• [asterisk-users] Tentative: [Invitation] VoIP Users Conference @ Fri Mar 712:00 - 13:00 ()   Kenneth T. Van Wie II
• [asterisk-users] AEL - SQL and TIMEDIFF()   Andreas Sikkema
• [asterisk-users] Declined: VoIP Users Conference   Evan Ruff
• [asterisk-users] chan_iax2.c:3904 iax2_trunk_queue: Maximum trunk data space exceeded   Brooks Bridges
• [asterisk-users] VoIP Users Conference for Friday March 7th @ 12 Noon EST   OCG Technical Support
• [asterisk-users] IAX user identification.   Thomas Kenyon
• [asterisk-users] Declined: VoIP Users Conference   Doug Lytle
• [asterisk-users] bristuff qozap support for beronet cards   stoffell
• [asterisk-users] VoIP Users Conference for Friday March 7th @ 12 Noon EST   Tzafrir Cohen
• [asterisk-users] Accepted: [Invitation] VoIP Users Conference @ Fri Mar 712:00 - 13:00 ()   Jeff Johnson
• [asterisk-users] {s} - extension   Noah Miller
• [asterisk-users] Declined: VoIP Users Conference   Darrick Hartman (lists)
• [asterisk-users] Declined: VoIP Users Conference   Dean Collins
• [asterisk-users] FXS channel banks   Chris Bagnall
• [asterisk-users] Declined: VoIP Users Conference   Doug Lytle
• [asterisk-users] FXS channel banks   Drew Gibson
• [asterisk-users] Asterisk based UNIX   Bill Andersen
• [asterisk-users] FXS channel banks   Jay R. Ashworth
• [asterisk-users] Asterisk based UNIX   Tzafrir Cohen
• [asterisk-users] FXS channel banks   Jay R. Ashworth
• [asterisk-users] Best Java library to interact with Asterisk   equis software
• [asterisk-users] Declined: VoIP Users Conference   www.IPKall.com
• [asterisk-users] Declined: VoIP Users Conference   www.IPKall.com
• [asterisk-users] format of UNIQUEID variable   Vieri
• [asterisk-users] Asterisk in the call center - how do you do it?   Norman Franke
• [asterisk-users] FXS channel banks   Steve Totaro
• [asterisk-users] format of UNIQUEID variable   Tilghman Lesher
• [asterisk-users] Call Manager as trunk   Aaron Fransen
• [asterisk-users] format of UNIQUEID variable   Tony Mountifield
• [asterisk-users] C compiler cannot create executables when building zaptel   CSB
• [asterisk-users] Declined: VoIP Users Conference   Mark Adams
• [asterisk-users] Passing variables between two DUNDi/IAX2 peers   Vieri
• [asterisk-users] Provider recommendation in USA   Vivek Shrivastava
• [asterisk-users] Allowguest=yes & language   Guilherme Loch Waltrick Góes
• [asterisk-users] Provider recommendation in USA   Jai Rangi
• [asterisk-users] C compiler cannot create executables when building zaptel   Tzafrir Cohen
• [asterisk-users] OT How to Change Polycom Web Admin User:Pass via Web   Yehavi Bourvine +972-8-9489444
• [asterisk-users] Best Java library to interact with Asterisk   Guilherme Loch Waltrick Góes
• [asterisk-users] C compiler cannot create executables when building zaptel   CSB
• [asterisk-users] zaptel compile question   Jerry Geis
• [asterisk-users] Net Neutrality   Dean Collins
• [asterisk-users] zaptel compile question   Shaun Ruffell
• [asterisk-users] Asterisk 1.4 w/ realtime static zapata   Edwin Lam
• [asterisk-users] DTMFR2- UNICALL   Jessica Gonzalez Arriagada
• [asterisk-users] zaptel compile question   Tzafrir Cohen
• [asterisk-users] en25.com   Philipp Kempgen
• [asterisk-users] format of UNIQUEID variable   Vieri
• [asterisk-users] Asterisk in the call center - how do you do it?   Jay R. Ashworth
• [asterisk-users] Cool New Website   Goran Donev
• [asterisk-users] Receiving double DTMF "if I pressed 1, then asterisk box recognize it 11"   bilal ghayyad
• [asterisk-users] Call Manager as trunk   Aaron Fransen
• [asterisk-users] Cool New Website   Dean Collins
• [asterisk-users] FXS channel banks   Lee, John (Sydney)
• [asterisk-users] OT: Upgrade Addpac AP200C   Pablo Almido
• [asterisk-users] Cool New Website   Brooks Bridges
• [asterisk-users] OT: Upgrade Addpac AP200C   Pablo Almido
• [asterisk-users] FXS channel banks   James Finstrom
• [asterisk-users] Had it with Dell Garbage   Steven Kurylo
• [asterisk-users] Cool New Website   Brooks Bridges
• [asterisk-users] FXS channel banks   Gordon Henderson
• [asterisk-users] Cool New Website   Michael Graves
• [asterisk-users] Newbie Polycom: IP600 Headset Problem   Paul Hales
• [asterisk-users] FXS channel banks   Tzafrir Cohen
• [asterisk-users] Had it with Dell Garbage   Ron Joffe
• [asterisk-users] DTMFR2- UNICALL   Moises Silva
• [asterisk-users] Had it with Dell Garbage   Paul Hales
• [asterisk-users] is this possible..   blackwater dev
• [asterisk-users] How to return the status of a call to the calling server?   Steve Edwards
• [asterisk-users] Newbie Polycom: IP600 Headset Problem   Lee, John (Sydney)
• [asterisk-users] is this possible..   Richard Lyman
• [asterisk-users] is this possible..   C F
• [asterisk-users] is this possible..   Adam Moffett
• [asterisk-users] is this possible..   Don Kelly
• [asterisk-users] Newbie Polycom: IP600 Headset Problem   Paul Hales
• [asterisk-users] Call flows of conference   preethy varghese
• [asterisk-users] Cool New Website   randulo
• [asterisk-users] Provider recommendation in USA   randulo
• [asterisk-users] Newbie MeetMe: How to control max users in conference?   Lee, John (Sydney)
• [asterisk-users] LDAP   Faraz Khan
• [asterisk-users] Newbie Polycom: IP600 Headset Problem   Rob Hillis
• [asterisk-users] Call flows of conference   Rob Hillis
• [asterisk-users] Silencing VoiceMail() app in * 1.4.10   Horwich IT Services (Godwin Stewart)
• [asterisk-users] Newbie Polycom: IP600 Headset Problem   Louwrens Benadé
• [asterisk-users] Newbie MeetMe: How to control max users in conference?   Tony Mountifield
• [asterisk-users] Silencing VoiceMail() app in * 1.4.10   Mindaugas Kezys
• [asterisk-users] Asterisk Realtime and SIP configuration   Stuart Ford
• [asterisk-users] Call flows of sequence   preethy varghese
• [asterisk-users] LDAP   Gonzalo Servat
• [asterisk-users] chan_sip.c:2918 auto congestion   srinivas Antarvedi
• [asterisk-users] Asterisk Realtime and SIP configuration   Julian Lyndon-Smith
• [asterisk-users] Asterisk Realtime and SIP configuration   Stuart Ford
• [asterisk-users] Silencing VoiceMail() app in * 1.4.10   Horwich IT Services (Godwin Stewart)
• [asterisk-users] Asterisk Realtime and SIP configuration   Stuart Ford
• [asterisk-users] LDAP   Gonzalo Servat
• [asterisk-users] LDAP   Faraz Khan
• [asterisk-users] Background: reading the digits correctly, buffering it, waiting the sound message to complete   bilal ghayyad
• [asterisk-users] Silencing VoiceMail() app in * 1.4.10   Mindaugas Kezys
• [asterisk-users] DTMFR2- UNICALL   Jessica Gonzalez Arriagada
• [asterisk-users] DTMFR2- UNICALL   Moises Silva
• [asterisk-users] WirelessIP5000 SIP registration problem   Jim Meehan
• [asterisk-users] load balancing   CSB
• [asterisk-users] load balancing   CSB
• [asterisk-users] Asterisk Voicemail for iPhone   Chris Carey
• [asterisk-users] How to get call back during attendant transfer?   Gary
• [asterisk-users] VoiceMail dialout context   Tony Plack
• [asterisk-users] sip show channels - gives a growing list of dead channels   Keith Hardee
• [asterisk-users] Motorola SBV5220   markgreene at gmail.com
• [asterisk-users] FXS channel banks   Jay R. Ashworth
• [asterisk-users] Sync Problem (astribank)   Grygoriy Dobrovolskyy
• [asterisk-users] FXS channel banks   Steve Totaro
• [asterisk-users] FXS channel banks   Steve Totaro
• [asterisk-users] FXS channel banks   Felipe Trevisan
• [asterisk-users] WirelessIP5000 SIP registration problem   Martin
• [asterisk-users] FXS channel banks   Tzafrir Cohen
• [asterisk-users] Sync Problem (astribank)   Tzafrir Cohen
• [asterisk-users] FXS channel banks   Steve Totaro
• [asterisk-users] Cisco 7960 SIP Upgrade   Mike Hammett
• [asterisk-users] NIN Ghosts music (free download) safe for MOH?   Tilghman Lesher
• [asterisk-users] sip show channels - gives a growing list of dead channels   Fons van der Beek
• [asterisk-users] Fwd: {s} - extension   Daniel Suleyman
• [asterisk-users] Fwd: {s} - extension   Tzafrir Cohen
• [asterisk-users] FXS channel banks   Tzafrir Cohen
• [asterisk-users] Fwd: {s} - extension   Daniel Suleyman
• [asterisk-users] NIN Ghosts music (free download) safe for MOH?   Tzafrir Cohen
• [asterisk-users] FXS channel banks   Faraz Khan
• [asterisk-users] FXS channel banks   Faraz Khan
• [asterisk-users] Experiences with grandstream GXW 4024 FXS?   Faraz Khan
• [asterisk-users] FXS channel banks   Raúl Gómez C.
• [asterisk-users] Fwd: {s} - extension   Tilghman Lesher
• [asterisk-users] FXS channel banks   Faraz Khan
• [asterisk-users] FXS channel banks   Tzafrir Cohen
• [asterisk-users] Customer complains of noise on line I cannot reproduce.   Royce Souther
• [asterisk-users] Background: reading the digits correctly, buffering it, waiting the sound message to complete   Noah Miller
• [asterisk-users] Customer complains of noise on line I cannot reproduce.   Steve Totaro
• [asterisk-users] FXS channel banks   Dumpolid Exeplish
• [asterisk-users] FXS channel banks   Steve Totaro
• [asterisk-users] replace astdb with a cluster-capable sql database engine   Vieri
• [asterisk-users] I am now on Refriendz!   Karthik Arumugam
• [asterisk-users] I am now on ___________   Tzafrir Cohen
• [asterisk-users] Sync Problem (astribank)   Grygoriy Dobrovolskyy
• [asterisk-users] NIN Ghosts music (free download) safe for MOH?   Jay R. Ashworth
• [asterisk-users] PRI suppliers in Switzerland   Chris Bagnall
• [asterisk-users] Customer complains of noise on line I cannot reproduce.   Royce Souther
• [asterisk-users] replace astdb with a cluster-capable sql database engine   Bruce Reeves
• [asterisk-users] Customer complains of noise on line I cannot reproduce.   Grygoriy Dobrovolskyy
• [asterisk-users] PRI suppliers in Switzerland   Grygoriy Dobrovolskyy
• [asterisk-users] Background: reading the digits correctly, buffering it, waiting the sound message to complete   bilal ghayyad
• [asterisk-users] PRI suppliers in Switzerland   Gordon Henderson
• [asterisk-users] PRI suppliers in Switzerland   Hans Witvliet
• [asterisk-users] Customer complains of noise on line I cannot reproduce.   Fons van der Beek
• [asterisk-users] How Do I continue after Dial Command ??   Jim Duda
• [asterisk-users] Customer complains of noise on line I cannot reproduce.   Grygoriy Dobrovolskyy
• [asterisk-users] PRI suppliers in Switzerland   Grygoriy Dobrovolskyy
• [asterisk-users] How Do I continue after Dial Command ??   Grygoriy Dobrovolskyy
• [asterisk-users] Customer complains of noise on line I cannot reproduce.   Royce Souther
• [asterisk-users] Customer complains of noise on line I cannot reproduce.   Royce Souther
• [asterisk-users] Fwd: {s} - extension   Daniel Suleyman
• [asterisk-users] Read function   Daniel Suleyman
• [asterisk-users] Customer complains of noise on line I cannot reproduce.   Fons van der Beek
• [asterisk-users] replace astdb with a cluster-capable sql database engine   Vieri
• [asterisk-users] Customer complains of noise on line I cannot reproduce.   Grygoriy Dobrovolskyy
• [asterisk-users] Customer complains of noise on line I cannot reproduce.   Grygoriy Dobrovolskyy
• [asterisk-users] PRI suppliers in Switzerland   Chris Bagnall
• [asterisk-users] Customer complains of noise on line I cannot reproduce.   Fons van der Beek
• [asterisk-users] PRI suppliers in Switzerland   Grygoriy Dobrovolskyy
• [asterisk-users] PRI suppliers in Switzerland   Administrator TOOTAI
• [asterisk-users] replace astdb with a cluster-capable sql database engine   Atis Lezdins
• [asterisk-users] replace astdb with a cluster-capable sql database engine   Tzafrir Cohen
• [asterisk-users] Silencing VoiceMail() app in * 1.4.10   Horwich IT Services (Godwin Stewart)
• [asterisk-users] Read function   Doug Lytle
• [asterisk-users] Read function   Niles Ingalls
• [asterisk-users] Read function   Daniel Suleyman
• [asterisk-users] Read function   Daniel Suleyman
• [asterisk-users] Silencing VoiceMail() app in * 1.4.10   Atis Lezdins
• [asterisk-users] Read function   Doug Lytle
• [asterisk-users] Read function   Daniel Suleyman
• [asterisk-users] Realtime error - PgSQL with 1.6 beta 5   Christian Svensson
• [asterisk-users] Read function   Doug Lytle
• [asterisk-users] Fwd: {s} - extension   Tilghman Lesher
• [asterisk-users] Fwd: {s} - extension   Tzafrir Cohen
• [asterisk-users] Read function   Daniel Suleyman
• [asterisk-users] replace astdb with a cluster-capable sql database engine (was: Re: asterisk-users Digest, Vol 44, Issue 22)   Matthew Rubenstein
• [asterisk-users] Fwd: {s} - extension   Daniel Suleyman
• [asterisk-users] Experiences with grandstream GXW 4024 FXS?   Darren Nickerson
• [asterisk-users] replace astdb with a cluster-capable sql database engine   Vieri
• [asterisk-users] replace astdb with a cluster-capable sql database engine   Tilghman Lesher
• [asterisk-users] Experiences with grandstream GXW 4024 FXS?   Faraz Khan
• [asterisk-users] FXS channel banks   Faraz Khan
• [asterisk-users] phones start ringing randomly with Grandstream GXW-40XX - solution!   Faraz Khan
• [asterisk-users] Silencing VoiceMail() app in * 1.4.10   Horwich IT Services (Godwin Stewart)
• [asterisk-users] FXS channel banks   Jay R. Ashworth
• [asterisk-users] replace astdb with a cluster-capable sql database engine   Rob Hillis
• [asterisk-users] Read function   Doug Lytle
• [asterisk-users] How Do I continue after Dial Command ??   Vincent
• [asterisk-users] Read function   Daniel Suleyman
• [asterisk-users] Silencing VoiceMail() app in * 1.4.10   Atis Lezdins
• [asterisk-users] replace astdb with a cluster-capable sql database engine   Vieri
• [asterisk-users] How Do I continue after Dial Command ??   Jim Duda
• [asterisk-users] Silencing VoiceMail() app in * 1.4.10   Horwich IT Services (Godwin Stewart)
• [asterisk-users] Read function   Doug Lytle
• [asterisk-users] How Do I continue after Dial Command ??   Jim Duda
• [asterisk-users] Dead Air on PF firewall   NOC ph
• [asterisk-users] Local music on hold -- mohinterpret=passthrough assymetrical ?   Ex Vito
• [asterisk-users] What replaces Macro() now? And how do you do the equivalent?   Philip Prindeville
• [asterisk-users] Microsoft Office Communications Server   Matt Riddell
• [asterisk-users] Had it with Dell Garbage   Matt Riddell
• [asterisk-users] best practice   Matt Riddell
• [asterisk-users] C compiler cannot create executables when building zaptel   Matt Riddell
• [asterisk-users] Fwd: {s} - extension   Noah Miller
• [asterisk-users] 1.6.beta5 (format 0x40 (slin))   marc+ast at milestonerdl.com
• [asterisk-users] How Do I continue after Dial Command ??   Vincent
• [asterisk-users] NIN Ghosts music (free download) safe for MOH?   John Faubion
• [asterisk-users] NIN Ghosts music (free download) safe for MOH?   Dave Cotton
• [asterisk-users] replace astdb with a cluster-capable sql database engine   Vieri
• [asterisk-users] Read function   Daniel Suleyman
• [asterisk-users] Fwd: {s} - extension   Daniel Suleyman
• [asterisk-users] NIN Ghosts music (free download) safe for MOH?   KodaK
• [asterisk-users] NIN Ghosts music (free download) safe for MOH?   Horwich IT Services (Godwin Stewart)
• [asterisk-users] NIN Ghosts music (free download) safe for MOH?   Anselm Martin Hoffmeister
• [asterisk-users] NIN Ghosts music (free download) safe for MOH?   randulo
• [asterisk-users] dialplan logic questions: macro not reaching DIALSTATUS based extension (s-ANSWER)   Michael Iedema
• [asterisk-users] dialplan logic questions: macro not reaching DIALSTATUS based extension (s-ANSWER)   Michael Iedema
• [asterisk-users] 1.6.beta5 (format 0x40 (slin))   Matt Riddell
• [asterisk-users] dialplan logic questions: macro not reaching DIALSTATUS based extension (s-ANSWER)   Matt Riddell
• [asterisk-users] Call forwarding-in india   sandeep
• [asterisk-users] call forward facility in INDIA   sandeep
• [asterisk-users] dialstatus and cancelled calls   Vieri
• [asterisk-users] Dead Air on PF firewall   Horwich IT Services (Godwin Stewart)
• [asterisk-users] Microsoft Office Communications Server   David Cook
• [asterisk-users] Dead Air on PF firewall   Michiel van Baak
• [asterisk-users] FaxBack Service with Asterisk   Dovid B
• [asterisk-users] Global Variables on Reload   Rob Schall
• [asterisk-users] Call forwarding-in india   Grygoriy Dobrovolskyy
• [asterisk-users] Microsoft Office Communications Server   Andrew Latham
• [asterisk-users] Had it with Dell Garbage   Grygoriy Dobrovolskyy
• [asterisk-users] Call forwarding-in india   Horwich IT Services (Godwin Stewart)
• [asterisk-users] Shared Extension   Tony Plack
• [asterisk-users] Global Variables on Reload   Tzafrir Cohen
• [asterisk-users] NIN Ghosts music (free download) safe for MOH?   John Novack
• [asterisk-users] Intermittent DTMF Problems   Brent Davidson
• [asterisk-users] Read function   Daniel Suleyman
• [asterisk-users] Queue Pickup   Rob Schall
• [asterisk-users] NIN Ghosts music (free download) safe for MOH?   Tzafrir Cohen
• [asterisk-users] Asterisk and Packetcable   Carlos Alberto Bernat Orozco
• [asterisk-users] Redirecting channels?   harry
• [asterisk-users] Asterisk and Packetcable   Mark Greene
• [asterisk-users] NIN Ghosts music (free download) safe for MOH?   Joshua Kinard
• [asterisk-users] Asterisk 1.6.0-beta5 Now Available   Michael Cargile
• [asterisk-users] NIN Ghosts music (free download) safe for MOH?   Brig C. McCoy
• [asterisk-users] Strange problem   Accursio Avona
• [asterisk-users] Global Variables on Reload   Edwin Lam
• [asterisk-users] display time on Cisco 79xx   Don Smith
• [asterisk-users] Asterisk and Packetcable   Consuelo Vega
• [asterisk-users] Queue calls drop to voicemail intermittantly   Chris Earle
• [asterisk-users] display time on Cisco 79xx   Michiel van Baak
• [asterisk-users] display time on Cisco 79xx   Chris Carey
• [asterisk-users] Intermittent DTMF Problems   bilal ghayyad
• [asterisk-users] display time on Cisco 79xx   Eric Wieling
• [asterisk-users] display time on Cisco 79xx   Doug Lytle
• [asterisk-users] Audiocodes MP124-FXS replying BUSY when line is not.   Atis Lezdins
• [asterisk-users] Intermittent DTMF Problems   Eric Wieling
• [asterisk-users] Queue Pickup   Justin Newman
• [asterisk-users] Read function   Daniel Suleyman
• [asterisk-users] display time on Cisco 79xx   Mark Johnson
• [asterisk-users] Want to know Frequency and lenght of Frame   sanjay.rajdev at featherstoneinformatics.com
• [asterisk-users] Disable SIP notify for peers   Adrian A
• [asterisk-users] sip show channels - gives a growing list of dead channels   Keith Hardee
• [asterisk-users] sip show channels - gives a growing list of dead channels   Raj Jain
• [asterisk-users] Global Variables on Reload   Grygoriy Dobrovolskyy
• [asterisk-users] Redirecting channels?   Grygoriy Dobrovolskyy
• [asterisk-users] Microsoft Office Communications Server   Kristian Kielhofner
• [asterisk-users] Intermittent DTMF Problems   Brent Davidson
• [asterisk-users] Intermittent DTMF Problems   Brent Davidson
• [asterisk-users] About CID with DTMF in Asterisk   José David Bravo Álvarez
• [asterisk-users] Microsoft Office Communications Server   mgraves at mstvp.com
• [asterisk-users] display time on Cisco 79xx   Don Smith
• [asterisk-users] Microsoft Office Communications Server   shadowym
• [asterisk-users] Shared Extension   Raj Jain
• [asterisk-users] Microsoft Office Communications Server   Michiel van Baak
• [asterisk-users] Microsoft Office Communications Server   Kristian Kielhofner
• [asterisk-users] FaxBack Service with Asterisk   Paul Hales
• [asterisk-users] Shared Extension   Tony Plack
• [asterisk-users] Intermittent DTMF Problems   Eric Wieling
• [asterisk-users] Microsoft Office Communications Server   Philipp von Klitzing
• [asterisk-users] replace astdb with a cluster-capable sql database engine   Philipp von Klitzing
• [asterisk-users] need * consultant in houston area   A_ Navone
• [asterisk-users] Microsoft Office Communications Server   Senad Jordanovic
• [asterisk-users] need * consultant in houston area   Bruce Reeves
• [asterisk-users] need * consultant in houston area   Ravichandran Rajagopal
• [asterisk-users] Little help with Conference   Ruben Zamora
• [asterisk-users] Newbie MeetMe: How to control max users in conference?   Lee, John (Sydney)
• [asterisk-users] need * consultant in houston area   Brian Capouch
• [asterisk-users] Newbie Polycom: IP601 console with expansion module   Lee, John (Sydney)
• [asterisk-users] codec_g729-v34 Builds Now Available   Bruce McAlister
• [asterisk-users] Newbie Polycom: IP601 console with expansion module   Rob Hillis
• [asterisk-users] Sending SMS   voip crazy
• [asterisk-users] WARNING[4294]: chan_zap.c:8851 zt_pri_error: 1 copying 5 bytes LLC   Ben Thompson
• [asterisk-users] Dead Air on PF firewall   NOC ph
• [asterisk-users] Dead Air on PF firewall   Michiel van Baak
• [asterisk-users] Central Asterisk with remote 'trunking' asterisk gateways   Faraz Khan
• [asterisk-users] dialstatus and cancelled calls   Vieri
• [asterisk-users] WARNING[4294]: chan_zap.c:8851 zt_pri_error: 1 copying 5 bytes LLC   Ben Thompson
• [asterisk-users] Dead Air on PF firewall   NOC ph
• [asterisk-users] CCM 6 and Asterisk routing again   Aaron Fransen
• [asterisk-users] E1 Card emulator?   Mark Welch
• [asterisk-users] Dead Air on PF firewall   Michiel van Baak
• [asterisk-users] CCM 6 and Asterisk routing again   Patrick
• [asterisk-users] Microsoft Office Communications Server   Razza
• [asterisk-users] CCM 6 and Asterisk routing again   Peter Pauly
• [asterisk-users] Meetme application on AstriskWin32   preethy varghese
• [asterisk-users] Microsoft Office Communications Server   Senad Jordanovic
• [asterisk-users] CCM 6 and Asterisk routing again   Aaron Fransen
• [asterisk-users] CCM 6 and Asterisk routing again   Aaron Fransen
• [asterisk-users] E1 Card emulator?   gincantalupo
• [asterisk-users] Microsoft Office Communications Server   Tzafrir Cohen
• [asterisk-users] E1 Card emulator?   Tzafrir Cohen
• [asterisk-users] Shared Extension   Raj Jain
• [asterisk-users] Call tracing - Asterisk 1.4   Louwrens Benadé
• [asterisk-users] Unison   Dean Collins
• [asterisk-users] Unison   Tzafrir Cohen
• [asterisk-users] E1 Card emulator?   Steve Totaro
• [asterisk-users] Microsoft Office Communications Server   Raj Jain
• [asterisk-users] CCM 6 and Asterisk routing again   Aaron Fransen
• [asterisk-users] Microsoft Office Communications Server   Milton Calnek
• [asterisk-users] Microsoft Office Communications Server   Senad Jordanovic
• [asterisk-users] About CID with DTMF in Asterisk   José David Bravo Álvarez
• [asterisk-users] Microsoft Office Communications Server   Tzafrir Cohen
• [asterisk-users] Unison   Ron Wellsted
• [asterisk-users] Microsoft Office Communications Server   Razza
• [asterisk-users] Unison   James Texter III
• [asterisk-users] Microsoft Office Communications Server   Senad Jordanovic
• [asterisk-users] Microsoft Office Communications Server   Razza
• [asterisk-users] Meetme application on AstriskWin32   Tony Mountifield
• [asterisk-users] Gartner Article   Dean Collins
• [asterisk-users] Unison   Terry Wilson
• [asterisk-users] About CID with DTMF in Asterisk   José David Bravo Álvarez
• [asterisk-users] Call tracing - Asterisk 1.4   Atis Lezdins
• [asterisk-users] dialstatus and cancelled calls   Ex Vito
• [asterisk-users] Microsoft Office Communications Server   Joe Pukepail
• [asterisk-users] T.38 passthrough in asterisk 1.4   Ricardo Carvalho
• [asterisk-users] Call tracing - Asterisk 1.4   David Van Ginneken
• [asterisk-users] setting callerid across servers   Jerry Geis
• [asterisk-users] Microsoft Office Communications Server   shadowym
• [asterisk-users] setting callerid across servers   Niles Ingalls
• [asterisk-users] Best alternative for getting prompts recorded.   sanjay.rajdev at featherstoneinformatics.com
• [asterisk-users] setting callerid across servers   Peter Eisch
• [asterisk-users] Best alternative for getting prompts recorded.   Doug
• [asterisk-users] Polycom IP 330 w/VLAN?   Ken D'Ambrosio
• [asterisk-users] Microsoft Office Communications Server   Hans Witvliet
• [asterisk-users] Best alternative for getting prompts recorded.   randulo
• [asterisk-users] Best alternative for getting prompts recorded.   Ron Joffe
• [asterisk-users] Polycom IP 330 w/VLAN?   Jay R. Ashworth
• [asterisk-users] Best alternative for getting prompts recorded.   Steve Edwards
• [asterisk-users] Asterisk 1.4.19-rc2 Now Available   The Asterisk Development Team
• [asterisk-users] Best alternative for getting prompts recorded.   sanjay.rajdev at featherstoneinformatics.com
• [asterisk-users] Best alternative for getting prompts recorded.   Tilghman Lesher
• [asterisk-users] Best alternative for getting prompts recorded.   Steve Prior
• [asterisk-users] Gartner Article (was: Re: asterisk-users Digest, Vol 44, Issue 32)   Matthew Rubenstein
• [asterisk-users] Best alternative for getting prompts recorded.   Steve Kennedy
• [asterisk-users] Sending Incoming Calls straight to park   Mark
• [asterisk-users] Best alternative for getting prompts recorded.   sanjay.rajdev at featherstoneinformatics.com
• [asterisk-users] Sending Incoming Calls straight to park   Paul Hales
• [asterisk-users] AGI - calling functions, CHANNEL STATUS broken?   Kevin DeGraaf
• [asterisk-users] Asterisk not transcoding between installed codecs   Gonzalo Servat
• [asterisk-users] Microsoft Office Communications Server   Matt Riddell
• [asterisk-users] Microsoft Office Communications Server   Matt Riddell
• [asterisk-users] Microsoft Office Communications Server   Matt Riddell
• [asterisk-users] Sending SMS   monim benayad
• [asterisk-users] TXFax/RXFax/AGX-Addons/SpanDSP Crashing   Matt Riddell
• [asterisk-users] AGI - calling functions, CHANNEL STATUS broken?   Matt Riddell
• [asterisk-users] Sending SMS   troxlinux
• [asterisk-users] MySQL Voicemail Storage Questions\Errors   Mike Hammett
• [asterisk-users] chanspy doesn't work properly   Bhrugu Mehta
• [asterisk-users] MySQL Voicemail Storage Questions\Errors   Tilghman Lesher
• [asterisk-users] S100I IAXy - Why is it discontinued?   Joseph
• [asterisk-users] S100I IAXy - Why is it discontinued?   Tilghman Lesher
• [asterisk-users] authentication number at the end of the number before calls go through.   Mark Quitoriano
• [asterisk-users] authentication number at the end of the number before calls go through.   Mark Quitoriano
• [asterisk-users] authentication number at the end of the number before calls go through.   Tilghman Lesher
• [asterisk-users] authentication number at the end of the number before calls go through.   Tilghman Lesher
• [asterisk-users] authentication number at the end of the number before calls go through.   Paul Hales
• [asterisk-users] DTMF problems while greeting is playing (Background())   James Lamanna
• [asterisk-users] authentication number at the end of the number before calls go through.   Mark Quitoriano
• [asterisk-users] authentication number at the end of the number before calls go through.   Mark Quitoriano
• [asterisk-users] Newbie Polycom: IP601 console with expansion module   Lee, John (Sydney)
• [asterisk-users] Call tracing - Asterisk 1.4   Louwrens Benadé
• [asterisk-users] include context on [globals]   Mark Quitoriano
• [asterisk-users] include context on [globals]   Tzafrir Cohen
• [asterisk-users] Strange problem Solved   Accursio Avona
• [asterisk-users] Expected iax user behaviour   Thomas Kenyon
• [asterisk-users] CCM 6 and Asterisk routing again   Aaron Fransen
• [asterisk-users] TXFax/RXFax/AGX-Addons/SpanDSP Crashing   Martin
• [asterisk-users] TXFax/RXFax/AGX-Addons/SpanDSP Crashing   Tzafrir Cohen
• [asterisk-users] S100I IAXy - Why is it discontinued?   Joseph
• [asterisk-users] S100I IAXy - Why is it discontinued?   Tilghman Lesher
• [asterisk-users] Newbie Polycom: IP601 console with expansion module   Jerry Jones
• [asterisk-users] What replaces Macro() now? And how do you do the equivalent?   Anthony Messina
• [asterisk-users] DTMF problems while greeting is playing (Background())   bilal ghayyad
• [asterisk-users] DTMF problems while greeting is playing (Background())   Brent Davidson
• [asterisk-users] snom official english forum   d tbsky
• [asterisk-users] Asterisk not transcoding between installed codecs   Brent Davidson
• [asterisk-users] Sending SMS   Jesus Mogollon
• [asterisk-users] Customer complains of noise on line I cannot reproduce.   Fons van der Beek
• [asterisk-users] Call tracing - Asterisk 1.4   Atis Lezdins
• [asterisk-users] DTMF problems while greeting is playing (Background())   James Lamanna
• [asterisk-users] AGI - calling functions, CHANNEL STATUS broken?   Steve Edwards
• [asterisk-users] IP650 console with expansion modules   Bill Andersen
• [asterisk-users] DTMF problems while greeting is playing (Background())   James Lamanna
• [asterisk-users] AGI - calling functions, CHANNEL STATUS broken?   Kevin DeGraaf
• [asterisk-users] DUNDi   Jeremy Mann
• [asterisk-users] Druid Open Source Edition   Joshua Wilson
• [asterisk-users] DUNDi   Jeremy Mann
• [asterisk-users] Druid Open Source Edition   Steve Totaro
• [asterisk-users] Druid Open Source Edition   Steve Totaro
• [asterisk-users] AGI - calling functions, CHANNEL STATUS broken?   David Van Ginneken
• [asterisk-users] Druid Open Source Edition   Joshua Wilson
• [asterisk-users] DUNDi   Chris Bagnall
• [asterisk-users] user web interface   Eve-Ellen
• [asterisk-users] SIP Registration!   Naveen Palani
• [asterisk-users] Druid Open Source Edition   Dean Collins
• [asterisk-users] Druid Open Source Edition   Anciso, Roy
• [asterisk-users] AGI - calling functions, CHANNEL STATUS broken?   Kevin DeGraaf
• [asterisk-users] does the meetme module still require an external timing source?   Dennis Christopher
• [asterisk-users] Druid Open Source Edition   Joshua Wilson
• [asterisk-users] Druid Open Source Edition   Joshua Wilson
• [asterisk-users] AGI - calling functions [SOLVED]   Kevin DeGraaf
• [asterisk-users] TXFax/RXFax/AGX-Addons/SpanDSP Crashing   Matt Riddell
• [asterisk-users] does the meetme module still require an external timing source?   Matt Riddell
• [asterisk-users] Problem sending CallerID Name to Dialogic based phone app   Mike Fedyk
• [asterisk-users] does the meetme module still require an external timing source?   Dennis Christopher
• [asterisk-users] does the meetme module still require an external timing source?   Steve Totaro
• [asterisk-users] does the meetme module still require anexternal timing source?   Mike Fedyk
• [asterisk-users] does the meetme module still require anexternal timing source?   Dennis Christopher
• [asterisk-users] does the meetme module still require an external timing source?   Michiel van Baak
• [asterisk-users] Druid Open Source Edition   Michiel van Baak
• [asterisk-users] What replaces Macro() now? And how do you do the equivalent?   Perssy Llamosas
• [asterisk-users] DTMF CCM Express   Consuelo Vega
• [asterisk-users] test please ignore   Christian
• [asterisk-users] Druid Open Source Edition   Joshua Wilson
• [asterisk-users] Druid Open Source Edition   Steve Totaro
• [asterisk-users] Druid Open Source Edition   Mike Fedyk
• [asterisk-users] Druid Open Source Edition   Joshua Wilson
• [asterisk-users] Druid Open Source Edition   Michiel van Baak
• [asterisk-users] Druid Open Source Edition   Joshua Wilson
• [asterisk-users] IP650 console with expansion modules   Noah Miller
• [asterisk-users] IP650 console with expansion modules   Klaverstyn, David C
• [asterisk-users] Newbie Polycom: IP601 console with expansion module   Paul Hales
• [asterisk-users] Newbie Polycom: IP601 console withexpansion module   Lee, John (Sydney)
• [asterisk-users] Druid Open Source Edition   Mike Clark
• [asterisk-users] Druid Open Source Edition   Tzafrir Cohen
• [asterisk-users] asterisk out of service   Rilawich Ango
• [asterisk-users] asterisk out of service   Mike Fedyk
• [asterisk-users] Newbie Polycom: IP601 console withexpansion module   Paul Hales
• [asterisk-users] Asterisk not transcoding between installed codecs   Gonzalo Servat
• [asterisk-users] TXFax/RXFax/AGX-Addons/SpanDSP Crashing   Martin
• [asterisk-users] T.38 SIP Issues   Andreas van dem Helge
• [asterisk-users] How to find out the IP of the calling party?   Gonzalo Servat
• [asterisk-users] Newbie One-touch Recording: Does not work   Lee, John (Sydney)
• [asterisk-users] Newbie One-touch Recording: Does not work (more info)   Lee, John (Sydney)
• [asterisk-users] Test - Please Ignore   Al Baker
• [asterisk-users] Newbie One-touch Recording: Does not work   Paul Hales
• [asterisk-users] Newbie One-touch Recording: Does not work   Paul Hales
• [asterisk-users] Newbie One-touch Recording: Does not work (more info)   Paul Hales
• [asterisk-users] Newbie One-touch Recording: Does not work   Lee, John (Sydney)
• [asterisk-users] Newbie One-touch Recording: Does not work   Lee, John (Sydney)
• [asterisk-users] Call tracing - Asterisk 1.4   Louwrens Benadé
• [asterisk-users] Newbie One-touch Recording: Does not work   Lee, John (Sydney)
• [asterisk-users] incoming call popup   Mike Diehl
• [asterisk-users] Newbie One-touch Recording: Does not work   Paul Hales
• [asterisk-users] Newbie One-touch Recording: Does not work   Paul Hales
• [asterisk-users] Newbie One-touch Recording: Does notwork (more info)   Lee, John (Sydney)
• [asterisk-users] Druid Open Source Edition   Michiel van Baak
• [asterisk-users] How to find out the IP of the calling party?   Ex Vito
• [asterisk-users] How to find out the IP of the calling party?   Ex Vito
• [asterisk-users] OT: RTP - NAT - SBC   Dovid B
• [asterisk-users] IP650 console with expansion modules   Noah Miller
• [asterisk-users] Newbie One-touch Recording: Does not work   Jared Smith
• [asterisk-users] MySQL Voicemail Storage Questions\Errors   Mike Hammett
• [asterisk-users] CCM 6 and Asterisk routing again   Aaron Fransen
• [asterisk-users] sip.conf help, inbound calls fall to last specified context   Mike Hammett
• [asterisk-users] queue log vs. cdr   Vieri
• [asterisk-users] SNOM on "Do Not Call" list????   Drew Gibson
• [asterisk-users] queue log vs. cdr   Atis Lezdins
• [asterisk-users] queue log vs. cdr   Rob Hillis
• [asterisk-users] SNOM on "Do Not Call" list????   d tbsky
• [asterisk-users] TXFax/RXFax/AGX-Addons/SpanDSP Crashing   lordfuknowsyou
• [asterisk-users] NIN Ghosts music (free download) safe for MOH?   Jay R. Ashworth
• [asterisk-users] NIN Ghosts music (free download) safe for MOH?   Jay R. Ashworth
• [asterisk-users] sip.conf help, inbound calls fall to last specified context   Mike Hammett
• [asterisk-users] dialstatus and cancelled calls   Vieri
• [asterisk-users] SNOM on "Do Not Call" list????   Usman Tahir
• [asterisk-users] need * consultant in st louis area   A_ Navone
• [asterisk-users] Error in Callback CDR   Douglas Garstang
• [asterisk-users] T.38 SIP Issues   Mindaugas Kezys
• [asterisk-users] CallerID setting issue with withheld numbers and mISDN ...   Gordon Henderson
• [asterisk-users] Application registration on Asterisk 1.4 and 1.6?   jonas boering
• [asterisk-users] CallerID setting issue with withheld numbers and mISDN ...   Doug Lytle
• [asterisk-users] CallerID setting issue with withheld numbers and mISDN ...   Gordon Henderson
• [asterisk-users] Druid Open Source Edition   Benny Amorsen
• [asterisk-users] Druid Open Source Edition   Joshua Wilson
• [asterisk-users] RedFone foneBRIDGE2 2e1 - anyone used it or another TDMoE bridge?   arkda
• [asterisk-users] RedFone foneBRIDGE2 2e1 - anyone used it or another TDMoE bridge?   Jonathan C. Bailey
• [asterisk-users] Application registration on Asterisk 1.4 and 1.6?   Kevin P. Fleming
• [asterisk-users] RedFone foneBRIDGE2 2e1 - anyone used it oranother TDMoE bridge?   Tom Moore
• [asterisk-users] RedFone foneBRIDGE2 2e1 - anyone used it oranother TDMoE bridge?   sales at failsafevoip.com
• [asterisk-users] Help: DTMF problem   Jarga Jallow
• [asterisk-users] Mail Server   Mike Hammett
• [asterisk-users] RedFone foneBRIDGE2 2e1 - anyone used it oranother TDMoE bridge?   Tzafrir Cohen
• [asterisk-users] Hardware Platform   Goran Donev
• [asterisk-users] Mail Server   Steve Prior
• [asterisk-users] Mail Server   Erik Anderson
• [asterisk-users] Mail Server   John Mason Jr
• [asterisk-users] Mail Server   Mike Hammett
• [asterisk-users] Mail Server   Mike Hammett
• [asterisk-users] Multiple clients registering on same definition in Realtime   Mike Hammett
• [asterisk-users] Multiple clients registering on same definition in Realtime   Matt Putnam
• [asterisk-users] RedFone foneBRIDGE2 2e1 - anyone used it oranother TDMoE bridge?   Ex Vito
• [asterisk-users] T.38 SIP Issues   Andreas van dem Helge
• [asterisk-users] asterisk out of service   Rilawich Ango
• [asterisk-users] Hardware Platform   Grygoriy Dobrovolskyy
• [asterisk-users] Customer complains of noise on line I cannot reproduce.   Grygoriy Dobrovolskyy
• [asterisk-users] Mail Server   Lyle Giese
• [asterisk-users] Asterisk 1.6   Paul Hales
• [asterisk-users] Anyone know of a pass through ATA   Thermal Wetland
• [asterisk-users] Group Listen on SIP Phone   Thermal Wetland
• [asterisk-users] Anyone know of a pass through ATA   W.Kenworthy
• [asterisk-users] Mail Server   Andreas van dem Helge
• [asterisk-users] Help: DTMF problem   Andreas van dem Helge
• [asterisk-users] Anyone know of a pass through ATA   Andreas van dem Helge
• [asterisk-users] Call tracing - Asterisk 1.4   Andreas van dem Helge
• [asterisk-users] Dialing patterns and "GSM" format numbers   Adrian Merwood
• [asterisk-users] Dialing patterns and "GSM" format numbers   Horwich IT Services (Godwin Stewart)
• [asterisk-users] VoIP Users Conference for Friday March 14th @ 12 Noon EDT   randulo
• [asterisk-users] Asterisk 1.6   Tzafrir Cohen
• [asterisk-users] Druid Open Source Edition   randulo
• [asterisk-users] Asterisk 1.6   Igor A. Goncharovsky
• [asterisk-users] CallerID(num) not showing on cli   Rizwan Hisham
• [asterisk-users] How to find out the IP of the calling party?   Rizwan Hisham
• [asterisk-users] Mail Server   love U.all
• [asterisk-users] Group Listen on SIP Phone   Faraz Khan
• [asterisk-users] Dialing patterns and "GSM" format numbers   Adrian Merwood
• [asterisk-users] queue log vs. cdr   Vieri
• [asterisk-users] Mail Server   Horwich IT Services (Godwin Stewart)
• [asterisk-users] T.38 SIP Issues   Mindaugas Kezys
• [asterisk-users] T.38 SIP Issues   Steve Underwood
• [asterisk-users] Mail Server   Felipe Trevisan
• [asterisk-users] Logs for Call generated by Manager API   sanjay.rajdev at featherstoneinformatics.com
• [asterisk-users] Dialing patterns and "GSM" format numbers   Remco Barendse
• [asterisk-users] Logs for Call generated by Manager API   sanjay.rajdev at featherstoneinformatics.com
• [asterisk-users] T.38 SIP Issues   Ricardo Carvalho
• [asterisk-users] Dialing patterns and "GSM" format numbers   Adrian Merwood
• [asterisk-users] T.38 SIP Issues   Steve Underwood
• [asterisk-users] How to find out the IP of the calling party?   Gonzalo Servat
• [asterisk-users] Anyone know of a pass through ATA   Jay R. Ashworth
• [asterisk-users] T.38 SIP Issues   Andreas van dem Helge
• [asterisk-users] Anyone know of a pass through ATA   Andreas van dem Helge
• [asterisk-users] Anyone know of a pass through ATA   Jay R. Ashworth
• [asterisk-users] Logs for Call generated by Manager API   Lee Jenkins
• [asterisk-users] DTMF problems while greeting is playing (Background())   bilal ghayyad
• [asterisk-users] Logs for Call generated by Manager API   Lee Jenkins
• [asterisk-users] Trouble with Incoming Callerid on Trixbox   Eric Rees
• [asterisk-users] FW: [asterisk-dev] Call failed, reason 0 explanation.   Mark Hamilton
• [asterisk-users] FW: [asterisk-dev] Hardware and CentOS tweaks.   Mark Hamilton
• [asterisk-users] Trouble with Incoming Callerid on Trixbox   Gordon Henderson
• [asterisk-users] DTMF problems while greeting is playing (Background())   James Lamanna
• [asterisk-users] Druid Open Source Edition   Jeff Johnson
• [asterisk-users] Druid Open Source Edition   Joshua Wilson
• [asterisk-users] Looking for a cheap SIP termination point.   Ken D'Ambrosio
• [asterisk-users] Logs for Call generated by Manager API   sanjay.rajdev at featherstoneinformatics.com
• [asterisk-users] Looking for a cheap SIP termination point.   Steve Edwards
• [asterisk-users] Logs for Call generated by Manager API   Mark Hamilton
• [asterisk-users] Logs for Call generated by Manager API   sanjay.rajdev at featherstoneinformatics.com
• [asterisk-users] Callerid Error- Causing All Zap Channels Busy   John Meksavan
• [asterisk-users] DTMF problems while greeting is playing (Background())   bilal ghayyad
• [asterisk-users] Logs for Call generated by Manager API   Lee Jenkins
• [asterisk-users] Logs for Call generated by Manager API   Lee Jenkins
• [asterisk-users] DID T1 PRI   broadband Voice
• [asterisk-users] Callerid Error- Causing All Zap Channels Busy   Lee Jenkins
• [asterisk-users] DID T1 PRI   Steve Totaro
• [asterisk-users] DID T1 PRI   broadband Voice
• [asterisk-users] TXFax/RXFax/AGX-Addons/SpanDSP Crashing   Edwin Lam
• [asterisk-users] Anyone know of a pass through ATA   Thermal Wetland
• [asterisk-users] DID T1 PRI   Darren Wright
• [asterisk-users] Anyone know of a pass through ATA   billk at iinet.net.au
• [asterisk-users] Anyone know of a pass through ATA   Thermal Wetland
• [asterisk-users] Anyone know of a pass through ATA   Andreas van dem Helge
• [asterisk-users] IAX ATA that can be set up and drop shipped to a US address?   randulo
• [asterisk-users] Sip phones for call centers   Mail list
• [asterisk-users] [asterisk-biz] Sip phones for call centers   Zoa
• [asterisk-users] DID T1 PRI   broadband Voice
• [asterisk-users] DID T1 PRI   broadband Voice
• [asterisk-users] LDAP   Gonzalo Servat
• [asterisk-users] Asterisk VOIP Jobs version 2 Launched!   Asterisk Jobs
• [asterisk-users] Calling a Macro with arguments in AstApplication/AstApplicationData   Gonzalo Servat
• [asterisk-users] DID T1 PRI   Darren Wright
• [asterisk-users] Calling a Macro with arguments in AstApplication/AstApplicationData   Gonzalo Servat
• [asterisk-users] Telemarketer Torture....   James Finstrom
• [asterisk-users] DID T1 PRI   Ron Joffe
• [asterisk-users] LDAP   Faraz Khan
• [asterisk-users] unsuscribe   Sébastien Mortier
• [asterisk-users] unsuscribe   Sam Tam
• [asterisk-users] DID T1 PRI   broadband Voice
• [asterisk-users] LDAP (was: Re: asterisk-users Digest, Vol 44, Issue 48)   Matthew Rubenstein
• [asterisk-users] Telemarketer Torture....   Lyle Giese
• [asterisk-users] DID T1 PRI   Ron Joffe
• [asterisk-users] Telemarketer Torture....   Horwich IT Services (Godwin Stewart)
• [asterisk-users] Telemarketer Torture....   Ade Vickers
• [asterisk-users] Telemarketer Torture....   John covici
• [asterisk-users] Telemarketer Torture....   randulo
• [asterisk-users] Druid Open Source Edition   Steve Totaro
• [asterisk-users] Telemarketer Torture....   Lee Jenkins
• [asterisk-users] Telemarketer Torture....   James Finstrom
• [asterisk-users] cordless usb handsets: Uniden Win1200?   sean darcy
• [asterisk-users] Telemarketer Torture....   Doug Lytle
• [asterisk-users] Telemarketer Torture.... (was: Re: asterisk-users Digest, Vol 44, Issue 49)   Matthew Rubenstein
• [asterisk-users] Druid Open Source Edition   Joshua Wilson
• [asterisk-users] Druid Open Source Edition   Tzafrir Cohen
• [asterisk-users] Druid Open Source Edition   Senad Jordanovic
• [asterisk-users] Druid Open Source Edition   Steve Totaro
• [asterisk-users] DUNDi or ENUM   Femi
• [asterisk-users] Druid Open Source Edition   Senad Jordanovic
• [asterisk-users] Problem with incoming calls on Broadvoice after upgrade to 1.4.18   Jon Miron
• [asterisk-users] dialstatus and cancelled calls   Matt Riddell
• [asterisk-users] Asterisk not transcoding between installed codecs   Jaswinder Singh
• [asterisk-users] Problem with incoming calls on Broadvoice after upgrade to 1.4.18   Raj Jain
• [asterisk-users] Mail Server   linuxian iandsd
• [asterisk-users] using the System() command to call a script   Jerry Geis
• [asterisk-users] DID T1 PRI   broadband Voice
• [asterisk-users] Problem with incoming calls on Broadvoice after upgrade to 1.4.18   Jon Miron
• [asterisk-users] Problem with incoming calls on Broadvoice after upgrade to 1.4.18   Raj Jain
• [asterisk-users] Asterisk VOIP Jobs version 2 Launched!   Matt Watson
• [asterisk-users] using the System() command to call a script   Steve Edwards
• [asterisk-users] Polycom IP 330 w/VLAN?   James Sneeringer
• [asterisk-users] DID T1 PRI   Ron Joffe
• [asterisk-users] Newbie ASTDB: cannot replicate among Asterisk servers?   Lee, John (Sydney)
• [asterisk-users] Redundant Voicemail   Ein Bielaczyc
• [asterisk-users] Newbie ASTDB: cannot replicate among Asterisk servers?   Doug Lytle
• [asterisk-users] Newbie ASTDB: cannot replicate among Asterisk servers?   Paul Hales
• [asterisk-users] Druid Open Source Edition   Steve Totaro
• [asterisk-users] Asterisk VOIP Jobs version 2 Launched!   Steve Totaro
• [asterisk-users] Newbie Polycom: DND answered as "on the phone" instead of "not available"   Lee, John (Sydney)
• [asterisk-users] update_call_counter: Call to peer '2509' rejected due to usage limit of 1?   Rajkumar S
• [asterisk-users] Desperately need help with Asterisk setup   Pete Kay
• [asterisk-users] Handling 3 different call ending causes   Tobias Ahlander
• [asterisk-users] Handling 3 different call ending causes   Alex Balashov
• [asterisk-users] Druid Open Source Edition   Senad Jordanovic
• [asterisk-users] DID T1 PRI   Tzafrir Cohen
• [asterisk-users] NVFaxDetect and Asterisk 1.6   Tobias Wolf
• [asterisk-users] Redundant Voicemail   Mindaugas Kezys
• [asterisk-users] Desperately need help with Asterisk setup   Anselm Martin Hoffmeister
• [asterisk-users] Desperately need help with Asterisk setup   Pete Kay
• [asterisk-users] ldap for sip users.   sasikala kala
• [asterisk-users] Druid Open Source Edition   Steve Totaro
• [asterisk-users] Desperately need help with Asterisk setup   Steve Totaro
• [asterisk-users] Druid Open Source Edition   Senad Jordanovic
• [asterisk-users] Druid Open Source Edition   Dean Collins
• [asterisk-users] Druid Open Source Edition   Outback Dingo
• [asterisk-users] Druid Open Source Edition   Outback Dingo
• [asterisk-users] Druid Open Source Edition   Senad Jordanovic
• [asterisk-users] The switch statement in extensions.conf   Rizwan Hisham
• [asterisk-users] update_call_counter: Call to peer '2509' rejected due to usage limit of 1?   Grygoriy Dobrovolskyy
• [asterisk-users] update_call_counter: Call to peer '2509' rejected due to usage limit of 1?   Grygoriy Dobrovolskyy
• [asterisk-users] Desperately need help with Asterisk setup   James Texter III
• [asterisk-users] Desperately need help with Asterisk setup   linuxian iandsd
• [asterisk-users] Desperately need help with Asterisk setup   Pete Kay
• [asterisk-users] Desperately need help with Asterisk setup   Pete Kay
• [asterisk-users] Asterisk VOIP Jobs version 2 Launched!   Rony Ron
• [asterisk-users] Desperately need help with Asterisk setup   Steve Totaro
• [asterisk-users] Weird NAT issue ...   Alan Williamson
• [asterisk-users] Weird NAT issue ...   Steve Davies
• [asterisk-users] Desperately need help with Asterisk setup   Pete Kay
• [asterisk-users] Telemarketer Torture....   Drew Gibson
• [asterisk-users] Desperately need help with Asterisk setup   Anselm Martin Hoffmeister
• [asterisk-users] Handling 3 different call ending causes   Tobias Ahlander
• [asterisk-users] Polycom IP 330 w/VLAN?   Noah Miller
• [asterisk-users] Weird NAT issue ...   Anselm Martin Hoffmeister
• [asterisk-users] Handling 3 different call ending causes   Arjan Kroon | Mobillion
• [asterisk-users] MeetMe option b   Jerry Geis
• [asterisk-users] update_call_counter: Call to peer '2509' rejected due to usage limit of 1?   Rajkumar S
• [asterisk-users] Asterisk VOIP Jobs version 2 Launched!   Alex Robar
• [asterisk-users] Desperately need help with Asterisk setup   Steve Edwards
• [asterisk-users] LDAP (was: Re: asterisk-users Digest, Vol 44, Issue 48)   Faraz Khan
• [asterisk-users] MeetMe option b   Jerry Geis
• [asterisk-users] MeetMe option b   Tony Mountifield
• [asterisk-users] LDAP   Faraz Khan
• [asterisk-users] ldap for sip users.   Faraz Khan
• [asterisk-users] MeetMe option b   Doug Lytle
• [asterisk-users] Desperately need help with Asterisk setup   Pete Kay
• [asterisk-users] Handling 3 different call ending causes   Tobias Ahlander
• [asterisk-users] MeetMe option b   Lee Jenkins
• [asterisk-users] Druid Open Source Edition   Steve Totaro
• [asterisk-users] Desperately need help with Asterisk setup   Pete Kay
• [asterisk-users] Desperately need help with Asterisk setup   Steve Totaro
• [asterisk-users] Turn off MusicOnHold for individual User   Adrian Marsh
• [asterisk-users] Druid Open Source Edition   Brett Crapser
• [asterisk-users] Druid Open Source Edition   Senad Jordanovic
• [asterisk-users] Druid Open Source Edition   Erik Anderson
• [asterisk-users] Druid Open Source Edition   Steve Totaro
• [asterisk-users] Druid Open Source Edition   Dean Collins
• [asterisk-users] Druid Open Source Edition   Steve Totaro
• [asterisk-users] php web chat + asterisk -> callcenter   Carlos Carvalhar
• [asterisk-users] Desperately need help with Asterisk setup   Steve Totaro
• [asterisk-users] Telemarketer Torture....   Mojo with Horan & Company, LLC
• [asterisk-users] Telemarketer Torture....   Dean Collins
• [asterisk-users] asterisk.conf uniquename or sysname for uniqueid field in CDR   Vieri
• [asterisk-users] RedFone foneBRIDGE2 2e1 - anyone used it or another TDMoE bridge?   Kevin DeGraaf
• [asterisk-users] Core dump?   Mark Hamilton
• [asterisk-users] Core dump?   Steve Totaro
• [asterisk-users] Order of queue member list   C. Chad Wallace
• [asterisk-users] update_call_counter: Call to peer '2509' rejected due to usage limit of 1?   Grygoriy Dobrovolskyy
• [asterisk-users] MeetMe option b   Jerry Geis
• [asterisk-users] Druid Open Source Edition   Jay R. Ashworth
• [asterisk-users] update_call_counter: Call to peer '2509' rejected due to usage limit of 1?   Atis Lezdins
• [asterisk-users] Desperately need help with Asterisk setup   Tzafrir Cohen
• [asterisk-users] Pre-pending certain digits (like 9) to an outbound call number   Joshua Kinard
• [asterisk-users] Pre-pending certain digits (like 9) to an outbound call number   Joshua Kinard
• [asterisk-users] Desperately need help with Asterisk setup   Mojo with Horan & Company, LLC
• [asterisk-users] Pre-pending certain digits (like 9) to an outbound call number   Mojo with Horan & Company, LLC
• [asterisk-users] Order of queue member list   Tilghman Lesher
• [asterisk-users] Newbie Polycom: DND answered as "on the phone" instead of "not available"   Paul Hales
• [asterisk-users] Order of queue member list   Atis Lezdins
• [asterisk-users] Subexpression usage in Asterisk Dialplan Regular Expressions   Mike Fedyk
• [asterisk-users] Way to hangup a call between two asterisk machines   Jerry Geis
• [asterisk-users] LCR in Asterisk   Ex Vito
• [asterisk-users] LCR in Asterisk   Tilghman Lesher
• [asterisk-users] php web chat + asterisk -> callcenter   Faraz Khan
• [asterisk-users] Desperately need help with Asterisk setup   Pete Kay
• [asterisk-users] call screening feature   Janu Mukherjee
• [asterisk-users] Newbie Queue: Simple Queue Problem   Lee, John (Sydney)
• [asterisk-users] dialstatus and cancelled calls   Vieri
• [asterisk-users] asterisk.conf uniquename or sysname for uniqueid field in CDR   Vieri
• [asterisk-users] Asterisk 1.4 reliability problems   Ben Willcox
• [asterisk-users] Newbie Queue: Simple Queue Problem   Doug Lytle
• [asterisk-users] Asterisk 1.4 reliability problems   Al Baker
• [asterisk-users] LDAP (was: Re: asterisk-users Digest, Vol 44, Issue 48)   Gavin Henry
• [asterisk-users] Asterisk 1.4 reliability problems   Doug Lytle
• [asterisk-users] Asterisk 1.4 reliability problems   Al Baker
• [asterisk-users] Newbie Queue: Simple Queue Problem   Robert Lister
• [asterisk-users] Asterisk 1.4 reliability problems   Doug Lytle
• [asterisk-users] Asterisk 1.4 reliability problems   Steve Totaro
• [asterisk-users] Asterisk 1.4 reliability problems   Patrick
• [asterisk-users] Sangoma FXO/FXS config   Paul Goodyear
• [asterisk-users] Asterisk 1.4 reliability problems   Gordon Henderson
• [asterisk-users] Asterisk 1.4 reliability problems   Matt Florell
• [asterisk-users] Sangoma FXO/FXS config   Tzafrir Cohen
• [asterisk-users] Using dedicated eth2 card for SIP trunk line to ISP provider - how to setup ?   Robert Rozman
• [asterisk-users] Problem with incoming calls on Broadvoice after upgrade to 1.4.18   Jon Miron
• [asterisk-users] call screening feature   Marco Mouta
• [asterisk-users] php web chat + asterisk -> callcenter   Marco Mouta
• [asterisk-users] Sangoma FXO/FXS config   Paul Goodyear
• [asterisk-users] Call signalling on BT FeatureLine Compact (Sangoma A200)   Paul Goodyear
• [asterisk-users] Call signalling on BT FeatureLine Compact (Sangoma A200)   Gordon Henderson
• [asterisk-users] Call signalling on BT FeatureLine Compact (Sangoma A200)   Tzafrir Cohen
• [asterisk-users] Asterisk 1.4 reliability problems   Norman Franke
• [asterisk-users] ztdummy problem causing playback () to fail   Pete Kay
• [asterisk-users] Call signalling on BT FeatureLine Compact (SangomaA200)   Steve Langstaff
• [asterisk-users] Asterisk 1.4 reliability problems   Ben Willcox
• [asterisk-users] Asterisk 1.4 reliability problems   Atis Lezdins
• [asterisk-users] Asterisk 1.4 reliability problems   Patrick
• [asterisk-users] ztdummy problem causing playback () to fail   Atis Lezdins
• [asterisk-users] ztdummy problem causing playback () to fail   RE Kushner List Account
• [asterisk-users] How is uniqueid computed   sanjay.rajdev at featherstoneinformatics.com
• [asterisk-users] ztdummy problem causing playback () to fail   Doug Lytle
• [asterisk-users] Call signalling on BT FeatureLine Compact (Sangoma A200)   Paul Goodyear
• [asterisk-users] Asterisk 1.4 reliability problems   Steve Totaro
• [asterisk-users] Asterisk 1.4 reliability problems   Matt Florell
• [asterisk-users] Turn off MusicOnHold for individual User   Adrian Marsh
• [asterisk-users] [SOLVED] GXP2000 and asterisk 1.0.9   Giordano Grandis
• [asterisk-users] Call signalling on BT FeatureLine Compact(Sangoma A200)   Ade Vickers
• [asterisk-users] How is uniqueid computed   Mindaugas Kezys
• [asterisk-users] ztdummy problem causing playback () to fail   Pete Kay
• [asterisk-users] Asterisk 1.4 reliability problems   Tzafrir Cohen
• [asterisk-users] ztdummy problem causing playback () to fail   Tzafrir Cohen
• [asterisk-users] Asterisk 1.4 reliability problems   Norman Franke
• [asterisk-users] Call signalling on BT FeatureLine Compact(Sangoma A200)   Paul Goodyear
• [asterisk-users] asterisk.conf uniquename or sysname for uniqueid field in CDR   Atis Lezdins
• [asterisk-users] Call forward on Telco line   Tim Litwiller
• [asterisk-users] How is uniqueid computed   sanjay.rajdev at featherstoneinformatics.com
• [asterisk-users] capacity   Eve-Ellen Cole
• [asterisk-users] capacity   Steve Totaro
• [asterisk-users] Call forward on Telco line   Steve Totaro
• [asterisk-users] capacity   Eve-Ellen Cole
• [asterisk-users] Newbie Queue: Simple Queue Problem   Benny Amorsen
• [asterisk-users] Asterisk 1.4 reliability problems   Benny Amorsen
• [asterisk-users] Asterisk 1.4 reliability problems   Matt Florell
• [asterisk-users] Sip Line Status/Pickup   Brent Davidson
• [asterisk-users] (Critical Updates) Asterisk 1.2.27, 1.4.18.1, 1.4.19-rc3, 1.6.0-beta6 Released   The Asterisk Development Team
• [asterisk-users] AEL2 Hint & Parking   Brent Davidson
• [asterisk-users] AST-2008-002: Two buffer overflows in RTP Codec Payload Handling   Asterisk Security Team
• [asterisk-users] call screening feature   Paul Hales
• [asterisk-users] AST-2008-003: Unauthenticated calls allowed from SIP channel driver   Asterisk Security Team
• [asterisk-users] Turn off MusicOnHold for individual User   Lee, John (Sydney)
• [asterisk-users] Newbie Queue: Simple Queue Problem   Paul Hales
• [asterisk-users] AST-2008-004: Format String Vulnerability in Logger and Manager   Asterisk Security Team
• [asterisk-users] AST-2008-005: HTTP Manager ID is predictable   Asterisk Security Team
• [asterisk-users] Asterisk and Avaya 4610 handset   Al lists
• [asterisk-users] Asterisk with lumenvox   Josué Conti
• [asterisk-users] Asterisk 1.4 reliability problems   Matt Riddell
• [asterisk-users] Newbie Queue: Simple Queue Problem   Lee, John (Sydney)
• [asterisk-users] capacity   Steve Totaro
• [asterisk-users] Hardphone SIP phone costs   Anciso, Roy
• [asterisk-users] Hardphone SIP phone costs   Michael Graves
• [asterisk-users] Call Screening feature using asterisk   Janu Mukherjee
• [asterisk-users] Hardphone SIP phone costs   John Faubion
• [asterisk-users] Getting config from SPA-941 or 942 phones   James Lamanna
• [asterisk-users] Deadair in queues.   Mark Hamilton
• [asterisk-users] Call Screening feature using asterisk   Paul Hales
• [asterisk-users] Newbie Queue: greetings when first joining queue   Lee, John (Sydney)
• [asterisk-users] Getting config from SPA-941 or 942 phones   Andres
• [asterisk-users] Newbie Queue: greetings when first joining queue   Mark Hamilton
• [asterisk-users] Newbie Queue: greetings when first joiningqueue   Lee, John (Sydney)
• [asterisk-users] FXS channel banks   Lee, John (Sydney)
• [asterisk-users] Newbie Queue: greetings when first joiningqueue   Mark Hamilton
• [asterisk-users] Druid Open Source Edition   Mark Hamilton
• [asterisk-users] Newbie Queue: greetings when first joiningqueue   Julian Lyndon-Smith
• [asterisk-users] Call signalling on BT FeatureLine Compact (Sangoma A200)   David Quinton
• [asterisk-users] British Summertime & Grandstream Phones   Gordon Henderson
• [asterisk-users] Call signalling on BT FeatureLine Compact (Sangoma A200)   David Quinton
• [asterisk-users] Handling 3 different call ending causes   Tobias Ahlander
• [asterisk-users] (Critical Updates) Asterisk 1.2.27, 1.4.18.1, 1.4.19-rc3, 1.6.0-beta6 Released   Adrià Vidal
• [asterisk-users] Query about Bluetooth Head phone   Kashif Naeem
• [asterisk-users] (Critical Updates) Asterisk 1.2.27, 1.4.18.1, 1.4.19-rc3, 1.6.0-beta6 Released   Tzafrir Cohen
• [asterisk-users] Handling 3 different call ending causes   Atis Lezdins
• [asterisk-users] Hardphone SIP phone costs   Steve Totaro
• [asterisk-users] Call signalling on BT FeatureLine Compact (Sangoma A200)   Gordon Henderson
• [asterisk-users] RedFone foneBRIDGE2 2e1 - anyone used it or another TDMoE bridge?   Al Baker
• [asterisk-users] capacity   Al Baker
• [asterisk-users] Limit calls when using autodial   Tong
• [asterisk-users] Limit calls when using autodial   Tong
• [asterisk-users] fxo tdm400p issue   Bhrugu Mehta
• [asterisk-users] Limit calls when using autodial   Fernando Berretta
• [asterisk-users] Limit calls when using autodial   Doug Lytle
• [asterisk-users] fxo tdm400p issue   Tzafrir Cohen
• [asterisk-users] Limit calls when using autodial   Tong
• [asterisk-users] question how to play wave file in meetme and exit   Jerry Geis
• [asterisk-users] capacity   Ron Joffe
• [asterisk-users] Desperately need help with Asterisk setup   Jared Smith
• [asterisk-users] Bug in voicemail's serveremail setting in 1.4.18   OCG Technical Support
• [asterisk-users] Bug in voicemail's serveremail setting in 1.4.18   OCG Technical Support
• [asterisk-users] Hardphone SIP phone costs   Drew Gibson
• [asterisk-users] Call signalling on BT FeatureLine Compact (Sangoma A200)   Paul Goodyear
• [asterisk-users] Limit calls when using autodial   Tong
• [asterisk-users] Limit calls when using autodial   Tong
• [asterisk-users] Limit calls when using autodial   Tong
• [asterisk-users] Limit calls when using autodial   Tilghman Lesher
• [asterisk-users] Limit calls when using autodial   Doug Lytle
• [asterisk-users] capacity   Steve Totaro
• [asterisk-users] Limit calls when using autodial   Steve Totaro
• [asterisk-users] Bug in voicemail's serveremail setting in 1.4.18   OCG Technical Support
• [asterisk-users] capacity   Ron Joffe
• [asterisk-users] British Summertime & Grandstream Phones   Drew Gibson
• [asterisk-users] How configure Voice mail for multi users.   Mian M Asif
• [asterisk-users] How to configure Voice mail for multi users.   Mian M Asif
• [asterisk-users] Handling 3 different call ending causes   Tobias Ahlander
• [asterisk-users] Hardphone SIP phone costs   Anciso, Roy
• [asterisk-users] How to configure Voice mail for multi users.   Eric Wieling
• [asterisk-users] Bug in voicemail's serveremail setting in 1.4.18   OCG Technical Support
• [asterisk-users] Limit calls when using autodial   Lee Jenkins
• [asterisk-users] capacity   Drew Gibson
• [asterisk-users] Can't play recording message wav file   Pete Kay
• [asterisk-users] capacity   Steve Totaro
• [asterisk-users] Hardphone SIP phone costs   Horwich IT Services (Godwin Stewart)
• [asterisk-users] capacity   Drew Gibson
• [asterisk-users] Is Asterisk ready for Prime-Time?   Bill Andersen
• [asterisk-users] Ribbit Demo   Dean Collins
• [asterisk-users] Call signalling on BT FeatureLine Compact (Sangoma A200)   Gordon Henderson
• [asterisk-users] Ribbit Demo   Steve Totaro
• [asterisk-users] RES: php web chat + asterisk -> callcenter   Carlos Carvalhar
• [asterisk-users] Is Asterisk ready for Prime-Time?   John Novack
• [asterisk-users] capacity   Steve Totaro
• [asterisk-users] Ribbit Demo   Dean Collins
• [asterisk-users] Hardphone SIP phone costs   Norman Franke
• [asterisk-users] Is Asterisk ready for Prime-Time?   Senad Jordanovic
• [asterisk-users] phpagi   Carlos Carvalhar
• [asterisk-users] Hardphone SIP phone costs   Gordon Henderson
• [asterisk-users] phpagi   Geraint Lee
• [asterisk-users] phpagi   Erik Anderson
• [asterisk-users] question on meetme   Jerry Geis
• [asterisk-users] Is Asterisk ready for Prime-Time?   Andrew Kohlsmith (lists)
• [asterisk-users] Limit calls when using autodial   Tong
• [asterisk-users] RES: phpagi   Carlos Carvalhar
• [asterisk-users] Is Asterisk ready for Prime-Time?   Robert Lister
• [asterisk-users] Is Asterisk ready for Prime-Time?   Drew Gibson
• [asterisk-users] Query about Bluetooth Head phone   zaib Younis
• [asterisk-users] Is Asterisk ready for Prime-Time?   Michael Collins
• [asterisk-users] RES: phpagi   Erik Anderson
• [asterisk-users] Is Asterisk ready for Prime-Time?   Gordon Henderson
• [asterisk-users] Is Asterisk ready for Prime-Time?   Senad Jordanovic
• [asterisk-users] Inband SIP DTMF   Norman Franke
• [asterisk-users] Is Asterisk ready for Prime-Time?   Tzafrir Cohen
• [asterisk-users] Is Asterisk ready for Prime-Time?   Mojo with Horan & Company, LLC
• [asterisk-users] Hardphone SIP phone costs   Norman Franke
• [asterisk-users] Is Asterisk ready for Prime-Time?   Ron Arts
• [asterisk-users] Is Asterisk ready for Prime-Time?   Norman Franke
• [asterisk-users] Limit calls when using autodial   Doug Lytle
• [asterisk-users] Chanspy from a call file   Jerry Geis
• [asterisk-users] Is Asterisk ready for Prime-Time?   Gordon Henderson
• [asterisk-users] Is Asterisk ready for Prime-Time?   Bill Andersen
• [asterisk-users] Is Asterisk ready for Prime-Time?   Steven Kurylo
• [asterisk-users] Multiple sites, same extension   Aaron Fransen
• [asterisk-users] Is Asterisk ready for Prime-Time?   Bill Andersen
• [asterisk-users] Is Asterisk ready for Prime-Time?   Drew Gibson
• [asterisk-users] Is Asterisk ready for Prime-Time?   Bill Andersen
• [asterisk-users] Is Asterisk ready for Prime-Time?   RE Kushner List Account
• [asterisk-users] Is Asterisk ready for Prime-Time?   Tim Nelson
• [asterisk-users] Is Asterisk ready for Prime-Time?   Senad Jordanovic
• [asterisk-users] Is Asterisk ready for Prime-Time?   Erik Anderson
• [asterisk-users] Asterisk 1.4 reliability problems   Benny Amorsen
• [asterisk-users] Is Asterisk ready for Prime-Time?   Mojo with Horan & Company, LLC
• [asterisk-users] Is Asterisk ready for Prime-Time?   Gordon Henderson
• [asterisk-users] Is Asterisk ready for Prime-Time?   Al Baker
• [asterisk-users] Is Asterisk ready for Prime-Time?   Steve Totaro
• [asterisk-users] Is Asterisk ready for Prime-Time?   Steven Kurylo
• [asterisk-users] Asterisk with lumenvox   Philipp von Klitzing
• [asterisk-users] Call All   Rupert Utteridge - Digital Techniques (Austalia) Limited
• [asterisk-users] Is Asterisk ready for Prime-Time?   Tom Moore
• [asterisk-users] Is Asterisk ready for Prime-Time?   Al Baker
• [asterisk-users] Is Asterisk ready for Prime-Time?   Al Baker
• [asterisk-users] Asterisk with lumenvox   Rodrigo Gonzalez
• [asterisk-users] Ribbit Demo   Steve Totaro
• [asterisk-users] Is Asterisk ready for Prime-Time?   Steve Totaro
• [asterisk-users] Is Asterisk ready for Prime-Time?   Steve Totaro
• [asterisk-users] Is Asterisk ready for Prime-Time?   Steve Totaro
• [asterisk-users] Is Asterisk ready for Prime-Time?   Mojo with Horan & Company, LLC
• [asterisk-users] Is Asterisk ready for Prime-Time?   Steve Totaro
• [asterisk-users] Is Asterisk ready for Prime-Time?   Mojo with Horan & Company, LLC
• [asterisk-users] Is Asterisk ready for Prime-Time?   Chris Bagnall
• [asterisk-users] Is Asterisk ready for Prime-Time?   Howard Leadmon
• [asterisk-users] Call All   Paul Hales
• [asterisk-users] Is Asterisk ready for Prime-Time?   Paul Hales
• [asterisk-users] Is Asterisk ready for Prime-Time?   Tzafrir Cohen
• [asterisk-users] Is Asterisk ready for Prime-Time?   Paul Hales
• [asterisk-users] Is Asterisk ready for Prime-Time?   Grygoriy Dobrovolskyy
• [asterisk-users] Newbie Asterisk: Disaster Recovery Proof Asterisk   Lee, John (Sydney)
• [asterisk-users] Asterisk with lumenvox   Josué Conti
• [asterisk-users] Is Asterisk ready for Prime-Time?   Andrew Kohlsmith (lists)
• [asterisk-users] Is Asterisk ready for Prime-Time?   Andrew Kohlsmith (lists)
• [asterisk-users] Mail Server   Al lists
• [asterisk-users] Using dedicated eth2 card for SIP trunk line to ISP provider - how to setup ?   Al lists
• [asterisk-users] Newbie IVR: How to read() before playback() is finished?   Lee, John (Sydney)
• [asterisk-users] Is Asterisk ready for Prime-Time?   Alex Balashov
• [asterisk-users] AMD timing issues   Drew Miller
• [asterisk-users] Is Asterisk ready for Prime-Time?   Jared Smith
• [asterisk-users] Is Asterisk ready for Prime-Time?   Alex Balashov
• [asterisk-users] Is Asterisk ready for Prime-Time?   Al Baker
• [asterisk-users] Is Asterisk ready for Prime-Time?   Al Baker
• [asterisk-users] Newbie IVR: How to read() before playback() is finished?   Paul Hales
• [asterisk-users] Newbie: Two problems with Asterisk Config, Please Help   Pete Kay
• [asterisk-users] Asterisk 1.4 reliability problems   Matt Florell
• [asterisk-users] Newbie Asterisk: Disaster Recovery Proof Asterisk   Al Baker
• [asterisk-users] Asterisk 1.4 reliability problems   Al Baker
• [asterisk-users] Is Asterisk ready for Prime-Time?   Tilghman Lesher
• [asterisk-users] Unable to build smsq on beta6 and x86_64.   William F. Acker WB2FLW +1-303-722-7209
• [asterisk-users] Is Asterisk ready for Prime-Time?   Tzafrir Cohen
• [asterisk-users] Call signalling on BT FeatureLine Compact (Sangoma A200)   David Quinton
• [asterisk-users] Handling 3 different call ending causes   Tobias Ahlander
• [asterisk-users] Is Asterisk ready for Prime-Time?   Tzafrir Cohen
• [asterisk-users] Unable to build smsq on beta6 and x86_64.   Tzafrir Cohen
• [asterisk-users] Is Asterisk ready for Prime-Time?   randulo
• [asterisk-users] Newbie Asterisk: Disaster Recovery Proof Asterisk   Tzafrir Cohen
• [asterisk-users] hint status unavailable   Stefan Schmidt
• [asterisk-users] Sip Line Status/Pickup   Stefan Schmidt
• [asterisk-users] Asterisk re-invites and billing   Jaswinder Singh
• [asterisk-users] How to configure Voice mail for multi users.   Mian M Asif
• [asterisk-users] Newbie IVR: How to read() before playback() is finished?   Tony Mountifield
• [asterisk-users] Handling 3 different call ending causes   Atis Lezdins
• [asterisk-users] Hardphone SIP phone costs   SIP
• [asterisk-users] Is Asterisk ready for Prime-Time?   Alex Balashov
• [asterisk-users] Is Asterisk ready for Prime-Time?   Horwich IT Services (Godwin Stewart)
• [asterisk-users] Is Asterisk ready for Prime-Time?   Tzafrir Cohen
• [asterisk-users] Is Asterisk ready for Prime-Time?   Andreas Sikkema
• [asterisk-users] Is Asterisk ready for Prime-Time?   Doug Lytle
• [asterisk-users] hint status unavailable   Atis Lezdins
• [asterisk-users] Newbie IVR: How to read() before playback() isfinished?   Gary
• [asterisk-users] hint status unavailable   Johansson Olle E
• [asterisk-users] Is Asterisk ready for Prime-Time?   Michael Graves
• [asterisk-users] question on app_conference()   Jerry Geis
• [asterisk-users] Multiple sites, same extension   Aaron Fransen
• [asterisk-users] hint status unavailable   Steve Davies
• [asterisk-users] Is Asterisk ready for Prime-Time?   John Faubion
• [asterisk-users] hint status unavailable   Atis Lezdins
• [asterisk-users] hint status unavailable   Watkins, Bradley
• [asterisk-users] Is Asterisk ready for Prime-Time?   John Faubion
• [asterisk-users] Is Asterisk ready for Prime-Time?   Tzafrir Cohen
• [asterisk-users] Multiple sites, same extension   Aaron Fransen
• [asterisk-users] Dialplan Help   Jeremy Mann
• [asterisk-users] Is Asterisk ready for Prime-Time?   Norman Franke
• [asterisk-users] Is Asterisk ready for Prime-Time?   Norman Franke
• [asterisk-users] Unable to build smsq on beta6 and x86_64.   William F. Acker WB2FLW +1-303-722-7209
• [asterisk-users] Is Asterisk ready for Prime-Time?   Louwrens Benadé
• [asterisk-users] More DTMF issues   Brent Davidson
• [asterisk-users] Is Asterisk ready for Prime-Time?   Tzafrir Cohen
• [asterisk-users] capacity   Eve-Ellen Cole
• [asterisk-users] Polycom 650   Brent Torrenga
• [asterisk-users] More DTMF issues   Brent Davidson
• [asterisk-users] Is Asterisk ready for Prime-Time?   John Novack
• [asterisk-users] Multiple sites, same extension   Anselm Martin Hoffmeister
• [asterisk-users] 423 "Interval Too Brief" and expiry settings in sip.conf   Robert Rozman
• [asterisk-users] BLF and Snom phones   Loic Didelot
• [asterisk-users] Is Asterisk ready for Prime-Time?   John Faubion
• [asterisk-users] Is Asterisk ready for Prime-Time?   Mojo with Horan & Company, LLC
• [asterisk-users] Is Asterisk ready for Prime-Time?   Anselm Martin Hoffmeister
• [asterisk-users] BLF and Snom phones   Stefan Schmidt
• [asterisk-users] Is Asterisk ready for Prime-Time?   Andrew Kohlsmith (lists)
• [asterisk-users] How to configure Voice mail for multi users.   Mojo with Horan & Company, LLC
• [asterisk-users] Want to know Frequency and lenght of Frame   Mojo with Horan & Company, LLC
• [asterisk-users] Is Asterisk ready for Prime-Time?   RE Kushner List Account
• [asterisk-users] polarity in zapata.conf   wassim darwish
• [asterisk-users] Is Asterisk ready for Prime-Time?   Steve Totaro
• [asterisk-users] Want to know Frequency and lenght of Frame   Zoa
• [asterisk-users] Is Asterisk ready for Prime-Time?   RE Kushner List Account
• [asterisk-users] polarity in zapata.conf   Tzafrir Cohen
• [asterisk-users] BLF on Cisco 7970   Matthew Gibson
• [asterisk-users] Is Asterisk ready for Prime-Time?   Gordon Henderson
• [asterisk-users] How to configure Voice mail for multi users.   Anthony Messina
• [asterisk-users] Is Asterisk ready for Prime-Time?   shadowym
• [asterisk-users] Newbie: Two problems with Asterisk Config, Please Help   Carlos Rojas
• [asterisk-users] Newbie: Two problems with Asterisk Config, Please Help   Steve Totaro
• [asterisk-users] Newbie: Two problems with Asterisk Config, Please Help   Steve Totaro
• [asterisk-users] Newbie: Two problems with Asterisk Config, Please Help   Pete Kay
• [asterisk-users] Hardphone SIP phone costs   Gordon Henderson
• [asterisk-users] DRUID/Voiceroute on VoIP Users COnference today Friday 2008-03-21 at 11:58 AM EDT   randulo
• [asterisk-users] Which command line is used to send emails to notify incoming voicemail ?   Olivier
• [asterisk-users] Which command line is used to send emails to notify incoming voicemail ?   Olivier
• [asterisk-users] Newbie: Two problems with Asterisk Config, Please Help   Steve Totaro
• [asterisk-users] update_call_counter: Call to peer '2509' rejected due to usage limit of 1?   Rajkumar S
• [asterisk-users] Hardphone SIP phone costs   John Faubion
• [asterisk-users] Hardphone SIP phone costs   Darrick Hartman (lists)
• [asterisk-users] Digium registration utility version 3.0.3 released   Terry Wilson
• [asterisk-users] Hardware supporting groundstart signalling   Tim Nelson
• [asterisk-users] Digium registration utility version 3.0.3 released   Darrick Hartman (lists)
• [asterisk-users] TxFax in asterisk 1.4   Giordano Grandis
• [asterisk-users] ----www.cdsportal.net---- wholesale voip provider --starting at 1.1 cent per min   Ignacio Ortega A.
• [asterisk-users] Hardphone SIP phone costs   Gordon Henderson
• [asterisk-users] Digium registration utility version 3.0.3 released   Michael Graves
• [asterisk-users] TxFax in asterisk 1.4   Jonn R Taylor
• [asterisk-users] ----www.cdsportal.net---- wholesale voip provider --starting at 1.1 cent per min   Tim Nelson
• [asterisk-users] R: TxFax in asterisk 1.4   Giordano Grandis
• [asterisk-users] ----www.cdsportal.net---- wholesale voip provider --starting at 1.1 cent per min   Outback Dingo
• [asterisk-users] Problem with user regsitration and ldap on SVN version   sylvain.desbureaux at orange-ftgroup.com
• [asterisk-users] ----www.cdsportal.net---- wholesale voip provider --starting at 1.1 cent per min   Tim Nelson
• [asterisk-users] Hardphone SIP phone costs   Olivier
• [asterisk-users] ----www.cdsportal.net---- wholesale voip provider --starting at 1.1 cent per min   Gonzalo Servat
• [asterisk-users] ----www.cdsportal.net---- wholesale voip provider --starting at 1.1 cent per min   Ignacio Ortega
• [asterisk-users] ----www.cdsportal.net---- wholesale voipprovider --starting at 1.1 cent per min   Joshua Kinard
• [asterisk-users] Hardphone SIP phone costs   Ira
• [asterisk-users] ----www.cdsportal.net---- wholesale voipprovider --starting at 1.1 cent per min   lists at infoway.net
• [asterisk-users] Calls to sip extensions not defined   Ricardo B.
• [asterisk-users] Calls to sip extensions not defined   Steve Edwards
• [asterisk-users] Want to know Frequency and lenght of Frame   Mojo with Horan & Company, LLC
• [asterisk-users] ----www.cdsportal.net---- wholesale voip provider   Matthew Warren
• [asterisk-users] ----www.cdsportal.net---- wholesale voip provider   Ignacio Ortega A.
• [asterisk-users] --POSIBLE SPAM-- Peticion on line Parque de atracciones mcdonals   Omar Lopez Limonta
• [asterisk-users] Want to know Frequency and lenght of Frame   Dan Austin
• [asterisk-users] ----www.cdsportal.net---- wholesale voipprovider   Joshua Kinard
• [asterisk-users] Newbie: Two problems with Asterisk Config, Please Help   Pete Kay
• [asterisk-users] how to detect redirect fax call   Pete Kay
• [asterisk-users] Anyone used Siemens SIP/Dect phones?   Alan Lord
• [asterisk-users] LDAP (was: Re: asterisk-users Digest, Vol 44, Issue 48)   Faraz Khan
• [asterisk-users] LDAP (was: Re: asterisk-users Digest, Vol 44, Issue 48)   Faraz Khan
• [asterisk-users] ----www.cdsportal.net---- wholesale voipprovider   Ignacio Ortega A.
• [asterisk-users] ----www.cdsportal.net---- wholesale voipprovider   Outback Dingo
• [asterisk-users] BLF and Snom phones   Philipp Kempgen
• [asterisk-users] ----www.cdsportal.net---- wholesale voipprovider   Ignacio Ortega A.
• [asterisk-users] Problem: Digium TDM400 with XOptionsFlex - Line Busy   Thomas Klettke
• [asterisk-users] Problem: Digium TDM400 with XOptionsFlex - Line Busy   John Novack
• [asterisk-users] Anyone used Siemens SIP/Dect phones?   Gordon Henderson
• [asterisk-users] Anyone used Siemens SIP/Dect phones?   Michael Graves
• [asterisk-users] Calls to sip extensions not defined   R. B.
• [asterisk-users] Hardphone SIP phone costs   Norman W. Franke
• [asterisk-users] Anyone used Siemens SIP/Dect phones?   Alan Lord
• [asterisk-users] Anyone used Siemens SIP/Dect phones?   Gordon Henderson
• [asterisk-users] Calls to sip extensions not defined   Steve Edwards
• [asterisk-users] G723 on asterisk 1.4.1   wassim darwish
• [asterisk-users] Which command line is used to send emails to notify incoming voicemail ?   Robert Lister
• [asterisk-users] Anyone used Siemens SIP/Dect phones?   Robert Lister
• [asterisk-users] Anyone used Siemens SIP/Dect phones?   Chris Bagnall
• [asterisk-users] Anyone used Siemens SIP/Dect phones?   Chris Bagnall
• [asterisk-users] G723 on asterisk 1.4.1   Martin
• [asterisk-users] Calls to sip extensions not defined   Stefan Schmidt
• [asterisk-users] voicemail and needed language to be selected   bilal ghayyad
• [asterisk-users] voicemail and needed language to be selected   Alex Balashov
• [asterisk-users] No audio on Sangoma A104.   Alex Balashov
• [asterisk-users] No audio on Sangoma A104.   Alex Balashov
• [asterisk-users] More Broadvoice woes. Who's fault could this be?   Jon Miron
• [asterisk-users] Best alternative for getting prompts recorded.   Francesco Peeters
• [asterisk-users] BLF and Snom phones   Bill Hackensack
• [asterisk-users] voicemail and needed language to be selected   Rob Hillis
• [asterisk-users] Best alternative for getting prompts recorded.   Rob Hillis
• [asterisk-users] BLF and Snom phones   Rob Hillis
• [asterisk-users] Access rights between AGI and Web server?   Vincent
• [asterisk-users] More Broadvoice woes. Who's fault could this be?   Outback Dingo
• [asterisk-users] BLF and Snom phones   Christian Stredicke
• [asterisk-users] Anyone used Siemens SIP/Dect phones?   Robert Lister
• [asterisk-users] Anyone used Siemens SIP/Dect phones?   Dave Cotton
• [asterisk-users] Anyone used Siemens SIP/Dect phones?   Gordon Henderson
• [asterisk-users] Access rights between AGI and Web server?   Al Baker
• [asterisk-users] G723 on asterisk 1.4.1   Eric Wieling
• [asterisk-users] Calls to sip extensions not defined   Eric Wieling
• [asterisk-users] No audio on Sangoma A104.   Matt Florell
• [asterisk-users] G723 on asterisk 1.4.1   Jaswinder Singh
• [asterisk-users] Problem: Digium TDM400 with XOptionsFlex - LineBusy   Martin
• [asterisk-users] G723 on asterisk 1.4.1   Eric Wieling
• [asterisk-users] Unable to capture CallerID through Zap   mark morreny
• [asterisk-users] G723 on asterisk 1.4.1   Martin
• [asterisk-users] How to detect if a call is fax or not   Pete Kay
• [asterisk-users] Unable to capture CallerID through Zap   randulo
• [asterisk-users] How to detect if a call is fax or not   Gordon Henderson
• [asterisk-users] Unable to capture CallerID through Zap   Gordon Henderson
• [asterisk-users] Dialplan Help   ram
• [asterisk-users] No audio on Sangoma A104.   Alex Balashov
• [asterisk-users] Unable to capture CallerID through Zap   mark morreny
• [asterisk-users] Storing voicemail in mysql   mark morreny
• [asterisk-users] Storing voicemail in mysql   Al Baker
• [asterisk-users] zap callerid problem   Mark Quitoriano
• [asterisk-users] Calls to sip extensions not defined   Ricardo B
• [asterisk-users] Problems with calls in asterisk.   Catalin S.
• [asterisk-users] zap callerid problem   Martin
• [asterisk-users] Which command line is used to send emails to notify incoming voicemail ?   Olivier
• [asterisk-users] Best alternative for getting prompts recorded.   Olivier
• [asterisk-users] Access rights between AGI and Web server?   Steve Edwards
• [asterisk-users] Access rights between AGI and Web server?   Chris Carey
• [asterisk-users] Access rights between AGI and Web server?   Chris Carey
• [asterisk-users] Access rights between AGI and Web server?   Steve Edwards
• [asterisk-users] How to capture destination number when receive call through ZAP   mark morreny
• [asterisk-users] G723 on asterisk 1.4.1   Tilghman Lesher
• [asterisk-users] How to capture destination number when receive call through ZAP   Eric Wieling
• [asterisk-users] More Broadvoice woes. Who's fault could this be?   John Faubion
• [asterisk-users] BLF and Snom phones   asterisk at anime.net
• [asterisk-users] How to capture destination number when receive call through ZAP   Rob Hillis
• [asterisk-users] Unable to obtain dialed number through ZAP   mark morreny
• [asterisk-users] How to capture destination number when receive call through ZAP   mark morreny
• [asterisk-users] More Broadvoice woes. Who's fault could this be?   Outback Dingo
• [asterisk-users] Unable to obtain dialed number through ZAP   Gordon Henderson
• [asterisk-users] How to capture destination number when receive call through ZAP   Rob Hillis
• [asterisk-users] Getting Exec Format Error when running AGI call   mark morreny
• [asterisk-users] g729 license for debian etch   kitti jaisong
• [asterisk-users] Greetings   raven sharkas
• [asterisk-users] Queue - Dynamic association   Rodrigo R Passos
• [asterisk-users] Calling extension from CLI?   Vincent
• [asterisk-users] Access rights between AGI and Web server?   Vincent
• [asterisk-users] Unable to obtain dialed number through ZAP   Andrew Kohlsmith (lists)
• [asterisk-users] More Broadvoice woes. Who's fault could this be?   Jon Miron
• [asterisk-users] Gold Mine CRM + Asterisk   Dovid B
• [asterisk-users] Gold Mine CRM + Asterisk   OCG Technical Support
• [asterisk-users] app_sms and smsq in germany   Tobias Wolf
• [asterisk-users] G723 on asterisk 1.4.1   Jared Smith
• [asterisk-users] More Broadvoice woes. Who's fault could thisbe?   John Faubion
• [asterisk-users] G.729 Copy Protection   Guilherme Loch Waltrick Góes
• [asterisk-users] More Broadvoice woes. Who's fault could thisbe?   Outback Dingo
• [asterisk-users] Dynamic meetme conference creation with Authenticate (Asterisk 1.4.0)   David Ruggles
• [asterisk-users] Problem with user regsitration and ldap on SVN version   Tilghman Lesher
• [asterisk-users] Getting Exec Format Error when running AGI call   Tilghman Lesher
• [asterisk-users] G.729 Copy Protection   Eric Wieling
• [asterisk-users] Dynamic meetme conference creation with Authenticate (Asterisk 1.4.0)   Tilghman Lesher
• [asterisk-users] app_sms and smsq in germany   Conrad Wood
• [asterisk-users] Dynamic meetme conference creation with Authenticate (Asterisk 1.4.0)   Richard Lyman
• [asterisk-users] Dynamic meetme conference creation withAuthenticate (Asterisk 1.4.0)   David Ruggles
• [asterisk-users] Dynamic meetme conference creation with Authenticate (Asterisk 1.4.0)   Richard Lyman
• [asterisk-users] G.729 Copy Protection   Guilherme Loch Waltrick Góes
• [asterisk-users] Slow compilation speed   Grygoriy Dobrovolskyy
• [asterisk-users] Calling extension from CLI?   Mojo with Horan & Company, LLC
• [asterisk-users] Storing voicemail in mysql   mark morreny
• [asterisk-users] Dynamic meetme conference creation with Authenticate (Asterisk 1.4.0)   Tilghman Lesher
• [asterisk-users] How to capture destination number when receive call through ZAP   Mojo with Horan & Company, LLC
• [asterisk-users] estimation on phone network capacity   mark morreny
• [asterisk-users] Problem: Digium TDM400 with XOptionsFlex - Solved   Thomas Klettke
• [asterisk-users] estimation on phone network capacity   Alex Balashov
• [asterisk-users] estimation on phone network capacity   Philipp Kempgen
• [asterisk-users] FYI about my Mona Vie business venture   BerkHolz, Steven
• [asterisk-users] estimation on phone network capacity   Greg Oliver
• [asterisk-users] estimation on phone network capacity   Steve Totaro
• [asterisk-users] OT: Re: Dynamic meetme conference creation with Authenticate (Asterisk 1.4.0)   Richard Lyman
• [asterisk-users] Access rights between AGI and Web server?   Mojo with Horan & Company, LLC
• [asterisk-users] Passing variables over IAX2 -- IAXVAR patch?   Chris Earle
• [asterisk-users] Newbie IVR: How to read() before playback() is finished?   Mojo with Horan & Company, LLC
• [asterisk-users] FYI about my Mona Vie business venture   BJ Weschke
• [asterisk-users] Access rights between AGI and Web server?   Vincent
• [asterisk-users] FYI about my Mona Vie business venture   Erik Anderson
• [asterisk-users] FYI about my Mona Vie business venture   John Novack
• [asterisk-users] FYI about my Mona Vie business venture   Brent Davidson
• [asterisk-users] Problem: Digium TDM400 with XOptionsFlex - Solved   John Novack
• [asterisk-users] g729 license for debian etch   Jeff Peeler
• [asterisk-users] FYI about my Mona Vie business venture   Drew Gibson
• [asterisk-users] estimation on phone network capacity   Hans Witvliet
• [asterisk-users] FYI about my Mona Vie business venture   Doug Lytle
• [asterisk-users] FYI about my Mona Vie business venture   Steve Prior
• [asterisk-users] FYI about my Mona Vie business venture   Yair Hakak
• [asterisk-users] FYI about my Mona Vie business venture - apology and rethink   BerkHolz, Steven
• [asterisk-users] Access rights between AGI and Web server?   Mojo with Horan & Company, LLC
• [asterisk-users] Passing variables over IAX2 -- IAXVAR patch?   Ron Arts
• [asterisk-users] Passing variables over IAX2 -- IAXVAR patch?   Al Baker
• [asterisk-users] Access rights between AGI and Web server?   Vincent
• [asterisk-users] FYI about my Mona Vie business venture   Mark Hamilton
• [asterisk-users] Passing variables over IAX2 -- IAXVAR patch?   Tilghman Lesher
• [asterisk-users] G.729 Copy Protection   Jacobson
• [asterisk-users] Passing variables over IAX2 -- IAXVAR patch?   Bill Hackensack
• [asterisk-users] More Broadvoice woes. Who's fault could thisbe?   Jon Miron
• [asterisk-users] Access rights between AGI and Web server?   Mojo with Horan & Company, LLC
• [asterisk-users] FYI about my Mona Vie business venture   C F
• [asterisk-users] Passing variables over IAX2 -- IAXVAR patch?   Watkins, Bradley
• [asterisk-users] G.729 Copy Protection   Ruben Zamora
• [asterisk-users] Passing variables over IAX2 -- IAXVAR patch?   Tilghman Lesher
• [asterisk-users] How to capture destination number when receive call through ZAP   Rob Hillis
• [asterisk-users] Passing variables over IAX2 -- IAXVAR patch?   Eric Wieling
• [asterisk-users] Unable to obtain dialed number through ZAP   Trevor Peirce
• [asterisk-users] Passing variables over IAX2 -- IAXVAR patch?   Tilghman Lesher
• [asterisk-users] Passing variables over IAX2 -- IAXVAR patch?   Bill Hackensack
• [asterisk-users] Unable to obtain dialed number through ZAP   Lacy Moore
• [asterisk-users] SIP carrier billing technicalities   Robert McNaught
• [asterisk-users] SIP carrier billing technicalities   Robert McNaught
• [asterisk-users] How to capture destination number when receive call through ZAP   Mojo with Horan & Company, LLC
• [asterisk-users] SIP carrier billing technicalities   Tom Moore
• [asterisk-users] FXS channel banks   Paul Hales
• [asterisk-users] Calls to sip extensions not defined   Mojo with Horan & Company, LLC
• [asterisk-users] Menuselect?   Kyle Gibbons
• [asterisk-users] SIP carrier billing technicalities   Grey Man
• [asterisk-users] Menuselect?   James M Kupernik
• [asterisk-users] Menuselect?   Paul Hales
• [asterisk-users] Access rights between AGI and Web server?   Tzafrir Cohen
• [asterisk-users] FYI about my Mona Vie business venture   Paul Hales
• [asterisk-users] Audio Problem...   Sneha Murganoor
• [asterisk-users] FYI about my Mona Vie business venture   Alex Balashov
• [asterisk-users] Redirect and free the channel   mark morreny
• [asterisk-users] SIP carrier billing technicalities   Alex Balashov
• [asterisk-users] Menuselect?   Rob Hillis
• [asterisk-users] SIP carrier billing technicalities   Luki
• [asterisk-users] Sip exten matching based on contact: sip header?   Mike Fedyk
• [asterisk-users] FXS channel banks   Lee, John (Sydney)
• [asterisk-users] Sip exten matching based on contact: sip header?   Grey Man
• [asterisk-users] New Release asterisk 1.6 Beta   Naveen Palani
• [asterisk-users] FXS channel banks   Paul Hales
• [asterisk-users] FXS channel banks   James Finstrom
• [asterisk-users] FXS channel banks   Lee, John (Sydney)
• [asterisk-users] FXS channel banks   Lee, John (Sydney)
• [asterisk-users] Problem with user regsitration and ldap on SVNversion   sylvain.desbureaux at orange-ftgroup.com
• [asterisk-users] Had it with Dell Garbage - HP Question   Al Baker
• [asterisk-users] Had it with Dell Garbage - HP Question   Steve Totaro
• [asterisk-users] Menuselect?   Grygoriy Dobrovolskyy
• [asterisk-users] Audio Problem...   Grygoriy Dobrovolskyy
• [asterisk-users] Getting Exec Format Error when running AGI call   Steve Davies
• [asterisk-users] BLF and Snom phones   Steve Davies
• [asterisk-users] Getting Exec Format Error when running AGI call   mark morreny
• [asterisk-users] force soft hangup   Vieri
• [asterisk-users] force soft hangup   Gordon Henderson
• [asterisk-users] force soft hangup   Steve Davies
• [asterisk-users] How to obtain SIPCHANINFO variables within custom application?   Mindaugas Kezys
• [asterisk-users] BLF and Snom phones   Yehavi Bourvine +972-8-9489444
• [asterisk-users] Delete voicemail messages on asterisk by replying to email   OCG Technical Support
• [asterisk-users] Problem with user regsitration and ldap on SVNversion   Tilghman Lesher
• [asterisk-users] CCM and multiple trunks   Aaron Fransen
• [asterisk-users] FXS channel banks   James Finstrom
• [asterisk-users] Asterisk parking hold and transferdigittimeout   Guido Hecken
• [asterisk-users] To what degree can Asterisk replace Cisco Unity?   Peter Pauly
• [asterisk-users] Problem with user regsitration and ldap onSVNversion   sylvain.desbureaux at orange-ftgroup.com
• [asterisk-users] How to capture destination number when receive call through ZAP   Håkan Källberg
• [asterisk-users] force soft hangup   Vieri
• [asterisk-users] force soft hangup   Vieri
• [asterisk-users] Auto-congest time for sip peers   Rizwan Hisham
• [asterisk-users] How to obtain SIPCHANINFO variables within custom application?   Tilghman Lesher
• [asterisk-users] force soft hangup   Tilghman Lesher
• [asterisk-users] voicemail.conf fromstring, emailbody - per context?   Chris Carey
• [asterisk-users] BLF and Snom phones   Steve Davies
• [asterisk-users] CCM and multiple trunks   Aaron Fransen
• [asterisk-users] CCM and multiple trunks   Dan Austin
• [asterisk-users] force soft hangup   Steve Davies
• [asterisk-users] voicemail.conf fromstring, emailbody - per context?   Johansson Olle E
• [asterisk-users] Slightly OT: Getting VOIP number into phone book   Jiffy Slides Leonard Burton
• [asterisk-users] force soft hangup   Vieri
• [asterisk-users] Application registration on Asterisk 1.4 and 1.6?   jonas boering
• [asterisk-users] Application registration on Asterisk 1.4 and 1.6?   Kevin P. Fleming
• [asterisk-users] [ [asterisk-ss7] libss7 2asterisk box   aymen warfalli
• [asterisk-users] Have problem with realtime sql   mark morreny
• [asterisk-users] How to obtain SIPCHANINFO variables within custom application?   Mindaugas Kezys
• [asterisk-users] force soft hangup   Vieri
• [asterisk-users] Slightly OT: Getting VOIP number into phone book   Chris Bagnall
• [asterisk-users] Slightly OT: Getting VOIP number into phone book   Lacy Moore
• [asterisk-users] RTP Payload Problem   Elliot Murdock
• [asterisk-users] Slightly OT: Getting VOIP number into phone book   Andrew Kohlsmith (lists)
• [asterisk-users] Send received fax to different email account   mark morreny
• [asterisk-users] Send received fax to different email account   Rodrigo Gonzalez
• [asterisk-users] RTP Payload Problem   Andres
• [asterisk-users] Slightly OT: Getting VOIP number into phone book   SIP
• [asterisk-users] RTP Payload Problem   Ex Vito
• [asterisk-users] Slightly OT: Getting VOIP number into phone book   Lacy Moore
• [asterisk-users] Have problem with realtime sql   Mike Fedyk
• [asterisk-users] SIP Domain Authentication   Robert Norton - SophTelecom LLC
• [asterisk-users] Have problem with realtime sql   mark morreny
• [asterisk-users] force soft hangup   Benny Amorsen
• [asterisk-users] Delete voicemail messages on asterisk by replying to email   OCG Technical Support
• [asterisk-users] Distorted Audio for incoming DTMF   Brent Davidson
• [asterisk-users] Had it with Dell Garbage - HP Question   Jesse Molina
• [asterisk-users] Asterisk parking hold and transferdigittimeout   Mojo with Horan & Company, LLC
• [asterisk-users] Had it with Dell Garbage - HP Question   Matthew Gibson
• [asterisk-users] Automatically reload/restart asterisk following IP change (dynamic IP)   OCG Technical Support
• [asterisk-users] Had it with Dell Garbage - HP Question   Al Baker
• [asterisk-users] Realtime replication!!!!!   Cheikhou DIAW
• [asterisk-users] FXS channel banks   Lee, John (Sydney)
• [asterisk-users] Menuselect?   Kyle Gibbons
• [asterisk-users] FYI about my Mona Vie business venture   Grygoriy Dobrovolskyy
• [asterisk-users] Menuselect?   Kyle Gibbons
• [asterisk-users] Realtime replication!!!!!   Edgar Guadamuz
• [asterisk-users] Menuselect?   Kyle Gibbons
• [asterisk-users] Realtime replication!!!!!   Tilghman Lesher
• [asterisk-users] FXS channel banks   James Finstrom
• [asterisk-users] Broadcast/Announce app   Justin Newman
• [asterisk-users] Had it with Dell Garbage - HP Question   Jesse Molina
• [asterisk-users] Broadcast/Announce app   Steve Edwards
• [asterisk-users] FXS channel banks   Lee, John (Sydney)
• [asterisk-users] Realtime replication!!!!!   Yehavi Bourvine +972-8-9489444
• [asterisk-users] FXS channel banks   Paul Hales
• [asterisk-users] Avantfax installation on Debian   Pete Kay
• [asterisk-users] SOLVED: Avantfax on Debian   Pete Kay
• [asterisk-users] Persistant Dialing   John Marvin
• [asterisk-users] Avantfax installation on Debian   Tzafrir Cohen
• [asterisk-users] how to detect redirect fax call   Mandeep Singh Bhabha
• [asterisk-users] what's a softphone can activer web browser   kitti jaisong
• [asterisk-users] Fax to DB   mark morreny
• [asterisk-users] BLF and Snom phones   Christian Stredicke
• [asterisk-users] UK GMT/BST settings   Adrian Marsh
• [asterisk-users] SIP Domain Authentication   Johansson Olle E
• [asterisk-users] UK GMT/BST settings   Adrian Marsh
• [asterisk-users] Realtime replication!!!!!   Al Baker
• [asterisk-users] Asterisk parking hold and transferdigittimeo ut   Guido Hecken
• [asterisk-users] BLF and Snom phones   Benny Amorsen
• [asterisk-users] Realtime replication!!!!!   Ricardo Carvalho
• [asterisk-users] BLF and Snom phones   Benny Amorsen
• [asterisk-users] Asterisk parking hold and transferdigittimeout   Jared Smith
• [asterisk-users] Test   Lima Martin
• [asterisk-users] Had it with Dell Garbage - HP Question   Darren Wright
• [asterisk-users] BLF and Snom phones   Johansson Olle E
• [asterisk-users] BLF and Snom phones   Steve Davies
• [asterisk-users] UK GMT/BST settings   Robert Lister
• [asterisk-users] BLF and Snom phones   Christian Stredicke
• [asterisk-users] Dialing off-hook with Polycom SoundPoint IP 430   Brig C. McCoy
• [asterisk-users] Realtime replication!!!!!   Tilghman Lesher
• [asterisk-users] UK GMT/BST settings   Adrian Marsh
• [asterisk-users] Dialing off-hook with Polycom SoundPoint IP 430   Jerry Jones
• [asterisk-users] Access rights between AGI and Web server?   Jay R. Ashworth
• [asterisk-users] Access rights between AGI and Web server?   Jay R. Ashworth
• [asterisk-users] Unable to obtain dialed number through ZAP   Jay R. Ashworth
• [asterisk-users] Menuselect?   Jason Parker
• [asterisk-users] Playing sound while talking   Artifex Maximus
• [asterisk-users] Dialing off-hook with Polycom SoundPoint IP 430   Brig C. McCoy
• [asterisk-users] Menuselect?   Tzafrir Cohen
• [asterisk-users] Unable to obtain dialed number through ZAP   John Novack
• [asterisk-users] Asterisk parking hold and transferdigittimeo ut   Guido Hecken
• [asterisk-users] Getting Exec Format Error when running AGI call   Jay R. Ashworth
• [asterisk-users] Asterisk parking hold and transferdigittimeo ut   Mojo with Horan & Company, LLC
• [asterisk-users] Application registration on Asterisk 1.4 and 1.6?   jonas boering
• [asterisk-users] Asterisk parking hold and transferdigittimeo ut   Guido Hecken
• [asterisk-users] customizing faxrcvd in PHP   mark morreny
• [asterisk-users] customizing faxrcvd in PHP   Tzafrir Cohen
• [asterisk-users] customizing faxrcvd in PHP   Kevin P. Fleming
• [asterisk-users] AGI-python script   equis software
• [asterisk-users] Had it with Dell Garbage - HP Question   shadowym
• [asterisk-users] Had it with Dell Garbage - HP Question   shadowym
• [asterisk-users] AGI-python script   Steve Edwards
• [asterisk-users] EvolutionPBX from Intuitive Voice   Bill Andersen
• [asterisk-users] Problem: Digium TDM400 with XOptionsFlex - Solved   Matthew Fredrickson
• [asterisk-users] DTMF suddenly stopped working on SIP channel   David Nedved
• [asterisk-users] DTMF suddenly stopped working on SIP channel   Eric Wieling
• [asterisk-users] DTMF suddenly stopped working on SIP channel   Darrick Hartman (lists)
• [asterisk-users] Strange RTP problem...   Carlos Chavez
• [asterisk-users] voicemail.conf fromstring, emailbody - per context?   Chris Carey
• [asterisk-users] Got SIP response 406 "Not Acceptable"   Al lists
• [asterisk-users] EvolutionPBX from Intuitive Voice   Paul Hales
• [asterisk-users] EvolutionPBX from Intuitive Voice   Bill Andersen
• [asterisk-users] Menuselect?   Paul Hales
• [asterisk-users] Got SIP response 406 "Not Acceptable"   Adrià Vidal
• [asterisk-users] Broadcast/Announce app   Mojo with Horan & Company, LLC
• [asterisk-users] Got SIP response 406 "Not Acceptable"   Al lists
• [asterisk-users] How to change Internal and external callerid   carl Lougher
• [asterisk-users] problem about voice when using TDM2400p with VPMADT032 echo canceller module   Vu AnhTuan
• [asterisk-users] Zapata Tormenta 2   Cody Jarrett
• [asterisk-users] Zapata Tormenta 2   Tzafrir Cohen
• [asterisk-users] Asterisk not hanging up after voicemail   mark morreny
• [asterisk-users] ADPCM codec and IAXy device   bilal ghayyad
• [asterisk-users] Calling users to the external domain using Asterisk   Aadilkhan Maniyar
• [asterisk-users] SPA400 vs Rhino/Digium card   Juan Antonio Ibañez Santorum
• [asterisk-users] IAXy device   bilal ghayyad
• [asterisk-users] SPA400 vs Rhino/Digium card   Steve Totaro
• [asterisk-users] Asterisk not hanging up after voicemail   Rob Hillis
• [asterisk-users] SPA400 vs Rhino/Digium card   Juan Antonio Ibañez Santorum
• [asterisk-users] SPA400 vs Rhino/Digium card   Grygoriy Dobrovolskyy
• [asterisk-users] SPA400 vs Rhino/Digium card   Louwrens Benadé
• [asterisk-users] Upgraded to 1.4.18 (from 1.2.27) and channels dropping on Zaptel and SIP   David Nedved
• [asterisk-users] Problem with socket_process: Call rejected by 127.0.0.1: Busy   mark morreny
• [asterisk-users] Had it with Dell Garbage - HP Question   Al Baker
• [asterisk-users] Question about PCI Slots for DIGIUMs Boards   Al Baker
• [asterisk-users] what's a softphone can activer web browser   Marc Charbonneau
• [asterisk-users] Cisco 7971   J. Oquendo
• [asterisk-users] Got SIP response 406 "Not Acceptable"   Adrià Vidal
• [asterisk-users] Had it with Dell Garbage - HP Question   Al Baker
• [asterisk-users] Astlinux Friday Mar 28 @12 Noon EDT VoIP Users Conference   randulo
• [asterisk-users] Unable to establish handshaking with fax machine   mark morreny
• [asterisk-users] Upgraded to 1.4.18 (from 1.2.27) and channels dropping on Zaptel and SIP   Darrick Hartman (lists)
• [asterisk-users] Question about PCI Slots for DIGIUMs Boards   Steve Totaro
• [asterisk-users] Upgraded to 1.4.18 (from 1.2.27) and channels dropping on Zaptel and SIP   Steve Totaro
• [asterisk-users] Had it with Dell Garbage - HP Question   Darren Wright
• [asterisk-users] Question about PCI Slots for DIGIUMs Boards   Jared Smith
• [asterisk-users] IAXy device   Eric Wieling
• [asterisk-users] AGI-python script   equis software
• [asterisk-users] Upgraded to 1.4.18 (from 1.2.27) and channels dropping on Zaptel and SIP   David Nedved
• [asterisk-users] Playing sound while talking   Artifex Maximus
• [asterisk-users] IAXy device   Sean Dennis
• [asterisk-users] Zapata Tormenta 2   Cody Jarrett
• [asterisk-users] Zapata Tormenta 2   Tzafrir Cohen
• [asterisk-users] DTMF suddenly stopped working on SIP channel   David Nedved
• [asterisk-users] Question about PCI Slots for DIGIUMs Boards   Horwich IT Services (Godwin Stewart)
• [asterisk-users] Had it with Dell Garbage - HP Question   Joshua Kinard
• [asterisk-users] Upgraded to 1.4.18 (from 1.2.27) and channels dropping on Zaptel and SIP   Steve Davies
• [asterisk-users] Question about PCI Slots for DIGIUMs Boards   Al Baker
• [asterisk-users] Had it with Dell Garbage - HP Question   Michiel van Baak
• [asterisk-users] Question about PCI Slots for DIGIUMs Boards   Al Baker
• [asterisk-users] callers in queue passed to agents who accept only one call at a time   Vieri
• [asterisk-users] Question about PCI Slots for DIGIUMs Boards   Watkins, Bradley
• [asterisk-users] Asterisk not picking up (some) calls due to zaptel detecting and clearing alarms   Gonzalo Servat
• [asterisk-users] Asterisk not picking up (some) calls due to zaptel detecting and clearing alarms   Tzafrir Cohen
• [asterisk-users] IAXy device   Mojo with Horan & Company, LLC
• [asterisk-users] ADPCM codec and IAXy device   Mojo with Horan & Company, LLC
• [asterisk-users] IAXy device   Eric Wieling
• [asterisk-users] Asterisk not picking up (some) calls due to zaptel detecting and clearing alarms   Gonzalo Servat
• [asterisk-users] IAXy device   Mojo with Horan & Company, LLC
• [asterisk-users] Question about PCI Slots for DIGIUMs Boards   Al Baker
• [asterisk-users] Question about PCI Slots for DIGIUMs Boards   Jared Smith
• [asterisk-users] Upgraded to 1.4.18 (from 1.2.27) and channels dropping on Zaptel and SIP   Jay R. Ashworth
• [asterisk-users] Calling users to the external domain using Asterisk   Mojo with Horan & Company, LLC
• [asterisk-users] Upgraded to 1.4.18 (from 1.2.27) and channels dropping on Zaptel and SIP   Steve Totaro
• [asterisk-users] Question about PCI Slots for DIGIUMs Boards   Steve Totaro
• [asterisk-users] IAXy device   Steve Totaro
• [asterisk-users] Upgraded to 1.4.18 (from 1.2.27) and channels dropping on Zaptel and SIP   Jay R. Ashworth
• [asterisk-users] Help with cisco 7960 phone   Jerry Geis
• [asterisk-users] Star Wars Echo Sound   Rob Schall
• [asterisk-users] Star Wars Echo Sound   Joshua Kinard
• [asterisk-users] Star Wars Echo Sound   Chris Earle
• [asterisk-users] Problem when leaving voicemail   Richard Open Source
• [asterisk-users] Help with cisco 7960 phone   Peder at NetworkOblivion
• [asterisk-users] Star Wars Echo Sound   Conrad Wood
• [asterisk-users] Cisco 7971   Matthew Gibson
• [asterisk-users] Developer Conference, Aug 5-7, Chicago   Michael Collins
• [asterisk-users] IAXy device   Andreas van dem Helge
• [asterisk-users] callers in queue passed to agents who accept only one call at a time   Atis Lezdins
• [asterisk-users] callers in queue passed to agents who accept only one call at a time   Rodrigo Gonzalez
• [asterisk-users] Unable to establish handshaking with fax machine   Lee Howard
• [asterisk-users] SPA-962+ SPA-932- blf function   John Meksavan
• [asterisk-users] problem about voice when using TDM2400p with VPMADT032 echo canceller module   Vu AnhTuan
• [asterisk-users] Newbie Polycom: DHCP/boot server supporting 2 models of phones   Lee, John (Sydney)
• [asterisk-users] Newbie Polycom: DHCP/boot server supporting 2 models of phones   Robert McNaught
• [asterisk-users] SPA-962+ SPA-932- blf function   Rob Hillis
• [asterisk-users] Newbie Polycom: DHCP/boot server supporting 2 models of phones   Rob Hillis
• [asterisk-users] Newbie Polycom: DHCP/boot server supporting 2 models of phones   Paul Hales
• [asterisk-users] Newbie Polycom: DHCP/boot server supporting 2 models of phones   Lee, John (Sydney)
• [asterisk-users] Need help with voicemail odbc   mark morreny
• [asterisk-users] IAXy device   randulo
• [asterisk-users] Call deflection on ISDN PRI in Sweden   Hanna Wallin
• [asterisk-users] IAX user register problem   Mian M Asif
• [asterisk-users] Calling users to the external domain using Asterisk   Ricardo Carvalho
• [asterisk-users] PRI error cause hangup calls   voip crazy
• [asterisk-users] sip.conf setvar option   Marcus Hunger
• [asterisk-users] Calling users to the external domain usingAsterisk   Aadilkhan Maniyar
• [asterisk-users] problem about voice when using TDM2400p with VPMADT032 echo canceller module   Jared Smith
• [asterisk-users] IAX user register problem   Mian M Asif
• [asterisk-users] SPA-962+ SPA-932- blf function   Jared Smith
• [asterisk-users] sip.conf setvar option   Jared Smith
• [asterisk-users] sip.conf setvar option   Johansson Olle E
• [asterisk-users] Calling users to the external domain usingAsterisk   Ricardo Carvalho
• [asterisk-users] sip.conf setvar option   Marcus Hunger
• [asterisk-users] wrong extension status when call-limit=1 is used   Vieri
• [asterisk-users] sip.conf setvar option   Jared Smith
• [asterisk-users] Two phones fail to agree on codec, asterisk at fault?   martin f krafft
• [asterisk-users] callers in queue passed to agents who accept only one call at a time   Vieri
• [asterisk-users] recommendable softphones / X-Lite / Zoiper for amd64?   martin f krafft
• [asterisk-users] sip.conf setvar option   Johansson Olle E
• [asterisk-users] wrong extension status when call-limit=1 is used   Johansson Olle E
• [asterisk-users] sip.conf setvar option   Marcus Hunger
• [asterisk-users] wrong extension status when call-limit=1 is used   Vieri
• [asterisk-users] how to register IAX user without password   Mian M Asif
• [asterisk-users] sip.conf setvar option   Johansson Olle E
• [asterisk-users] SPA-962+ SPA-932- blf function   Sean Dennis
• [asterisk-users] problem about voice when using TDM2400p with VPMADT032 echo canceller module   Matthew Fredrickson
• [asterisk-users] recommendable softphones / X-Lite / Zoiper for amd64?   zoa
• [asterisk-users] how to register IAX user without password   sanjay.rajdev at featherstoneinformatics.com
• [asterisk-users] jingle with Asterisk + PSTN   Ali Jawad
• [asterisk-users] IAXy device   Matthew Fredrickson
• [asterisk-users] SPA-962+ SPA-932- blf function   John Meksavan
• [asterisk-users] recommendable softphones / X-Lite / Zoiper for amd64?   Tim Nelson
• [asterisk-users] Newbie Polycom: DHCP/boot server supporting 2 models of phones   Steve Johnson
• [asterisk-users] Call deflection on ISDN PRI in Sweden   Matthew Fredrickson
• [asterisk-users] SPA-962+ SPA-932- blf function   Jared Smith
• [asterisk-users] sip.conf setvar option   Johansson Olle E
• [asterisk-users] recommendable softphones / X-Lite / Zoiper for amd64?   martin f krafft
• [asterisk-users] overlap calls from NT-BRI timeout problem   Thomas Winter
• [asterisk-users] Grandstream BLF and Call-limit   Peder at NetworkOblivion
• [asterisk-users] Question on Dynamic Queue and Agent   mark morreny
• [asterisk-users] SPA-962+ SPA-932- blf function   John Meksavan
• [asterisk-users] Question on Dynamic Queue and Agent   Mark Michelson
• [asterisk-users] voicemail custom greeting   Mark Quitoriano
• [asterisk-users] More info on my previous dynamic queue question   mark morreny
• [asterisk-users] Question on Dynamic Queue and Agent   mark morreny
• [asterisk-users] Question on Dynamic Queue and Agent   mark morreny
• [asterisk-users] Star Wars Echo Sound   Rob Schall
• [asterisk-users] Cisco 7971   J. Oquendo
• [asterisk-users] Newbie Polycom: DHCP/boot server supporting 2 models of phones   Scott Plante
• [asterisk-users] Polycom 650   Scott Plante
• [asterisk-users] how to register IAX user without password for any user   Mian M Asif
• [asterisk-users] Newbie Polycom: DND answered as "on the phone" instead of "not available"   Scott Plante
• [asterisk-users] Newbie Polycom: DND answered as "on the phone" instead of "not available"   Andreas van dem Helge
• [asterisk-users] Call deflection on ISDN PRI in Sweden   Andreas van dem Helge
• [asterisk-users] Star Wars Echo Sound   Anthony Francis
• [asterisk-users] Two phones fail to agree on codec, asterisk at fault?   Brent Davidson
• [asterisk-users] Star Wars Echo Sound   Rob Schall
• [asterisk-users] New Tutorial: Asterisk on EPIA VIA C3   Lenz
• [asterisk-users] Application registration on Asterisk 1.4 and 1.6?   jonas boering
• [asterisk-users] voicemail custom greeting   Marc Charbonneau
• [asterisk-users] Asterisk 1.4.19-rc4 and 1.6.0-beta7 Now Available   The Asterisk Development Team
• [asterisk-users] voicemail custom greeting   Mojo with Horan & Company, LLC
• [asterisk-users] Two phones fail to agree on codec, asterisk at fault?   Mojo with Horan & Company, LLC
• [asterisk-users] SPA-962+ SPA-932- blf function   Rob Hillis
• [asterisk-users] Finding iaxy's (iaxies?)   Steve Edwards
• [asterisk-users] voicemail custom greeting   Mark Quitoriano
• [asterisk-users] Question about PCI Slots for DIGIUMs Boards   Al Baker
• [asterisk-users] Question about PCI Slots for DIGIUMs Boards   Al Baker
• [asterisk-users] Had it with Dell Garbage - HP Question   Al Baker
• [asterisk-users] Had it with Dell Garbage - HP Question   Al Baker
• [asterisk-users] Had it with Dell Garbage - HP Question   Tzafrir Cohen
• [asterisk-users] Slightly OT: Getting VOIP number into phonebook   John van Oppen
• [asterisk-users] Finding iaxy's (iaxies?)   Stelios Koroneos
• [asterisk-users] Had it with Dell Garbage - HP Question   Michiel van Baak
• [asterisk-users] Cisco 7971   Matthew Gibson
• [asterisk-users] Had it with Dell Garbage - HP Question   Steve Totaro
• [asterisk-users] Question about PCI Slots for DIGIUMs Boards   Steve Totaro
• [asterisk-users] New Tutorial: Asterisk on EPIA VIA C3   Alan Lord
• [asterisk-users] Two phones fail to agree on codec, asterisk at fault?   Karsten Wemheuer
• [asterisk-users] Had it with Dell Garbage - HP Question   Darren Wright
• [asterisk-users] Asterisk and Digium HP thin clients compatibility   aymen warfalli
• [asterisk-users] freecall.com - has anybody tried it?   Charles Wang
• [asterisk-users] Two phones fail to agree on codec, asterisk at fault?   martin f krafft
• [asterisk-users] how to register IAX user without password for any user   Tim H. Panton
• [asterisk-users] Cisco 7971   Patrick
• [asterisk-users] freecall.com - has anybody tried it?   Matt
• [asterisk-users] e164.org   Grey Man
• [asterisk-users] e164.org   Armin Schindler
• [asterisk-users] Question about PCI Slots for DIGIUMs Boards   Nick Seraphin
• [asterisk-users] Cisco 7971   Matthew Gibson
• [asterisk-users] audio disappeared after ztdummy install   ronald ramos
• [asterisk-users] How many maximum SIP Registrations can Asterisk Handle   Abid Saleem Choudhary
• [asterisk-users] breaking DNID into country code, area code, and local code   mark morreny
• [asterisk-users] How many maximum SIP Registrations can Asterisk Handle   Grey Man
• [asterisk-users] breaking DNID into country code, area code, and local code   Anselm Martin Hoffmeister
• [asterisk-users] How many maximum SIP Registrations can Asterisk Handle   Zoa
• [asterisk-users] e164.org   Michiel van Baak
• [asterisk-users] Langugae issue   aymen warfalli
• [asterisk-users] Langugae issue   Mike Trest - Personal
• [asterisk-users] audio disappeared after ztdummy install   Eric Wieling
• [asterisk-users] audio disappeared after ztdummy install   Tzafrir Cohen
• [asterisk-users] Langugae issue   Anselm Martin Hoffmeister
• [asterisk-users] Langugae issue   Tzafrir Cohen
• [asterisk-users] Langugae issue   Mike Trest - Personal
• [asterisk-users] ztdummy / RTC error   Drew Miller
• [asterisk-users] Langugae issue   Tzafrir Cohen
• [asterisk-users] Need help with voicemail odbc   Tilghman Lesher
• [asterisk-users] audio disappeared after ztdummy install   Norman W. Franke
• [asterisk-users] audio disappeared after ztdummy install   Tzafrir Cohen
• [asterisk-users] Asterisk not picking up (some) calls due to zaptel detecting and clearing alarms   sean darcy
• [asterisk-users] New Tutorial: Asterisk on EPIA VIA C3   Darrick Hartman (lists)
• [asterisk-users] Tests in VMWare   Ein Bielaczyc
• [asterisk-users] audio disappeared after ztdummy install   ronald ramos
• [asterisk-users] Advice on queue setup needed please.   Mark Hamilton
• [asterisk-users] audio disappeared after ztdummy install   david
• [asterisk-users] Tests in VMWare   Alex Robar
• [asterisk-users] Star Wars Echo Sound   Matt Riddell
• [asterisk-users] Ping   Alexey Shimeshov
• [asterisk-users] Had it with Dell Garbage - HP Question   Al Baker
• [asterisk-users] Tests in VMWare   Tzafrir Cohen
• [asterisk-users] Had it with Dell Garbage - HP Question   Al Baker
• [asterisk-users] Had it with Dell Garbage - HP Question   Michiel van Baak
• [asterisk-users] Question about PCI Slots for DIGIUMs Boards   Al Baker
• [asterisk-users] Had it with Dell Garbage - HP Question   Tzafrir Cohen
• [asterisk-users] Clustering Meetme over multiple boxes?   Gopal krishnan
• [asterisk-users] New Tutorial: Asterisk on EPIA VIA C3   Alan Lord
• [asterisk-users] applicationmap in features.conf Asterisk 1.2 is ignoring DIAL tT options   Thomas Winter
• [asterisk-users] No voice in one direction, SIP, call manager   Martin Edlman
• [asterisk-users] New Tutorial: Asterisk on EPIA VIA C3   Lenz
• [asterisk-users] The most efficient way to know SIP phones IP addresses ?   Olivier
• [asterisk-users] Broken calls during conversation   Administrator TOOTAI
• [asterisk-users] New Tutorial: Asterisk on EPIA VIA C3   Tzafrir Cohen
• [asterisk-users] The most efficient way to know SIP phones IP addresses ?   Simon Elliston Ball
• [asterisk-users] The most efficient way to know SIP phones IP addresses ?   Olivier
• [asterisk-users] The most efficient way to know SIP phones IP addresses ?   Simon Elliston Ball
• [asterisk-users] New Tutorial: Asterisk on EPIA VIA C3   Alan Lord
• [asterisk-users] Cisco 7971   J. Oquendo
• [asterisk-users] ENUMLOOKUP question.   Aadilkhan Maniyar
• [asterisk-users] UK FXO hangup detection with a twist   Steve Davies
• [asterisk-users] transfer call   Borko Jankovic
• [asterisk-users] UK FXO hangup detection with a twist   Conrad Wood
• [asterisk-users] jingle with Asterisk + PSTN   Philippe Sultan
• [asterisk-users] Newbie Polycom: DND answered as "on the phone" instead of "not available"   Scott Plante
• [asterisk-users] How to customize voicemail greeting   mark morreny
• [asterisk-users] Tests in VMWare (was: Re: asterisk-users Digest, Vol 44, Issue 104)   Matthew Rubenstein
• [asterisk-users] UK FXO hangup detection with a twist   Mike Dent
• [asterisk-users] jingle with Asterisk + PSTN   Ali Jawad
• [asterisk-users] Problem with VoiceMailMain   mark morreny
• [asterisk-users] asterisk-users Digest, Vol 44, Issue 104   Norman Franke
• [asterisk-users] AGI-python script   equis software
• [asterisk-users] UK FXO hangup detection with a twist   Steve Davies
• [asterisk-users] jingle with Asterisk + PSTN   Philippe Sultan
• [asterisk-users] UK FXO hangup detection with a twist   David Boyd
• [asterisk-users] Simple Question   Drew Miller
• [asterisk-users] UK FXO hangup detection with a twist   Steve Davies
• [asterisk-users] asterisk-users Digest, Vol 44, Issue 104   Tzafrir Cohen
• [asterisk-users] Simple Question   sanjay.rajdev at featherstoneinformatics.com
• [asterisk-users] SIP proxy screwing up peer addresses.   martin f krafft
• [asterisk-users] New Tutorial: Asterisk on EPIA VIA C3   Gordon Henderson
• [asterisk-users] UK FXO hangup detection with a twist   Gordon Henderson
• [asterisk-users] AsterPas ObjectPascal Based FastAGI Server goes Open Source   Lee Jenkins
• [asterisk-users] Question about PCI Slots for DIGIUMs Boards   Peter Lindquist
• [asterisk-users] AsterPas ObjectPascal Based FastAGI Server goes Open Source   Lee Jenkins
• [asterisk-users] PRI cards, Digium vs. Sangoma   Octavio Ruiz
• [asterisk-users] How to give user a prompt before connecting the call   Pete Kay
• [asterisk-users] PRI cards, Digium vs. Sangoma   Doug Lytle
• [asterisk-users] Control of RTP open ports   Alejandro Cabrera Obed
• [asterisk-users] CDR Timestamps (cdr-custom)   Kelvin Williams
• [asterisk-users] PRI cards, Digium vs. Sangoma   Tzafrir Cohen
• [asterisk-users] How to give user a prompt before connecting thecall   Dean Collins
• [asterisk-users] New Tutorial: Asterisk on EPIA VIA C3   Alan Lord
• [asterisk-users] PRI cards, Digium vs. Sangoma   Octavio Ruiz
• [asterisk-users] Gentilini, Paul is out of the office.   PGentilini at intellicare.com
• [asterisk-users] PRI cards, Digium vs. Sangoma   Andres
• [asterisk-users] Cisco 7965 SIP Firmware   Razza
• [asterisk-users] New Tutorial: Asterisk on EPIA VIA C3   Gordon Henderson
• [asterisk-users] Cisco 7965 SIP Firmware   J. Oquendo
• [asterisk-users] Need some input for Quad T1 and channel banks.   Al lists
• [asterisk-users] Cisco 7965 SIP Firmware   Razza
• [asterisk-users] Cisco 7965 SIP Firmware   Greg Oliver
• [asterisk-users] Need some input for Quad T1 and channel banks.   Doug Lytle
• [asterisk-users] Need some input for Quad T1 and channel banks.   Don Pobanz
• [asterisk-users] Unicall + incomplete DNIS on international calls   Iván Reyes Tejera
• [asterisk-users] Need some input for Quad T1 and channel banks.   Steve Totaro
• [asterisk-users] How to give user a prompt before connecting thecall   Paul Hales
• [asterisk-users] Need some input for Quad T1 and channel banks.   Doug Lytle
• [asterisk-users] How to give user a prompt before connecting thecall   Jeremy Mann
• [asterisk-users] Need some input for Quad T1 and channel banks.   Lyle Giese
• [asterisk-users] Need some input for Quad T1 and channel banks.   Al lists
• [asterisk-users] How to give user a prompt before connecting thecall   Paul Hales

Last message date: Mon Mar 31 21:34:07 CDT 2008
Archived on: Mon Mar 31 21:34:19 CDT 2008

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