November 2006 Archives by date

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Starting: Wed Nov 1 00:45:03 MST 2006
Ending: Thu Nov 30 23:43:20 MST 2006
Messages: 2920

• [asterisk-users] Audiocodes MP-114 noise   Jason Kim
• [asterisk-users] Need help connecting Alcatel 4400 PBX to Asterisk   Shweta Jain
• [asterisk-users] Example Polycom function key config   Jamie Heckford
• [asterisk-users] FXO Cards vs. Channel bank with T1   Ed
• [asterisk-users] SIP realtime issues   Don
• [asterisk-users] Manager API - Originate Call - Need Help   Ehsan Khosrowshahi
• [asterisk-users] Manager API - Originate Call - Need Help   Conrad Wood
• [asterisk-users] Manager API - Originate Call - Need Help   mitcheloc
• [asterisk-users] Help me on Call parking   raviprakash sunkara
• [asterisk-users] ${CALLERIDNUM}   Scott Pinhorne
• [asterisk-users] Re: Manager API - Originate Call - Need Help   Tony Mountifield
• [asterisk-users] IAX Realtime MD5 authentication   Roland Ndaka Fru
• [asterisk-users] Help me on Call parking   Doug Lytle
• [asterisk-users] Re: Newbie Questions   Andrew Latham
• [asterisk-users] ${CALLERIDNUM}   C F
• [asterisk-users] ${CALLERIDNUM}   Michiel van Baak
• [asterisk-users] ${CALLERIDNUM}   Conrad Wood
• [asterisk-users] AEL2 - CUT function usage   yusuf
• [asterisk-users] siemens hipath interoperability - PRI/Q.SIG - cardrecommendation   Pavel Jezek
• [asterisk-users] ${CALLERIDNUM}   yusuf
• [asterisk-users] ${CALLERIDNUM}   scott
• [asterisk-users] wav format isn't compatible with Windows Media Player   René Christensen
• [asterisk-users] Re: Asterisk and Panasonic KX Model   ggonzalez at telviso.com.ar
• [asterisk-users] Neat Application for Text to Speech   Dean Collins
• [asterisk-users] Snom or Cisco Phones?   Tom Vile
• [asterisk-users] Fxo box for asterisk ?   f6hqz-m at hamwlan.net
• [asterisk-users] FXO Cards vs. Channel bank with T1   Ejay Hire
• [asterisk-users] T.38 faxing with spandsp and Grandstream HT.486   Henning Holtschneider
• [asterisk-users] Snom or Cisco Phones?   Rosli Sukri
• [SPAM HEADER] - Re: [asterisk-users] Snom or Cisco Phones? - Email found in subject   Cory Andrews
• [asterisk-users] Help me on Call parking   Ejay Hire
• [SPAM HEADER] - Re: [asterisk-users] Snom or Cisco Phones? - Email found in subject   Andrew Latham
• [asterisk-users] Re: Newbie Questions - Grandstorm phones?   Ken Williams
• [asterisk-users] compilation problem with asterisk-addons   Russell Bryant
• [asterisk-users] Which IP phones have best voice quality, preferably under $150   Zeeshan Zakaria
• [SPAM HEADER] - RE: [asterisk-users] Re: Newbie Questions - Grandstorm phones? - Email found in subject   Cory Andrews
• [asterisk-users] Asterisk manager   Ed Nuñez
• [SPAM HEADER] - [asterisk-users] Which IP phones have best voice quality, preferably under $150 - Email found in subject   Cory Andrews
• [asterisk-users] Upgrading from 1.0.9 to 1.2.6   Matt
• [asterisk-users] DTMF over IAX   Jason Walker
• [asterisk-users] Which IP phones have best voice quality, preferably under $150   Joe Dennick
• [asterisk-users] ${CALLERIDNUM}   Ira
• [asterisk-users] Snom or Cisco Phones?   Jessee J Holmes
• [asterisk-users] Re: Newbie Questions - Grandstorm phones?   Tom Vile
• [asterisk-users] Asterisk manager   Ed Nuñez
• [asterisk-users] Audiocodes MP-114 noise   Jessee J Holmes
• [asterisk-users] Which IP phones have best voice quality, preferably under $150   Eric "ManxPower" Wieling
• [asterisk-users] Upgrading from 1.0.9 to 1.2.6   Eric "ManxPower" Wieling
• AW: [asterisk-users] Which IP phones have best voice quality, preferably under $150   Christian Stredicke
• [asterisk-users] Still no CLI in 1.4 branch (OSX)   Martin Joseph
• [asterisk-users] Which IP phones have best voice quality, preferably under $150   Bryan Mahin
• [asterisk-users] Re: Asterisk both behind a NAT and outside at the same time   Benny Amorsen
• [asterisk-users] DTMF over IAX   Andrew Joakimsen
• [asterisk-users] Which IP phones have best voice quality, preferably under $150   Kristian Kielhofner
• [asterisk-users] Which IP phones have best voice quality, preferably under $150   Todd- Asterisk
• [asterisk-users] Re: DTMF over IAX   Steven
• [asterisk-users] Re: Newbie Questions - Grandstorm phones?   Jason Walker
• [asterisk-users] Polycom Managment tools   Neider, Clint
• [asterisk-users] Cisco 7960 password/shared secret problem --- Related to OS X ?   Mark Engelhardt
• [asterisk-users] simultaneous ring - call groups or queues orsomething else?   Stephen Bosch
• [asterisk-users] Re: DTMF over IAX   Martin Joseph
• [asterisk-users] FXO Cards vs. Channel bank with T1   Tzafrir Cohen
• [asterisk-users] Re: 1.4 branch on OSX?   Joshua Colp
• [asterisk-users] Remote-Party-Id and Attended Transfers   Douglas Garstang
• [asterisk-users] Upgrading from 1.0.9 to 1.2.6   Matt
• [asterisk-users] Re: Opinions on the best wholesale origination/term providers   Andrew Joakimsen
• [asterisk-users] Java WEB Phone   Vladimir Montealegre Estailes
• [asterisk-users] Diva server 4bri and Portuguese BRI   Pedro Silva
• [asterisk-users] Re: Newbie Questions - Grandstorm phones?   Henry.L.Coleman
• [asterisk-users] Which IP phones have best voice quality, preferably under $150   Henry.L.Coleman
• [asterisk-users] a2billing   Jeremy McNamara
• [asterisk-users] imap on debian   Tzafrir Cohen
• [asterisk-users] Still no CLI in 1.4 branch (OSX)   Joshua Colp
• [asterisk-users] simultaneous ring - call groups or queues orsomething else?   Tzafrir Cohen
• [asterisk-users] FXO Cards vs. Channel bank with T1   Jay R. Ashworth
• [asterisk-users] Remote-Party-Id and Attended Transfers   Douglas Garstang
• [asterisk-users] Re: Asterisk both behind a NAT and outside at the same time   Brad Templeton
• [asterisk-users] Which IP phones have best voice quality, preferably under $150   Zeeshan Zakaria
• [asterisk-users] Opinions on the best wholesale origination/term providers   Marcel Eric Loiselle
• [asterisk-users] DTMF over IAX   Eric "ManxPower" Wieling
• [asterisk-users] Upgrading from 1.0.9 to 1.2.6   Eric "ManxPower" Wieling
• [asterisk-users] Re: 1.4 branch on OSX?   Martin Joseph
• [asterisk-users] Remote-Party-Id and Attended Transfers   Eric "ManxPower" Wieling
• [asterisk-users] Re: Still no CLI in 1.4 branch (OSX)   Martin Joseph
• [asterisk-users] Java Web Phone   Vladimir Montealegre Estailes
• [asterisk-users] Diva server 4bri and Portuguese BRI   Armin Schindler
• [asterisk-users] Polycom Managment tools   Kristian Kielhofner
• [asterisk-users] Which IP phones have best voice quality, preferably under $150   Zeeshan Zakaria
• [asterisk-users] Polycom Managment tools   Kristian Kielhofner
• [asterisk-users] Can I use Realtime entries to do multiple registers to same trunk/peer   Tom Browning
• [asterisk-users] simultaneous ring - call groups or queues orsomething else?   John Novack
• [asterisk-users] TE110P Card Little help   Julian Varanini
• [asterisk-users] imap on debian   Michiel van Baak
• [asterisk-users] Opinions on the best wholesale origination/termproviders   Ron McLeod
• [asterisk-users] connecting internal line with external line   Ekkard Gerlach
• [asterisk-users] Sound breaking. Because of Tormenta2 PRI Interface Card or something else   Zeeshan Zakaria
• [asterisk-users] Upgrading from 1.0.9 to 1.2.6   Matt
• [asterisk-users] Echo Issues   Matt
• [asterisk-users] Remote-Party-Id and Attended Transfers   Kevin Collins
• [asterisk-users] Snom or Cisco Phones?   cp
• [asterisk-users] Re: Newbie Questions - Grandstorm phones?   Zeeshan Zakaria
• [asterisk-users] Java Web Phone   Dean Collins
• [asterisk-users] Realtime, DUNDi and regexten   Douglas Garstang
• [asterisk-users] Remote-Party-Id and Attended Transfers   Douglas Garstang
• [asterisk-users] Re: Newbie Questions - Grandstorm phones?   Henry.L.Coleman
• [asterisk-users] PURE OUTBOUND setup (how do I proceed from here?)   Zak Kinion
• [asterisk-users] a2billing   Khaled
• [asterisk-users] Fwd: Benachrichtung zum +ANw-bermittlungsstatus (Fehlgeschlagen)   Andrew Joakimsen
• [asterisk-users] a2billing   Khaled
• [asterisk-users] Java Web Phone   Carlos Rojas
• [asterisk-users] Java Web Phone   Al Bochter
• [asterisk-users] IAX problem   Itamar Lavender
• [asterisk-users] Snom or Cisco Phones?   Klaverstyn, David C
• [asterisk-users] Two Sipura 3000s   Marlin Unruh
• [asterisk-users] Re: Asterisk and Panasonic KX Model   C F
• [asterisk-users] New Dell range   Paul Hales
• [asterisk-users] Problem with libpri?   equis software
• [asterisk-users] channel.c: Unable to request channel ZAP   Forrest Beck
• [asterisk-users] My Phone Review- Large Scale Corp Deployment.   Matthew Mackes (Webmail)
• [asterisk-users] Java Web Phone   Dean Collins
• [asterisk-users] PURE OUTBOUND setup (how do I proceed from here?)   William Piper
• [asterisk-users] simultaneous ring - call groups or queuesorsomething else?   Dovid B
• [asterisk-users] My Phone Review- Large Scale Corp Deployment.   Eddie Johnson Jr
• [asterisk-users] echo with spa-3000   James Harper
• [asterisk-users] Asterisk manager   Dovid B
• [asterisk-users] My Phone Review- Large Scale Corp Deployment.   Steve Edwards
• [asterisk-users] Realtime, DUNDi and regexten   Andrew Joakimsen
• [SPAM??] Re: [asterisk-users] My Phone Review- Large Scale Corp Deployment.   Matthew Mackes (Webmail)
• [asterisk-users] My Phone Review- Large Scale Corp Deployment.   Matthew Mackes (Webmail)
• [asterisk-users] My Phone Review- Large Scale Corp Deployment.   Maxx Lobo
• [asterisk-users] Videoconferencing solutions with Asterisk-   Maxx Lobo
• [asterisk-users] channel.c: Unable to request channel ZAP   Shweta Jain
• [asterisk-users] Asterisk Manager and Ruby   Rajkumar S
• [SPAM??] Re: [asterisk-users] My Phone Review- Large Scale Corp Deployment.   Stephen Davies
• [SPAM??] Re: [asterisk-users] My Phone Review- Large Scale Corp Deployment.   Matthew Mackes (Webmail)
• [asterisk-users] Two Sipura 3000s   Leo Ann Boon
• [asterisk-users] Using asterisk as a call router between pbxs   Christopher Chan
• [asterisk-users] My Phone Review- Large Scale Corp Deployment.   Steve Edwards
• [asterisk-users] Polycom 601 Phone can not find TFTP server   Klaverstyn, David C
• [asterisk-users] My Phone Review- Large Scale Corp Deployment.   Steve Edwards
• [asterisk-users] SIP v IAX2   Jon Farmer
• [asterisk-users] IAX Realtime MD5 authentication   Roland Ndaka Fru
• [asterisk-users] FXO Cards vs. Channel bank with T1   Ed
• [asterisk-users] Asterisk Manager and Ruby   snacktime
• [asterisk-users] mpg123 new version   Frank Liu
• [asterisk-users] Sound breaking. Because of Tormenta2 PRI Interface Card or something else   Tzafrir Cohen
• [asterisk-users] Re: Asterisk Call Statistics   Tomislav Parčina
• [asterisk-users] Re: Asterisk Call Statistics   Tomislav Parčina
• [asterisk-users] Thomson ST2030 and Asterisk BLF   Alberto Pastore
• [asterisk-users] My Phone Review- Large Scale Corp Deployment.   Olivier
• [asterisk-users] ZAPtel channel dance   Florian Hars
• [asterisk-users] IAX problem   Tim Panton
• [asterisk-users] Sound breaking. Because of Tormenta2 PRI Interface Card or something else   Tim Panton
• [asterisk-users] Which IP phones have best voice quality, preferably under $150   Tim Panton
• [asterisk-users] Auto dial out and auto answer   Michel
• [asterisk-users] ZAPtel channel dance   Tzafrir Cohen
• [asterisk-users] echo with spa-3000   Dave Cotton
• [asterisk-users] Extending a call limited by L in Dial app   Rajkumar S
• [asterisk-users] My Phone Review- Large Scale Corp Deployment.   Matthias Fechner
• [asterisk-users] Example Polycom function key config   Jamie Heckford
• [asterisk-users] Sound breaking. Because of Tormenta2 PRI Interface Card or something else   Zeeshan Zakaria
• [asterisk-users] echo with spa-3000   James Harper
• [asterisk-users] sound-files not playing?   Evert
• [asterisk-users] Wait for an extension and dial. Why does this not work?   Evert
• [asterisk-users] Macro variables and redirects   Wildheart
• [asterisk-users] blindtransfer and initiator hangup   Vitaly Oborsky
• [asterisk-users] Diva server 4bri and Portuguese BRI   Pedro Silva
• [asterisk-users] Lucent TNT Help   Corey Frang
• [asterisk-users] ZAPtel channel dance   Bob Chiodini
• [asterisk-users] How to clear trixbox configuration   Pedro Silva
• [asterisk-users] Can some moderator kick this person out of the list   Zeeshan Zakaria
• [asterisk-users] VM Language   Al Bochter
• [asterisk-users] Which IP phones have best voice quality, preferably under $150   Bob Chiodini
• [asterisk-users] Re: ${CALLERIDNUM}   Benny Amorsen
• [asterisk-users] Videoconferencing solutions with Asterisk-   Dean Collins
• [asterisk-users] Example Polycom function key config   Doug Lytle
• [asterisk-users] SIP v IAX2   Henry.L.Coleman
• [asterisk-users] Lucent TNT Help   Don
• [asterisk-users] ZAPtel channel dance   Tzafrir Cohen
• [asterisk-users] connecting internal line with external line   Ekkard Gerlach
• [asterisk-users] Error installing asterisk, module zaptel not found   phoneman at wildblue.net
• [asterisk-users] ZAPtel channel dance   Florian Hars
• [asterisk-users] SIP v IAX2   Al Bochter
• [asterisk-users] Asterisk and Panasonic KX Model   ggonzalez at telviso.com.ar
• [asterisk-users] How to determine which version is running   René Christensen
• [asterisk-users] Re: Java Web Phone   Steven
• [asterisk-users] SIP v IAX2   Dean Collins
• [asterisk-users] Auto dial out and auto answer   Dovid B
• [asterisk-users] Java Web Phone   Dovid B
• [asterisk-users] Subscriptions and call back on busy problems with Snom phones   Andreas Haardt
• [asterisk-users] How to determine which version is running   Eric "ManxPower" Wieling
• [asterisk-users] Example Polycom function key config   Jamie Heckford
• [asterisk-users] Re: Java Web Phone   Dean Collins
• [asterisk-users] SIP v IAX2   Al Bochter
• [asterisk-users] How to determine which version is running   Tzafrir Cohen
• [asterisk-users] light web user interface   Jonathan Rivera
• [asterisk-users] Example Polycom function key config   Doug Lytle
• [asterisk-users] VOIP Bandwidth questions   mail-lists
• [asterisk-users] IAX problem   Itamar Lavender
• [asterisk-users] SIP v IAX2   Dean Collins
• [asterisk-users] Grandstream HandyTone-488 with Asterisk ?   Noc Phibee
• [asterisk-users] Running asterisk with 'sudo'   Asterisk
• [asterisk-users] AGI Problems   Jordan Kirby
• [asterisk-users] AstLinux 0.4.4 Released!   Kristian Kielhofner
• [asterisk-users] Java Web Phone   Guillermo Salas M.
• [asterisk-users] Snom or Cisco Phones?   Jessee J Holmes
• [asterisk-users] Asterisk architecure   jezzzz .
• [asterisk-users] Static Realtime Select from Database   Douglas Garstang
• [asterisk-users] Polycom 601 Phone can not find TFTP server   Neider, Clint
• [asterisk-users] Polycom 601 Phone can not find TFTP server   Jessee J Holmes
• [asterisk-users] Realtime, DUNDi and regexten   Aaron Daniel
• [asterisk-users] SIP v IAX2   maka
• [asterisk-users] Running asterisk with 'sudo'   Guillermo Salas M.
• [asterisk-users] SIP v IAX2   Tim Panton
• [asterisk-users] Re: Newbie Questions - Grandstorm phones?   Gordon Henderson
• [asterisk-users] Re: Grandstream HandyTone-488 with Asterisk ?   Martin Joseph
• [asterisk-users] Re: VOIP Bandwidth questions   Martin Joseph
• [asterisk-users] Problem with MYSQL commands in dialplan   Mike
• AW: [asterisk-users] Snom or Cisco Phones?   mail-lists
• [asterisk-users] echo with spa-3000   Bruce Ferrell
• [asterisk-users] Running asterisk with 'sudo'   Tod Detre (CampusEAI Consortium)
• [asterisk-users] Voicemail issues   Jason Walker
• [asterisk-users] AstLinux 0.4.4 Released!   Barry Fawthrop
• [asterisk-users] Polycom 601 Phone can not find TFTP server   Mailing List
• [asterisk-users] testing   Forum
• [asterisk-users] Asterisk architecure   Vikki
• [asterisk-users] SIP v IAX2   Dean Collins
• [asterisk-users] Out Dial Interface for Asterisk   Shawn Kelley
• [asterisk-users] light web user interface   Curt Shaffer
• [asterisk-users] regexten & regcontext broken for SIP?   Andrew Joakimsen
• [asterisk-users] Still no CLI in 1.4 branch (OSX)   Tim Panton
• [asterisk-users] AGI Problems   Eric "ManxPower" Wieling
• [asterisk-users] Tampa Bay Asterisk Users Meetup on Monday   Matt Florell
• [asterisk-users] Re: VOIP Bandwidth questions   mail-lists
• [asterisk-users] Polycom latest version   Mike
• [asterisk-users] VOIP Bandwidth questions   Erick Perez
• [asterisk-users] VOIP Bandwidth questions   Erick Perez
• [asterisk-users] Re: VOIP Bandwidth questions   Vikki
• [asterisk-users] Realtime, DUNDi and regexten   Michael Collins
• [asterisk-users] Asterisk 1.2.16 AIX2 - SIP Attended transfer   David Parcerisa
• [asterisk-users] Out Dial Interface for Asterisk   Mojo with Horan & Company, LLC
• [asterisk-users] Out Dial Interface for Asterisk   Julian Lyndon-Smith
• [asterisk-users] Polycom 501 supports now FTPS?   Mike
• [asterisk-users] Error installing asterisk, module zaptel not found   Erick Perez
• [asterisk-users] Out Dial Interface for Asterisk   Conrad Wood
• [asterisk-users] regexten & regcontext broken for SIP?   Bruce Reeves
• [asterisk-users] regexten & regcontext broken for SIP?   Watkins, Bradley
• [asterisk-users] ZAPtel channel dance   Bob Chiodini
• [asterisk-users] Polycom latest version   Jessee J Holmes
• [asterisk-users] VOIP Bandwidth questions   Jorge Alayon
• [asterisk-users] Polycom latest version   Eric "ManxPower" Wieling
• [asterisk-users] Re: VOIP Bandwidth questions   Eric "ManxPower" Wieling
• [asterisk-users] VOIP Bandwidth questions   Steve Kennedy
• [asterisk-users] Polycom latest version   Bruce Reeves
• [asterisk-users] Out Dial Interface for Asterisk   Mojo with Horan & Company, LLC
• [asterisk-users] Out Dial Interface for Asterisk   Héctor Maldonado
• [asterisk-users] regexten & regcontext broken for SIP?   Watkins, Bradley
• Recall: [asterisk-users] regexten & regcontext broken for SIP?   Watkins, Bradley
• [asterisk-users] echo with spa-3000   James Harper
• [asterisk-users] regexten & regcontext broken for SIP?   Andrew Joakimsen
• [asterisk-users] regexten & regcontext broken for SIP?   Bruce Reeves
• [asterisk-users] Realtime, DUNDi and regexten   Andrew Joakimsen
• [asterisk-users] Asterisk architecture   jezzzz .
• [asterisk-users] Re: ${CALLERIDNUM}   Leo Ann Boon
• [asterisk-users] DTMF over IAX   Leo Ann Boon
• [asterisk-users] My Phone Review- Large Scale Corp Deployment.   Vitalie Apostu
• [asterisk-users] Re: Newbie Questions - Grandstorm phones?   Vitalie Apostu
• [asterisk-users] SIP v IAX2   maka
• [asterisk-users] Re: VOIP Bandwidth questions   Julio Arruda
• [asterisk-users] a extension intentionally dropped in favor of * ?   Damon Estep
• [asterisk-users] How to clear trixbox configuration   Zeeshan Zakaria
• [asterisk-users] My Phone Review- Large Scale Corp Deployment.   Matthew Mackes (Webmail)
• AW: [asterisk-users] Snom or Cisco Phones?   Zeeshan Zakaria
• [asterisk-users] Java Web Phone   Brad Templeton
• [asterisk-users] Two Sipura 3000s   Joseph
• [asterisk-users] mpg123 new version   Eddie Johnson Jr
• [asterisk-users] Error installing asterisk, module zaptel not found   joe a.
• [asterisk-users] Realtime, DUNDi and regexten   Bruce Reeves
• [asterisk-users] Voicemail issues   Dovid B
• [asterisk-users] ZAPtel channel dance   Tzafrir Cohen
• [asterisk-users] Java Web Phone   Dean Collins
• [asterisk-users] mpg123 new version   Tzafrir Cohen
• [asterisk-users] Running asterisk with 'sudo'   Tzafrir Cohen
• [asterisk-users] Re: mpg123 new version   Frank Liu
• [asterisk-users] fax eater   James Harper
• [asterisk-users] Error installing asterisk, module zaptel not found   Tzafrir Cohen
• [asterisk-users] How to clear trixbox configuration   Alex Robar
• [asterisk-users] fax eater   Alex Robar
• [asterisk-users] ip address in CDR   Benjamin Jacob
• [asterisk-users] is IAX required for firewall and router?   Thufir
• [asterisk-users] Audiocodes MP-114 noise   Jason Kim
• [asterisk-users] Re: Newbie Questions - Grandstorm phones?   Henry.L.Coleman
• [asterisk-users] Re: fax eater   Martin Joseph
• [asterisk-users] Re: ZAPtel channel dance   Martin Joseph
• [asterisk-users] Cisco 7960 - Fast dial   Tomislav Parčina
• [asterisk-users] Java Web Phone   Administrator TOOTAI
• [asterisk-users] Re: Can some moderator kick this person out of the list   Tomislav Parčina
• SV: [asterisk-users] ip address in CDR   Jon Schøpzinsky
• [asterisk-users] Re: VM Language   Tomislav Parčina
• [asterisk-users] Re: Grandstream HandyTone-488 with Asterisk ?   Tomislav Parčina
• [asterisk-users] PAP2 to use on my asterisk.   twanny at line.sytes.net
• SV: [asterisk-users] ip address in CDR   Benjamin Jacob
• [asterisk-users] Re: Can some moderator kick this person out of the list   Zeeshan Zakaria
• [asterisk-users] Re: Can some moderator kick this person out of the list   Tijl Van den Broeck
• [asterisk-users] Re: Can some moderator kick this person out of the list   Zeeshan Zakaria
• [asterisk-users] Asterisk IAX Trunk and Queues   mbodbg at gmx.net
• [asterisk-users] PAP2 to use on my asterisk.   Marnus van Niekerk
• [asterisk-users] Re: fax eater   James Harper
• [asterisk-users] How to reduce latency and first real ring timing?   Zeeshan Zakaria
• *****SPAM***** [asterisk-users] Meetme Conference Rooms   Dean Bath
• [Asterisk-Users] The most cost effective and Asterisk friendly T.38 gateway ?   Olivier
• [asterisk-users] How do i redirect a call without answering it? SIP channel   Marco Mouta
• [asterisk-users] Audio goes one way during the call for a few seconds. Is it RTP, NAT, dyndns, or what it is?   Zeeshan Zakaria
• [asterisk-users] Re: Audio goes one way during the call for a few seconds. Is it RTP, NAT, dyndns, or what it is?   Zeeshan Zakaria
• [asterisk-users] Running asterisk with 'sudo'   Asterisk
• [asterisk-users] Asterisk Manager and Ruby   Dominique Dartois
• [asterisk-users] Help for registration with "sipdiscount"   Crazy Boy
• [asterisk-users] Error updating bootrom on Polycom phones..doesn't even download the bootrom!   Mike
• [asterisk-users] Pass-through any codecs   Szabó András
• [asterisk-users] Re: PAP2 to use on my asterisk.   Thufir
• [asterisk-users] TDM400 hungup problem   Ale
• [asterisk-users] Caller ID 1.2.10   Barry Fawthrop
• [asterisk-users] some simple newbie help with dialplan needed...   Evert
• [asterisk-users] Error updating bootrom on Polycom phones..doesn'teven download the bootrom!   Mike
• [asterisk-users] Re: Cisco 7960 - Fast dial   David Cook
• [asterisk-users] Re: fax eater   Lee Howard
• [asterisk-users] Nortel Option 11C and SIP gateway integration   Heison Chak
• [asterisk-users] Re: Nortel Option 11C and SIP gateway integration   Heison Chak
• [asterisk-users] Echo Issues   Garth van Sittert
• [asterisk-users] Re: Nortel Option 11C and SIP gateway integration   Watkins, Bradley
• [asterisk-users] Re: Nortel Option 11C and SIP gateway integration   Joe Dennick
• [asterisk-users] In bound SIP context issue   nrbwpi at gmail.com
• [asterisk-users] Clearing Outgoing Call Queue   Matthew Rubenstein
• [asterisk-users] TDM400 hungup problem   Giorgio Incantalupo
• [asterisk-users] In bound SIP context issue   Alex Robar
• [asterisk-users] billing   Doug Lytle
• [asterisk-users] In bound SIP context issue   nrbwpi at gmail.com
• [asterisk-users] AEL2 in 1.2   Aaron Daniel
• [asterisk-users] Problems Overwriting CallerID with True ANI   Steve Totaro
• [asterisk-users] Asterisk does not bridge zap channels on outgoing calls   Alexandru Voinescu
• [Asterisk-Users] Is fax bridging with TDM2400 working (or about to work) ?   Olivier
• [asterisk-users] Re: fax eater   Jay R. Ashworth
• [asterisk-users] Monitor, MixMonitor and volume levels   Steve Davies
• [asterisk-users] Example Polycom function key config   Jamie Heckford
• [asterisk-users] billing   Jeremy McNamara
• [asterisk-users] Configure Max TNT PRI to SIP with Asterisk   JR Richardson
• [asterisk-users] Binding a peer context to a specific IP address   Tomer Horn
• [Asterisk-Users] Is fax bridging with TDM2400 working (or about to work) ?   Lee Howard
• [asterisk-users] TDM400 hungup problem   Ale
• AW: [asterisk-users] Snom or Cisco Phones?   Dave Schardin
• [asterisk-users] billing   Don MacArthur
• [asterisk-users] Audiocodes MP-114 noise   Jessee J Holmes
• [asterisk-users] TDM400 hungup problem   Giorgio Incantalupo
• [asterisk-users] Clicking Noise on Pure Voip Calls   Noah Miller
• AW: [asterisk-users] Snom or Cisco Phones?   Jamie Heckford
• [asterisk-users] AEL2 in 1.2   Aaron Daniel
• [asterisk-users] SOLVED Problems Overwriting CallerID with True ANI   Steve Totaro
• [asterisk-users] TDM400 hungup problem   Ale
• [asterisk-users] Problem with Realtime/ODBC   Maurizio Pederneschi
• [asterisk-users] SIP - IAX Attended transfer   David Parcerisa
• [asterisk-users] Extension Spy   Ed Nuñez
• [asterisk-users] Re: AEL2 in 1.2   Steve Murphy
• [asterisk-users] IVR problem   Doug Crompton
• [asterisk-users] International dialing with GPX-2000 and "early dial"   Anthony Kepler
• [asterisk-users] Clearing Outgoing Call Queue   Steve Totaro
• [asterisk-users] Extension Spy   Julian Lyndon-Smith
• [asterisk-users] Echo Issues   Matt
• [asterisk-users] Random 'no audio' problem   Matt
• [asterisk-users] Extension Spy   Matt
• [asterisk-users] How do i redirect a call without answering it? SIP channel   William Piper
• [asterisk-users] Extension Spy   Ed Nuñez
• [asterisk-users] Echo Issues   Matthew Fredrickson
• [asterisk-users] some simple newbie help with dialplan needed...   William Piper
• [asterisk-users] Experiment: Dialplan size vs. Speed   Steve Murphy
• [asterisk-users] Problems Overwriting CallerID with True ANI   William Piper
• [Asterisk-Users] Is fax bridging with TDM2400 working (or about to work) ?   Matthew Fredrickson
• [asterisk-users] compilation problem with asterisk-addons   Erick Perez
• [asterisk-users] IVR problem   Bruce Reeves
• [asterisk-users] fax eater   Michelle Dupuis
• [asterisk-users] Polycom provisioning and Pure-FTP : problems   Mike
• [asterisk-users] Random 'no audio' problem   Zeeshan Zakaria
• [asterisk-users] IVR problem   Doug Crompton
• [Asterisk-Users] Is fax bridging with TDM2400 working (or about to work) ?   Steve Underwood
• [asterisk-users] Unicall's MFCR2 with Asterisk 1.4   Barzilai Spinak
• [asterisk-users] Asterisk architecture   Christopher Chan
• [asterisk-users] SendDTMF() behaves strangely   Stephen Bosch
• [asterisk-users] billing   Khaled
• [asterisk-users] Random 'no audio' problem   Matt
• [asterisk-users] Random 'no audio' problem   Zeeshan Zakaria
• [asterisk-users] Why only one out of many IP Phones re-registering every one minute   Zeeshan Zakaria
• [asterisk-users] Unicall's MFCR2 with Asterisk 1.4   Moises Silva
• [asterisk-users] SendDTMF() behaves strangely   Moises Silva
• [asterisk-users] DID with extensions   Don
• [asterisk-users] Re: My Phone Review- Large Scale Corp Deployment.   Doug Meredith
• [asterisk-users] Patton 1400   Kevin Withnall
• [asterisk-users] Polycom SIP 2.0.2 firmware   Kevin Bockman
• [asterisk-users] Re: My Phone Review- Large Scale Corp Deployment.   Doug Crompton
• [asterisk-users] Asterisk upgrade from 1.0.9 to 1.2.6 not working   Matt
• [asterisk-users] Help for registration with "sipdiscount"   Ron Wellsted
• [asterisk-users] SIP - IAX Attended transfer   David Parcerisa
• [asterisk-users] iax2 qualify - false "peer unreachable"   Pavel Jezek
• [asterisk-users] Patton 1400   Guido Hecken
• [asterisk-users] Polycom SIP 2.0.2 firmware   Eric Bishop
• [asterisk-users] best gui   Zeeshan Zakaria
• [asterisk-users] My first Asterisk - Not recognizing X100P clone   Poul Moller
• [asterisk-users] best gui   Rob Hillis
• [asterisk-users] app_prepaid won't load - undefined symbol mysql_num_fields   Mosiuoa Tsietsi
• [asterisk-users] Problems Overwriting CallerID with True ANI   Kevin Bockman
• [asterisk-users] Hairpinning problems using IAX2 and SIP   hugolivude
• [asterisk-users] Re: Asterisk upgrade from 1.0.9 to 1.2.6 not working   Matt
• [asterisk-users] I suggest using TFTP.   Matthew Mackes (Webmail)
• [asterisk-users] Asterlink Down?   Matt
• [asterisk-users] PRI issues   Doug Lytle
• [asterisk-users] Re: Polycom provisioning and Pure-FTP : problems   David Cook
• [asterisk-users] Asterisk architecture   Vikki
• [asterisk-users] My first Asterisk - Not recognizing X100P clone   Vikki
• [asterisk-users] Upgrading from 1.2.12.1 to 1.2.13   Henrik Woffinden
• [asterisk-users] SendDTMF() behaves strangely   Stephen Bosch
• [asterisk-users] app_prepaid won't load - undefined symbol mysql_num_fields   Tzafrir Cohen
• [asterisk-users] PRI issues   Scott Keagy
• [asterisk-users] Re: PRI issues   Doug Lytle
• [asterisk-users] Re: PRI issues   Doug Lytle
• [asterisk-users] app_prepaid won't load - undefined symbol mysql_num_fields   Mosiuoa Tsietsi
• [asterisk-users] app_prepaid won't load - undefined symbol mysql_num_fields   Rodrigo Gonzalez
• [asterisk-users] app_prepaid won't load - undefined symbol mysql_num_fields   Mosiuoa Tsietsi
• [asterisk-users] app_prepaid won't load - undefined symbol mysql_num_fields   Tzafrir Cohen
• [asterisk-users] Re: PRI issues   Bruce Reeves
• [asterisk-users] app_prepaid won't load - undefined symbol mysql_num_fields   Mosiuoa Tsietsi
• [asterisk-users] best gui   joe a.
• [asterisk-users] Redirect problems using IAX2 and SIP   hugolivude
• [asterisk-users] SendDTMF() behaves strangely   Stephen Bosch
• [asterisk-users] Redirect problems using IAX2 and SIP   hugolivude
• [asterisk-users] Pass through   Szabó András
• [asterisk-users] best gui   Zeeshan Zakaria
• [asterisk-users] Re: Audio goes one way during the call for a few seconds. Is it RTP, NAT, dyndns, or what it is?   Zeeshan Zakaria
• [asterisk-users] g729 codec help   programming dept
• [asterisk-users] Only one out of 10 remote extensions expiring registry   Zeeshan Zakaria
• [asterisk-users] best gui   Tom Vile
• [asterisk-users] FXO lines taking several rings to answer, always two   Jordan Novak
• [asterisk-users] Unicall's MFCR2 with Asterisk 1.4   Barzilai Spinak
• [asterisk-users] best gui   Zeeshan Zakaria
• [asterisk-users] g729 codec help   Hermann Wecke
• [asterisk-users] Only one out of 10 remote extensions expiring registry   Nic Bellamy
• [asterisk-users] Only one out of 10 remote extensions expiring registry   Zeeshan Zakaria
• [asterisk-users] FXO lines taking several rings to answer, always two   Tom Vile
• [asterisk-users] Advice on GUI   Diego Andres Asenjo G.
• [asterisk-users] Need help choosing cards that support detail Hangup Cause   King Ho
• [asterisk-users] best gui   Diego Andres Asenjo G.
• [asterisk-users] Hairpinning problems using IAX2 and SIP   Andrew Joakimsen
• [asterisk-users] International dialing with GPX-2000 and "early dial"   Andrew Joakimsen
• [asterisk-users] SPA3k wired to PAP2 for echo testing   James Harper
• [asterisk-users] Need help choosing cards that support detail Hangup Cause   C F
• [asterisk-users] Newbie questions about Voice mail   bdk at unb.ca
• [asterisk-users] Newbie questions about Voice mail   Dean Collins
• 回覆: [asterisk-users] Need help choosing cards that support detailHangup Cause   King Ho
• [asterisk-users] [FIXED] NAT issue ?   Dovid B
• [asterisk-users] Hang up on SIP calls if connected to long   Dovid B
• [asterisk-users] Re: PRI issues   Eric "ManxPower" Wieling
• [asterisk-users] Anybody used Asterfax?   Zeeshan Zakaria
• [asterisk-users] skype and SIP hardware for linux   Thufir
• [asterisk-users] Asterisk and FXO Digium Card for Analog line   Noc Phibee
• [asterisk-users] skype and SIP hardware for linux   Peter Bowyer
• [asterisk-users] Re: Audio goes one way during the call for a few seconds. Is it RTP, NAT, dyndns, or what it is?   Matt Koscica
• [asterisk-users] Reading Voicemail Config from MySQL   Mosiuoa Tsietsi
• [asterisk-users] call transfer problem   Colin MacMillan
• [asterisk-users] Asterisk and FXO Digium Card for Analog line   Dovid B
• [asterisk-users] Re: skype and SIP hardware for linux   Thufir
• [asterisk-users] Only one out of 10 remote extensions expiring registry   Zeeshan Zakaria
• [asterisk-users] Only one out of 10 remote extensions expiring registry   Zeeshan Zakaria
• [asterisk-users] Re: skype and SIP hardware for linux   Thufir
• [asterisk-users] Re: skype and SIP hardware for linux   Thufir
• [asterisk-users] Re: skype and SIP hardware for linux   Dovid B
• [asterisk-users] Audio goes one way during the call for a few seconds. Is it RTP, NAT, dyndns, or what it is?   Matt
• [asterisk-users] Hang up on SIP calls if connected to long   Matt
• [asterisk-users] Definity Asterisk CallerID Issue   cp
• [asterisk-users] SPA3k wired to PAP2 for echo testing   Bob Chiodini
• Free PBX, was - Re: [asterisk-users] best gui   joe a.
• [asterisk-users] Voicemail.conf multi languages   Guerid Salim
• [asterisk-users] Newbie questions about Voice mail   bdk at unb.ca
• [asterisk-users] SPA3k wired to PAP2 for echo testing   Stephen Davies
• [asterisk-users] Newbie questions about Voice mail   Dean Collins
• [asterisk-users] SIP v IAX2   Stephen Davies
• [asterisk-users] Newbie questions about Voice mail   bdk at unb.ca
• [asterisk-users] Call Quality Issues with IAX?   Aaron J. Angel
• [asterisk-users] Definity Asterisk CallerID Issue   Steve Totaro
• [asterisk-users] Re: Re: skype and SIP hardware for linux   Thufir
• [asterisk-users] light web user interface   kjcsb
• [asterisk-users] best gui   embrow at cwjamaica.com
• [asterisk-users] Asterisk upgrade from 1.0.9 to 1.2.6 not working   Steve Totaro
• [asterisk-users] Definity Asterisk CallerID Issue   Steve Totaro
• [asterisk-users] Problems Overwriting CallerID with True ANI   Steve Totaro
• [asterisk-users] Very high translation costs for g729   Avi Miller
• [asterisk-users] Very high translation costs for g729   Julian J. M.
• [asterisk-users] names of SIP aware firewalls   Erick Perez
• [asterisk-users] Newbie questions about Voice mail   Dean Collins
• [asterisk-users] Reading Voicemail Config from MySQL [+ ODBC]   Mosiuoa Tsietsi
• [asterisk-users] names of SIP aware firewalls   Dovid B
• [asterisk-users] Definity Asterisk Caller ID Issue   mavince at optonline.net
• [asterisk-users] Call Quality Issues with IAX?   hugolivude
• [asterisk-users] names of SIP aware firewalls   Carla Schroder
• [asterisk-users] Very high translation costs for g729   Avi Miller
• [asterisk-users] SPA3k wired to PAP2 for echo testing   James Harper
• [asterisk-users] Hairpinning problems using IAX2 and SIP   hugolivude
• [asterisk-users] Anybody used Asterfax?   Warrick Zedi
• [asterisk-users] Experiment: Dialplan size vs. Speed   Nick Hoffman
• [asterisk-users] TDM400 hungup problem   Shaun Hofer
• [asterisk-users] Very high translation costs for g729   Jason
• [asterisk-users] Use astbill to bill Trixbox   Matt Arnilo S. Baluyos (Mailing Lists)
• [asterisk-users] Re: Newbie Questions - Grandstorm phones?   Jesús Méndez Román
• [asterisk-users] xfsound=beep is not beeping   Klaverstyn, David C
• [asterisk-users] Re: Newbie Questions - Grandstorm phones?   Doug Crompton
• [asterisk-users] SPA3k wired to PAP2 for echo testing   Doug Crompton
• [asterisk-users] Polycom SIP 2.0.2 firmware   Lacy Moore - Aspendora
• [asterisk-users] names of SIP aware firewalls   Lacy Moore - Aspendora
• [asterisk-users] Anyone got a dialplan for SPA ATAs for ISN?   kjcsb
• [asterisk-users] asterisk DTMF detection   Brent Addis
• [asterisk-users] names of SIP aware firewalls   Jerry Jones
• [asterisk-users] names of SIP aware firewalls   Kristian Kielhofner
• [asterisk-users] Tampa Bay Asterisk Users Meetup on Monday   Kristian Kielhofner
• [asterisk-users] names of SIP aware firewalls   Leo Ann Boon
• [asterisk-users] Re: some simple newbie help with dialplan needed...   Evert
• [asterisk-users] some simple newbie help with dialplan needed...   Evert
• [asterisk-users] some simple newbie help with dialplan needed...   Leo Ann Boon
• [Asterisk-Users] Is fax bridging with TDM2400 working (or about to work) ?   Olivier
• [asterisk-users] some simple newbie help with dialplan needed...   Angel Heart
• [asterisk-users] Fast detection of unreachable SIP clients?   Dmitry Ivanov
• [asterisk-users] Re: some simple newbie help with dialplan needed...   Evert
• [asterisk-users] best gui   Gregory Duchatelet
• [asterisk-users] DTMF Tones occuring randomly   Stefan Agethen
• [asterisk-users] best gui   Tzafrir Cohen
• [asterisk-users] Audio goes one way during the call for a fewseconds. Is it RTP, NAT, dyndns, or what it is?   Jamie Heckford
• [asterisk-users] SPA3k wired to PAP2 for echo testing   Bob Chiodini
• [asterisk-users] Ring locally when home or roadwarrior via IAX when away   Arik Raffael Funke
• [asterisk-users] best gui   Gregory Duchatelet
• [asterisk-users] Tampa Bay Asterisk Users Meetup on Monday   Barry Fawthrop
• [asterisk-users] Re: some simple newbie help with dialplan needed...   Evert
• [asterisk-users] Definity Asterisk Caller ID Issue   cp
• [asterisk-users] Re: Newbie Questions - Grandstorm phones?   Gordon Henderson
• [asterisk-users] Ring locally when home or roadwarrior via IAX when away   Anselm Martin Hoffmeister
• [asterisk-users] Is it possible have multiple ip numbers for an extension?   William Kenworthy
• [asterisk-users] Port Range   Matt
• [asterisk-users] Port Range   Zoa
• [asterisk-users] Port Range   Matt
• [asterisk-users] Re: Echo Issues   Steven
• [asterisk-users] Re: Echo Issues   Steven
• [asterisk-users] Re: Port Range   Evert
• [asterisk-users] Re: Echo Issues   Matt
• [asterisk-users] Is it possible have multiple ip numbers for an extension?   Bob Chiodini
• [asterisk-users] Fast detection of unreachable SIP clients?   Matt
• [asterisk-users] Re: Port Range   Matt
• [asterisk-users] Re: Echo Issues   Tzafrir Cohen
• [asterisk-users] Audio goes one way during the call for a fewseconds. Is it RTP, NAT, dyndns, or what it is?   Zeeshan Zakaria
• [asterisk-users] Amending CLI in Dialplan   Scott Pinhorne
• [asterisk-users] Re: Echo Issues   Matt
• [asterisk-users] Unicall's MFCR2 with Asterisk 1.4   Moises Silva
• [asterisk-users] Page system using the sound card   Xue Liangliang
• [asterisk-users] SendDTMF() behaves strangely   Moises Silva
• [asterisk-users] Register vs. Host=IPADDR   Forrest Beck
• [asterisk-users] several behind NAT   Todd- Asterisk
• [asterisk-users] Anybody used Asterfax?   Zeeshan Zakaria
• [asterisk-users] Amending CLI in Dialplan   Anselm Martin Hoffmeister
• [asterisk-users] Port Range   Dovid B
• [asterisk-users] Re: Port Range   Zeeshan Zakaria
• [asterisk-users] several behind NAT   Zeeshan Zakaria
• [asterisk-users] several behind NAT   David M. Zendzian
• [asterisk-users] several behind NAT   Tim Panton
• [asterisk-users] asterisk DTMF detection   Zeeshan Zakaria
• [asterisk-users] several behind NAT   Tomer Horn
• [asterisk-users] Re: Asterisk and Panasonic KX Model   ggonzalez at telviso.com.ar
• [asterisk-users] names of SIP aware firewalls   Ira
• [asterisk-users] Fedora Core 6 (FC6) and Asterisk-1.2.13 and Zaptel-1.2.10 compile problems   Bruno Wolff III
• [asterisk-users] Queue time out   rachid
• [asterisk-users] SendDTMF() behaves strangely   Stephen Bosch
• [asterisk-users] receptionist - large number of concurrent calls - example needed   Colin MacMillan
• [asterisk-users] Echo Issues   Ejay Hire
• [asterisk-users] Tampa Bay Asterisk Users Meetup on Monday   Kristian Kielhofner
• [asterisk-users] is IAX required for firewall and router?   Mojo with Horan & Company, LLC
• [asterisk-users] Asterisk servers being greedy and not letting go of the media path. (using IAX2 channels)   Mat Stace
• [asterisk-users] My first Asterisk - Not recognizing X100P clone   Poul Moller
• [asterisk-users] several behind NAT   Dovid B
• [asterisk-users] Hairpinning problems using IAX2 and SIP   Andrew Joakimsen
• [asterisk-users] Hairpinning problems using IAX2 and SIP   Andrew Joakimsen
• [asterisk-users] Tampa Bay Asterisk Users Meetup on Monday   Andrew Joakimsen
• [asterisk-users] Register vs. Host=IPADDR   Andrew Joakimsen
• [asterisk-users] Re: Echo Issues   Tzafrir Cohen
• [asterisk-users] Callmanager 3.3(5) and Asterisk with ooh323   Pavel Jezek
• [asterisk-users] Re: Definity ISDN PRI   mavince at optonline.net
• [asterisk-users] Polycom unable to answer more than 3 calls at a time   Dovid B
• [asterisk-users] Re: Re: Echo Issues   Steven
• [asterisk-users] Nortel Option 11C and SIP gateway integration   Anthony Rodgers
• [asterisk-users] OT: BarCamp USA   Dean Collins
• [asterisk-users] Re: Ring locally when home or roadwarrior via IAX when away   Arik Raffael Funke
• [asterisk-users] DTMF Tones occuring randomly   Stefan Agethen
• [asterisk-users] receptionist - large number of concurrent calls - example needed   Andrew Joakimsen
• [asterisk-users] several behind NAT   Andrew Joakimsen
• [asterisk-users] Queue time out   Andrew Joakimsen
• [asterisk-users] Disappearing voicemail?   Ken D'Ambrosio
• [asterisk-users] Hairpinning problems using IAX2 and SIP   hugolivude
• [asterisk-users] Callmanager 3.3(5) and Asterisk with ooh323   Pavel Jezek
• [asterisk-users] DTMF Tones occuring randomly   Eric "ManxPower" Wieling
• [asterisk-users] asterisk DTMF detection   Brent Addis
• [asterisk-users] Unicall's MFCR2 with Asterisk 1.4   Barzilai Spinak
• [asterisk-users] TrixBox and MP104 FXO (AudioCodes GW)   Yuval Yogev
• [asterisk-users] IAX FWD - down, running own proxy or stun server   Joseph
• [asterisk-users] Queue time out   Darrell Davis
• [asterisk-users] Tampa Bay Asterisk Users Meetup on Monday   Kristian Kielhofner
• [asterisk-users] Follow Me problems   shadowym
• [asterisk-users] Do my messages come through?   Christian
• [asterisk-users] how to indicate an non-existent number?   Louis-David Mitterrand
• [asterisk-users] Callmanager 3.3(5) and Asterisk with ooh323   Pavel Jezek
• [asterisk-users] Asterisk + E1 with MFC/R2 in Argentina?   Ilan Rabinovitch
• [asterisk-users] names of SIP aware firewalls   Rich Adamson
• [asterisk-users] Tampa Bay Asterisk Users Meetup on Monday   Matt Florell
• [asterisk-users] Polycom dealers in Toronto/London ON   Alvin Austin
• [asterisk-users] Unicall's MFCR2 with Asterisk 1.4   Moises Silva
• [asterisk-users] names of SIP aware firewalls   Dovid B
• [asterisk-users] dial D option with w for wait?   kjcsb
• [asterisk-users] names of SIP aware firewalls   joe a.
• [asterisk-users] Echo Issues   Matt
• [asterisk-users] Callmanager 3.3(5) and Asterisk with ooh323   Dan Austin
• [asterisk-users] Page system using the sound card   Doug Lytle
• [asterisk-users] help for recording   Taylan KIRAN
• [asterisk-users] Question on Aastra phones and Astrisk   Matt
• [asterisk-users] Re: Question on Aastra phones and Astrisk   Matt
• [asterisk-users] how to indicate an non-existent number?   Eric "ManxPower" Wieling
• [asterisk-users] Question on Aastra phones and Astrisk   Curt Shaffer
• [asterisk-users] Question on Aastra phones and Astrisk   Curt Shaffer
• [asterisk-users] Polycom autoprovision behind a NAT   Curt Shaffer
• [asterisk-users] Re: Re: DTMF Tones occuring randomly   Stefan Agethen
• [asterisk-users] asterisk 1,4 and google talk   Mani Sridhar
• [asterisk-users] Polycom autoprovision behind a NAT   Russ Beaupre
• [asterisk-users] Re: Port Range   Leo Ann Boon
• [asterisk-users] Re: Port Range   Zeeshan Zakaria
• [asterisk-users] Polycom dealers in Toronto/London ON   Zeeshan Zakaria
• [asterisk-users] Polycom autoprovision behind a NAT   Curt Shaffer
• [asterisk-users] Polycom autoprovision behind a NAT   Russ Beaupre
• [asterisk-users] Polycom autoprovision behind a NAT   Curt Shaffer
• [asterisk-users] Polycom unable to answer more than 3 calls at a time   Noah Miller
• [asterisk-users] Polycom autoprovision behind a NAT   Russ Beaupre
• [asterisk-users] Polycom autoprovision behind a NAT   Curt Shaffer
• [asterisk-users] Polycom autoprovision behind a NAT   end1r at comcast.net
• [asterisk-users] Re: Port Range   Tom Vile
• [asterisk-users] Re: Port Range   Leo Ann Boon
• [asterisk-users] best gui   Diego Andres Asenjo G.
• [asterisk-users] silencedetecthangup=   Dovid B
• [asterisk-users] some simple newbie help with dialplan needed...   William Piper
• [asterisk-users] Re: some simple newbie help with dialplan needed...   William Piper
• [asterisk-users] some simple newbie help with dialplan needed...   William Piper
• [asterisk-users] Add yourself!   meric mara
• [asterisk-users] Re: Port Range   Eric "ManxPower" Wieling
• [asterisk-users] Zap channel shows "answered" as soon as outbound ringing starts   shadowym
• [asterisk-users] Zap channel shows "answered" as soon as outbound ringing starts   Leo Ann Boon
• [asterisk-users] how to indicate an non-existent number?   Alejandro Kauffmann
• [asterisk-users] Re: Port Range   Tzafrir Cohen
• [asterisk-users] Zap channel shows "answered" as soon as outbound ringing starts   yusuf
• [asterisk-users] Re: Port Range   Tom Vile
• [asterisk-users] My Phone Review- Large Scale Corp Deployment.   RR
• [asterisk-users] Dial plan Question   Doug Crompton
• [asterisk-users] My Phone Review- Large Scale Corp Deployment.   RR
• [asterisk-users] Re: some simple newbie help with dialplan needed...   Evert
• [asterisk-users] Re: Do my messages come through?   Martin Joseph
• [asterisk-users] Desired apps   Justin Newman
• [asterisk-users] Asterisk and FreeTDS 0.64 or >0.63   RR
• [asterisk-users] Fast detection of unreachable SIP clients?   Dmitry Ivanov
• [asterisk-users] Snom 360 flickering screen   Nick Hoffman
• [asterisk-users] Asterisk Showing 404 not found when calling from third party SIP server (newbie question)   Alok Mohapatra
• [asterisk-users] Dial plan Question   Anselm Martin Hoffmeister
• [asterisk-users] Upgrading sox   René Christensen
• [asterisk-users] Mapping CLI'S in Dialplan   Scott Pinhorne
• [asterisk-users] Mapping CLI'S in Dialplan   Benjamin Jacob
• [asterisk-users] Upgrading sox   Tzafrir Cohen
• [asterisk-users] Re: Pass through   Szabó András
• [asterisk-users] Queue time out   rachid
• [asterisk-users] Asterisk + E1 with MFC/R2 in Argentina?   R.R. Libera
• [asterisk-users] Problem: 2 second silence at the beginning of most calls   Mike
• *****SPAM***** RE: [asterisk-users] Queue time out   Dean Bath
• [asterisk-users] Problem: 2 second silence at the beginning of mostcalls   Steve Langstaff
• [asterisk-users] names of SIP aware firewalls   Tom Rymes
• [asterisk-users] Problem: 2 second silence at the beginning ofmostcalls   Mike
• [asterisk-users] Asterisk SMS: Experience with EMS?   Anselm Martin Hoffmeister
• [asterisk-users] Question on Aastra phones and Astrisk   Matt
• [asterisk-users] Snom 360 flickering screen   Chris Mazuc
• [asterisk-users] Question on Aastra phones and Astrisk   Gareth Owen
• [asterisk-users] Follow Me problems   Time Bandit
• [asterisk-users] My Phone Review- Large Scale Corp Deployment.   Andrew Joakimsen
• [asterisk-users] Dial plan Question   Doug Crompton
• [asterisk-users] Follow Me problems   David M. Zendzian
• [asterisk-users] capiAnswerFax   Pedro Silva
• [asterisk-users] Mapping CLI'S in Dialplan   scott
• [asterisk-users] capiAnswerFax   Michiel van Baak
• [asterisk-users] How do I make this stop? (Bridging of IAX channels?)   Matt
• [asterisk-users] capiAnswerFax   Armin Schindler
• [asterisk-users] failed to authenticate on invite   Damon Estep
• [asterisk-users] billing   Khaled
• [asterisk-users] Snom 360 flickering screen   Dovid B
• [asterisk-users] capiAnswerFax   Pedro Silva
• [asterisk-users] hicecaller ID   Nik Engel
• [asterisk-users] "Sticky" Polycom 501 keys and handset   Mike
• [asterisk-users] connect Sipura with Asterisk - both behind NAT   Joseph
• [asterisk-users] "Sticky" Polycom 501 keys and handset   Rick Smith
• [asterisk-users] Generating Recall/Flash using Zaptel   Dululu Ululu
• [asterisk-users] Reading Voicemail Config from MySQL [+ ODBC]   RR
• [asterisk-users] Audiocodes MP-114 noise   Jessee J Holmes
• [asterisk-users] operator console   Stephen Wingfield
• [asterisk-users] Grandstream TFTP system wide settings   Zeeshan Zakaria
• [asterisk-users] How do I make this stop? (Bridging of IAX channels?)   Joshua Colp
• [asterisk-users] g729   Joshua Colp
• [asterisk-users] g729   Tzafrir Cohen
• [asterisk-users] Queues and multiple lines   Michael Sampson
• [asterisk-users] capiAnswerFax   Michiel van Baak
• [asterisk-users] "Sticky" Polycom 501 keys and handset   Mike
• [asterisk-users] connect Sipura with Asterisk - both behind NAT   Juan Manuel Sá
• [asterisk-users] "Sticky" Polycom 501 keys and handset   Mike
• [asterisk-users] loosening voicemail file permissions for msg????.txt and msg????.wav   Scott Keagy
• [asterisk-users] hicecaller ID   Nik Engel
• [asterisk-users] Re: Port Range   shadowym
• [asterisk-users] How is ANI "Usually" sent on an ISDN PRI?   Steve Totaro
• [asterisk-users] "Sticky" Polycom 501 keys and handset   Bill Gibbs
• [asterisk-users] Asterisk and Max TNT Passing calls SIP to PRI, not PRI to SIP Authentication Issue   JR Richardson
• [asterisk-users] connect Sipura with Asterisk - both behind NAT   Alyed Tzompa
• [asterisk-users] Pressing "*" makes Asterisk destroy my call   Stefan Agethen
• [asterisk-users] Queues and multiple lines   Wes Baehr
• [asterisk-users] My Phone Review- Large Scale Corp Deployment.   shadowym
• [asterisk-users] Question on Aastra phones and Astrisk   shadowym
• [asterisk-users] incoming call destination: IVR not working   Mark Bryant
• [asterisk-users] How do I make this stop? (Bridging of IAX channels?)   Matt
• [asterisk-users] My Phone Review- Large Scale Corp Deployment.   Steve Edwards
• [asterisk-users] astertest   Victor Toofic
• [asterisk-users] Re: Asterisk and Max TNT Passing calls SIP to PRI, not PRI to SIP Authentication Issue   JR Richardson
• [asterisk-users] Microsoft will enter VoIP market in earnest next year, says Ballmer   Dean Collins
• [asterisk-users] test message please ignore   Christian
• [asterisk-users] [resolved] asterisk 1,4 and google talk   Mani Sridhar
• [asterisk-users] Why dont my messages get through   Christian
• [asterisk-users] Fax & Digium   Ken Williams
• [asterisk-users] Glitches in sound every time that Asterisk receives reINVITEs   rcarvalho at iric.up.pt
• [asterisk-users] Queues and multiple lines   brandon kruz
• [asterisk-users] RE: Queues and multiple lines   Shane O'Cain
• [asterisk-users] Help with latest Asterisk on latest Debian   Christian
• [asterisk-users] test please ignore   Christian
• [asterisk-users] "Sticky" Polycom 501 keys and handset   Rick Smith
• [asterisk-users] Asterisk and Max TNT Passing calls SIP to PRI, not PRI to SIP Authentication Issue   Barry Fawthrop
• [asterisk-users] Asterisk and Max TNT Passing calls SIP to PRI, not PRI to SIP Authentication Issue   Scott Keagy
• [asterisk-users] "Sticky" Polycom 501 keys and handset   Darryl Dunkin
• [asterisk-users] Dial plan Question   Doug Crompton
• [asterisk-users] Fax & Digium   Hadley Rich
• [asterisk-users] Why dont my messages get through   Alex Robar
• [asterisk-users] Microsoft will enter VoIP market in earnest next year, says Ballmer   Alex Robar
• [asterisk-users] Why dont my messages get through   Nick Hoffman
• [asterisk-users] Why dont my messages get through   Alex Robar
• [asterisk-users] Help with latest Asterisk on latest Debian   Tzafrir Cohen
• [asterisk-users] Fax & Digium   Dovid B
• [asterisk-users] Microsoft will enter VoIP market in earnest nextyear, says Ballmer   Curt Shaffer
• [asterisk-users] Snom 360 flickering screen   Nick Hoffman
• [asterisk-users] Why dont my messages get through   Dovid B
• [asterisk-users] Why dont my messages get through   Andrew Furey
• [asterisk-users] hicecaller ID   William Piper
• [asterisk-users] Re: Asterisk and Max TNT Passing calls SIP to PRI, not PRI to SIP Authentication Issue   Enrique Martinez
• [asterisk-users] "Sticky" Polycom 501 keys and handset   Darryl Dunkin
• [asterisk-users] My Phone Review- Large Scale Corp Deployment.   RR
• [asterisk-users] My Phone Review- Large Scale Corp Deployment.   Steve Edwards
• [asterisk-users] My Phone Review- Large Scale Corp Deployment.   Steve Edwards
• [asterisk-users] RxFAX - How to catch errors in the dialplan   Håkan Källberg
• [asterisk-users] Follow Me problems   Philippe Lindheimer
• [asterisk-users] My Phone Review- Large Scale Corp Deployment.   RR
• [asterisk-users] loosening voicemail file permissions formsg????.txt and msg????.wav   Guerid Salim
• [asterisk-users] Audiocodes MP-114 noise   Jason Kim
• [asterisk-users] Operating queues with clients on a legacy PABX   Rob Hillis
• [asterisk-users] flash transfer problem in asterisk integration with old PBX   Andrea Giuliani
• [asterisk-users] Queue forks asterisk and then leaves the extra processes lying around   Nigel Roberts
• [asterisk-users] channel.c: Unable to request channel ZAP   Asterisk
• [asterisk-users] Queue forks asterisk and then leaves the extraprocesses lying around   Lee Archer
• [asterisk-users] Queue forks asterisk and then leaves the extraprocesses lying around   Nigel Roberts
• [asterisk-users] Queue forks asterisk and then leaves theextraprocesses lying around   Lee Archer
• [asterisk-users] Agents that handle calls from multiple queues   Ardjan Zwartjes
• [asterisk-users] Glitches in sound every time that Asteriskreceives reINVITEs   Andreas Sikkema
• [asterisk-users] Newbie questions about Voice mail   Stephen Wingfield
• [asterisk-users] Asterisk CTI - SAP R/3 Intergration Certification   Silvia Gallego
• [asterisk-users] Glitches in sound every time that Asteriskreceives reINVITEs   Ricardo Carvalho
• [asterisk-users] DTMF Tones occuring randomly   Stefan Agethen
• [asterisk-users] no sound when bridging 2 asterisk SIP connections   Louis-David Mitterrand
• [asterisk-users] Asterisk servers being greedy and not letting goof the media path. (using IAX2 channels)   Mat Stace
• [asterisk-users] faxing times!   Benjamin Jacob
• [asterisk-users] asterisk and peep tone (network tone)   Giedrius Augys
• [asterisk-users] flash transfer problem in asterisk integration with old PBX   Dululu Ululu
• [asterisk-users] Why dont my messages get through   Bob Chiodini
• [asterisk-users] How to reboot a Polycom phone remotely   Mike
• [asterisk-users] Odd results from fxotune?   James Dyer
• [asterisk-users] Asterisk 1.4 and Queues RealTime   Gregory Duchatelet
• [asterisk-users] Re: Queues and multiple lines   David Cook (Canada)
• [asterisk-users] DTMF Tones occuring randomly   Eric "ManxPower" Wieling
• [asterisk-users] Queues and multiple lines   Wes Baehr
• [asterisk-users] Question on Aastra phones and Astrisk   Matt
• [asterisk-users] Pressing "*" makes Asterisk destroy my call   Matt
• [asterisk-users] I need (some) help in configuring PAP2.   twanny at line.sytes.net
• [asterisk-users] Why dont my messages get through   Christian
• [asterisk-users] Ringing phones   Matt
• [asterisk-users] how to indicate an non-existent number?   Louis-David Mitterrand
• [asterisk-users] Asterisk 1.2.x and video   Jorge Mendoza
• [asterisk-users] operator console   Forrest Beck
• [asterisk-users] HANGUPCAUSE for unalocated number?   Louis-David Mitterrand
• [asterisk-users] Performance issues in Realtime   Andrea Spadaccini
• [asterisk-users] Ringing phones   Doug Crompton
• [asterisk-users] Digium Asterisk-GUI problem   Adam Robins
• [asterisk-users] VLANs and Quality   Barry Fawthrop
• [asterisk-users] Ringing phones   Matt
• [asterisk-users] SIP CANCEL NOT WORKING   Mario Fernández Alonso
• [asterisk-users] jpeglib   Pedro Silva
• [asterisk-users] flash transfer problem in asterisk integration with old PBX   Andrew Joakimsen
• [asterisk-users] Ringing phones   Andrew Joakimsen
• [asterisk-users] How do I make this stop? (Bridging of IAX channels?)   Matt
• [asterisk-users] How do I make this stop? (Bridging of IAX channels?)   Matt
• [asterisk-users] Microsoft will enter VoIP market in earnestnextyear, says Ballmer   Gary G. Hendershot
• [asterisk-users] g729   Khaled
• [asterisk-users] VLANs and Quality   Conrad Wood
• [asterisk-users] my future count on this please help   Khaled
• [asterisk-users] FIC-GTA001   Dean Collins
• [asterisk-users] Audiocodes MP-114 noise   Jessee J Holmes
• [asterisk-users] DTMF Tones occuring randomly   Stefan Agethen
• [asterisk-users] Ringing phones   Time Bandit
• [asterisk-users] RE: FIC-GTA001   Dean Collins
• [asterisk-users] Ringing phones   Matt
• [asterisk-users] Delay between DTMF Down & Detected Digit   Jonathan Campbell
• [asterisk-users] Re: Performance issues in Realtime   JR Richardson
• [asterisk-users] Agents that handle calls from multiple queues   Douglas Garstang
• [asterisk-users] Re: Asterisk and Max TNT PRI to SIP Authentication Issue   JR Richardson
• [asterisk-users] Re: Asterisk and Max TNT PRI to SIP Authentication Issue   JR Richardson
• [asterisk-users] Re: Echo Issues   Matthew Fredrickson
• [asterisk-users] DTMF Tones occuring randomly   John covici
• [asterisk-users] Re: HANGUPCAUSE for unalocated number?   Steven
• [asterisk-users] Ringing phones   Noah Miller
• [asterisk-users] Ringing phones   Eric "ManxPower" Wieling
• [asterisk-users] Ringing phones   Matt
• [asterisk-users] One-Way-Audio After placing call on hold   Matt
• [asterisk-users] Re: asterisk iax2 monitoring   David Thomas
• [asterisk-users] Re: One-Way-Audio After placing call on hold   Matt
• [asterisk-users] Re: Re: Echo Issues   Steven
• [asterisk-users] Asterisk and Solaris   Jorge Alayon
• [asterisk-users] talking caller ID   Christian
• [asterisk-users] Re: One-Way-Audio After placing call on hold   Matt
• [asterisk-users] Microsoft will enter VoIP market in earnestnextyear, says Ballmer   Michiel van Baak
• [asterisk-users] DTMF Corruption Problem   Justin Tunney
• [asterisk-users] Re: Asterisk and Max TNT PRI to SIP Authentication Issue, a little closer   JR Richardson
• [asterisk-users] Microsoft will enter VoIP market in earnest   Dean Collins
• [asterisk-users] Microsoft will enter VoIP market in earnest   Curt Shaffer
• [asterisk-users] Warning: "Channel does not have a CDR" when doing ForkCDR   Michael Collins
• [asterisk-users] DTMF Corruption Problem   Justin Tunney
• [asterisk-users] Off-Site Extensions That Would Show As In-Use?   Alexander Burke
• [asterisk-users] Asterisk and Solaris   Andrew Joakimsen
• [asterisk-users] Operating queues with clients on a legacy PABX   Andrew Joakimsen
• [asterisk-users] Re: Re: Echo Issues   Steven
• [asterisk-users] DTMF Corruption Problem   Kristian Kielhofner
• [asterisk-users] sms script on receive   Stephen Farrell
• [asterisk-users] [FC5] How to update kernel/kernel-develop for Athlon?   Vincent Delporte
• [asterisk-users] International dialing with GPX-2000 and "early dial"   Anthony Kepler
• [asterisk-users] I LOVE IT   Ken Williams
• [asterisk-users] Still problems with Asterisk on latest Debian   Christian
• [asterisk-users] Reg errors? Other anomalies? Check those capacitors!   Ronald Lewis
• [asterisk-users] Reg errors? Other anomalies? Check those capacitors!   Doug Crompton
• [asterisk-users] Queues: member order vs. defines in queues.conf   lists.digium.com at tgice.com
• [asterisk-users] Ask users.conf   mrdlnf
• [asterisk-users] Auto record a call?   Michael Collins
• [asterisk-users] Re: Reg errors? Other anomalies? Check thosecapacitors!   Steven
• [asterisk-users] Unknown caller id problem   Jay Lee
• [asterisk-users] DID billing with a2billing   Al Bochter
• [asterisk-users] Re: I need (some) help in configuring PAP2.   Martin Joseph
• [asterisk-users] talking caller ID   Philippe Lindheimer
• [asterisk-users] DTMF Corruption Problem   Stephen Bosch
• [asterisk-users] Re: Port Range   Martin Joseph
• [asterisk-users] OT - Polycom https provisioning   Philippe Lindheimer
• [asterisk-users] Re: I LOVE IT   Martin Joseph
• [asterisk-users] Problem with CDR interpretation   Michał Niklas
• [asterisk-users] connect Sipura with Asterisk - both behind NAT   Mark Coccimiglio
• [asterisk-users] TDM, loopstart and modules GSM Nokia32   Paco Brufal
• [asterisk-users] Still problems with Asterisk on latest Debian   Tzafrir Cohen
• [asterisk-users] Asterisk and Solaris   Akash Singh
• [hylafax-users] [asterisk-users] asterisk, iaxmodem, hylafax quality problem   Artifex Maximus
• [asterisk-users] Re: I LOVE IT   Steve Totaro
• [asterisk-users] SRTP   yusuf
• [asterisk-users] Alcatel trunk with asterisk problem on dialing digit-by-digit   Frederico Madeira
• [asterisk-users] Alcatel trunk with asterisk problem on dialing digit-by-digit   Marco Mouta
• [asterisk-users] Re: Reg errors? Other anomalies? Check thosecapacitors!   Bruce Reeves
• [asterisk-users] Re: Monitor, MixMonitor and volume levels   Steve Davies
• [asterisk-users] Alcatel trunk with asterisk problem on dialing digit-by-digit   Frederico Madeira
• [asterisk-users] Alcatel trunk with asterisk problem on dialing digit-by-digit   Leonardo Gomes Figueira
• [asterisk-users] Alcatel trunk with asterisk problem on dialing digit-by-digit   Marco Mouta
• [asterisk-users] DTMF Tones occuring randomly   Stefan Agethen
• [asterisk-users] DTMF Corruption Problem   mail-lists
• [asterisk-users] DTMF Corruption Problem   Erick Perez
• [asterisk-users] Alcatel trunk with asterisk problem on dialing digit-by-digit   Frederico Madeira
• [asterisk-users] Still problems with Asterisk on latest Debian   Christian
• [asterisk-users] DTMF Corruption Problem   mail-lists
• [asterisk-users] Problem with register command in SIP.conf   Frederico Madeira
• [asterisk-users] Still problems with Asterisk on latest Debian   Tzafrir Cohen
• [asterisk-users] Re: I LOVE IT   Mike Dent
• [asterisk-users] Voxee lag problems ?   Vicky
• [asterisk-users] DUNDi precache   Douglas Garstang
• [asterisk-users] Re: Re: Reg errors? Other anomalies? Checkthosecapacitors!   Steven
• [asterisk-users] several behind NAT   Todd- Asterisk
• [asterisk-users] Problem with register command in SIP.conf   Anselm Martin Hoffmeister
• [asterisk-users] DTMF Corruption Problem   Erick Perez
• [asterisk-users] unsubscribe   Adam Mattina
• [asterisk-users] Question on Aastra phones and Astrisk   shadowym
• [asterisk-users] asterisk and norstar   Gustavo Berman
• [asterisk-users] Re: Reg errors? Other anomalies? Check thosecapacitors!   Ira
• [asterisk-users] special characters in alphanumeric extensions   Ricardo Carvalho
• [asterisk-users] Voxee lag problems ?   mail-lists
• [asterisk-users] Bug ???   phil.dawson at marnock.com
• [asterisk-users] DUNDi precache   Aaron Daniel
• [asterisk-users] Quick Q...   Jay Moore
• [asterisk-users] Digium B410P   King Ho
• [asterisk-users] A couple of new tutorials: installing * 1.4 and the Asterisk GUI   Lenz
• [asterisk-users] special characters in alphanumeric extension s   Colin Anderson
• [asterisk-users] DUNDi precache   Douglas Garstang
• [asterisk-users] unsubscribe   Anthony Rodgers
• [asterisk-users] Station Voip Brazil   Felipe Amaral
• [asterisk-users] DUNDi precache   Bruce Reeves
• [asterisk-users] SRTP   Kristian Kielhofner
• [asterisk-users] DUNDi precache   Michiel van Baak
• [asterisk-users] DUNDi precache   Aaron Daniel
• [asterisk-users] Auto record a call?   Ed Nuñez
• [asterisk-users] New Asterisk 1.4 GUI   Curt Shaffer
• [asterisk-users] wip5000 roaming   Altus Snyman
• [asterisk-users] New Asterisk 1.4 GUI   Lenz
• [asterisk-users] DUNDi precache   Douglas Garstang
• [asterisk-users] wip5000 roaming   Andrew Joakimsen
• [asterisk-users] DUNDi precache   Douglas Garstang
• [asterisk-users] wip5000 roaming   Altus Snyman
• [asterisk-users] Voxee lag problems ?   marcelobiz at comcast.net
• [asterisk-users] DUNDi precache   Douglas Garstang
• [asterisk-users] SRTP   Patrick
• [asterisk-users] register suddenly fails   Norbert Zawodsky
• [asterisk-users] Harris 20-20   Alyed Tzompa
• [asterisk-users] DUNDi precache   Bruce Reeves
• [asterisk-users] Re: DTMF Corruption Problem   Justin Tunney
• [asterisk-users] porting numbers away from packet 8?   Dean Collins
• [asterisk-users] Re: DTMF Corruption Problem   Eric "ManxPower" Wieling
• [asterisk-users] wip5000 roaming   Andrew Joakimsen
• [asterisk-users] Re: Asterisk and Max TNT PRI to SIP Authentication Issue, a little closer   Andres
• [asterisk-users] DUNDi precache   Aaron Daniel
• [asterisk-users] Zaptel 1.2.11 released   Asterisk Development Team
• [asterisk-users] Modprobe Zaptel   Julian Varanini
• [asterisk-users] Mitel 5224 & Asterisk Distinctive Ring -- Anyone have it working?   Christopher Aloi
• [asterisk-users] Modprobe Zaptel   Eric "ManxPower" Wieling
• [asterisk-users] Microsoft will enter VoIP market in earnest nextyear, says Ballmer   Dean Collins
• [asterisk-users] Latest Debian and latest zaptel   Christian
• [asterisk-users] asterisk and norstar   Jorge Mendoza
• [asterisk-users] asterisk and norstar   Jorge Mendoza
• [asterisk-users] Re: DTMF Corruption Problem   Justin Tunney
• [asterisk-users] Station Voip Brazil   Kristian Kielhofner
• [asterisk-users] porting numbers away from packet 8?   Dovid B
• [asterisk-users] Quick Q...   Dovid B
• [asterisk-users] Voxee lag problems ?   Dovid B
• [asterisk-users] Off-Site Extensions That Would Show As In-Use?   Dovid B
• [asterisk-users] porting numbers away from packet 8?   Dean Collins
• [asterisk-users] Modprobe Zaptel   Julian Varanini
• [asterisk-users] Modprobe Zaptel   Eric "ManxPower" Wieling
• [asterisk-users] Modprobe Zaptel   Darryl Dunkin
• [asterisk-users] Powering SNOM 200 phones?   Brad Templeton
• [asterisk-users] DTMF problems with IVR - What DMTF Tx method   Bruce Ferrell
• [asterisk-users] Powering SNOM 200 phones?   Christopher Aloi
• [asterisk-users] Ask users.conf   mrdlnf
• [asterisk-users] Off-Site Extensions That Would Show As In-Use?   Alexander Burke
• [asterisk-users] Powering SNOM 200 phones?   Paul Hales
• [asterisk-users] Powering SNOM 200 phones?   Brad Templeton
• [asterisk-users] DTMF problems with IVR - What DMTF Tx method   Eric "ManxPower" Wieling
• [asterisk-users] DID billing with a2billing   Al Bochter
• [asterisk-users] announcing inbound PSTN calls   Jeronimo Romero
• [asterisk-users] Zaptel 1.2.11 released   Alberto Pastore
• [asterisk-users] Latest Debian and latest zaptel   Michiel van Baak
• [asterisk-users] Voxee lag problems ?   Ira
• [asterisk-users] Voxee lag problems ?   Vicky
• [asterisk-users] Asterisk BlindTransfer behaves differently in version 1.0 and 1.2   Xue Liangliang
• [asterisk-users] register suddenly fails   Vicky
• [asterisk-users] register suddenly fails   Vicky
• [asterisk-users] SRTP   Khaled
• [asterisk-users] EuroISDN+ and Callers name   Dave Cotton
• [asterisk-users] announcing inbound PSTN calls   Marco Mouta
• [asterisk-users] EuroISDN+ and Callers name   Anselm Martin Hoffmeister
• [asterisk-users] Station Voip Brazil   Felipe Amaral
• [asterisk-users] Dropping Connections   Mike Heininger
• SV: [asterisk-users] Dropping Connections   Jon Schøpzinsky
• [asterisk-users] Looking for IP phone / ATA that has builtin VPN support   M. Salaque
• [asterisk-users] Question on Aastra phones and Astrisk   Matt
• [asterisk-users] Microsoft will enter VoIP market in earnest nextyear, says Ballmer   Zach Fine
• [asterisk-users] Dropping Connections   Anselm Martin Hoffmeister
• [asterisk-users] Queues and Timeouts.   Thomas Kenyon
• [asterisk-users] Stable clock with 2.6 and without Digium hardware.   Anton Tinchev
• [asterisk-users] Stable clock with 2.6 and without Digium hardware.   Zoa
• [asterisk-users] Dropping Connections   Mike Heininger
• [asterisk-users] Stable clock with 2.6 and without Digium hardware.   Anton Tinchev
• [asterisk-users] VM notification to pager and phone   BerkHolz, Steven
• [asterisk-users] Presence-awareness in Asterisk   Ondrej Valousek
• [asterisk-users] Quick Q...   Jay Moore
• [asterisk-users] Pointers/suggestions?   Pierre Fortin
• [asterisk-users] Outgoing problem on PRI   Mohamed A. Gombolaty
• [asterisk-users] asterisk and norstar   Gustavo Berman
• [asterisk-users] Re: Delay between DTMF Down & Detected Digit   Steven
• [asterisk-users] Presence-awareness in Asterisk   Anselm Martin Hoffmeister
• [asterisk-users] announcing inbound PSTN calls   Gustavo Berman
• [asterisk-users] announcing inbound PSTN calls   bails
• [asterisk-users] Presence-awareness in Asterisk   Ondrej Valousek
• [asterisk-users] announcing inbound PSTN calls   Marco Mouta
• [asterisk-users] announcing inbound PSTN calls   Anselm Martin Hoffmeister
• [asterisk-users] Presence-awareness in Asterisk   Anselm Martin Hoffmeister
• [asterisk-users] Looking for IP phone / ATA that has builtin VPN support   Noah Miller
• [asterisk-users] Question about Mitel phones   Christian
• [asterisk-users] Outgoing problem on PRI   Andres
• [asterisk-users] Re: Asterisk and Max TNT SIP Authentication Issue, WORKING   JR Richardson
• [asterisk-users] Latest Debian and latest zaptel   Tzafrir Cohen
• [asterisk-users] Re: Reg errors? Other anomalies? Check thosecapacitors!   Tim Panton
• [asterisk-users] Modprobe Zaptel   Tzafrir Cohen
• [asterisk-users] Modprobe Zaptel   Tzafrir Cohen
• [asterisk-users] Choppy sound in voicemail using Asterisk 1.2.11 on CENTOS4 guest on vmware server   Mario François Jauvin
• [asterisk-users] Voxee lag problems ?   Jonathan Borden
• [asterisk-users] How to get CDR to show answered calls only   shadowym
• [asterisk-users] Need to automatically park an incoming call and then connect to an extension.   Mauro Zanin
• [asterisk-users] Modprobe Zaptel   Doug Lytle
• [asterisk-users] How to get CDR to show answered calls only   mail-lists
• [asterisk-users] Modprobe Zaptel   Eric "ManxPower" Wieling
• [asterisk-users] [resolved] asterisk 1,4 and google talk   Mik Cheez
• [asterisk-users] Presence-awareness in Asterisk   Andrea Spadaccini
• [asterisk-users] Stable clock with 2.6 and without Digium hardware.   Tzafrir Cohen
• [asterisk-users] Need to automatically park an incoming call and then connect to an extension.   C F
• [asterisk-users] Realtime & sippeers using NAT   Mik Cheez
• [asterisk-users] Stable clock with 2.6 and without Digium hardware.   Tzafrir Cohen
• [asterisk-users] Returncode from command   Andre Courchesne - Consultant
• [asterisk-users] Re: IAX2 goes "one way audio" when lag gets bad   Pavel Jezek
• [asterisk-users] Harris picking up before extension   Alyed Tzompa
• [asterisk-users] app_pppd - Could not read send data   Stefan Tichy
• [asterisk-users] Harris picking up before extension   Eric "ManxPower" Wieling
• [asterisk-users] Queues and Timeouts.   Thomas Kenyon
• [asterisk-users] WIFI phones on asterisk   Jerry Geis
• [asterisk-users] WIFI phones on asterisk   Jay R. Ashworth
• [asterisk-users] config template for Grandstreams   Todd- Asterisk
• [asterisk-users] Stable clock with 2.6 and without Digium hardware.   Dan Austin
• [asterisk-users] Latest Debian and latest zaptel   Christian
• [asterisk-users] Voxee lag problems ?   Rajeev Natarajan
• [asterisk-users] Latest Debian and latest zaptel   John covici
• [asterisk-users] monitor-join does not seem to work.   Carlos Alberto Hastenreiter Assumpção
• [asterisk-users] Latest Debian and latest zaptel   Christian
• [asterisk-users] (no subject)   Stas Khromoy
• [asterisk-users] Latest Debian and latest zaptel   John covici
• [asterisk-users] Latest Debian and latest zaptel   Michiel van Baak
• [asterisk-users] Microsoft will enter VoIP market in earnest next year, says Ballmer   Brodie Macleod
• [asterisk-users] Voxee lag problems ?   Tim Panton
• [asterisk-users] Re: Monitor, MixMonitor and volume levels   Leo Ann Boon
• [asterisk-users] Microsoft will enter VoIP market in earnest nextyear, says Ballmer   Dean Collins
• [asterisk-users] Microsoft will enter VoIP market in earnest next year, says Ballmer   Matt Birmingham
• [asterisk-users] Latest Debian and latest zaptel   Christian
• [asterisk-users] (no subject)   Tom Vile
• [asterisk-users] WIFI phones on asterisk   Andrew Joakimsen
• [asterisk-users] SPA-941 (and others ) Transmit Sound Quality   Ron Winograd
• [asterisk-users] Push to Talk settings.   Jonson Player
• [asterisk-users] Dialing from "Placed Calls" on Polycom IP501 doesn't always work   Anthony Rodgers
• [asterisk-users] Voxee lag problems ?   Tom Lynn
• [asterisk-users] Latest Debian and latest zaptel   Christian
• [asterisk-users] app_swift: Failed to set voice   Earle Clubb
• [asterisk-users] Latest Debian and latest zaptel   brandon kruz
• [asterisk-users] Soundfiles adding during phone calls   Ronald Wiplinger
• [asterisk-users] Latest Debian and latest zaptel   Tzafrir Cohen
• [asterisk-users] chansp core dump   Thirumal Saminathan
• [asterisk-users] config template for Grandstreams   Alberto Pastore
• [asterisk-users] A couple of new tutorials: installing * 1.4 and the Asterisk GUI   Thirumal Saminathan
• [asterisk-users] A couple of new tutorials: installing * 1.4 and the Asterisk GUI   Tzafrir Cohen
• [asterisk-users] asterisk and norstar   Jorge Mendoza
• [asterisk-users] WIFI phones on asterisk   Jorge Mendoza
• [asterisk-users] Soundfiles adding during phone calls   Andrew Joakimsen
• [asterisk-users] Call file: CallerID problem   Arik Raffael Funke
• [asterisk-users] Latest Debian and latest zaptel   Christian
• [asterisk-users] Latest Debian and latest zaptel   Bob Chiodini
• [asterisk-users] Voxee lag problems ?   Guillermo Salas M.
• [asterisk-users] can't hear MusicOnHold when zap answers   Louis-David Mitterrand
• [asterisk-users] Call file: CallerID problem   Doug Lytle
• [asterisk-users] Latest Debian and latest zaptel   Tzafrir Cohen
• [asterisk-users] Dialing from "Placed Calls" on Polycom IP501 doesn't always work   Noah Miller
• [asterisk-users] Need to automatically park an incoming call andthen connect to an extension.   Dovid B
• [asterisk-users] Re: WIFI phones on asterisk   Martin Joseph
• [asterisk-users] Latest Debian and latest zaptel   Christian
• [asterisk-users] Presence-awareness in Asterisk   Rajeev Natarajan
• [asterisk-users] Voxee lag problems ?   Vicky
• [asterisk-users] Latest Debian and latest zaptel   Tzafrir Cohen
• [asterisk-users] Presence-awareness in Asterisk   Tzafrir Cohen
• [asterisk-users] Latest Debian and latest zaptel   Bob Chiodini
• [asterisk-users] sip forward behind a nat   nik600
• [asterisk-users] A couple of new tutorials: installing * 1.4 and the Asterisk GUI   Lenz
• [asterisk-users] Re: Choppy sound in voicemail using Asterisk 1.2.11 on CENTOS4 guest on vmware server   Martin Joseph
• [asterisk-users] Latest Debian and latest zaptel   Christian
• [asterisk-users] Latest Debian and latest zaptel   Tzafrir Cohen
• [asterisk-users] Presence-awareness in Asterisk   Michiel van Baak
• [asterisk-users] Voxee lag problems ?   Tom Lynn
• [asterisk-users] Voxee lag problems ?   Tom Lynn
• [asterisk-users] Re: Choppy sound in voicemail using Asterisk 1.2.11 on CENTOS4 guest on vmware server   Mauro Crovato
• [asterisk-users] Soundfiles adding during phone calls   Ronald Wiplinger
• [asterisk-users] No sounds in svn version?   Christian
• [asterisk-users] Latest Debian and latest zaptel   Tzafrir Cohen
• [asterisk-users] No sounds in svn version?   Bruce Reeves
• [asterisk-users] CLI message: remote unix connection disconnected   Yu Safin
• [asterisk-users] Re: CLI message: remote unix connection disconnected   Yu Safin
• [asterisk-users] Shared Line Appearances in 1.4   Lacy Moore - Aspendora
• [asterisk-users] Soundfiles adding during phone calls   Tom Lynn
• [asterisk-users] Soundfiles adding during phone calls   Ronald Wiplinger
• [asterisk-users] sip forward behind a nat   Rosli Sukri
• [asterisk-users] sip forward behind a nat   Vicky
• [asterisk-users] Knowing when an answerphone answers   Nic Hughes
• [asterisk-users] Re: Outgoing problem on PRI   Mohamed A. Gombolaty
• [asterisk-users] sip forward behind a nat   nik600
• [asterisk-users] Asterisk architecture   G(P)L
• [asterisk-users] Knowing when an answerphone answers   Anselm Martin Hoffmeister
• [asterisk-users] dynamically modifying the dialplan?   Norbert Zawodsky
• [asterisk-users] dynamically modifying the dialplan?   Vicky
• [asterisk-users] dynamically modifying the dialplan?   Michiel van Baak
• [asterisk-users] Example Polycom function key config   Doug Lytle
• [asterisk-users] Latest Debian and latest zaptel   Christian
• [asterisk-users] asterisk-addons 1.4 SVN fails to compile   Robert La Ferla
• [asterisk-users] Speeding up SayDigits?   Robert La Ferla
• [asterisk-users] Looking for a simple TFTP server for Linux   Christian
• [asterisk-users] Looking for a simple TFTP server for Linux   Doug Lytle
• [asterisk-users] Asterisk Media Gateway   Osama Kamal
• [asterisk-users] Looking for a simple TFTP server for Linux   Christian
• [asterisk-users] Looking for a simple TFTP server for Linux   Tzafrir Cohen
• [asterisk-users] Re: dynamically modifying the dialplan?   Benny Amorsen
• [asterisk-users] Re: dynamically modifying the dialplan?   Benny Amorsen
• [asterisk-users] Some pictures from Astricon 2006 in Dallas   Rob Lith
• [asterisk-users] CLI message: remote unix connection disconnected   Dovid B
• [asterisk-users] Some pictures from Astricon 2006 in Dallas   Dean Collins
• [asterisk-users] VM problems...   Alexander Topolanek
• [asterisk-users] Looking for a simple TFTP server for Linux   Andrew Joakimsen
• [asterisk-users] Re: Choppy sound in voicemail using Asterisk1.2.11 on CENTOS4 guest on vmware server   Webster, Andrew
• [asterisk-users] outgoing works, incoming fails on asterisk passthrough to inter-tel   Nathan Bell
• [asterisk-users] Determine if Call is from a cellular phone   Dovid B
• [asterisk-users] Asterisk billing   Vicky
• [asterisk-users] Determine if Call is from a cellular phone   Andrew Joakimsen
• [asterisk-users] Determine if Call is from a cellular phone   John Novack
• [asterisk-users] same extension on softphones and hardphones   Jeronimo Romero
• [asterisk-users] Random 'no audio' problem   Jordi Nelissen
• [asterisk-users] operator console   Jordi Nelissen
• [asterisk-users] Determine if Call is from a cellular phone   Dovid B
• [asterisk-users] Determine if Call is from a cellular phone   Ron McLeod
• [asterisk-users] same extension on softphones and hardphones   Anselm Martin Hoffmeister
• [asterisk-users] best gui   Jordi Nelissen
• [asterisk-users] Determine if Call is from a cellular phone   Dovid B
• [asterisk-users] Re: Question about Mitel phones   Jesse Peterson
• [asterisk-users] cadences zapata.conf   joe a.
• [asterisk-users] IAX2 one way audio   joe a. (joea at j4computers.com)
• [asterisk-users] same extension on softphones and hardphones   Jeronimo Romero
• [asterisk-users] same extension on softphones and hardphones   Cullin J. Wible
• [asterisk-users] Headaches with Video over SIP   Peter Howard
• [asterisk-users] Headaches with Video over SIP   Peter Howard
• [asterisk-users] Headaches with Video over SIP   Patrick
• [asterisk-users] Determine if Call is from a cellular phone   Alexander Lopez
• [asterisk-users] Headaches with Video over SIP   Peter Howard
• [asterisk-users] Zaptel compile problems   Traue, Paul
• [asterisk-users] Slow to get dialtone when going off hook - big problem for me :(   Jim Archer
• [asterisk-users] Slow to get dialtone when going off hook - big problem for me :(   Steve Totaro
• [asterisk-users] Slow to get dialtone when going off hook - big problem for me :(   Jim Archer
• [asterisk-users] Slow to get dialtone when going off hook -big problem for me :(   Dovid B
• [asterisk-users] Trixbox dialout problems   Tim Uckun
• [asterisk-users] Slow to get dialtone when going off hook - big problem for me :(   John Novack
• [asterisk-users] Slow to get dialtone when going off hook - bigproblem for me :(   Dovid B
• [asterisk-users] Slow to get dialtone when going off hook - bigproblem for me :(   John Novack
• [asterisk-users] Asterisk VM with Cisco routing   Curt Shaffer
• [asterisk-users] Slow to get dialtone when going off hook - bigproblem for me :(   Jim Archer
• [asterisk-users] Headaches with Video over SIP   Rosli Sukri
• [asterisk-users] Problem with CDR interpretation   Michał Niklas
• [asterisk-users] Headaches with Video over SIP   Peter Howard
• [asterisk-users] Zaptel compile problems   Tzafrir Cohen
• [asterisk-users] Moh stops immediately   zen Perry
• [asterisk-users] Can i have two asterisk vcersions running on same PC??   Sri Keerthy
• [asterisk-users] Can i have two asterisk versions running on same PC??   Sri Keerthy
• [asterisk-users] Re: Moh stops immediately   Martin Joseph
• [asterisk-users] Re: Moh stops immediately   zen Perry
• [asterisk-users] Re: IAX2 one way audio   Martin Joseph
• [asterisk-users] Can i have two asterisk vcersions running on same PC??   Tzafrir Cohen
• [asterisk-users] operator console   Vicky
• [asterisk-users] IAX2 one way audio   Vicky
• [asterisk-users] Sending '#' with Dial   Emil Thelin
• [asterisk-users] several behind NAT   kjcsb
• [asterisk-users] Can i have two asterisk vcersions running on same PC??   Rob Hillis
• [asterisk-users] Re: Moh stops immediately   Vicky
• [asterisk-users] Slow to get dialtone when going off hook - bigproblem for me :(   Vicky
• [asterisk-users] Voicemail and realtime : the emailbody option ...   Jean-Baptiste Bellet
• [asterisk-users] Dial : Executing context/priority after bridge?   Yuri Veremeyenko
• [asterisk-users] Dial : Executing context/priority after bridge?   Vicky
• [asterisk-users] Desktop integration   Ondrej Valousek
• [asterisk-users] Asterisk IVR functionality   nik600
• [asterisk-users] Can i have two asterisk vcersions running on same PC??   Pavel Siderov
• [asterisk-users] Desktop integration   Vij
• [asterisk-users] Desktop integration   Michał Niklas
• [asterisk-users] Asterisk IVR functionality   Pavel Siderov
• Fwd: [asterisk-users] Asterisk IVR functionality   Rajkumar S
• [asterisk-users] Asterisk IVR functionality   Brian Rogan
• [asterisk-users] operator console   Jordi Nelissen
• [asterisk-users] Asterisk IVR functionality   nik600
• [asterisk-users] Asterisk IVR functionality   nik600
• [asterisk-users] Question about the GUI for 1.4   Christian
• [asterisk-users] Desktop integration   Ondrej Valousek
• [asterisk-users] sip forward behind a nat   nik600
• [asterisk-users] Presence-awareness in Asterisk   Olivier
• FW: [asterisk-users] Desktop integration   Ondrej Valousek
• [asterisk-users] How to get CDR to show answered calls only   Olivier
• [asterisk-users] Problem with internet down   Andre Luiz Martins
• [asterisk-users] Problem with internet down   Al Bochter
• [asterisk-users] Problem with internet down   Andre Luiz Martins
• [asterisk-users] Problem with internet down   Al Bochter
• [asterisk-users] Problem with internet down   Steve Langstaff
• FW: [asterisk-users] Desktop integration   mitcheloc
• FW: [asterisk-users] Desktop integration   Tom Vile
• [asterisk-users] asterisk and norstar   Gustavo Berman
• FW: [asterisk-users] Desktop integration   Ondrej Valousek
• [asterisk-users] How to get CDR to show answered calls only   Vicky
• [asterisk-users] Defunct / zombie AGI after some execution time   Mark
• [asterisk-users] Problem with internet down   Andre Luiz Martins
• FW: [asterisk-users] Desktop integration   bails
• [asterisk-users] Custom voicemail extension greeting   joe a.
• [asterisk-users] operator console   Vicky
• [asterisk-users] Re: Desktop integration   Steven
• FW: [asterisk-users] Desktop integration   Michiel van Baak
• [asterisk-users] Re: Monitor, MixMonitor and volume levels   Steve Davies
• [asterisk-users] Problem with internet down   Time Bandit
• [asterisk-users] problem with redirects   Ricardo Carvalho
• [asterisk-users] Problem with internet down   Steve Langstaff
• [asterisk-users] Asterisk IVR functionality   Tzafrir Cohen
• [asterisk-users] Random 'no audio' problem   Matt
• FW: [asterisk-users] Desktop integration   Dave Cotton
• [asterisk-users] Re: Desktop integration   Yu Safin
• [asterisk-users] ChanSpy problems in Unicall to SIP calls   Guillermo Freige
• [asterisk-users] Custom voicemail extension greeting   Anselm Martin Hoffmeister
• [asterisk-users] Slow playback of sound prompts   Roger Lewau
• [asterisk-users] Recording outbound analog calls with X100P   Matthew J. Roth
• [asterisk-users] Custom voicemail extension greeting   joe a.
• [asterisk-users] 2 * servers Host=ip - doesn't work Host=dynamic with register is OK, why?   Marco Mouta
• [asterisk-users] Re: Desktop integration   Andrew Kohlsmith
• [asterisk-users] Asterisk Call Statistics   omar parihuana
• [asterisk-users] Voicemail multiple languages   Guerid Salim
• [asterisk-users] Modprobe Zaptel   Julian Varanini
• FW: [asterisk-users] Desktop integration   Michiel van Baak
• [asterisk-users] newbie question   blackwater dev
• [asterisk-users] problem with redirects   Ricardo Carvalho
• [asterisk-users] Fast Busy with autodial using a call file   James Hammer
• [asterisk-users] Music on hold question   Christian
• [asterisk-users] DSl and more then 1 call   Kelly Opal
• [asterisk-users] Mysql 6 second rounding   Vicky
• [asterisk-users] asterisk as a Media Gateway   Osama Kamal
• [asterisk-users] Problem with internet down   Vicky
• [asterisk-users] Can AGI do this?   Bret Schuhmacher
• [asterisk-users] sip forward behind a nat   Vicky
• [asterisk-users] Mysql 6 second rounding   James Coberly
• [asterisk-users] DSl and more then 1 call   Vicky
• [asterisk-users] Modprobe Zaptel   Julian Varanini
• [asterisk-users] Asterisk with ss7 and sip-t   Josué Conti
• [asterisk-users] FAX using T38   Ricardo Carvalho
• [asterisk-users] Voicemail argument size limit   Donald Stahl
• [asterisk-users] Mysql 6 second rounding   Damon Estep
• [asterisk-users] Survey: In what ways do you use Asterisk at your house?   Earle Clubb
• [asterisk-users] Mysql 6 second rounding   Vicky
• [asterisk-users] DSl and more then 1 call   Kelly Opal
• [asterisk-users] Native TDM Bridge   Forrest Beck
• [asterisk-users] Voicemail argument size limit   Doug Lytle
• [asterisk-users] Mysql 6 second rounding   Marnus van Niekerk
• [asterisk-users] MWI not working in 1.4   Mark Hulber
• [asterisk-users] Dial/Continue/Announce   Matthew Rubenstein
• [asterisk-users] Voicemail argument size limit   Donald Stahl
• [asterisk-users] SIP Ports (1000 to 2000 works)   Al Bochter
• [asterisk-users] Mysql 6 second rounding   Vicky
• [asterisk-users] Voicemail argument size limit   Doug Lytle
• [asterisk-users] SIP Ports (1000 to 2000 works)   Vicky
• [asterisk-users] Voicemail argument size limit   Donald Stahl
• [asterisk-users] Voicemail argument size limit   Doug Lytle
• [asterisk-users] Survey: In what ways do you use Asterisk at your house?   Dave Fullerton
• [asterisk-users] Voicemail argument size limit   Donald Stahl
• [asterisk-users] Survey: In what ways do you use Asterisk at your house?   Michiel van Baak
• [asterisk-users] Question about MySQL Fetch foundRow from the dial plan   Tom Vile
• [asterisk-users] Survey: In what ways do you use Asterisk at your house?   mitcheloc
• [asterisk-users] Music on hold question   Tzafrir Cohen
• [asterisk-users] Mysql 6 second rounding   Steve Edwards
• [asterisk-users] Survey: In what ways do you use Asterisk at your house?   Yu Safin
• [asterisk-users] Survey: In what ways do you use Asterisk at your house?   Colin Anderson
• [asterisk-users] Music on hold question   Christian
• [asterisk-users] Headaches with Video over SIP   Peter Howard
• [asterisk-users] SIP Ports (1000 to 2000 works)   Al Bochter
• [asterisk-users] Zaptel   Julian Varanini
• [asterisk-users] Headaches with Video over SIP   Peter Howard
• [asterisk-users] "Username/auth name mismatch" + SIP phone can't connect?   Fred
• [asterisk-users] SIP Ports (1000 to 2000 works)   Vicky
• [asterisk-users] SIP Ports (1000 to 2000 works)   Peter Bowyer
• [asterisk-users] Recommend software version for Cisco IOS Gateways and SIP Phones?   Heyer, JohnX
• [asterisk-users] Headaches with Video over SIP   Peter Howard
• [asterisk-users] STUN with one public and one private IP?   Steve Sobol
• [asterisk-users] Survey: In what ways do you use Asterisk at your house?   mitcheloc
• [asterisk-users] "Username/auth name mismatch" + SIP phone can't connect?   Anselm Martin Hoffmeister
• [asterisk-users] Bellsouth issue ?   Dovid B
• Problem found Re: [asterisk-users] Headaches with Video over SIP   Peter Howard
• [asterisk-users] Survey: In what ways do you use Asterisk at your house?   Michael Graves
• [asterisk-users] Survey: In what ways do you use Asterisk at your house?   Anselm Martin Hoffmeister
• [asterisk-users] establish meetme limit for a single room   Damon Estep
• [asterisk-users] SIP Ports (1000 to 2000 works)   Al Bochter
• [asterisk-users] STUN with one public and one private IP?   Al Bochter
• [asterisk-users] How to get CDR to show answered calls only   Damon Estep
• [asterisk-users] Dual Wan Router with Failover   Dovid B
• [asterisk-users] Data over zaptel   Andrew Joakimsen
• [asterisk-users] Survey: In what ways do you use Asterisk at yourhouse?   Dovid B
• [asterisk-users] "Username/auth name mismatch" + SIP phone can'tconnect?   Dovid B
• [asterisk-users] CDR shows NO ANSWER when call is really ANSWERED   voiplist
• [asterisk-users] Linksys doesn't resync properly and doesn't get provisioning from TFTP and HTTP   Zeeshan Zakaria
• [asterisk-users] Application Directory question   Tom Vile
• [asterisk-users] Load balance Asterisk servers?   voiplist
• [asterisk-users] Re: Linksys doesn't resync properly and doesn't get provisioning from TFTP and HTTP   Zeeshan Zakaria
• [asterisk-users] Bellsouth issue ?   Bob Chiodini
• [asterisk-users] Load balance Asterisk servers?   Douglas Garstang
• [asterisk-users] Newbie Questions . . .   Jason Flatt
• [asterisk-users] Newbie Questions . . .   Sharon Lim
• [asterisk-users] chanspy -coredump( asterisk 1.4)   Thirumal Saminathan
• [asterisk-users] Dialing from "Placed Calls" on Polycom IP501 doesn't always work   James Andrewartha
• [asterisk-users] Newbie Questions . . .   Tom Lynn
• [asterisk-users] Definity Asterisk CallerID Issue   Tom Lynn
• [asterisk-users] SIP Ports (1000 to 2000 works)   Vicky
• [asterisk-users] Installation of Unicall for MFC/R2   Angel Heart
• [asterisk-users] bindport   Khaled
• [asterisk-users] Load balance Asterisk servers?   Stelios Koroneos
• [asterisk-users] Music on hold question   Tzafrir Cohen
• [asterisk-users] sip forward behind a nat   nik600
• [asterisk-users] Redirecting Calls   Jason Frisch
• [asterisk-users] SIP Ports (1000 to 2000 works)   Al Bochter
• [asterisk-users] Re: Linksys doesn't resync properly and doesn't get provisioning from TFTP and HTTP   Tzafrir Cohen
• [asterisk-users] Asterisk IVR functionality   Tim Panton
• [asterisk-users] Desktop integration   Tim Panton
• [asterisk-users] Is asterisk able to integrate with MS SQL   Sharon Lim
• [asterisk-users] Re: Linksys doesn't resync properly and doesn't get provisioning from TFTP and HTTP   Zeeshan Zakaria
• [asterisk-users] asterisk as a Media Gateway   Osama Kamal
• [asterisk-users] Survey: In what ways do you use Asterisk at yourhouse?   mitcheloc
• Problem found Re: [asterisk-users] Headaches with Video over SIP   Steve Langstaff
• [asterisk-users] Is asterisk able to integrate with MS SQL   Vicky
• [asterisk-users] Re: Is asterisk able to integrate with MS SQL   Tony Mountifield
• [asterisk-users] sip forward behind a nat   Vicky
• [asterisk-users] Re: Is asterisk able to integrate with MS SQL   Tony Mountifield
• [asterisk-users] Stable clock with 2.6 and without Digium hardware.   Anton Tinchev
• [asterisk-users] Stable clock with 2.6 and without Digium hardware.   Anton Tinchev
• [asterisk-users] Re: Is asterisk able to integrate with MS SQL   Stéphane LASSERRE
• [asterisk-users] Re: Is asterisk able to integrate with MS SQL   Vicky
• [asterisk-users] Stable clock with 2.6 and without Digium hardware.   zoachien at securax.org
• [asterisk-users] [Voicemail] Change the format of the VM_DATE   Jean-Baptiste Bellet
• [asterisk-users] Can I disable send e-mail feature in the voicemail application?   Ma Zhiyong
• [asterisk-users] DSl and more then 1 call   Bob Chiodini
• [asterisk-users] Desktop integration   Michał Niklas
• [asterisk-users] Polycom - how to 'buddy watch' trunks?   Robert Jenkins
• [asterisk-users] Zaptel and limiting number off channels channels   Rodrigo Ricardo Passos
• [asterisk-users] Dropping Connections   Mike Heininger
• [asterisk-users] Can I disable send e-mail feature in the voicemail application?   Vicky
• [asterisk-users] (no subject)   Phillip Jackson
• [asterisk-users] Dual Wan Router with Failover   Todd- Asterisk
• [asterisk-users] 900 rules   Doug Crompton
• [asterisk-users] Dual Wan Router with Failover   Dean Collins
• [asterisk-users] 900 rules   Eric "ManxPower" Wieling
• [asterisk-users] OutCall Release   Stephen Wingfield
• [asterisk-users] Desktop integration   Senad Jordanovic
• [asterisk-users] 900 rules   Doug Crompton
• [asterisk-users] Dual Wan Router with Failover   Dean Collins
• [asterisk-users] Broken Call Screening   Gary T. Giesen
• [asterisk-users] Dual Wan Router with Failover   Mailing List
• [asterisk-users] Load balance Asterisk servers?   David Thomas
• [asterisk-users] Fax killed on all zaptel devices   Mohamed A. Gombolaty
• [asterisk-users] 900 rules   Tim Sharp
• [asterisk-users] 900 rules   Chris Mazuc
• [asterisk-users] Re: 900 rules   Steven
• [asterisk-users] 900 rules   Eric "ManxPower" Wieling
• [asterisk-users] 900 rules   Steve Sobol
• [asterisk-users] Dual Wan Router with Failover   George Gardiner
• [asterisk-users] Dual Wan Router with Failover   Jordan Kirby
• [asterisk-users] Dual Wan Router with Failover   Dean Collins
• [asterisk-users] Problem with FXS ports of TDM400P   Gustavo Felisberto
• [asterisk-users] Dual Wan Router with Failover   Jason
• [asterisk-users] Problems with voicemail   Christian
• [asterisk-users] Re: 900 rules   Steven
• [asterisk-users] Load balance Asterisk servers?   Aaron Daniel
• [asterisk-users] Dual Wan Router with Failover   Mailing List
• [asterisk-users] Dual Wan Router with Failover   Matthew Rubenstein
• [asterisk-users] Dual Wan Router with Failover   Jason
• [asterisk-users] Dialplan options   joe a.
• [asterisk-users] Retain call control: Avoid letting call get into cellular voicemail   Nilesh Londhe
• [asterisk-users] asterisk sip doesn't see other asterisk-sip   Louis-David Mitterrand
• [asterisk-users] config template for Grandstreams   Gordon Henderson
• [asterisk-users] 900 rules   Jay R. Ashworth
• [asterisk-users] Dual Wan Router with Failover   Dean Collins
• [asterisk-users] Dual Wan Router with Failover   Jeronimo Romero
• [asterisk-users] Re: Broken Call Screening   Justin Newman
• [asterisk-users] 900 rules   Brian Kaye
• [asterisk-users] Load balance Asterisk servers?   Douglas Garstang
• [asterisk-users] config template for Grandstreams   Todd- Asterisk
• [asterisk-users] Newbie Questions . . .   Henry.L.Coleman
• [asterisk-users] config template for Grandstreams   Todd- Asterisk
• [asterisk-users] Load balance Asterisk servers?   David Thomas
• [asterisk-users] config template for Grandstreams   Bob Chiodini
• [asterisk-users] 900 rules   Ron McLeod
• [asterisk-users] Retain call control: Avoid letting call get into cellular voicemail (was: Dialplan options)   Anselm Martin Hoffmeister
• [asterisk-users] Dialing from "Placed Calls" on Polycom IP501doesn't always work   Anthony Rodgers
• [asterisk-users] Polycom - how to 'buddy watch' trunks?   Anthony Rodgers
• [asterisk-users] Survey: In what ways do you use Asterisk at your house?   Ira
• [asterisk-users] DUNDi Asterisk Cluster   David Thomas
• [asterisk-users] OutCall Release   Tzafrir Cohen
• [asterisk-users] In the beginning-The first question.   James R. Stevens
• Problem found Re: [asterisk-users] Headaches with Video over SIP   Peter Howard
• [asterisk-users] In the beginning-The first question.   Doug Lytle
• [asterisk-users] OutCall Release   Senad Jordanovic
• [asterisk-users] Problem with FXS ports of TDM400P   Gustavo Felisberto
• [asterisk-users] unable to get channel lock BAD BAD BAD   Tim Uckun
• [asterisk-users] Load balance Asterisk servers?   Aaron Daniel
• [asterisk-users] Re: "Username/auth name mismatch" + SIP phone can't connect?   Fred
• [asterisk-users] DUNDi Asterisk Cluster   Aaron Daniel
• [asterisk-users] How do I change the rtp packet size in a Cisco 7490 from 10ms to 20ms   Naija Man
• [asterisk-users] Problem with FXS ports of TDM400P   Gustavo Felisberto
• [asterisk-users] Voice mail transfer between 2 asterisk servers   Naija Man
• [asterisk-users] Problem with FXS ports of TDM400P   Robert Jenkins
• [asterisk-users] In the beginning-The first question.   James R. Stevens
• [asterisk-users] In the beginning-The first question.   Doug Lytle
• [asterisk-users] How to use Sipura SPA3k POTS line to dial Asterisk SIP phones?   Larry Alkoff
• [asterisk-users] Re: Is asterisk able to integrate with MS SQL   Sharon Lim
• [asterisk-users] Caller ID in Sweden not working and looking for and voices   Mattias Andersson
• [asterisk-users] Caller ID in Sweden not working and looking for and voices   Mattias Andersson
• [asterisk-users] Caller ID in Sweden not working and looking for and voices   Anselm Martin Hoffmeister
• [asterisk-users] Re: Grandstream TFTP system wide settings   Zeeshan Zakaria
• [asterisk-users] Re: Why only one out of many IP Phones re-registering every one minute   Zeeshan Zakaria
• [asterisk-users] Problem with FXS ports of TDM400P   Dovid B
• [asterisk-users] Retain call control: Avoid letting call get intocellular voicemail   Dovid B
• [asterisk-users] Defunct / zombie AGI after some execution time   Andrew Joakimsen
• [asterisk-users] Can AGI do this?   Octavio Ruiz (Ta^3)
• [asterisk-users] Re: DUNDi Asterisk Cluster   JR Richardson
• [asterisk-users] ATA with reliable FAX?   Curt Shaffer
• [asterisk-users] Re: Grandstream TFTP system wide settings   Todd- Asterisk
• [asterisk-users] ATA with reliable FAX?   George Pajari
• [asterisk-users] Dual Wan Router with Failover   Dovid B
• [asterisk-users] Can I disable send e-mail feature in the voicemailapplication?   Dovid B
• [asterisk-users] Desktop integration   Dovid B
• [asterisk-users] Reception Console   Paul Hales
• [asterisk-users] Problem with FXS ports of TDM400P   Gustavo Felisberto
• [asterisk-users] Installation of Unicall for MFC/R2   Moises Silva
• [asterisk-users] Call log reveals redundant calls!   Mike Benjamin
• [asterisk-users] Installation of Unicall for MFC/R2   Angel Heart
• [asterisk-users] Call log reveals redundant calls!   Dovid B
• [asterisk-users] trixbox + agi   blackwater dev
• [asterisk-users] Desktop integration   Matthew Rubenstein
• [SPAM HEADER] - [asterisk-users] trixbox + agi - Email found in subject   Cory Andrews
• [asterisk-users] trixbox + agi   Tim Uckun
• [asterisk-users] trixbox + agi   Tom Vile
• [asterisk-users] Add Apps to Asterisk?   Matthew Rubenstein
• [asterisk-users] Caller Initiated Conference   R.Linga Reddy
• [asterisk-users] TDD - stops receiving characters   Warrick Zedi
• [asterisk-users] trixbox + agi   Tim Uckun
• [asterisk-users] How to use Sipura SPA3k POTS line to dial Asterisk SIP phones?   Doug Crompton
• [asterisk-users] Retain call control: Avoid letting call get   Philippe Lindheimer
• [asterisk-users] [RESOLUTION] Polycom microbrowser issue Error HTTP 406 withIIS   Steve Gladden
• [asterisk-users] Add Apps to Asterisk?   Darryl Dunkin
• [asterisk-users] Caller ID in Sweden not working and looking for and voices   Mattias Andersson
• [asterisk-users] Caller ID in Sweden not working and looking for and voices   Mattias Andersson
• [asterisk-users] Condensing queue CDRs into single entry   Rajkumar S
• [asterisk-users] Re: Moh stops immediately   Martin Joseph
• Problem found Re: [asterisk-users] Headaches with Video over SIP   Steve Langstaff
• [asterisk-users] In the beginning-The first question.   Steve Langstaff
• [asterisk-users] How to use Voipjet or any Voip provider Trunk from my mobile through fxo and fxs ports?   Crazy Boy
• [asterisk-users] How to do the Call Snooping   raviprakash sunkara
• [asterisk-users] Add Apps to Asterisk?   Andrea Spadaccini
• [asterisk-users] Add Apps to Asterisk?   Mattias Andersson
• [asterisk-users] A question on ISDN cards... (in the UK)   Gordon Henderson
• [asterisk-users] How to do the Call Snooping   Vij
• [asterisk-users] A question on ISDN cards... (in the UK)   Senad Jordanovic
• [asterisk-users] Time Based Voicemail Messages   Wildheart
• [asterisk-users] Time Based Voicemail Messages   Wildheart
• [asterisk-users] Time Based Voicemail Messages   Marnus van Niekerk
• [asterisk-users] Time Based Voicemail Messages   Benjamin Jacob
• [asterisk-users] Asterisk as a SIP client, Need to auto-answer   Ehsan Khosrowshahi
• [asterisk-users] A question on ISDN cards... (in the UK)   Conrad Wood
• [asterisk-users] Time Based Voicemail Messages   Wildheart
• [asterisk-users] Re: Is asterisk able to integrate with MS SQL   Tony Mountifield
• [asterisk-users] A question on ISDN cards... (in the UK)   Marco Mouta
• [asterisk-users] A question on ISDN cards... (in the UK)   Steve Kennedy
• [asterisk-users] Zaptel configurations in India   K Y Iyer
• [asterisk-users] T38 problem   Tomislav Parčina
• [asterisk-users] Re: unable to get channel lock BAD BAD BAD   Tomislav Parčina
• [asterisk-users] some questions about atxfer usage   Antonio Almodóvar
• [asterisk-users] Re: Is asterisk able to integrate with MS SQL   Bruce Reeves
• [asterisk-users] Re: ATA with reliable FAX?   Tomislav Parčina
• [Asterisk-Users] The best available CAPI BRI card for Asterisk ?   Olivier
• [asterisk-users] Re: Re: Voicemail Press '0'   Brian Roy
• [asterisk-users] Asterisk - big installation   doki_cti
• [asterisk-users] How to disable the 482 Loop Detected messages sent by Asterisk   Ricardo Carvalho
• [asterisk-users] How to disable the 482 Loop Detected messages sent by Asterisk   Eric "ManxPower" Wieling
• [asterisk-users] Re: Is asterisk able to integrate with MS SQL   Wes Baehr
• [asterisk-users] trixbox + agi   blackwater dev
• [asterisk-users] SIP NOTIFY routing problem   Steve Langstaff
• [asterisk-users] Time Based Voicemail Messages   C F
• [asterisk-users] State of a public number   Giordano Grandis
• [asterisk-users] Soundfiles adding during phone calls   Ronald Wiplinger
• [asterisk-users] Time Based Voicemail Messages   tracinet
• [asterisk-users] ODBC Voicemail Storage   Edwin Horton
• [asterisk-users] Add Apps to Asterisk?   Matthew Rubenstein
• [asterisk-users] trixbox + agi   Tom Vile
• [asterisk-users] Re: Broken Call Screening   Gary T. Giesen
• [asterisk-users] Setting the CallerID   Tobias Wolf
• [asterisk-users] Disabling Features Temporarily   Mailing List
• [asterisk-users] Problems with language support   Diego Andres Asenjo G.
• [asterisk-users] ODBC Voicemail Storage   Dan Austin
• [asterisk-users] some questions about atxfer usage   Conrad Wood
• [asterisk-users] Page() Function Timeout   Ken Williams
• [asterisk-users] Setting the CallerID   Conrad Wood
• [asterisk-users] Recording outbound analog calls with X100P   Matthew J. Roth
• [asterisk-users] Recording outbound analog calls with X100P   Conrad Wood
• [asterisk-users] Asterisk - big installation   Conrad Wood
• [asterisk-users] Page() Function Timeout   Ken Williams
• [asterisk-users] dtmf tones not always recognized   Don Pobanz
• [asterisk-users] Got 200 OK on REGISTER that isn't a register   Stas Khromoy
• [asterisk-users] quadbri + kernel 2.6.18.1   Paco Brufal
• [asterisk-users] Page() Function Timeout   Steven Ringwald
• [asterisk-users] Handy tip for intercom with FreePBX & Grandstream phones   Ken Williams
• [asterisk-users] Got 200 OK on REGISTER that isn't a register   Stas Khromoy
• [asterisk-users] Got 200 OK on REGISTER that isn't a register   Ron McLeod
• [Asterisk-Users] Got 200 OK on REGISTER that isn't a register   Stas Khromoy
• [asterisk-users] safe_asterisks pawning multiple asterisk process???   Andre Courchesne - Consultant
• [asterisk-users] PHPAGI example usage of input.php   Tom Vile
• [asterisk-users] Recording outbound analog calls with X100P   Hadley Rich
• [asterisk-users] PHPAGI example usage of input.php   Jay Moore
• [asterisk-users] PHPAGI example usage of input.php   Tom Vile
• [asterisk-users] Problem with FXS ports of TDM400P   Noah Miller
• [asterisk-users] Dialing from "Placed Calls" on Polycom IP501doesn't always work   Noah Miller
• [asterisk-users] safe_asterisks pawning multiple asterisk process???   Vicky
• [asterisk-users] Question about TFTPD server   Christian
• [asterisk-users] PHPAGI example usage of input.php   Jay Moore
• [asterisk-users] Recording outbound analog calls with X100P   Time Bandit
• [asterisk-users] Question about TFTPD server   Ron McLeod
• [asterisk-users] Huawei Videophone   Nabeel Jafferali
• [asterisk-users] PHPAGI example usage of input.php   Tom Vile
• [asterisk-users] PortSip and Astericks new install   Charlie Grosvenor
• [asterisk-users] trixbox + agi   Tim Uckun
• [asterisk-users] Re: unable to get channel lock BAD BAD BAD   Tim Uckun
• [asterisk-users] Monitor Zap Status   Ken Williams
• [asterisk-users] Monitor Zap Status - Full E-mail...   Ken Williams
• [asterisk-users] PortSip and Astericks new install   Vicky
• [asterisk-users] PortSip and Astericks new install   Doug Lytle
• [asterisk-users] Set port to which Asterisk should send its answer   Andre Kirchner
• [asterisk-users] trixbox + agi   Michael Collins
• [asterisk-users] web interface to control zap interface   Jordan Novak
• [asterisk-users] Monitor Zap Status - Full E-mail...   Ken Williams
• [asterisk-users] Sending '#' with Dial   Mark Hulber
• [asterisk-users] Set port to which Asterisk should send its answer   Eric "ManxPower" Wieling
• [asterisk-users] Re: safe_asterisks pawning multiple asterisk process???   Steve Murphy
• [asterisk-users] Caller ID in Sweden not working and looking for and voices   Mattias Andersson
• [asterisk-users] Page() Function Timeout   David Gagnon
• [asterisk-users] Set port to which Asterisk should send its answer   Ron McLeod
• [asterisk-users] Dialing from "Placed Calls" on PolycomIP501doesn't always work   Anthony Rodgers
• [asterisk-users] Set port to which Asterisk should send its answer   Ron McLeod
• [asterisk-users] Auth Issue using Asterisk as Voicemail AND as Normal SIP Extension.   Jean-Marc Salsa
• [asterisk-users] Grandstream Programmable Buttons & Retrieving On Hold Lines   Ken Williams
• [asterisk-users] Queue - how to provide a caller ringing tone when some agent become available   David Hajek
• [asterisk-users] PortSip and Astericks new install   Charlie Grosvenor
• [asterisk-users] Attempting native bridge of   Victor Toofic
• [asterisk-users] Question about TFTPD server   Edwin Lam
• [asterisk-users] How to use Sipura SPA3k POTS line to dial Asterisk SIP phones?   Larry Alkoff
• Problem found Re: [asterisk-users] Headaches with Video over SIP   Peter Howard
• [asterisk-users] STUN with one public and one private IP?   Steve Sobol
• [asterisk-users] simple mainmenu ivr tones not recognized   Troy Perkins
• [asterisk-users] PortSip and Astericks new install   Doug Lytle
• [asterisk-users] chan_unicall.c isntallation problem   Angel Heart
• [asterisk-users] Sending '#' with Dial   mauro at crovato.com.ar
• [asterisk-users] Grandstream GXP2000 -- What's the Catch?   Jeronimo Romero
• [asterisk-users] Intercom function on eyebeam xten softphones.   Jeronimo Romero
• [asterisk-users] T38 problem   Craig Guy
• [asterisk-users] simple mainmenu ivr tones not recognized   Jeronimo Romero
• [asterisk-users] Grandstream GXP2000 -- What's the Catch?   Tom Vile
• [asterisk-users] Grandstream GXP2000 -- What's the Catch?   Jeronimo Romero
• [asterisk-users] Grandstream GXP2000 -- What's the Catch?   Edwin Lam
• [asterisk-users] Grandstream GXP2000 -- What's the Catch?   Tom Vile
• [asterisk-users] Regular audio fade-out fade-in on IAX2 calls Asterisk 1.2.4 Hi all, One of my users has a problem with many of his calls via my Asterisk™ server. He describes the problem as having the sound slowly fade out and then fade back at a regular frequency. Has anyone experienced this before? Can you suggest a potential cause? The client is an IAX2 soft phone running on Windows XP. Asterisk version is 1.2.4. Codec is G.711-ulaw. The user is connecting from long distance away (from client phone in New Zealand to Asterisk™ server in Australia). We've confirmed that roundtrip times are fairly consistent and that the user's PC is in good working order. Would appreciate any advice. Thanks. -- Lucas   Lucas Barbuto
• [asterisk-users] How to use Sipura SPA3k POTS line to dial Asterisk SIP phones?   Doug Crompton
• [asterisk-users] Grandstream GXP2000 -- What's the Catch?   Doug Crompton
• [asterisk-users] Grandstream GXP2000 -- What's the Catch?   Alex Robar
• [asterisk-users] Anyone using the directory.agi app in AGI perl   Tom Vile
• [asterisk-users] Installing Ztdummy on Fedora Core 5   Rogers Ochieng
• [asterisk-users] Grandstream GXP2000 -- What's the Catch?   Henry.L.Coleman
• [asterisk-users] Do Not Call List   Matthew Rubenstein
• [asterisk-users] Sipura SPA3000   Larry Alkoff
• [asterisk-users] Found GSM version, but any better WAV or ULAW recordings of "Steve" or "Ian" out there?   Steve Prior
• [asterisk-users] Grandstream GXP2000 -- What's the Catch?   Ira
• [asterisk-users] Grandstream GXP2000 -- What's the Catch?   Doug Crompton
• [asterisk-users] Do Not Call List   Dean Collins
• [asterisk-users] Zaptel 1.4.0-beta2 compile error   William M. Sandiford
• [asterisk-users] Found GSM version, but any better WAV or ULAW recordings of "Steve" or "Ian" out there?   Lacy Moore - Aspendora
• [asterisk-users] Do Not Call List   Anselm Martin Hoffmeister
• [asterisk-users] Sending '#' with Dial   Emil Thelin
• [asterisk-users] Question about TFTPD server   Tzafrir Cohen
• [asterisk-users] some questions about atxfer usage   Alberto Pastore
• [asterisk-users] Re: unable to get channel lock BAD BAD BAD   Tomislav Parčina
• [asterisk-users] Re: T38 problem   Tomislav Parčina
• [asterisk-users] Grandstream GXP2000 -- What's the Catch?   Michiel van Baak
• [asterisk-users] Re: State of a public number   Tomislav Parčina
• [asterisk-users] quadbri + kernel 2.6.18.1   Tzafrir Cohen
• [asterisk-users] queue management   nik600
• [asterisk-users] Trunk outcall line ?   Noc Phibee
• [asterisk-users] Re: Regular audio fade-out fade-in on IAX2 calls Asterisk 1.2.4 Hi all, One of my users has a problem with many of his calls via my Asterisk™ server. He describes the problem as having the sound slowly fade out and then fade back at a regular frequency. Has anyone experienced this before? Can you suggest a potential cause? The client is an IAX2 soft phone running on Windows XP. Asterisk version is 1.2.4. Codec is G.711-ulaw. The user is connecting from long distance away (from client phone in New Zealand to Asterisk™ server in Australia). We've confirmed that roundtrip times are fairly consis   Martin Joseph
• [asterisk-users] queue management   Lenz
• [asterisk-users] Grandstream GXP2000 -- What's the Catch?   Zeeshan Zakaria
• [asterisk-users] Setting the CallerID   Tobias Wolf
• [asterisk-users] Cannot call Alcatel PBX extn from SIP   Shweta Jain
• [asterisk-users] Page() Function Timeout   Dinesh Nair
• [asterisk-users] chanspy crash the asterisk 1.4   Thirumal Saminathan
• [asterisk-users] Question about TFTPD server   JOAO CARLOS MOURA
• [asterisk-users] Problems with language support   Tzafrir Cohen
• [asterisk-users] spandsp 0.0.3 RxFax fax reception crashes bristuffed asterisk 1.2.13   Marcel van der Boom
• [asterisk-users] Defunct / zombie AGI after some execution time   Mark
• [asterisk-users] Recording outbound analog calls with X100P   Conrad Wood
• [asterisk-users] Grandstream GXP2000 -- What's the Catch?   Matthias Fechner
• [asterisk-users] Re: spandsp 0.0.3 RxFax fax reception crashes bristuffed asterisk 1.2.13   Nenad Radosavljevic
• [asterisk-users] spandsp 0.0.3 RxFax fax reception crashes bristuffed asterisk 1.2.13   DRi at b-w-computer.de
• [asterisk-users] T.38 - make conclusion   Tomislav Parčina
• [asterisk-users] turning off DTMF detection on Zap channels   Lenz
• [asterisk-users] spandsp 0.0.3 RxFax fax reception crashes bristuffed asterisk 1.2.13   Marcel van der Boom
• [asterisk-users] Re: ATA with reliable FAX?   George Pajari
• [asterisk-users] Sipura SPA3000   Bob Chiodini
• [asterisk-users] safe_asterisks pawning multiple asterisk process???   Tzafrir Cohen
• [asterisk-users] quadbri + kernel 2.6.18.1   Paco Brufal
• [asterisk-users] Installing Ztdummy on Fedora Core 5   Tzafrir Cohen
• [asterisk-users] Dialing from "Placed Calls" on Polycom IP501doesn't always work   John Marvin
• [asterisk-users] spandsp 0.0.3 RxFax fax reception crashes bristuffed asterisk 1.2.13   DRi at b-w-computer.de
• [asterisk-users] dialing channel late   Tamas Cseke
• [asterisk-users] T.38 - make conclusion   Doug Lytle
• [asterisk-users] chanspy crash the asterisk 1.4   Vicky
• [asterisk-users] Re: unable to get channel lock BAD BAD BAD   Benny Amorsen
• [asterisk-users] Attempting native bridge of   Vicky
• [asterisk-users] Fax pdf attachment naming badly when no caller id on zaptel   Warren (mailing lists)
• [asterisk-users] Sangoma A101 gives 'no PRI configured on span 1' error   Zeeshan Zakaria
• [asterisk-users] STUN with one public and one private IP?   Zeeshan Zakaria
• [asterisk-users] Newbie Questions . . .   Jason Flatt
• [asterisk-users] Re: unable to get channel lock BAD BAD BAD   Steve Davies
• [asterisk-users] Question about TFTPD server   mitcheloc
• [asterisk-users] Sangoma A101 gives 'no PRI configured on span 1' error   yusuf
• [asterisk-users] upgading to install-misdn 0.3.1-rc23 broke dtmf detection on some calls   Giorgio Incantalupo
• [asterisk-users] installing asterisk for Ubuntu Synaptic   blackwater dev
• [asterisk-users] Asterisk call recording   Vicky
• [asterisk-users] Backup and mail on trixox   Mattias Andersson
• [asterisk-users] installing asterisk for Ubuntu Synaptic   Brian Rogan
• [asterisk-users] installing asterisk for Ubuntu Synaptic   Brian Rogan
• [asterisk-users] Found GSM version, but any better WAV or ULAW recordings of "Steve" or "Ian" out there?   Steve Prior
• [asterisk-users] Setting the CallerID   Conrad Wood
• [asterisk-users] POS Terminals   Christopher Aloi
• [asterisk-users] Soundfiles adding during phone calls   Ronald Wiplinger
• [asterisk-users] make: execvp: build_tools/make_svn_branch_name: Permission denied   Erick Perez
• [asterisk-users] call from cisco router to asterisk gets auto attendant   SWhite at littleme.com
• R: [asterisk-users] Re: State of a public number   Giordano Grandis
• [asterisk-users] zaptel, bristuff zaphfc, and florz question   Steve Davies
• [asterisk-users] Re: zaptel, bristuff zaphfc, and florz question   Steve Davies
• [asterisk-users] STUN with one public and one private IP?   Al Bochter
• [asterisk-users] POS Terminals   Al Bochter
• [asterisk-users] jitterbuffer in pure voip (sip/iax) - what is best practice   Pavel Jezek
• [asterisk-users] Problems with language support   Diego Andres Asenjo G.
• [asterisk-users] Attempting native bridge of   Victor Toofic
• [asterisk-users] Setting the CallerID   Tobias Wolf
• [asterisk-users] Asterisk as an analog call recording solution (was: Recording outbound analog calls with X100P)   Matthew J. Roth
• [asterisk-users] Asterisk as an analog call recording solution (was: Recording outbound analog calls with X100P)   Time Bandit
• [asterisk-users] hosted asterisk   Dean Collins
• [asterisk-users] Celliax LiveCD 0.0.32 released (Asterisk managing cellular phones, and Skype calls to/from cellphones, via chan_celliax)   Giovanni Maruzzelli
• [asterisk-users] Asterisk as an analog call recording solution (was: Recording outbound analog calls with X100P)   John Novack
• [asterisk-users] POS Terminals   Christopher Aloi
• [asterisk-users] Attempting native bridge of   Victor Toofic
• [asterisk-users] FXO PCI Master abort   Jerry Rasmussen
• [asterisk-users] Re: unable to get channel lock BAD BAD BAD   Tim Uckun
• [asterisk-users] Asterisk 1.2.13 can't load module app_curl.so   Eric Pinet
• [asterisk-users] Re: unable to get channel lock BAD BAD BAD   Doug Lytle
• [asterisk-users] Re: unable to get channel lock BAD BAD BAD   Eric "ManxPower" Wieling
• [asterisk-users] asterisk billing software   Chris Mazuc
• [asterisk-users] PortSip and Astericks new install   Charlie Grosvenor
• [asterisk-users] AEL2 Confusion   Bart Fisher
• [asterisk-users] PortSip and Astericks new install   Doug Lytle
• [asterisk-users] Re: unable to get channel lock BAD BAD BAD   Al Bochter
• [asterisk-users] AEL2 Confusion   Eric "ManxPower" Wieling
• [asterisk-users] Re: unable to get channel lock BAD BAD BAD   Doug Lytle
• [asterisk-users] hosted asterisk   Eric Bishop
• [asterisk-users] asterisk billing software   Andrew Joakimsen
• [asterisk-users] T.38 - make conclusion   Andrew Joakimsen
• [asterisk-users] Nokia E70   Michiel van Baak
• [asterisk-users] Asterisk 1.2.13 can't load module app_curl.so   Alyed Tzompa
• [asterisk-users] asterisk billing software   Chris Mazuc
• [asterisk-users] Nokia E70   Alberto Pastore
• [asterisk-users] dialplan "*" and "0" key detection, not working   joe a.
• [asterisk-users] queue management   Octavio Ruiz (Ta^3)
• [asterisk-users] Regular audio fade-out fade-in on IAX2 calls Asterisk 1.2.4   Lucas Barbuto
• [asterisk-users] Re: Regular audio fade-out fade-in on IAX2 calls Asterisk 1.2.4   Lucas Barbuto
• [asterisk-users] PortSip and Astericks new install   Charlie Grosvenor
• [asterisk-users] Re: unable to get channel lock BAD BAD BAD   Benny Amorsen
• [asterisk-users] Attempting native bridge of   Vicky
• [asterisk-users] STUN with one public and one private IP?   Steve Sobol
• [asterisk-users] STUN with one public and one private IP?   Steve Sobol
• [asterisk-users] Multi-site Redundancy. Possible?   Heyer, JohnX
• [asterisk-users] Re: unable to get channel lock BAD BAD BAD   Tim Uckun
• [asterisk-users] Re: unable to get channel lock BAD BAD BAD   Tim Uckun
• [asterisk-users] Re: unable to get channel lock BAD BAD BAD   Tim Uckun
• [asterisk-users] dialplan "*" and "0" key detection, not working   Eric "ManxPower" Wieling
• [asterisk-users] Re: unable to get channel lock BAD BAD BAD   BJ Weschke
• [asterisk-users] installing asterisk for Ubuntu Synaptic   blackwater dev
• [asterisk-users] asterisk OSX -astmasters site is gone   blackwater dev
• [asterisk-users] Queue - how to provide a caller ringing tone when some agent become available   BJ Weschke
• [asterisk-users] PortSip and Astericks new install   Doug Lytle
• [asterisk-users] Asterisk on Solars?   Andrew Joakimsen
• [asterisk-users] Call forwarding....   Leeno Jose .P.A
• [asterisk-users] POS Terminals   Paul Hales
• [asterisk-users] Soundfiles adding during phone calls   bails
• [asterisk-users] Re: unable to get channel lock BAD BAD BAD   Tomislav Parčina
• [asterisk-users] queue management   nik600
• [asterisk-users] sip forward behind a nat   nik600
• [Asterisk-Users] Siemens Gigaset SL75   Olivier
• [asterisk-users] Nokia E70   Administrator TOOTAI
• [asterisk-users] splitting a PRI using TDMoE   King Ho
• [asterisk-users] Re: T.38 - make conclusion   Tomislav Parčina
• [asterisk-users] dialplan "*" and "0" key detection, not working   joe a.
• [asterisk-users] Problem with Asterisk 1.4.0-beta3 and Digium TE405P   Alex Lake
• [asterisk-users] spandsp 0.0.3 RxFax fax reception crashes bristuffed asterisk 1.2.13   Marcel van der Boom
• [asterisk-users] Need help on Music on Hold   gc
• [asterisk-users] Asterisk on Solars?   J. Oquendo
• [asterisk-users] Need help on Music on Hold   bas at datarack.nl
• [asterisk-users] Asterisk on Solars?   J. Oquendo
• [asterisk-users] redhat enterprise 3   nik600
• [asterisk-users] Asterisk on Solars?   Andrew Joakimsen
• [asterisk-users] Asterisk on Solars?   J. Oquendo
• [asterisk-users] Asterisk on Solaris (last message)   J. Oquendo
• [asterisk-users] Attempting native bridge of   Victor Toofic
• [asterisk-users] 1 FXO termination device   Jean-Michel Hiver
• [asterisk-users] Understanding the CDR with forwards...   Mike
• [asterisk-users] specify codec by domain?   Mark Price
• [asterisk-users] 1 FXO termination device   Administrator TOOTAI
• [asterisk-users] 1 FXO termination device   aLaN SaNcHeZ
• [asterisk-users] Asterisk as an analog call recording solution (was: Recording outbound analog calls with X100P)   Matthew J. Roth
• [asterisk-users] some questions about atxfer usage   Antonio Almodóvar
• [asterisk-users] some questions about atxfer usage   Antonio Almodóvar
• [asterisk-users] voice quality of Aastra 480i CT and cordless   Scott Keagy
• [asterisk-users] voice quality of Aastra 480i CT and cordless   Cory Andrews
• [asterisk-users] Freepbx changes dont reflect in asterisk   Pedro Silva
• [asterisk-users] voice quality of Aastra 480i CT and cordless   Curt Shaffer
• [asterisk-users] metermaid and 1.2.13?   BerkHolz, Steven
• [asterisk-users] voice quality of Aastra 480i CT and cordless   Michael Graves
• [asterisk-users] Freepbx changes dont reflect in asterisk   Bart Fisher
• [asterisk-users] Freepbx changes dont reflect in asterisk   Alex Robar
• [asterisk-users] wget from within asterisk?   Damon Estep
• [asterisk-users] wget from within asterisk?   Russ Beaupre
• [asterisk-users] wget from within asterisk?   Matteo Brancaleoni
• [asterisk-users] wget from within asterisk?   Damon Estep
• [asterisk-users] wget from within asterisk?   Russ Beaupre
• [asterisk-users] wget from within asterisk?   Damon Estep
• [asterisk-users] wget from within asterisk?   Russ Beaupre
• [asterisk-users] TDM2400p and HW echo canceller   Webster, Andrew
• [asterisk-users] automated response   Stayt, Mark
• [asterisk-users] wget from within asterisk?   Damon Estep
• [asterisk-users] wget from within asterisk?   Damon Estep
• [asterisk-users] Extension Response Slow   Phil Jackson
• [asterisk-users] wget from within asterisk?   Russ Beaupre
• [asterisk-users] wget from within asterisk?   Damon Estep
• [asterisk-users] Asterisk - Do Not Call List   Ed Nuñez
• [asterisk-users] Do Not Call List   Don Fanning
• [asterisk-users] TDM2400p and HW echo canceller   Doug Lytle
• [asterisk-users] wget from within asterisk?   Damon Estep
• [asterisk-users] Extension Response Slow   Doug Lytle
• [asterisk-users] Destar release!   Diego Andres Asenjo G.
• [asterisk-users] Do Not Call List   Michael Collins
• [asterisk-users] Do Not Call List   Dean Collins
• [asterisk-users] Do Not Call List   Don Fanning
• [asterisk-users] Ringing a group of phones but not if they are busy   Carlos Chavez
• [asterisk-users] Ringing a group of phones but not if they are busy   Steven Ringwald
• [asterisk-users] outgoing works, incoming fails on asterisk passthrough to inter-tel   Nathan Bell
• [asterisk-users] Freepbx changes dont reflect in asterisk   Pedro Silva
• [asterisk-users] Do Not Call List   Don Fanning
• [asterisk-users] Ringing a group of phones but not if they are busy   Carlos Chavez
• [asterisk-users] Jitter Buffers in Zapata   Andres
• [asterisk-users] Freepbx changes dont reflect in asterisk   Zeeshan Zakaria
• [asterisk-users] voice quality of Aastra 480i CT and cordless   Zeeshan Zakaria
• [asterisk-users] Need help on Music on Hold   Zeeshan Zakaria
• [asterisk-users] specify codec by domain?   Michael Strelnikov
• [asterisk-users] Freepbx changes dont reflect in asterisk   Alex Robar
• [asterisk-users] Soundfiles adding during phone calls   Ronald Wiplinger
• [asterisk-users] strip + sign from incoming ${EXTEN} var?   voiplist
• [asterisk-users] Freepbx changes dont reflect in asterisk   voiplist
• [asterisk-users] strip + sign from incoming ${EXTEN} var?   Eric "ManxPower" Wieling
• [asterisk-users] spc.exe   Andrew Joakimsen
• [asterisk-users] H323 no audio   Jason Kim
• [asterisk-users] Re: strip + sign from incoming ${EXTEN} var?   Benny Amorsen
• [asterisk-users] Ringing a group of phones but not if they arebusy   Chris Bagnall
• [asterisk-users] Ringing a group of phones but not if they are busy   Alberto Pastore
• [asterisk-users] spc.exe   Alberto Sagredo
• [asterisk-users] PortSip and Astericks new install   Charlie Grosvenor
• [asterisk-users] Re: zaptel, bristuff zaphfc, and florz question   Paul Hewlett
• [asterisk-users] If of external small box supply fxs Isdn and E1 ?   Noc Phibee
• [asterisk-users] AdvancedVoIP Billing ?   Noc Phibee
• [asterisk-users] Freepbx changes dont reflect in asterisk   Pedro Silva
• [asterisk-users] AdvancedVoIP Billing ?   Al Bochter
• [asterisk-users] Re: zaptel, bristuff zaphfc, and florz question   Steve Davies
• [asterisk-users] AdvancedVoIP Billing ?   Noc Phibee
• [asterisk-users] AdvancedVoIP Billing ?   Hermann Wecke
• [asterisk-users] wget from within asterisk?   Russ Beaupre
• [asterisk-users] Using ChanSpy for spying voicemail   Anselm Martin Hoffmeister
• [asterisk-users] Hardware Echo cancelation   Carlos Prieto
• [asterisk-users] spc.exe   Andrew Joakimsen
• [asterisk-users] AdvancedVoIP Billing ?   Noc Phibee
• [asterisk-users] spc.exe   Cory Andrews
• [asterisk-users] Hardware Echo cancelation   Doug Lytle
• [asterisk-users] Asterisk Manager: equivalent of 'show channels'?   Marco Mouta
• [asterisk-users] Asterisk Manager: equivalent of 'show channels'?   Michael Collins
• [asterisk-users] Asterisk Manager: equivalent of 'show channels'?   Nico Busch
• [asterisk-users] TDD/TTY device for the deaf   Michael Welter
• [asterisk-users] AdvancedVoIP Billing ?   Vicky
• [asterisk-users] AdvancedVoIP Billing ?   Darren Wiebe
• [asterisk-users] Asterisk Manager: equivalent of 'show channels'?   Earl Terwilliger
• [asterisk-users] Asterisk Manager: equivalent of 'show channels'?   Richard Lyman
• [asterisk-users] AdvancedVoIP Billing ?   Vicky
• [asterisk-users] Cisco 2801 and asterisk   bas at datarack.nl
• [asterisk-users] AdvancedVoIP Billing ?   Darren Wiebe
• [asterisk-users] AdvancedVoIP Billing ?   Al Bochter
• [asterisk-users] Dialout Conferences?   Matthew Rubenstein
• [asterisk-users] Need help with a function   Christian
• [asterisk-users] spc.exe   Andrew Joakimsen
• [asterisk-users] How to use Sipura SPA3k POTS line to dial Asterisk SIP phones?   Larry Alkoff
• [asterisk-users] Statistics on Number of Minutes   broadbandvoice at comcast.net
• [asterisk-users] Freepbx changes dont reflect in asterisk   Dumpolid Exeplish
• [asterisk-users] How to use Sipura SPA3k POTS line to dial Asterisk SIP phones?   Doug Crompton
• [asterisk-users] Asterisk Manager: equivalent of 'show channels'?   Michael Collins
• [asterisk-users] How to use Sipura SPA3k POTS line to dial Asterisk SIP phones?   Larry Alkoff
• [asterisk-users] Freepbx changes dont reflect in asterisk   Pedro Silva
• [asterisk-users] How to use Sipura SPA3k POTS line to dial Asterisk SIP phones?   Larry Alkoff
• [asterisk-users] Re: Asterisk to listen for sip traffic on 80 and 5060   kjcsb
• [asterisk-users] Re: Asterisk to listen for sip traffic on 80 and 5060   Leo Ann Boon
• [asterisk-users] Dialout Conferences?   Matt
• [asterisk-users] spc.exe   Matt
• [asterisk-users] Dial/Continue/Announce   Matt
• [asterisk-users] odd issue with IP tables   Curt Shaffer
• [asterisk-users] odd issue with IP tables   Alex Robar
• [asterisk-users] odd issue with IP tables   Ron McLeod
• [asterisk-users] odd issue with IP tables   Curt Shaffer
• [asterisk-users] odd issue with IP tables   Ron McLeod
• [asterisk-users] Ringing a group of phones but not if they are busy   C F
• [asterisk-users] odd issue with IP tables   Curt Shaffer
• [asterisk-users] Cant record phone calls   Christian
• [asterisk-users] spc.exe   Andrew Joakimsen
• [asterisk-users] How to use Sipura SPA3k POTS line to dial Asterisk SIP phones?   Larry Alkoff
• [asterisk-users] odd issue with IP tables   Jay R. Ashworth
• [asterisk-users] Re: odd issue with IP tables   Benny Amorsen
• [asterisk-users] Question on CDR Database   Noc Phibee
• [asterisk-users] G723 pass-through and codec negotiation   Ray Jackson
• [asterisk-users] MS-GSM codec issues - Anybody seen anything similar?   Ray Jackson
• [asterisk-users] wget from within asterisk?   Marnus van Niekerk
• [asterisk-users] Question on CDR Database   Vicky
• [asterisk-users] Question on CDR Database   Al Bochter
• [asterisk-users] What card for E1R2?   Lito Lampitoc
• [asterisk-users] What card for E1R2?   Lito Lampitoc
• [asterisk-users] Re: unable to get channel lock BAD BAD BAD   Tzafrir Cohen
• [asterisk-users] wget from within asterisk?   Damon Estep
• [asterisk-users] G723 pass-through and codec negotiation   Andrew Joakimsen
• [asterisk-users] WaitExten only reading 1 digit.   Simon Tennant
• [asterisk-users] Do Not Call List   Kevin Bockman
• [asterisk-users] Do Not Call List   Dean Collins
• [asterisk-users] PHP to .call file   zero massive
• [asterisk-users] WaitExten only reading 1 digit.   Ron Wellsted
• [asterisk-users] WaitExten only reading 1 digit.   Doug Lytle
• [asterisk-users] WaitExten only reading 1 digit.   Doug Lytle
• [asterisk-users] Do Not Call List   Don Fanning
• [asterisk-users] Do Not Call List   Don Fanning
• [asterisk-users] Freepbx changes dont reflect in asterisk   Mike Dent
• [asterisk-users] WaitExten only reading 1 digit.   Simon Tennant
• [asterisk-users] Desktop integration   kjcsb
• [asterisk-users] Desktop integration   Joe Dennick
• [asterisk-users] Desktop integration   Michiel van Baak
• [asterisk-users] G723 pass-through and codec negotiation   Ray Jackson
• [asterisk-users] What card for E1R2?   Josué Conti
• [asterisk-users] Re: Dialout Conferences?   Tony Mountifield
• [asterisk-users] Desktop integration   Joe Dennick
• [asterisk-users] wget from within asterisk?   Andrew Joakimsen
• [asterisk-users] Asterisk - Do Not Call List   Michelle Dupuis
• [asterisk-users] What card for E1R2?   Lito Lampitoc
• [asterisk-users] A question on ISDN cards... (in the UK)   Nigel Godfrey
• [asterisk-users] switching trunks based on quality   Curt Shaffer
• [asterisk-users] switching trunks based on quality   Luki
• [asterisk-users] switching trunks based on quality   Curt Shaffer
• [asterisk-users] make: execvp: build_tools/make_svn_branch_name: Permission denied   Tzafrir Cohen
• [asterisk-users] wget from within asterisk?   Tzafrir Cohen
• [asterisk-users] Vonage uses Cisco   Curt Shaffer
• [asterisk-users] FXO PCI Master abort   Tzafrir Cohen
• [asterisk-users] Freepbx changes dont reflect in asterisk   Tzafrir Cohen
• [asterisk-users] Trunk outcall line ?   Noc Phibee
• [asterisk-users] reduce dialtone volume on zap channel.   Don Pobanz
• [asterisk-users] Linksys doesn't resync properly and doesn't get provisioning from TFTP and HTTP   Andrew Joakimsen
• [asterisk-users] What card for E1R2?   Josué Conti
• [asterisk-users] Desktop integration   Senad Jordanovic
• [asterisk-users] Asterisk Manager: equivalent of 'show channels'?   Tim Panton
• [asterisk-users] reduce dialtone volume on zap channel.   Doug Lytle
• [asterisk-users] G723 pass-through and codec negotiation   Tim Panton
• [Asterisk-Users] Siemens Gigaset SL75   Joao Pereira
• [asterisk-users] Trunk outcall line ?   Marco Mouta
• [asterisk-users] G723 pass-through and codec negotiation   Eric "ManxPower" Wieling
• [asterisk-users] Recording g729   Marnus van Niekerk
• [Asterisk-Users] Siemens Gigaset SL75   Armin Schindler
• [asterisk-users] g729 registered   Ralph Liebessohn
• [Asterisk-Users] Unicall MFC problems in 0.0.3+asterisk 1.2   Facundo Ameal
• [asterisk-users] Call-limit   René Enskat
• [asterisk-users] g729 registered   Alex Robar
• [asterisk-users] reduce dialtone volume on zap channel.   Don Pobanz
• [asterisk-users] Freepbx changes dont reflect in asterisk   Tom Rymes
• [asterisk-users] reduce dialtone volume on zap channel.   Doug Lytle
• [asterisk-users] wget from within asterisk?   Damon Estep
• [asterisk-users] .call file unable to hear or speak   zero massive
• [asterisk-users] Compilation problem   Greg D
• [asterisk-users] Help me to configure my NOKIA E70 Mobile with my Asterisk server   Crazy Boy
• [asterisk-users] reduce dialtone volume on zap channel.   Don Pobanz
• [asterisk-users] Help me to configure my NOKIA E70 Mobile with my Asterisk server   Guillermo Salas M.
• [asterisk-users] Help me to configure my NOKIA E70 Mobile with my Asterisk server   Peter Bowyer
• [asterisk-users] g729 registered   Ralph Liebessohn
• [asterisk-users] reduce dialtone volume on zap channel.   Doug Lytle
• [asterisk-users] spandsp 0.0.3 RxFax fax reception crashes bristuffed asterisk 1.2.13   DRi at b-w-computer.de
• [asterisk-users] SNMP-based monitor for Asterisk (Was: Asterisk Manager: equivalent of 'show channels'?)   Michael Collins
• [asterisk-users] International dialing with GPX-2000 and "early dial"   Andrew Joakimsen
• [asterisk-users] reduce dialtone volume on zap channel.   Don Pobanz
• [asterisk-users] T.38 - By reinvitation only?   Alex Lake
• [asterisk-users] reduce dialtone volume on zap channel.   Eric "ManxPower" Wieling
• [asterisk-users] SIP Multi-Domain   Jean-Marc Salsa
• [asterisk-users] Vonage uses Cisco   Matthew Rubenstein
• [asterisk-users] T.38 - By reinvitation only?   Alex Lake
• [asterisk-users] Call limits and VoIP providers   Mike
• [asterisk-users] How to accept All incomings calls from One Special Host (like a proxy)   Jean-Marc Salsa
• [asterisk-users] ODBC Voicemail Storage   Edwin Horton
• [asterisk-users] Call limits and VoIP providers   Vicky
• [asterisk-users] Set a feature within AgentLogin   Gregory Duchatelet
• email etiquette (was: Re: [Asterisk-Users] Unicall MFC problems in 0.0.3+asterisk 1.2)   Matthew Rubenstein
• [asterisk-users] Call limits and VoIP providers   Mike
• [asterisk-users] Re: email etiquette   Jay Moore
• email etiquette (was: Re: [Asterisk-Users] Unicall MFC problems in 0.0.3+asterisk 1.2)   Jay R. Ashworth
• [asterisk-users] Call limits and VoIP providers   Jeremy McNamara
• [asterisk-users] Call limits and VoIP providers   mail-lists
• [asterisk-users] Call limits and VoIP providers   Jeremy McNamara
• [asterisk-users] Call limits and VoIP providers   tracinet
• [asterisk-users] reduce dialtone volume on zap channel.   Don Pobanz
• [asterisk-users] iSoftSwitch:: Asterisk & GnuGK Integretion   Mohammad Emran
• [asterisk-users] Auto recording calls?   Jay Moore
• [asterisk-users] Auto recording calls?   Tim C. Lewis
• [asterisk-users] Auto recording calls?   Doug
• [asterisk-users] Auto recording calls?   Doug
• [asterisk-users] Auto recording calls?   zero massive
• [asterisk-users] Auto recording calls?   Jay Moore
• [asterisk-users] Auto recording calls?   Jeronimo Romero
• [asterisk-users] Auto recording calls?   Jay Moore
• [asterisk-users] Snom 360 Multiple calls on hold help   Ron McCarthy
• [asterisk-users] Call limits and VoIP providers   Mike
• [asterisk-users] Auto recording calls?   Doug
• [asterisk-users] Snom 360 Multiple calls on hold help   Patrick
• [asterisk-users] Snom 360 Multiple calls on hold help   Dr. Michael J. Chudobiak
• AW: [asterisk-users] Snom 360 Multiple calls on hold help   Christian Stredicke
• [asterisk-users] Auto recording calls?   Jay Moore
• [asterisk-users] reduce dialtone volume on zap channel.   Eric "ManxPower" Wieling
• [asterisk-users] Reset Extension Preferens   Santiago Hoyos R.
• [asterisk-users] Auto recording calls?   Doug
• [asterisk-users] Auto recording calls?   Jay Moore
• [asterisk-users] install asterisk ad zaptel from source on red hat enterprise 3   nik600
• [asterisk-users] alert_info + Linksys 9xx + custom ringtone   Joshua Citron
• [asterisk-users] Snom 360 Multiple calls on hold help   Ron McCarthy
• [asterisk-users] Spandsp rxfax txtax fails no errors   daveasterisk
• [asterisk-users] reduce dialtone volume on zap channel.   Don Pobanz
• R: [asterisk-users] Snom 360 Multiple calls on hold help   Giordano Grandis
• R: [asterisk-users] Snom 360 Multiple calls on hold help   Ron McCarthy
• [asterisk-users] How to secure access to PSTN line through Linksys gateway?   Vincent Delporte
• [asterisk-users] International dialing with GPX-2000 and "early dial"   Anthony Kepler
• [asterisk-users] Problem with FXS ports of TDM400P   Gustavo Felisberto
• [asterisk-users] is there a queue size call limit on the ACD?   MF
• [asterisk-users] CAPI (Eicon Diva V-4BRI), Hylafax & IAXModem   MBIT Technologies
• [asterisk-users] TDM400 native bridge echo   Shaun Kruger
• [asterisk-users] CAPI (Eicon Diva V-4BRI), Hylafax & IAXModem   Avi Miller
• [asterisk-users] CAPI (Eicon Diva V-4BRI), Hylafax & IAXModem   MBIT Technologies
• [asterisk-users] CAPI (Eicon Diva V-4BRI), Hylafax & IAXModem   Avi Miller
• [asterisk-users] CAPI (Eicon Diva V-4BRI), Hylafax & IAXModem   MBIT Technologies
• [asterisk-users] Re: Snom 360 Multiple calls on hold help   Ron McCarthy
• [asterisk-users] TDM400 native bridge echo   Mervyn Yeo
• [asterisk-users] CAPI (Eicon Diva V-4BRI), Hylafax & IAXModem   Avi Miller
• [asterisk-users] TDM400 native bridge echo   Shaun Kruger
• [asterisk-users] QB intergreation   Dovid B
• [asterisk-users] reduce dialtone volume on zap channel.   Eric "ManxPower" Wieling
• [asterisk-users] Reset Extension Preferens   Eric "ManxPower" Wieling
• [asterisk-users] SIP Multi-Domain   Eric "ManxPower" Wieling
• [asterisk-users] Trunk outcall line ?   Eric "ManxPower" Wieling
• [asterisk-users] switching trunks based on quality   Eric "ManxPower" Wieling
• [asterisk-users] Reliable European SIP/IAX Providers?   Nate Kapi
• [asterisk-users] TDM400 native bridge echo   Tzafrir Cohen
• [asterisk-users] How to secure access to PSTN line through Linksys gateway?   Tzafrir Cohen
• [asterisk-users] SIP Multi-Domain   Jean-Marc Salsa
• [asterisk-users] switching trunks based on quality   Olivier
• [asterisk-users] Vonage uses Cisco   Marnus van Niekerk
• [asterisk-users] switching trunks based on quality   Tzafrir Cohen
• [asterisk-users] Can i have two asterisk versions running on samePC??   Thomas Deillon
• [asterisk-users] Reliable European SIP/IAX Providers?   Zoa
• [asterisk-users] RE: Snom 360 Multiple calls on hold help   Usman Tahir
• [asterisk-users] Reliable European SIP/IAX Providers?   Yair Hakak
• [asterisk-users] Hairping calls and Originating CLI   Adrian Marsh
• [asterisk-users] Help me to configure my NOKIA E70 Mobile with my Asterisk server   Tim Panton
• [asterisk-users] Forums   Paul Hales
• [asterisk-users] Callback agents without chan_agent issues (queue recording)   Andrey Solovjov
• [asterisk-users] Handle Options Method   Thomas Deillon
• [asterisk-users] Handle Options Method   Thomas Deillon
• [asterisk-users] Signalogic SigC5561 PCI card   cleviton.araujo at caixa.gov.br
• [asterisk-users] Handle Options Method   Thomas Deillon
• [asterisk-users] Freepbx changes dont reflect in asterisk   Alejandro Acosta
• [asterisk-users] Re: How to secure access to PSTN line through Linksys gateway?   Vincent Delporte
• [asterisk-users] RE: Snom 360 Multiple calls on hold help   Ron McCarthy
• [asterisk-users] Re: How to secure access to PSTN line through Linksys gateway?   Noah Miller
• [asterisk-users] spc.exe   Matt
• [asterisk-users] Re: How to secure access to PSTN line through Linksys gateway?   Tzafrir Cohen
• [asterisk-users] RE: Snom 360 Multiple calls on hold help   Noah Miller
• [Asterisk-Users] Which reliable source for ToIP security alerts ?   Olivier
• [asterisk-users] spc.exe   Bruce Reeves
• [asterisk-users] Nortel CS1000 Asterisk with SIP   Koen Van Impe
• [asterisk-users] SNMP-based monitor for Asterisk (Was: Asterisk Manager: equivalent of 'show channels'?)   Tim Panton
• [asterisk-users] switching trunks based on quality   Olivier
• [asterisk-users] Auto recording calls?   Jay Moore
• [asterisk-users] RE: Snom 360 Multiple calls on hold help   Usman Tahir
• [asterisk-users] FW: CISCO 7960G & Asterisk   Jeronimo Romero
• [asterisk-users] Cisco media gateways in general   Bas van der Veen
• [asterisk-users] Diva Server, chan_capi and tone detection   Gregory Duchatelet
• [asterisk-users] spc.exe   Nabeel Jafferali
• [asterisk-users] Auto recording calls?   Brett Crapser
• [asterisk-users] FW: CISCO 7960G & Asterisk   Anthony LaMantia
• [asterisk-users] Auto recording calls?   Jay Moore
• [asterisk-users] spc.exe   Andrew Joakimsen
• [Asterisk-Users] Siemens Gigaset SL75   jbauer at teec.com
• [asterisk-users] Cisco media gateways in general   Scott Keagy
• [asterisk-users] alert_info + Linksys 9xx + custom ringtone   Andrew Joakimsen
• [asterisk-users] How to accept All incomings calls from One Special Host (like a proxy)   Andrew Joakimsen
• [asterisk-users] Recording g729   Andrew Joakimsen
• [asterisk-users] FW: CISCO 7960G & Asterisk   Ejay Hire
• [asterisk-users] QueueMetrics 1.3.1 released today   Lenz
• [asterisk-users] FW: CISCO 7960G & Asterisk   Lorell Hathcock
• [asterisk-users] Setting the default AMAflag   Mike
• [asterisk-users] RESOLVED - Snom 360 Multiple calls on hold help   Ron McCarthy
• [asterisk-users] Parking on an extension   Kelly Opal
• [asterisk-users] Cisco media gateways in general   Bas van der Veen
• [asterisk-users] Cisco media gateways in general   Bill Gibbs
• [asterisk-users] spc.exe   Andrew Joakimsen
• [asterisk-users] Re: Choppy sound in voicemail usingAsterisk1.2.11 on CENTOS4 guest on vmware server   Mario François Jauvin
• [asterisk-users] FW: CISCO 7960G & Asterisk   Bill Hackensack
• [asterisk-users] Prefix/Suffix/StripLSD/StripMSD gone?   Paul Schenkeveld
• [asterisk-users] Parking on an extension   Doug Lytle
• [asterisk-users] Parking on an extension   Andrew Joakimsen
• [asterisk-users] IAX access to FWD broken?   Brian Capouch
• [asterisk-users] Recording g729   Matthew Rubenstein
• [asterisk-users] Parking on an extension   Ron McLeod
• [asterisk-users] Hairping calls and Originating CLI   Tim Panton
• [asterisk-users] Prefix/Suffix/StripLSD/StripMSD gone?   Don Pobanz
• [asterisk-users] Parking on an extension   Kelly Opal
• [asterisk-users] voice breaking problems   ram
• [asterisk-users] spc.exe   Matt
• [asterisk-users] Answer Machine Detection   Matheus Rossato
• [asterisk-users] Call to disconnected number on PRI just rings   courchea at net-forces.com
• [asterisk-users] Prefix/Suffix/StripLSD/StripMSD gone?   Paul Schenkeveld
• [asterisk-users] QB intergreation   C F
• [asterisk-users] Parking on an extension   Doug Lytle
• [asterisk-users] Answer Machine Detection   Matt Florell
• [asterisk-users] QoS on Linksys SWR208P?   Curt Shaffer
• [asterisk-users] QB intergreation   Darrick Hartman
• [asterisk-users] spc.exe   Brian Capouch
• [asterisk-users] Re: Call limits and VoIP providers   Benny Amorsen
• [asterisk-users] Prefix/Suffix/StripLSD/StripMSD gone?   Tzafrir Cohen
• [asterisk-users] receptionist - large number of concurrent calls - example needed   Paul Hales
• [asterisk-users] Prefix/Suffix/StripLSD/StripMSD gone?   Don Pobanz
• [asterisk-users] VM mail notification and locale   Dominique Dartois
• [asterisk-users] cmd Record   Michael Welter
• [asterisk-users] Cisco media gateways in general   Eric "ManxPower" Wieling
• [asterisk-users] VM mail notification and locale   Tzafrir Cohen
• [asterisk-users] cmd Record   Tzafrir Cohen
• [asterisk-users] RE: Snom 360 Multiple calls on hold help   Ron McCarthy
• [asterisk-users] RE: Snom 360 Multiple calls on hold help   Ron McCarthy
• [asterisk-users] OT: Reflash Mitel 5220 from MiNet to SIP   Colin Anderson
• [asterisk-users] Re: Spandsp rxfax txtax fails no errors   daveasterisk
• [asterisk-users] Is this possible?   shadowym
• [asterisk-users] IP601 Expansion Module HELP!!!   Ron McCarthy
• [asterisk-users] IP601 Expansion Module HELP!!!   Klaverstyn, David C
• [asterisk-users] Hints no longer working in 1.4beta3 with Polycoms   Anthony Rodgers
• [asterisk-users] Hints no longer working in 1.4beta3 with Polycoms   John Lange
• [asterisk-users] Why Aastra uses 48V whereas other IP Phones use much less, i.e. 5-12V   Zeeshan Zakaria
• [asterisk-users] IP601 Expansion Module HELP!!!   Andrew Joakimsen
• [asterisk-users] Why Aastra uses 48V whereas other IP Phones use much less, i.e. 5-12V   Julien Goodwin
• [asterisk-users] IP601 Expansion Module HELP!!!   Klaverstyn, David C
• [asterisk-users] Can anyone enlighten me as to what this means?   Matt
• [asterisk-users] Is this a PRI problem, *, or the phone???   Ron McCarthy
• [asterisk-users] codec information   rilawich ango
• [asterisk-users] Attn:Peter, Gsalas, Tim-Help me to configure my NOKIA E70 Mobile with my Asterisk server   Crazy Boy
• [asterisk-users] Can anyone enlighten me as to what this means?   Paul Hales
• [Asterisk-Users] Siemens Gigaset SL75   Olivier
• [asterisk-users] Spandsp rxfax txtax fails no errors   Tzafrir Cohen
• [asterisk-users] Welcome to Join Asterisk MSN Groups!   Mayson.Wang
• [asterisk-users] RE: Snom 360 Multiple calls on hold help   Tzafrir Cohen
• [asterisk-users] Diva Server, chan_capi and tone detection   Gregory Duchatelet
• [asterisk-users] Agent Channel SIP transfer   Xue Liangliang
• [asterisk-users] snom subscriptions issue on WRT   tommaso.carrara
• [asterisk-users] asterisk-cluster with one database   René Enskat
• [asterisk-users] IAX Codecs & IAX Realtime   Marian Rychtecky
• [asterisk-users] Re: Spandsp rxfax txtax fails no errors   Håkan Källberg
• [asterisk-users] Is it easy to route SIP/SDP and SIP/RTP through different routes ?   Olivier
• [asterisk-users] reduce dialtone volume on zap channel.   Tom Rymes
• [asterisk-users] SOLVED: Digium TE405 card and Matra PBX   Jan Marek
• [asterisk-users] Zaptel error   ram
• [asterisk-users] about voicemail setting   rilawich ango
• [asterisk-users] Request for working config for DISA   Crazy Boy
• [asterisk-users] Ast 1.4 and B410p   yusuf
• [asterisk-users] qualify=yes   Enrico Pasqualotto
• [asterisk-users] qualify=yes   Enrico Pasqualotto
• [asterisk-users] Can anyone enlighten me as to what this means?   Matt
• [asterisk-users] about voicemail setting   Conrad Wood
• [asterisk-users] RE: Snom 360 Multiple calls on hold help   Alban
• [asterisk-users] VM mail notification and locale   Dominique Dartois
• [asterisk-users] Re: Call limits and VoIP providers   Mike
• [asterisk-users] Why Aastra uses 48V whereas other IP Phones use much less, i.e. 5-12V   Andrew Latham
• [asterisk-users] snom subscriptions issue on WRT   Andrew Latham
• [asterisk-users] RE: Snom 360 Multiple calls on hold help   Steve Davies
• [asterisk-users] spc.exe   Andrew Latham
• [asterisk-users] Hints no longer working in 1.4beta3 with Polycoms   Andrew Latham
• [asterisk-users] help in Call parking......   raviprakash sunkara
• [asterisk-users] Rewriting caller ID from database?   Vincent Delporte
• [asterisk-users] RE: Snom 360 Multiple calls on hold help   Alban
• [asterisk-users] Recordings.   jjmvermeeren at tiscali.nl
• [asterisk-users] Welcome to Join Asterisk MSN Groups!   Al Bochter
• [asterisk-users] qualify=yes   Eric "ManxPower" Wieling
• [asterisk-users] reduce dialtone volume on zap channel.   Don Pobanz
• [asterisk-users] DTMF detection during Call   chrigu at lorraine.ch
• [asterisk-users] Zaptel error   Patrick
• [asterisk-users] IP601 Expansion Module HELP!!!   Dave Fullerton
• [asterisk-users] Recordings.   Marcus Franke
• [asterisk-users] Asterisk incoming call behaviour   Vicky
• [asterisk-users] Hairping calls and Originating CLI   Adrian Marsh
• [asterisk-users] qualify=yes   Vicky
• [asterisk-users] Recordings.   Michael Welter
• [asterisk-users] Recordings.   Vicky
• [asterisk-users] iax2 - wildiax phone & myself puzzled   Alberto Pastore
• [asterisk-users] Send event from dialplan   Gregory Duchatelet
• [asterisk-users] Answer Machine Detection   Wes Baehr
• [asterisk-users] Recordings for VR analysis   Michael Welter
• [asterisk-users] Recordings.   Marcus Franke
• [asterisk-users] Send event from dialplan   Gregory Duchatelet
• [asterisk-users] Is this possible?   C F
• [asterisk-users] Recordings.   Marcus Franke
• [asterisk-users] How ecord all calls?   Eugeniy Khvastunov
• [asterisk-users] qualify=yes   Eric "ManxPower" Wieling
• [asterisk-users] How ecord all calls?   Eric "ManxPower" Wieling
• [asterisk-users] DTMF detection during Call   Eric "ManxPower" Wieling
• [asterisk-users] Recordings for VR analysis   Brett Crapser
• [asterisk-users] qualify=yes   Pavel Jezek
• [asterisk-users] Cisco media gateways in general   Pavel Jezek
• [asterisk-users] How ecord all calls?   Eric "ManxPower" Wieling
• [asterisk-users] RE: Snom 360 Multiple calls on hold help   Ron McCarthy
• [asterisk-users] G729 issues on 1.4 beta 3   Jason Adams
• [asterisk-users] Cisco media gateways in general   Scott Keagy
• [Asterisk-Users] Siemens Gigaset SL75   jbauer at teec.com
• [asterisk-users] Hints no longer working in 1.4beta3 with Polycoms   Anthony Rodgers
• [asterisk-users] Re: Rewriting caller ID from database?   Steven
• [asterisk-users] Re: Rewriting caller ID from database?   Marco Mouta
• [asterisk-users] channel_find_locked: Avoided deadlock ... messages - What to do?   Jim Rice
• [asterisk-users] Why Aastra uses 48V whereas other IP Phones use much less, i.e. 5-12V   Huib van Wees
• [asterisk-users] Send event from dialplan   Richard Lyman
• [asterisk-users] Hints no longer working in 1.4beta3 with Polycoms   Anthony Rodgers
• [asterisk-users] Cisco media gateways in general   SWhite at littleme.com
• [asterisk-users] Asterisk On FreeBSD   J. Oquendo
• [asterisk-users] Request for working config for DISA   zero massive
• [asterisk-users] Asterisk On FreeBSD   Dumpolid Exeplish
• [asterisk-users] Asterisk On FreeBSD   Rick Smith
• [asterisk-users] Asterisk On FreeBSD   Emil Thelin
• [asterisk-users] Can anyone enlighten me as to what this means?   Tristan
• [asterisk-users] TE110P and TDM400P   Henk Dick
• [asterisk-users] Why Aastra uses 48V whereas other IP Phones use much less, i.e. 5-12V   Steve Kennedy
• [asterisk-users] TE110P and TDM400P   Lincoln Zuljewic Silva
• [asterisk-users] Why Aastra uses 48V whereas other IP Phones use much less, i.e. 5-12V   Michael Graves
• [asterisk-users] More than one asterisk process   Ard
• [asterisk-users] TE110P and TDM400P   Henk Dick
• [asterisk-users] Diva Server, chan_capi and tone detection   Armin Schindler
• [asterisk-users] Call park on Linksys 922 and similar phones?   Brad Templeton
• [asterisk-users] TE110P and TDM400P   Lincoln Zuljewic Silva
• [asterisk-users] Why Aastra uses 48V whereas other IP Phones use much less, i.e. 5-12V   Shaun Kruger
• [asterisk-users] Recordings.   Vicky
• [asterisk-users] aastra 480i configuration help   marvin horst
• [asterisk-users] Call to disconnected number on PRI just rings   Kevin Bockman
• [asterisk-users] cmd Record   Kevin Bockman
• [asterisk-users] Re: Why Aastra uses 48V whereas other IP Phones use much less, i.e. 5-12V   Steve Murphy
• [asterisk-users] Asterisk On FreeBSD   Jeronimo Romero
• [asterisk-users] snom subscriptions issue on WRT   Jorge Mendoza
• [asterisk-users] Dialing from "Placed Calls" on Polycom IP501 doesn't always work   Anthony Rodgers
• [asterisk-users] Terrible, horrible firewall issues in * to * setup   Lachek Butalek
• [asterisk-users] Rewriting caller ID from database?   Doug Lytle
• [asterisk-users] Rewriting caller ID from database?   Time Bandit
• [asterisk-users] Zaptel - make b410p fails on Ubuntu 6.10   Timothy Parez
• [asterisk-users] How to park calls on a specific extension   Steve Sobol
• [asterisk-users] Powering SNOM 200 phones?   Brad Templeton
• [asterisk-users] Regular audio fade-out fade-in on IAX2 calls Asterisk 1.2.4 Hi all, One of my users has a problem with many of his calls via my Asterisk™ server. He describes the problem as having the sound slowly fade out and then fade back at a regular frequency. Has anyone experienced this before? Can you suggest a potential cause? The client is an IAX2 soft phone running on Windows XP. Asterisk version is 1.2.4. Codec is G.711-ulaw. The user is connecting from long distance away (from client phone in New Zealand to Asterisk™ server in Australia). We've confirmed that roundtrip times are fairly consistent and that the user's PC is in good working order. Would appreciate any advice. Thanks. -- Lucas   Paul Hales
• [asterisk-users] How to park calls on a specific extension   Ira
• [asterisk-users] qualify=yes   Julian J. M.
• [asterisk-users] Sipura phone does not ring   Larry Alkoff
• [asterisk-users] Cisco media gateways in general   Leo Ann Boon
• [asterisk-users] queuemetrics   pdhales at optusnet.com.au
• [asterisk-users] Recordings.   Leo Ann Boon
• [asterisk-users] Why Aastra uses 48V whereas other IP Phones use much less, i.e. 5-12V   Zeeshan Zakaria
• [asterisk-users] Can anyone enlighten me as to what this means?   Rob McKrill
• [asterisk-users] Why Aastra uses 48V whereas other IP Phones use much less, i.e. 5-12V   Michael Graves
• [asterisk-users] G722?   Michael Graves
• [asterisk-users] Hold calling channel and ask called channel before connect???   Nigel J. Terry
• [asterisk-users] Rewriting caller ID from database?   Jay Milk
• [asterisk-users] gotoiftime and blocking calls   Tom Vile
• [asterisk-users] Why Aastra uses 48V whereas other IP Phones usemuch less, i.e. 5-12V   Michelle Dupuis
• [asterisk-users] Zaptel error   ram
• [asterisk-users] in Asterisk Manger its Unauthentication User and Host ..........   raviprakash sunkara
• [asterisk-users] Zaptel - make b410p fails on Ubuntu 6.10   Tzafrir Cohen
• [asterisk-users] TE110P and TDM400P   Tzafrir Cohen
• [asterisk-users] More than one asterisk process   Tzafrir Cohen
• [asterisk-users] How ecord all calls?   Tzafrir Cohen
• [asterisk-users] gotoiftime and blocking calls   Alberto Pastore
• [asterisk-users] Exact definition of ASR   asterisk at frameweb.it
• [asterisk-users] OriginateEvent reason codes.   Jan du Toit
• [asterisk-users] Re: OriginateEvent reason codes.   Jan du Toit
• [asterisk-users] Calls "from asterisk"   Eric Bishop
• [asterisk-users] asterisk 1.4 chan_h323, help please...   Jason Kim
• [asterisk-users] Re: Is asterisk able to integrate with MS SQL   Sharon Lim
• [asterisk-users] How to kill a meet me room at midnight   Eric Bishop
• [asterisk-users] snom subscriptions issue on WRT (2)   tommaso.carrara
• [asterisk-users] Why Aastra uses 48V whereas other IP Phones usemuch less, i.e. 5-12V   Brad Templeton
• [asterisk-users] How to kill a meet me room at midnight   Michiel van Baak
• [asterisk-users] qualify=yes   Pavel Jezek
• [asterisk-users] Diva Server, chan_capi and tone detection   Gregory Duchatelet
• [asterisk-users] Diva Server, chan_capi and tone detection   Armin Schindler
• [asterisk-users] Request for working config for DISA   Crazy Boy
• [asterisk-users] How to change IAX default port 4569 to some other port   Zeeshan Zakaria
• [asterisk-users] Why Aastra uses 48V whereas other IP Phones usemuch less, i.e. 5-12V   Steve Kennedy
• [asterisk-users] Re: How to change IAX default port 4569 to some other port :Debug Message Attached   Zeeshan Zakaria
• [asterisk-users] TE110P and TDM400P   Marco Mouta
• [asterisk-users] AGI info   Artifex Maximus
• [asterisk-users] Request for working config for DISA   Tzafrir Cohen
• [asterisk-users] TE110P and TDM400P   Tzafrir Cohen
• [asterisk-users] (OT) HylaFAX, IAXModem, Asterisk   Steve Totaro
• [asterisk-users] queuemetrics   Steve Totaro
• [asterisk-users] More than one asterisk process   Ard
• [asterisk-users] Recordings.   Steve Totaro
• [asterisk-users] Can anyone enlighten me as to what this means?   Matt
• [asterisk-users] RE: Snom 360 Multiple calls on hold help   Sven Fischer
• [asterisk-users] Asterisk with SER   Arun Kumar
• [asterisk-users] (OT) HylaFAX, IAXModem, Asterisk   Doug Lytle
• [asterisk-users] Survey: In what ways do you use Asterisk at your house?   Neil Cherry
• [asterisk-users] Cisco 7970   David Parcerisa
• [asterisk-users] (OT) HylaFAX, IAXModem, Asterisk   Steve Totaro
• [asterisk-users] Error uninstalling freepbx-panel   Diego Quintana Cruz
• [asterisk-users] Asterisk incoming call behaviour   Time Bandit
• [asterisk-users] TE110P and TDM400P   Tzafrir Cohen
• [asterisk-users] More than one asterisk process   Tzafrir Cohen
• [asterisk-users] TE110P and TDM400P   Lincoln Zuljewic Silva
• [asterisk-users] FOP is not displaying all my SIP extensions neither all E1 channels , why?   Marco Mouta
• [asterisk-users] Terrible, horrible firewall issues in * to * setup   Drew Gibson
• [asterisk-users] Re: G722?   Benny Amorsen
• [asterisk-users] Re: G722?   Julio Arruda
• [asterisk-users] Cisco 7970   David Parcerisa
• [asterisk-users] Digium through Octasic   Heidi Mendoza
• [asterisk-users] festival problem using IAX (chan_iax2.c:2995 iax2_read)   Itamar Lavender
• [asterisk-users] Cisco 7970   david parcerisa
• [asterisk-users] Zaptel error   Anthony Rodgers
• [asterisk-users] Calls "from asterisk"   Anthony Rodgers
• [asterisk-users] Asterisk with SER   Marnus van Niekerk
• [asterisk-users] Re: Rewriting caller ID from database?   David Cook (Canada)
• [asterisk-users] Cisco 7970 SIP upgrade issues   Admin at TheAdmiralNelson.Com
• [asterisk-users] When does voicemail authentication take place?   jezzzz .
• [asterisk-users] FREE DOWNLOAD - PRI / T1 Circuit monitoring   Paul
• [asterisk-users] More than one asterisk process   Ard
• [asterisk-users] Hairping calls and Originating CLI   Tim Panton
• [asterisk-users] Asterisk 1.4 Error   Richard
• [asterisk-users] Call Transfers in SER + Asterisk architecture   Ricardo Carvalho
• [asterisk-users] Store voicemal data in mysql DB   Norbert Zawodsky
• [asterisk-users] Re: asterisk-users Digest, Vol 28, Issue 122   Vincent Delporte
• [asterisk-users] Hairping calls and Originating CLI   Steve Kennedy
• [asterisk-users] Passing arguments to AGI script   Esteban Guana-Jarrin
• [asterisk-users] Asterisk voicemail and hotel software integration   Erick Perez
• [asterisk-users] FW: CISCO 7960G & Asterisk   Scott Keagy
• [asterisk-users] asterisk-users, Matt has invited you to open a Google mail account   Matt
• [asterisk-users] asterisk 1.4 variable list   Roi Stork
• [asterisk-users] MWI from ITSP   Tom Vile
• [asterisk-users] Direct UA to UA RTP connection   Mario François Jauvin
• [asterisk-users] aastra 480i configuration help   Zeeshan Zakaria
• [asterisk-users] FREE DOWNLOAD - PRI / T1 Circuit monitoring   Steve Totaro
• [asterisk-users] Asterisk and TDM400P ?   Noc Phibee
• [asterisk-users] Re: Is asterisk able to integrate with MS SQL   Steve Totaro
• [asterisk-users] asterisk and MISDN on a core2 Duo x64 system   Markus Amann
• [asterisk-users] (no subject)   Imran M Yousuf
• [asterisk-users] (no subject)   Paul Hales
• [asterisk-users] Dial() cmd seams unable to detect caller hangup   Matt
• [asterisk-users] Asterisk and TDM400P ?   Giorgio Incantalupo
• [asterisk-users] Asterisk and TDM400P ?   Noc Phibee
• [asterisk-users] Installing the b410p card, unable to install mISDN   Timothy Parez
• [asterisk-users] Server Configuration for E1's   Imran M Yousuf
• [asterisk-users] Asterisk and TDM400P ?   Giorgio Incantalupo
• [asterisk-users] Asterisk and TDM400P ?   Leo Ann Boon
• [asterisk-users] How to change IAX default port 4569 to some other port   Tim Panton
• [Asterisk-Users] Siemens Gigaset SL75   Olivier
• [asterisk-users] Asterisk and TDM400P ?   Noc Phibee
• [asterisk-users] Asterisk and TDM400P ?   Leo Ann Boon
• [asterisk-users] Re: How to change IAX default port 4569 to some other port :Debug Message Attached   Tim Panton
• [asterisk-users] Terrible, horrible firewall issues in * to * setup   Tim Panton
• [asterisk-users] Fwd: Cisco 7970   David Parcerisa
• [asterisk-users] Asterisk and TDM400P ?   Noc Phibee
• [asterisk-users] Asterisk and TDM400P ?   Pranav Peshwe
• [asterisk-users] Call Transfers in SER + Asterisk architecture   Marco Mouta
• [asterisk-users] Asterisk and TDM400P ?   Noc Phibee
• [asterisk-users] upgraded polycom to 2.0.1.0291 and...   Shaun
• [asterisk-users] DB9 e1 to RJ45 pinout   Giordano Grandis
• [asterisk-users] Request for help with DISA (Not taking my input number correctly?)   Crazy Boy
• [asterisk-users] Asterisk and TDM400P ?   Giorgio Incantalupo
• [asterisk-users] Caller Id not propagated to the analog line   Anton Frolov
• [asterisk-users] Asterisk and TDM400P ?   Pranav Peshwe
• [asterisk-users] More than one asterisk process   Tzafrir Cohen
• [asterisk-users] Asterisk and TDM400P ?   Tzafrir Cohen
• [asterisk-users] Installing the b410p card, unable to install mISDN   Tzafrir Cohen
• [asterisk-users] Asterisk and TDM400P ?   Anton Frolov
• [asterisk-users] Re: Installing Ztdummy on Fedora Core 5   Axel Thimm
• [asterisk-users] Asterisk and TDM400P ?   Noc Phibee
• [asterisk-users] Encrypted password for voicemail   jezzzz .
• [asterisk-users] Asterisk and TDM400P ?   Ing. Germán González B.
• [asterisk-users] Snom 360 / firmware 6.5.1 / registration problems with Asterisk   Koopmann, Jan-Peter
• [asterisk-users] Cisco 7970   david parcerisa
• [asterisk-users] Re: How to change IAX default port 4569 to some other port :Debug Message Attached   Zeeshan Zakaria
• [asterisk-users] DB9 e1 to RJ45 pinout   Tim Panton
• [asterisk-users] Call Transfers in SER + Asterisk architecture   Ricardo Carvalho
• [asterisk-users] Asterisk and TDM400P ?   Anton Frolov
• [asterisk-users] Call Transfers in SER + Asterisk architecture   Marco Mouta
• [asterisk-users] DB9 e1 to RJ45 pinout   Erick Perez
• [asterisk-users] DB9 e1 to RJ45 pinout   Erick Perez
• [asterisk-users] Junk faxes   Doug Lytle
• [asterisk-users] Junk faxes   Colin Anderson
• [asterisk-users] Doubling up; redunancy with DUNDi   Gavin Hamill
• [asterisk-users] Junk faxes   Joe Greco
• [asterisk-users] Junk faxes   Doug Lytle
• [asterisk-users] Asterisk and TDM400P ?   Time Bandit
• [asterisk-users] Asterisk and TDM400P ?   Noc Phibee
• [asterisk-users] DB9 e1 to RJ45 pinout   Alban
• [asterisk-users] Asterisk and TDM400P ?   Tzafrir Cohen
• [asterisk-users] Encrypted password for voicemail   Tzafrir Cohen
• [asterisk-users] Auto recording calls?   Tim Panton
• [asterisk-users] Monitoring awareness   nick005 at free.fr
• [asterisk-users] Server Configuration for E1's   Steve Totaro
• [asterisk-users] cisco 7961 , asterisk and busy lamp   Max Bergmann
• [asterisk-users] Request for help with DISA (Not taking my input number correctly?)   Steve Totaro
• [asterisk-users] Junk faxes   Steve Totaro
• [asterisk-users] mfcr/R2   Alyed Tzompa
• [asterisk-users] Monitoring awareness   Steve Totaro
• [asterisk-users] Junk faxes   Steve Totaro
• [asterisk-users] FS: Sangoma 10 port FXO card   Mark Phillips
• [asterisk-users] FS: Sangoma 10 port FXO card   Anthony Rodgers
• [asterisk-users] Junk faxes   Doug Lytle
• [asterisk-users] Junk faxes   Doug Lytle
• [asterisk-users] FS: Sangoma 10 port FXO card   Mark Phillips
• [asterisk-users] Re: Rewriting caller ID from database?   Vincent Delporte
• [asterisk-users] Re: Rewriting caller ID from database?   Anselm Martin Hoffmeister
• [asterisk-users] Terrible, horrible firewall issues in * to * setup   Lachek Butalek
• [asterisk-users] Terrible, horrible firewall issues in * to * setup   Lachek Butalek
• [asterisk-users] Terrible, horrible firewall issues in * to * setup   Lachek Butalek
• [asterisk-users] When does voicemail authentication take place?   Luki
• [asterisk-users] Asterisk and UK ISDN 30   Neil Tancock
• [asterisk-users] Asterisk and UK ISDN 30   Steve Kennedy
• [asterisk-users] Card don't hangup but Asterisk hangup   Jesus Jimenez
• [asterisk-users] Re:Call Transfers in SER + Asterisk   M.Emran
• [asterisk-users] Card don't hangup but Asterisk hangup   Leo Ann Boon
• [asterisk-users] Re: Rewriting caller ID from database?   Daryl Jones
• [asterisk-users] Correct syntax to access a shell variable?   Larry Alkoff
• [asterisk-users] How to park calls on a specific extension   marvin horst
• [asterisk-users] Re:Call Transfers in SER + Asterisk   Arun Kumar
• [asterisk-users] How to change IAX default port 4569 to some other port   William Piper
• [asterisk-users] How to kill a meet me room at midnight   Eric Bishop
• [asterisk-users] How to kill a meet me room at midnight   John covici
• [asterisk-users] Help needed - Can anyone please explain to me what is causing this - TDM2400P   Naija Man
• [asterisk-users] FOP is not displaying all my SIP extensions neither all E1 channels , why?   Nicolás Gudiño
• [asterisk-users] How to park calls on a specific extension   Brad Templeton
• [asterisk-users] RE: Snom 360 Multiple calls on hold help   Ron McCarthy
• [asterisk-users] Request for help with DISA (Not taking my input number correctly?)   Crazy Boy
• [asterisk-users] Card don't hangup but Asterisk hangup   Jesus Jimenez
• [asterisk-users] FREE DOWNLOAD - PRI / T1 Circuit monitoring   Tzafrir Cohen
• [asterisk-users] Card don't hangup but Asterisk hangup   Tzafrir Cohen
• [asterisk-users] Correct syntax to access a shell variable?   Dominique Dartois
• [asterisk-users] Correct syntax to access a shell variable?   Tzafrir Cohen
• [asterisk-users] Error uninstalling freepbx-panel   Tzafrir Cohen
• [asterisk-users] Card don't hangup but Asterisk hangup   txus
• [asterisk-users] Modem and TDM400P   joe a. (joea at j4computers.com)
• [asterisk-users] How to park calls on a specific extension   marvin horst
• [asterisk-users] VOIP Consultants wanted to build a Scalable ITSP Architecture Using OpenSource Softwares   Shanti kishore Balusu
• [asterisk-users] How to do Call barging with SIP channel   raviprakash sunkara
• [asterisk-users] Re: asterisk-users Digest, Vol 28, Issue 131   DOUGLAS LEBER
• [asterisk-users] Re: Rewriting caller ID from database?   Michelle Dupuis
• [asterisk-users] Re: Rewriting caller ID from database?   Time Bandit
• [asterisk-users] Asterisk and UK ISDN 30   Neil Tancock
• [asterisk-users] Asterisk and UK ISDN 30   Tim Panton
• [asterisk-users] Card don't hangup but Asterisk hangup   Carlos Rojas
• [asterisk-users] DID Provider   broadbandvoice at comcast.net
• [asterisk-users] dialing with different speed   Androtech
• [asterisk-users] 1.4 svn voicemail bug / crash   Robert La Ferla
• [asterisk-users] dialing with different speed   Brett Crapser
• [asterisk-users] Sipura phone does not ring   Fran Oliveira
• [asterisk-users] Problems with sound quality   Ullas
• [asterisk-users] Re: asterisk-users Digest, Vol 28, Issue 132   DOUGLAS LEBER
• [asterisk-users] CallerID number not being displayed on SIP phones   Daryl Jones
• [asterisk-users] Linking Asterisk Servers using SIP instead of IAX   Paul
• [asterisk-users] How to park calls on a specific extension   Brad Templeton
• [asterisk-users] MeetMe, background agi and playing sounds   Jan Eirik Sandnes
• [asterisk-users] Sip reinvite   Vicky
• [asterisk-users] DID Provider   Alex
• [asterisk-users] DID Provider   broadbandvoice at comcast.net
• [asterisk-users] FREE DOWNLOAD - PRI / T1 Circuit monitoring   Dovid B
• [asterisk-users] Re:VOIP Consultants wanted to build a Scalable ITSP Architecture   M.Emran
• [asterisk-users] 1.4 svn voicemail bug / crash   Robert La Ferla
• [asterisk-users] Re: cisco 7961 , asterisk and busy lamp : solved   Max Bergmann
• [asterisk-users] Asterisknow   Carlos Rojas
• [asterisk-users] Digium Iaxy S100 Factory Default?   Matt Gibson
• [asterisk-users] Passing PRI traffic to remote * over IAX   Darren Wright
• [asterisk-users] Passing PRI traffic to remote * over IAX   C F
• [asterisk-users] SOLVED - 1.4 svn voicemail bug / crash   Robert La Ferla
• [asterisk-users] Passing PRI traffic to remote * over IAX   Tzafrir Cohen
• [asterisk-users] 1.4 svn voicemail bug / crash   Tzafrir Cohen
• [asterisk-users] G729 issues on 1.4 beta 3   Russell Bryant
• [asterisk-users] How to change IAX default port 4569 to some other port   Zeeshan Zakaria
• [asterisk-users] Streaming MoH working example required   Zeeshan Zakaria
• [asterisk-users] Streaming MoH working example required   Tzafrir Cohen
• [asterisk-users] DB9 e1 to RJ45 pinout   Niklas Larsson
• [asterisk-users] asterisk 1.4 chan_h323, help please...   Jason Kim
• [asterisk-users] DID Provider   broadbandvoice at comcast.net
• [asterisk-users] Terrible, horrible firewall issues in * to * setup   Tim Panton
• [asterisk-users] Setting up FastAGI in Asterisk?   Bret Schuhmacher
• [asterisk-users] way to get extension's hint?   Nikola Ciprich
• [asterisk-users] Setting up FastAGI in Asterisk?   Jean-Michel Hiver
• [asterisk-users] Re: Rewriting caller ID from database?   Tom Lynn
• [asterisk-users] Auto recording calls?   Tom Lynn
• [asterisk-users] Re: Re: Rewriting caller ID from database?   Vincent Delporte
• [asterisk-users] dialing with different speed   Androtech
• [asterisk-users] Looking for toll-free US did   Vicky
• [asterisk-users] DID Provider   Dovid B
• [asterisk-users] FS: Sangoma 10 port FXO card   Dovid B
• [asterisk-users] Looking for toll-free US did   Dovid B
• [asterisk-users] Re: Re: Rewriting caller ID from database?   Tom Lynn
• [asterisk-users] Re: Rewriting caller ID from database?   Vincent Delporte
• [asterisk-users] Re: Rewriting caller ID from database?   Vincent Delporte
• [asterisk-users] Re: Rewriting caller ID from database?   Vincent Delporte
• [asterisk-users] Re: Re: Rewriting caller ID from database?   Pavel Jezek
• [asterisk-users] dialing with different speed   Brett Crapser
• [asterisk-users] spandsp   Warrick Zedi
• [asterisk-users] Asterisk 1.4 : Gentoo, MYSQL CDR, and GUI howto   Matt Gibson
• [asterisk-users] MWI - Message Waiting Indicator free software routines   Paul
• [asterisk-users] Asterisk 1.4 : AstDB Callerid Rolodex Tool   Matt Gibson
• [asterisk-users] Can anyone enlighten me as to what this means?   Matt
• [asterisk-users] Odd blip when playinv IVR over IAX   Matt
• [asterisk-users] Looking for toll-free US did   Al Bochter
• [asterisk-users] Dialout to Meetme Fails?   Matthew Rubenstein
• [asterisk-users] 3xx redirect from asterisk?   Benjamin Jacob
• [asterisk-users] calls hang up even after Background() message eventhough response timeout is set to 10 sec   Jeronimo Romero
• [asterisk-users] calls hang up even after Background() message eventhough response timeout is set to 10 sec   Doug Crompton
• [asterisk-users] Junghanns Bristuff PRI indication   Kevin Boddy
• [asterisk-users] Re: 3xx redirect from asterisk?   Nick Adams
• [asterisk-users] Click to dial apps always show from "asterisk"   Eric Bishop
• [asterisk-users] Click to dial apps always show from "asterisk"   mitcheloc
• [asterisk-users] bristuff error: "received SETUP message for call that is not a new call"   Louis-David Mitterrand
• [asterisk-users] Re: Junghanns Bristuff PRI indication   Louis-David Mitterrand
• [asterisk-users] How to park calls on a specific extension   Steve Langstaff
• [asterisk-users] How to park calls on a specific extension   Brad Templeton
• [asterisk-users] Odd blip when playinv IVR over IAX   Tim Panton
• [asterisk-users] Asterisk is taking the first digit of my entered number twice. Why?   Crazy Boy
• [asterisk-users] How to park calls on a specific extension   Steve Langstaff
• [asterisk-users] Odd blip when playinv IVR over IAX   Matt
• [asterisk-users] Odd blip when playinv IVR over IAX   Tim Panton
• [asterisk-users] AgentCallbackLogin deprecated?   Miguel Paolino
• [asterisk-users] flash transfer problem in asterisk with old PBX   Andrea Giuliani
• [asterisk-users] Re: Digium through Octasic   Steven
• [asterisk-users] calls hang up even after Background() messageeventhough response timeout is set to 10 sec   Jeronimo Romero
• [asterisk-users] SIP group management   nik600
• [asterisk-users] SIP group management   Marnus van Niekerk
• [asterisk-users] wip5000 crash AP   Altus Snyman
• [asterisk-users] Asterisk Feature Codes won't work   Mattias Andersson
• [asterisk-users] Announce Queue Position variable   bivio
• [asterisk-users] SIP group management   nik600
• [asterisk-users] Asterisk Feature Codes won't work   Marco Mouta
• [asterisk-users] Incoming calls don't arrive for correct number   Frederico Madeira
• [asterisk-users] Trunk Alcatel - Ring problem and call disconnection   Frederico Madeira
• [asterisk-users] RE: Snom 360 Multiple calls on hold help   Jamie Heckford
• [asterisk-users] Script on hold   Idris AVCI
• [asterisk-users] Incoming calls don't arrive for correct number   Marco Mouta
• [asterisk-users] (OT) HylaFAX, IAXModem, Asterisk   mail-lists
• [asterisk-users] Trunk Alcatel - Ring problem and call disconnection   K Y Iyer
• [asterisk-users] SIP group management   J. Oquendo
• [Asterisk-Users] Siemens Gigaset SL75   jbauer at teec.com
• [asterisk-users] BRIcard not sending DTMF   Giorgio Incantalupo
• [asterisk-users] [VoIP Trunk] No such host   Vincent Delporte
• [Asterisk-Users] Siemens Gigaset SL75   Andrew Joakimsen
• [asterisk-users] port rtp problem   Khaled
• [asterisk-users] wip5000 crash AP   Anselm Martin Hoffmeister
• [asterisk-users] DID Provider   Eric "ManxPower" Wieling
• [asterisk-users] Feature and multiple application   Nicolas
• [asterisk-users] CTI   Hernany Oliveira
• [asterisk-users] Queues and Flash/SendDTMF in hybrid PBX   Vieri
• [Asterisk-Users] Siemens Gigaset SL75   Zoa
• [asterisk-users] RE: Snom 360 Multiple calls on hold help   Ron McCarthy
• [asterisk-users] registration ip address   Anselm Martin Hoffmeister
• [asterisk-users] How to kill a meet me room at midnight   Noah Miller
• [asterisk-users] upgraded polycom to 2.0.1.0291 and...   Noah Miller
• [asterisk-users] Re: Digium through Octasic   Aaron Daniel
• [asterisk-users] Multi-site Redundancy. Possible?   Stephen Wingfield
• [asterisk-users] Busy signal from IAXy when not connecting to my Asterisk box   Frank Tarczynski
• [asterisk-users] 3xx redirect from asterisk?   Ron McLeod
• [asterisk-users] Voicemail, SQL & ODBC   Peder at NetworkOblivion
• [asterisk-users] Can anyone enlighten me as to what this means?   Matt
• [asterisk-users] (OT) HylaFAX, IAXModem, Asterisk   Lacy Moore - Aspendora
• [asterisk-users] Can anyone enlighten me as to what this means?   Eric "ManxPower" Wieling
• [asterisk-users] Sangoma & Dell 750   Duracom Lists
• [asterisk-users] Sangoma & Dell 750   Chad Osmond
• [asterisk-users] Caller ID issues   Bruce Reeves
• [asterisk-users] CTI   Bret Schuhmacher
• [asterisk-users] Memory leak   Mitch Thompson
• [asterisk-users] Asterisk server reports   Frederico Madeira
• [asterisk-users] Manage Users in LDAP   Steven Baker
• RES: [asterisk-users] CTI   Hernany Oliveira
• [asterisk-users] CTI   Matt Florell
• [asterisk-users] CTI   Cory Andrews
• [asterisk-users] Voicemail, SQL & ODBC   Norbert Zawodsky
• [asterisk-users] Memory leak   Michael Collins
• [asterisk-users] Caller ID issues   Anton Frolov
• [asterisk-users] Caller ID issues   Eric "ManxPower" Wieling
• [asterisk-users] Caller ID issues   Bruce Reeves
• [asterisk-users] alert_info + Linksys 9xx + custom ringtone   Joshua Citron
• [asterisk-users] Caller ID issues   Eric "ManxPower" Wieling
• [asterisk-users] How to kill a meet me room at midnight   Nathan Bowyer
• [asterisk-users] registration ip address   slavon at bigtelecom.ru
• [asterisk-users] Manage Users in LDAP   Gary Richardson
• [asterisk-users] Modprobe Zaptel   Julian Varanini
• [asterisk-users] alert_info + Linksys 9xx + custom ringtone   Andrew Joakimsen
• [asterisk-users] alert_info + Linksys 9xx + custom ringtone   Joshua Citron
• [asterisk-users] Manage Users in LDAP   Colin Anderson
• [asterisk-users] Caller ID issues   Bruce Reeves
• [asterisk-users] Caller ID issues   Eric "ManxPower" Wieling
• [asterisk-users] Busy signal from IAXy when not connecting to my Asterisk box   Jens Vagelpohl
• [asterisk-users] Manage Users in LDAP   Anil Ramsingh
• [asterisk-users] Busy signal from IAXy when not connecting to my Asterisk box   Andrew Joakimsen
• [asterisk-users] Asterisk server reports   Andrew Joakimsen
• [asterisk-users] Looking for toll-free US did   Andrew Joakimsen
• [asterisk-users] Encrypted password for voicemail   jezzzz .
• [asterisk-users] When does voicemail authentication take place?   jezzzz .
• [asterisk-users] When does voicemail authentication take place?   Luki
• [asterisk-users] How to kill a meet me room at midnight   Eric Bishop
• [asterisk-users] Click to dial apps always show from "asterisk"   Eric Bishop
• [asterisk-users] AsteriskNow console access   Geoff Karl
• [asterisk-users] SayDecimal Number   Sharon Lim
• [asterisk-users] How to park calls on a specific extension   Brad Templeton
• [asterisk-users] Do extra CPU's help?   Eric Bishop
• [asterisk-users] How to park calls on a specific extension   Andrew Joakimsen
• [asterisk-users] Re: AsteriskNow console access   Geoff Karl
• [asterisk-users] Voicemail, SQL & ODBC   RR
• [asterisk-users] SRTP   Khaled
• [asterisk-users] Do extra CPU's help?   Don
• [asterisk-users] How to park calls on a specific extension   Brad Templeton
• [asterisk-users] Do extra CPU's help?   Eric Bishop
• [asterisk-users] Click to dial apps always show from "asterisk"   mitcheloc
• [asterisk-users] Do extra CPU's help?   Don
• [asterisk-users] How to kill a meet me room at midnight   Noah Miller
• [asterisk-users] vm_change_password shell?   jezzzz .
• [asterisk-users] Encrypted password for voicemail   Tzafrir Cohen
• [asterisk-users] Modprobe zaptel reports FATAL: Module zaptel not found   kjcsb
• [asterisk-users] vm_change_password shell?   Tzafrir Cohen
• [asterisk-users] Modprobe zaptel reports FATAL: Module zaptel not found   Tzafrir Cohen
• [asterisk-users] AGI Script with parameters   Olivier Saulnier
• [asterisk-users] Re: upgraded polycom to 2.0.1.0291 and...   Shaun
• [asterisk-users] AGI Script with parameters   Anton Frolov
• [asterisk-users] AGI Script with parameters   Olivier Saulnier
• [asterisk-users] AGI Script with parameters   Anton Frolov
• [asterisk-users] Click to dial apps always show from "asterisk"   Tim Panton
• [asterisk-users] AGI Script with parameters   Tim Panton
• [asterisk-users] AGI Script with parameters   Olivier Saulnier
• [asterisk-users] Attn: DISA Experts(Strange problem with DISA)   Crazy Boy
• [asterisk-users] Use of VPNs   Barry Fawthrop
• [asterisk-users] Different click2call?   Darko
• [asterisk-users] Use of VPNs   Conrad Wood
• [asterisk-users] Symbian Softphone   Marnus van Niekerk
• [asterisk-users] Manage Users in LDAP   marvin horst
• [asterisk-users] Symbian Softphone   Klaus Darilion
• RES: [asterisk-users] CTI   Hernany Oliveira
• [asterisk-users] Zaptel drivers for Solaris?   Frank Tarczynski
• [asterisk-users] Anybody used Asterfax?   Klaus Darilion
• [asterisk-users] Can anyone enlighten me as to what this means?   Rob McKrill
• [asterisk-users] Anybody used Asterfax?   Zeeshan Zakaria
• [asterisk-users] AsteriskNow console access   James Willing
• [asterisk-users] Voicemail, SQL & ODBC   Norbert Zawodsky
• [asterisk-users] aastra 480i xml interface for "comedian" mail   marvin horst
• [asterisk-users] registration ip address   Khaled
• [asterisk-users] Zaptel drivers for Solaris?   Steve Kennedy
• [asterisk-users] Zaptel drivers for Solaris?   Matt Florell
• [asterisk-users] Symbian Softphone   Marnus van Niekerk
• [asterisk-users] CTI   Matt Florell
• [asterisk-users] cmd Record doesn't resume Dialplan if phone Hangs-Up.   Jean-Marc Salsa
• [asterisk-users] How to kill a meet me room at midnight   Richard Lyman
• [asterisk-users] Manage Users in LDAP   Ejay Hire
• [asterisk-users] Do extra CPU's help?   Colin Anderson
• [asterisk-users] Symbian Softphone   Emil Thelin
• [asterisk-users] Re: cisco 7961 , asterisk and busy lamp : solved   John Reynolds
• [asterisk-users] cmd Record doesn't resume Dialplan if phone Hangs-Up.   Anselm Martin Hoffmeister
• [asterisk-users] Manage Users in LDAP   Walt Reed
• [asterisk-users] Asterisk "Generating" SIP 486   Jonathan Palley
• [asterisk-users] Zaptel drivers for Solaris?   Jason Parker
• [asterisk-users] AgentCallbackLogin deprecated?   Jason Parker
• [asterisk-users] SIP ATA Device Problems   Jerry Rasmussen
• [asterisk-users] AgentCallbackLogin deprecated?   Gavin Hamill
• [asterisk-users] Encrypted password for voicemail   jezzzz .
• [asterisk-users] AGI Script with parameters   Steve Edwards
• [asterisk-users] Modprobe zaptel reports FATAL: Module zaptel notfound   kjcsb
• [asterisk-users] AGI and some informations   Olivier Saulnier
• [asterisk-users] Different click2call?   Tim Panton
• [asterisk-users] Zaptel drivers for Solaris?   Andrew Joakimsen
• [asterisk-users] AGI and some informations   Anton Frolov
• [asterisk-users] Return codes   Doug Crompton
• [asterisk-users] Return codes   Tim Panton
• [asterisk-users] Zaptel drivers for Solaris?   Matt Florell
• [asterisk-users] hang up detection   Matic
• [asterisk-users] WANTED : Zaptel Patch - Dtmfthreshold   Stefan Agethen
• [asterisk-users] Best email client for Asterfax   Olivier
• [asterisk-users] Encrypted password for voicemail   Tzafrir Cohen
• [asterisk-users] Modprobe zaptel reports FATAL: Module zaptel notfound   Tzafrir Cohen
• [asterisk-users] hang up detection   Tzafrir Cohen
• [asterisk-users] Why is * continually "destroying call"   Larry Alkoff
• [asterisk-users] Do extra CPU's help?   Tzafrir Cohen
• [asterisk-users] Call recording filename   Vicky
• [asterisk-users] VoiceOne 0.4.0 released: a new web-based and open source GUI   Nicolas S.
• [asterisk-users] Why is * continually "destroying call"   Colin Anderson
• [asterisk-users] hang up detection   Matic
• [asterisk-users] AGI and some informations   Olivier Saulnier
• [asterisk-users] IAX access to FWD broken?   Timothy Parez
• [asterisk-users] AGI and some informations   Anton Frolov
• [asterisk-users] channel ending with /n   Nuria Fernández Mingo
• [asterisk-users] Bad Voice Quality - IAX2 redirect   hugolivude
• [asterisk-users] Bad Voice Quality - IAX2 redirect   Andrew Joakimsen
• [asterisk-users] SIP ATA Device Problems   Andrew Joakimsen
• [asterisk-users] AgentCallbackLogin deprecated?   Octavio Ruiz (Ta^3)
• [asterisk-users] No sound: X-Lite -> Asterisk -> VoIP Provider -> Cellphone   Vincent Delporte
• [asterisk-users] Asterisknow   Dumpolid Exeplish
• [asterisk-users] AsteriskNow console access   Dumpolid Exeplish
• [asterisk-users] Re: Re: Re: Re: Rewriting caller ID from database?   Vincent Delporte
• [asterisk-users] Billing software with reseller accounts   Guillermo Salas M.
• [asterisk-users] Voicemail, SQL & ODBC   Derek Whitten
• [asterisk-users] Voicemail, SQL & ODBC   Noah Miller
• [asterisk-users] Zaptel drivers for Solaris?   Derek Whitten
• [asterisk-users] IAX access to FWD broken?   jason
• [asterisk-users] Zaptel drivers for Solaris?   Andrew Joakimsen
• [asterisk-users] Re: newbie question-asterisk username/password   blackwater dev
• [asterisk-users] accountcode= placement in zapata.conf   Damon Estep
• [asterisk-users] accountcode= placement in zapata.conf   Andrew Joakimsen
• [asterisk-users] SIP Port 5060   lists at infoway.net
• [asterisk-users] accountcode= placement in zapata.conf   Andrew Joakimsen
• [asterisk-users] SIP Port 5060   Andrew Joakimsen
• [asterisk-users] SIP Port 5060   Patrick
• [asterisk-users] Best text to speech program   Hall, Eric M.
• [asterisk-users] Encrypted password for voicemail   jezzzz .
• [asterisk-users] When does voicemail authentication take place?   jezzzz .
• [asterisk-users] SIP Port 5060   Tom Lynn
• [asterisk-users] Best text to speech program   Tom Lynn
• [asterisk-users] Survey: In what ways do you use Asterisk at your house?   Tom Lynn
• [asterisk-users] Encrypted password for voicemail   Tzafrir Cohen
• [asterisk-users] Encrypted password for voicemail   jezzzz .
• [asterisk-users] vm_change_password shell?   jezzzz .
• [asterisk-users] SIP Port 5060   Joseph
• [asterisk-users] Custom Voicemail Notification Email   RR
• [asterisk-users] Custom Voicemail Notification Email   Marnus van Niekerk
• [asterisk-users] Custom Voicemail Notification Email   Marnus van Niekerk
• [asterisk-users] iptables example   Jeronimo Romero
• [asterisk-users] AGI and some informations   Olivier Saulnier
• [asterisk-users] Custom Voicemail Notification Email   RR
• [asterisk-users] Which SIP transport from France and termination services in the Nederlands   Olivier
• Resolved: Re: [asterisk-users] Modprobe zaptel reports FATAL: Module zaptel notfound   kjcsb
• [asterisk-users] Custom Voicemail Notification Email   Marnus van Niekerk
• [asterisk-users] Custom Voicemail Notification Email   Marnus van Niekerk
• [asterisk-users] Play an announcement while receiving DTMF?   Tony Mountifield
• [asterisk-users] iptables example   Scott Wolfe
• [asterisk-users] Something similar or better than HUD Pro?   Zeeshan Zakaria
• [asterisk-users] SIP Port 5060   Zeeshan Zakaria
• [asterisk-users] Click to dial apps always show from "asterisk"   Eric Bishop
• [asterisk-users] sendmail or postfix?   Zeeshan Zakaria
• [asterisk-users] Custom Voicemail Notification Email   RR
• [asterisk-users] Re: SIP group management   Benny Amorsen
• [asterisk-users] sendmail or postfix?   Tzafrir Cohen
• [asterisk-users] Which SIP transport from France and termination services in the Nederlands   Alban
• [asterisk-users] chan_misdn on a junghanns card   Louis-David Mitterrand
• [asterisk-users] Monitoring an asterisk server during off hours   Olivier
• [asterisk-users] Re: chan_misdn on a junghanns card   Louis-David Mitterrand
• [asterisk-users] Cisco 7970 SIP upgrade issues   Paul
• [asterisk-users] Monitoring an asterisk server during off hours   Mattias Andersson
• [asterisk-users] Cisco 7970 SIP upgrade issues   Mattias Andersson
• [asterisk-users] Siemens Gigaset C450 IP vs S450 IP   Eugen Leitl
• [asterisk-users] Cisco 7970 SIP upgrade issues   Paul A Brown
• [asterisk-users] Re: SIP Port 5060 (Tom Lynn)   Leonardo B
• [asterisk-users] mISDN   Timothy Parez
• [asterisk-users] IAX access to FWD broken?   Timothy Parez
• [asterisk-users] IAX access to FWD broken?   Timothy Parez
• [asterisk-users] Cisco 7970 SIP upgrade issues   Mattias Andersson
• [asterisk-users] Siemens Gigaset C450 IP vs S450 IP   Florian Overkamp
• [asterisk-users] AGI and some informations   Olivier Saulnier
• [asterisk-users] Siemens Gigaset C450 IP vs S450 IP   Eugen Leitl
• [asterisk-users] Cisco 7970 SIP upgrade issues   Paul A Brown
• [asterisk-users] Desktop application for zap/agent call control   Jordan Novak
• [asterisk-users] keep line on hook   Marco Vescovi
• [asterisk-users] Answer Supervision problem   Timur S. Sattarov
• [asterisk-users] Cisco 7970 SIP upgrade issues   Mattias Andersson
• [asterisk-users] mISDN   Patrick
• [asterisk-users] Re: IAX access to FWD broken?   Jim Lawson
• [asterisk-users] AGI PHP Issues (Not new to Asterisk but new to AGI)   Chris Blunt
• [asterisk-users] mISDN   Giorgio Incantalupo
• [asterisk-users] Re: IAX access to FWD broken?   Al Bochter
• [asterisk-users] Re: IAX access to FWD broken?   Michael Graves
• [asterisk-users] Cisco 7970 SIP upgrade issues   Paul A Brown
• [asterisk-users] Voicemail, SQL & ODBC   Norbert Zawodsky
• [asterisk-users] Voicemail, SQL & ODBC   Norbert Zawodsky
• [asterisk-users] Re: IAX access to FWD broken?   Timothy Parez
• [asterisk-users] Blind transfer # not working for forwarded or picked calls   Roger Lewau
• [asterisk-users] mISDN   Timothy Parez
• [asterisk-users] Asterisk + Avaya S8700   Tomer Horn
• [asterisk-users] mISDN   Timothy Parez
• [asterisk-users] IAX access to FWD broken?   Derek Whitten
• [asterisk-users] mISDN   Timothy Parez
• [asterisk-users] b410p hangup detection - Portugal   Nuno Pais Fernandes
• [asterisk-users] Re: IAX access to FWD broken?   Zed
• [asterisk-users] Getting app_cepstral to work with Asterisk 1.4.0-beta3   Hall, Eric M.
• [asterisk-users] Asterisk + Avaya S8700   Michel R Vaillancourt
• [asterisk-users] Loosing IAX connection between offices   DM
• [asterisk-users] Sipura phone does not ring   Larry Alkoff
• [asterisk-users] g729 registered   Ralph Liebessohn
• [asterisk-users] AGI PHP Issues (Not new to Asterisk but new to AGI)   Time Bandit
• [asterisk-users] What's up with the Manager Interface?!?!   Douglas Garstang
• [asterisk-users] SIP Port 5060   lists at infoway.net
• [asterisk-users] SIP Port 5060   lists at infoway.net
• [asterisk-users] SIP Port 5060   lists at infoway.net
• [asterisk-users] What's up with the Manager Interface?!?!   Michael Collins
• [asterisk-users] Call Recording and Call Transfers   Stephen Kratzer
• [asterisk-users] Getting app_cepstral to work with Asterisk 1.4.0-beta3   Earle Clubb
• [asterisk-users] I am unable to find any included rpms with hudlite...   Jordan Novak
• [asterisk-users] What's up with the Manager Interface?!?!   Richard Lyman
• [asterisk-users] Getting app_cepstral to work with Asterisk 1.4.0-beta3   Bob Chiodini
• [asterisk-users] What's up with the Manager Interface?!?!   Douglas Garstang
• [asterisk-users] Asterisk connection to a PBX   asterisk-robert
• [asterisk-users] NEED ASTERISK DEVELPER : OH323-asterisk driver and openh323   Oliver Vermeulen
• [asterisk-users] What's up with the Manager Interface?!?!   Matt Florell
• [asterisk-users] What's up with the Manager Interface?!?!   Douglas Garstang
• [asterisk-users] Voicemail, SQL & ODBC   Noah Miller
• [asterisk-users] No sound: X-Lite -> Asterisk -> VoIP Provider -> Cellphone   Noah Miller
• [asterisk-users] mISDN   Patrick
• [asterisk-users] What's up with the Manager Interface?!?!   Michael Collins
• [asterisk-users] Voicemail, SQL & ODBC   Norbert Zawodsky
• Spam? Re: [asterisk-users] Getting app_cepstral to work withAsterisk 1.4.0-beta3   Hall, Eric M.
• [asterisk-users] Playing streaming MOH in Asterisk   Matt
• [asterisk-users] Polycom 601 Second Incoming Call   Dovid B
• [asterisk-users] What's up with the Manager Interface?!?!   Douglas Garstang
• [asterisk-users] Re: Spandsp rxfax txtax fails no errors   daveasterisk
• [asterisk-users] voicemail.conf locking problem   jezzzz .
• [asterisk-users] What's up with the Manager Interface?!?!   Steve Edwards
• [asterisk-users] What's up with the Manager Interface?!?!   James Texter
• [asterisk-users] voicemail.conf locking problem   Scott Keagy
• [asterisk-users] Billing software with reseller accounts   Dovid B
• [asterisk-users] beeping noise in background   Kim Jones
• [asterisk-users] What's up with the Manager Interface?!?!   Douglas Garstang
• [asterisk-users] g726 voice prompts   Eric Bishop
• [asterisk-users] Re: What's up with the Manager Interface?!?!   Tony Mountifield
• [asterisk-users] What's up with the Manager Interface?!?!   Douglas Garstang
• [asterisk-users] Cisco 7940 Firmware 8.2   James R. Stevens
• [asterisk-users] voicemail.conf locking problem   Michiel van Baak
• [asterisk-users] What's up with the Manager Interface?!?!   Richard Lyman
• [asterisk-users] Call recording with Asterisk BE   Ed Nuñez
• [asterisk-users] voicemail.conf locking problem   Tzafrir Cohen
• [asterisk-users] How to park calls on a specific extension   Steve Sobol
• [asterisk-users] What's up with the Manager Interface?!?!   Douglas Garstang
• [asterisk-users] How to park calls on a specific extension   Steve Sobol
• [asterisk-users] Call dropping   Ed Nuñez
• [asterisk-users] SIP Port 5060   Joseph
• [asterisk-users] What's up with the Manager Interface?!?!   Douglas Garstang
• [asterisk-users] Setting RTP ports for Asterisk?   Vincent Delporte
• [asterisk-users] Re: What's up with the Manager Interface?!?!   Richard Lyman
• [asterisk-users] What's up with the Manager Interface?!?!   Richard Lyman
• [asterisk-users] How to park calls on a specific extension   Ira
• [asterisk-users] What's up with the Manager Interface?!?!   Ira
• [asterisk-users] SIP Port 5060   Andrew Joakimsen
• [asterisk-users] Siemens Gigaset C450 IP vs S450 IP   Andrew Joakimsen
• [asterisk-users] Attn: DISA Experts(Strange problem with DISA)   Andrew Joakimsen
• [asterisk-users] SIP Port 5060   Zeeshan Zakaria
• [asterisk-users] What's up with the Manager Interface?!?!   Sig Lange
• [asterisk-users] Conferencing Issue please help   Ishanka Anuradha Ranasooriya
• [asterisk-users] Re: asterisk-users Digest, Vol 28, Issue 152   Ishanka Anuradha Ranasooriya
• [asterisk-users] Trouble using 2 IAX2 DiDs provided by different ITSPs   hugolivude
• [asterisk-users] Asterisk connection to a PBX   Tom Lynn
• [asterisk-users] Asterisk connection to a PBX   asterisk-robert
• [asterisk-users] Setting RTP ports for Asterisk?   Noah Miller
• [asterisk-users] Siemens Gigaset C450 IP vs S450 IP   Cory Andrews
• [asterisk-users] How to park calls on a specific extension   Brad Templeton
• [asterisk-users] register history   rilawich ango
• [asterisk-users] SIP Port 5060   Brad Templeton
• [asterisk-users] How to park calls on a specific extension   Lacy Moore - Aspendora
• [asterisk-users] SIP Port 5060   lists at infoway.net
• [asterisk-users] SIP Port 5060   lists at infoway.net
• [asterisk-users] How to park calls on a specific extension   Brian Capouch
• [asterisk-users] SIP Port 5060   Leo Ann Boon
• [asterisk-users] extension launch into AGI   Roy Kidder
• [asterisk-users] Return code - How to?   Doug Crompton
• [asterisk-users] Voicemail, SQL & ODBC   Benjamin Jacob
• [asterisk-users] MeetMe announcements and SIP channels   Mike
• [asterisk-users] How to park calls on a specific extension   Lacy Moore - Aspendora
• [asterisk-users] Voicemail, SQL & ODBC   RR
• [asterisk-users] voicemail.conf locking problem   jezzzz .
• [asterisk-users] How to park calls on a specific extension   Ira
• [asterisk-users] Siemens Gigaset C450 IP vs S450 IP   Dave Cotton
• [asterisk-users] Cut function on semicolon separator   Koen Van Impe
• [asterisk-users] What's up with the Manager Interface?!?!   Dovid B
• [asterisk-users] What's up with the Manager Interface?!?!   Dovid B
• [asterisk-users] Distinctive ring   Tomislav Parčina
• [asterisk-users] Cut function on semicolon separator   Peter Lindquist
• [asterisk-users] mISDN   nuria fernandezm
• [asterisk-users] Digium TE405P dtmf issue   leonimar cape
• [asterisk-users] Trouble with regexten   Russell Brown
• [asterisk-users] 200+ analog phones connected to FXS modules   Vieri
• [asterisk-users] 200+ analog phones connected to FXS modules   Carlo Taguinod
• [asterisk-users] 200+ analog phones connected to FXS modules   Paco Brufal
• [asterisk-users] Cut function on semicolon separator   Koen Van Impe
• [asterisk-users] 200+ analog phones connected to FXS modules   Jon Schøpzinsky
• [asterisk-users] AGI PHP Issues (AGI script runs but phone hangs up too quickly)   Chris Blunt
• [asterisk-users] Re: Cisco 7940 Firmware 8.2   Tomislav Parčina
• [asterisk-users] Cut function on semicolon separator   Artifex Maximus
• [asterisk-users] How to park calls on a specific extension   Brad Templeton
• [asterisk-users] AgentCallbackLogin deprecated?   Gavin Hamill
• [asterisk-users] Cut function on semicolon separator   Peter Boehm
• [asterisk-users] codec error message   rilawich ango
• [asterisk-users] Cut function on semicolon separator   Koen Van Impe
• [asterisk-users] 200+ analog phones connected to FXS modules   John covici
• [asterisk-users] 200+ analog phones connected to FXS modules   Jon Schøpzinsky
• [asterisk-users] 200+ analog phones connected to FXS modules   Zoa
• [asterisk-users] 200+ analog phones connected to FXS modules   John covici
• [asterisk-users] Setting RTP ports for Asterisk?   Derek Whitten
• [asterisk-users] Monitoring awareness   Ondrej Valousek
• [asterisk-users] Server Compatibility questions... IBM and Dell   Mark Edwards
• [asterisk-users] Voicemail, SQL & ODBC   Norbert Zawodsky
• [asterisk-users] Asterisk connection to a PBX   Mark Edwards
• [asterisk-users] Voicemail, SQL & ODBC   Derek Whitten
• [asterisk-users] Polycom 601 Second Incoming Call   Jerry Jones
• [asterisk-users] Polycom 601 Second Incoming Call   Walt Reed
• [asterisk-users] AgentCallbackLogin deprecated?   Gavin Hamill
• [asterisk-users] Server Compatibility questions... IBM and Dell   Joe Dennick
• [asterisk-users] SIP transfer from agent fails   Damon Estep
• [asterisk-users] Server Compatibility questions... IBM and Dell   Mark Edwards
• [asterisk-users] extension launch into AGI   Time Bandit
• [asterisk-users] Voicemail, SQL & ODBC   RR
• [asterisk-users] Trouble with regexten   Watkins, Bradley
• [asterisk-users] Trouble using 2 IAX2 DiDs provided by different ITSPs   hugolivude
• [asterisk-users] Loosing IAX connection between offices   DM
• [asterisk-users] Loosing IAX connection between offices   jason
• [asterisk-users] IP call to extensions off my server   Jerry Geis
• [asterisk-users] Call recording with Asterisk BE   Noah Miller
• [asterisk-users] Monitoring awareness   Nicolas
• [asterisk-users] extension launch into AGI   Roy Kidder
• [asterisk-users] zombie SIP channels after CURL cnam lookup   Damon Estep
• [asterisk-users] IP call to extensions off my server   Damon Estep
• [asterisk-users] 200+ analog phones connected to FXS modules   C F
• [asterisk-users] meetme monitoring   Tamas Cseke
• [asterisk-users] Billing Software   lists at infoway.net
• [asterisk-users] Modprobe Zaptel   Tzafrir Cohen
• [asterisk-users] Re: MeetMe announcements and SIP channels   Tony Mountifield
• [asterisk-users] Re: AGI PHP Issues (AGI script runs but phone hangs up too quickly)   Tony Mountifield
• [asterisk-users] Trouble with regexten   Andrew Joakimsen
• [asterisk-users] Loosing IAX connection between offices   DM
• [asterisk-users] zaptel compilation problems with linux 2.6.19   Roman Yeryomin
• [asterisk-users] Re: Spandsp rxfax txtax fails no errors   daveasterisk
• [asterisk-users] T1's in St. Lucia   Forum
• [asterisk-users] Problem with ZapRAS and asterisk   Achille.Sogliani at selex-comms.com
• [asterisk-users] Live call monitoring   Yaakov Menken
• [asterisk-users] 200+ analog phones connected to FXS modules   Andrew Kohlsmith
• [asterisk-users] Trouble with regexten   Watkins, Bradley
• [asterisk-users] Digium through Octasic   Andrew Kohlsmith
• [asterisk-users] Asterisk 1.4 : App_Swift (Cepstral) Howto   Matt Gibson
• [asterisk-users] Trouble with regexten   Watkins, Bradley
• [asterisk-users] Loosing IAX connection between offices   Dave Fullerton
• [asterisk-users] Live call monitoring   Time Bandit
• [asterisk-users] 200+ analog phones connected to FXS modules   Zoa
• [asterisk-users] Re: AGI PHP Issues (AGI script runs but phone hangs up too quickly)   Ove Aursand
• [asterisk-users] Asterisk 1.4 : App_Swift (Cepstral) Howto   Hall, Eric M.
• [asterisk-users] T1's in St. Lucia   Chris Mason (Lists)
• [asterisk-users] Asterisk 1.4 : App_Swift (Cepstral) Howto   Michael Collins
• [asterisk-users] Loosing IAX connection between offices   jason
• [asterisk-users] Asterisk 1.4 : App_Swift (Cepstral) Howto   Hall, Eric M.
• [asterisk-users] zaptel compilation problems with linux 2.6.19   Matthew Rubenstein
• [asterisk-users] zaptel compilation problems with linux 2.6.19   Tzafrir Cohen
• [asterisk-users] How to park calls on a specific extension   Tom Rymes
• [asterisk-users] Voicemail callback bug?   Damon Estep
• [asterisk-users] zaptel compilation problems with linux 2.6.19   Matthew Rubenstein
• [asterisk-users] zaptel compilation problems with linux 2.6.19   Matthew Rubenstein
• [asterisk-users] Loosing IAX connection between offices   Time Bandit
• [asterisk-users] Loosing IAX connection between offices   DM
• [asterisk-users] 200+ analog phones connected to FXS modules   Tzafrir Cohen
• [asterisk-users] 2nd attempt - Return code - How to?   Doug Crompton
• [asterisk-users] 2nd attempt - Return code - How to?   Time Bandit
• [asterisk-users] Pickup *8 with CallerID   Nik Engel
• [asterisk-users] voicemailmain   John Hill
• [asterisk-users] Trouble using 2 IAX2 DiDs provided by different ITSPs   hugolivude
• [asterisk-users] 200+ analog phones connected to FXS modules   Dovid B
• [asterisk-users] incominglimit and outgoinglimit   Nik Engel
• [asterisk-users] PAP2 and Asterisk   phil.dawson at marnock.com
• [asterisk-users] re:voicemailmain   John Hill
• [asterisk-users] VoIP GSM Gateways   Sam Tam
• [asterisk-users] Pickup *8 with CallerID   Andrew Joakimsen
• [asterisk-users] voicemailmain   Dovid B
• [asterisk-users] Cisco 7970 SIP upgrade issues   Paul A Brown
• [asterisk-users] zaptel compilation problems with linux 2.6.19   Tzafrir Cohen
• [asterisk-users] VoIP GSM Gateways   Steve Kennedy
• [asterisk-users] Force re-read of sip.conf   Michelle Dupuis
• [asterisk-users] Force re-read of sip.conf   Darryl Dunkin
• [asterisk-users] Asterisk 1.4 : App_Swift (Cepstral) Howto   Matt Gibson
• [asterisk-users] Pickup *8 with CallerID   Matt
• [asterisk-users] How to park calls on a specific extension   Brad Templeton
• [asterisk-users] upgrading grandstream GXP-2000 from 1.0.2.13 to 1.1.1.14   Scott Keagy
• [asterisk-users] 1.4beta3 help   Doug Crompton
• [asterisk-users] Trouble with regexten   Andrew Joakimsen
• [asterisk-users] How to park calls on a specific extension   Lacy Moore - Aspendora
• [asterisk-users] 1.4beta3 help   Matt Gibson
• [asterisk-users] Force re-read of sip.conf   Mervyn Yeo
• [asterisk-users] 1.4beta3 help   Doug Crompton
• [asterisk-users] Polycom 601 Second Incoming Call   James Andrewartha
• [asterisk-users] How to park calls on a specific extension   Andrew Joakimsen
• [asterisk-users] mfcr/R2   Moises Silva
• [asterisk-users] PHP to .call file   Moises Silva
• [asterisk-users] 1.4beta3 help   Thirumal Saminathan
• [asterisk-users] Answer Supervision problem   Timur S. Sattarov
• [asterisk-users] CAPI module issue   Esteban Guana-Jarrin
• [asterisk-users] Pickup *8 with CallerID   yusuf
• [asterisk-users] zaptel compilation problems with linux 2.6.19   Matthew Rubenstein