October 2006 Archives by date

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Starting: Sun Oct 1 01:09:05 MST 2006
Ending: Tue Oct 31 23:56:30 MST 2006
Messages: 3146

• [asterisk-users] Dial-9 (was Extension Numbering)   Matthew Thompson
• [asterisk-users] Real-time LDAP config   Richi Plana
• [asterisk-users] Install issues on Freebsd 6.1 with Asterisk 1.4.0-beta2   Artifex Maximus
• [asterisk-users] polycomphone   clive.chan(Alpha Trilogies Networks)
• [asterisk-users] Dial-9 (was Extension Numbering)   Gordon Henderson
• [asterisk-users] WiFi SIP handset with Bluetooth required   Andy Green
• [asterisk-users] WiFi SIP handset with Bluetooth required   Andrew Kohlsmith
• [asterisk-users] WiFi SIP handset with Bluetooth required   Zeb Fletcher
• [asterisk-users] Dial-9 (was Extension Numbering)   Eric "ManxPower" Wieling
• [asterisk-users] extensions.conf strangeness   Marco Mouta
• [asterisk-users] detecting busy on queue transfer   Lenz
• [asterisk-users] detecting busy on queue transfer   Adam Goryachev
• [asterisk-users] detecting busy on queue transfer   Lenz
• [Asterisk-Users] Help, please help -- IAX2 softphone to server on LAN   Naija Man
• [asterisk-users] Dial-9 (was Extension Numbering)   Jay R. Ashworth
• [asterisk-users] polycomphone   Alex Robar
• [asterisk-users] Dial-9 (was Extension Numbering)   Eric "ManxPower" Wieling
• [asterisk-users] Dial-9 (was Extension Numbering)   Jay R. Ashworth
• [asterisk-users] Fax detection ...   Lee Howard
• [asterisk-users] Fax detection ...   Jay R. Ashworth
• [asterisk-users] Fax detection ...   Marco Mouta
• [asterisk-users] Another Issue with 1.4   Shidan
• [asterisk-users] detecting busy on queue transfer   Marco Mouta
• [asterisk-users] Fax detection ...   Lee Howard
• [asterisk-users] polycomphone (SOLVED)   clive.chan(Alpha Trilogies Networks)
• [asterisk-users] DSL router with integrated SIP proxy?   Ed Greenberg
• [asterisk-users] WiFi SIP handset with Bluetooth required   Cory Andrews
• [asterisk-users] Fax detection ...   Steve Underwood
• [asterisk-users] G726 prompts   RR
• [asterisk-users] G729 Codec Loading Error   David Gagnon
• [asterisk-users] detecting busy on queue transfer   lenz
• [asterisk-users] Bristuff vs. vISDN vs. mISDN for hfc card ?   Robert Rozman
• [asterisk-users] channel_find_locked   Alexandru Voinescu
• [asterisk-users] Bristuff vs. vISDN vs. mISDN for hfc card ?   Michiel van Baak
• [asterisk-users] cisco 2600   Tijl Van den Broeck
• [asterisk-users] cisco 2600   Idris AVCI
• I: [asterisk-users] Sip answer one side , ring other side   antonio
• [asterisk-users] Siemens Hipath <-> asterisk, pri problem   bivio
• R: [asterisk-users] Siemens Hipath <-> asterisk, pri problem   Giordano Grandis
• [asterisk-users] detecting busy on queue transfer   Marco Mouta
• [asterisk-users] Siemens Hipath <-> asterisk, pri problem   Marco Mouta
• [asterisk-users] asterisk-oh323   Christian Gatti
• [asterisk-users] suggest a configuration   stan ford
• [asterisk-users] Siemens Hipath <-> asterisk, pri problem   bivio
• [asterisk-users] Siemens Hipath <-> asterisk, pri problem   Viktor Tatianin
• R: [asterisk-users] Siemens Hipath <-> asterisk, pri problem   Giordano Grandis
• [asterisk-users] Siemens Hipath <-> asterisk, pri problem   Marco Mouta
• [asterisk-users] Building the Perfect Box   adebayo omo-dare
• [asterisk-users] Siemens Hipath <-> asterisk, pri problem   bivio
• [asterisk-users] Where are the kernel sources?   Jim Lynch
• [asterisk-users] Can't get second line of Sangoma A200 to work.   Jim Lynch
• [asterisk-users] Call Quality / Echo / Problems   Barry Fawthrop
• [asterisk-users] can't transcode ilbc   James Harper
• [asterisk-users] Spying a channel in a meetme   Eduard Martínez
• [asterisk-users] attended transfer unreliable (2nd try)   Stefan Friedrich
• [asterisk-users] attended transfer unreliable (2nd try)   Doug Lytle
• [asterisk-users] Dial and connect to sip provider works, but no audio.   Jim Lynch
• [asterisk-users] Issues with calling certain phone numbers...   Luca Corti
• [asterisk-users] attended transfer unreliable (2nd try)   Florian Hars
• [asterisk-users] asterisk queues with SER, aka "sip show peers"   Mark Price
• [asterisk-users] 480i phone: Is there a trick to registering with *?? <--Solved   Colin Anderson
• [asterisk-users] Issues with calling certain phone numbers...   Marco Mouta
• [asterisk-users] Issues with calling certain phone numbers...   Marco Mouta
• [asterisk-users] 480i phone: Is there a trick to registering with *?? <--Solved, first impressions   Colin Anderson
• [asterisk-users] Asterisk 1.2.10 and SCCP   Scott Higginbotham
• [asterisk-users] Fax detection ...   Jay R. Ashworth
• [asterisk-users] G726 prompts   Jay R. Ashworth
• [asterisk-users] recommended application for salesman using asterisk   Yu Safin
• [asterisk-users] Fax detection ...   Lee Howard
• [asterisk-users] t1 voip to failover pri   stan ford
• [asterisk-users] !! No channel map, no channel, and no ds1? What am I supposed to identify?   Frederico Madeira
• [asterisk-users] t1 voip to failover pri   Colin Anderson
• [asterisk-users] t1 voip to failover pri   Steve Glaus
• [asterisk-users] Conversations Mix   Panitaxx
• [asterisk-users] 480i phone: Is there a trick to registering with *??   Mark Hulber
• [asterisk-users] Re: can't transcode ilbc   Martin Joseph
• [asterisk-users] Re: can't transcode ilbc   Martin Joseph
• [asterisk-users] Re: WiFi SIP handset with Bluetooth required   Martin Joseph
• [asterisk-users] Queue AddQueueMember()   Douglas Garstang
• [asterisk-users] t1 voip to failover pri   Jay R. Ashworth
• [asterisk-users] g729 Codec for AMD Sempron   sdcharly at gmail.com
• [asterisk-users] Fax detection ...   phil.dawson at marnock.com
• [asterisk-users] Passing Arguments to FastAGI   Douglas Garstang
• [asterisk-users] Fax detection ...   phil.dawson at marnock.com
• [asterisk-users] t1 voip to failover pri   adebayo omo-dare
• [asterisk-users] t1 voip to failover pri   stan ford
• [asterisk-users] Fax detection ...   Lee Howard
• [asterisk-users] Minexpiry time - how to set this   Cavanna, Richard
• [asterisk-users] Fax detection ...   James
• [asterisk-users] How do I reset a password?   George Masgras
• [asterisk-users] extensions.conf strangeness   Brian Candler
• [asterisk-users] Configuration / dialplan problem   Mark Muffett
• [asterisk-users] Problems with Tormenta 2 quad card   Tomasz Paszkowski
• [asterisk-users] Fax detection ...   Michelle Dupuis
• [asterisk-users] Trunks and Outbound Routes   Dakota Burns
• [asterisk-users] Call Quality / Echo / Problems   Alexander Lopez
• [asterisk-users] Fax detection ...   James
• [asterisk-users] Polycom Buddy Watch Broken with 2.0.1 Firmware?   Paul Dugas
• [asterisk-users] Re: WiFi SIP handset with Bluetooth required   Sam Tam
• [asterisk-users] Fax detection ...   Steve Underwood
• [asterisk-users] Spying a channel in a meetme   Nicolás Gudiño
• [asterisk-users] Fax detection ...   Jay R. Ashworth
• [asterisk-users] Polycom Buddy Watch Broken with 2.0.1 Firmware?   Douglas Garstang
• [asterisk-users] Polycom Buddy Watch Broken with 2.0.1 Firmware?   Scott Higginbotham
• [asterisk-users] Polycom Buddy Watch Broken with 2.0.1 Firmware?   Alexander Lopez
• [asterisk-users] Polycom Buddy Watch Broken with 2.0.1 Firmware?   Paul Dugas
• [asterisk-users] TDM2400P wiring.   C F
• [asterisk-users] tools/techniques/metrics for measurement of end-point quality   Nilesh Londhe
• [asterisk-users] Configuration / dialplan problem   Kevin Smith
• [asterisk-users] G726 prompts   RR
• [asterisk-users] Polycom Buddy Watch Broken with 2.0.1 Firmware?   Eric Bishop
• [asterisk-users] Asterisk Advice   K Y Iyer
• [asterisk-users] Asterisk Advice   yusuf
• [asterisk-users] Re: Voip Buster - CID   Tomislav Parčina
• [asterisk-users] Re: Voip Buster - CID   Tomislav Parčina
• [asterisk-users] Re: RPID   Tomislav Parčina
• [asterisk-users] tools/techniques/metrics for measurement of end-point quality   Olivier
• [asterisk-users] Polycom Buddy Watch Setup help request   Robert Jenkins
• [asterisk-users] Asterisk Advice   Tim Panton
• [asterisk-users] Fax detection ...   Olivier
• [asterisk-users] ZyXEL desktop ethernet switch for QoS   Janto Dreijer
• [asterisk-users] [ast-users] bridging active channels together   Roberto Sottile
• [asterisk-users] Configuration / dialplan problem   Marco Mouta
• [asterisk-users] pbx call setup to asterisk, behavior context dependend   Christian Gatti
• [asterisk-users] RE: Asterisk Advice   K Y Iyer
• [asterisk-users] Unknown signalling method 'pri_cpe'   Eugeniy Khvastunov
• [asterisk-users] realtimeupdate error   unplug
• [asterisk-users] Unknown signalling method 'pri_cpe'   yusuf
• [asterisk-users] Which IP Phone is good to use at reception desk?   Zeeshan Zakaria
• [asterisk-users] sip provider not working   Jim Lynch
• [asterisk-users] Unknown signalling method 'pri_cpe'   Eugeniy Khvastunov
• [asterisk-users] Defining sip users through mysql   Arkaitz
• [asterisk-users] RE: Asterisk Advice   Cory Andrews
• [asterisk-users] Spying a channel in a meetme   Eduard Martínez
• [asterisk-users] Asterisk manager   Voipers Portugal
• [asterisk-users] Asterisk manager   Voipers Portugal
• [asterisk-users] How to add new codec support?   Rawat Anshuman-cbf738
• [asterisk-users] Asterisk manager   Dean Collins
• [asterisk-users] Asterisk manager   Bill Gibbs
• [asterisk-users] Screen pop based on incoming DID   Greg Delgado
• [asterisk-users] Asterisk manager   Voipers Portugal
• [asterisk-users] Zaphfc woth florz patch   Giordano Grandis
• [asterisk-users] Problems with automon   Clif Jones
• SV: [asterisk-users] Screen pop based on incoming DID   Jon Schøpzinsky
• [asterisk-users] Re: Two phones, same number   Benny Amorsen
• [asterisk-users] Query on Call Parking   Chan Kwang Mien
• [asterisk-users] TDM2400P wiring.   f6hqz-m at hamwlan.net
• [asterisk-users] g729 Codec for AMD Sempron   sdcharly at gmail.com
• SV: [asterisk-users] Screen pop based on incoming DID   Joe Dennick
• [asterisk-users] Bandwidth usage   Steve Glaus
• R: [asterisk-users] Zaphfc woth florz patch   Giordano Grandis
• [asterisk-users] Screen pop based on incoming DID   Rajkumar S
• [asterisk-users] Problems with automon   Michael Neuhauser
• [asterisk-users] Caller ID on Zap not always working   nrbwpi at gmail.com
• [asterisk-users] Polycom Buddy Watch Setup help request   Scott Higginbotham
• [asterisk-users] Cisco 7961 - Presence Example?   Scott Higginbotham
• [asterisk-users] Bandwidth usage   Jay R. Ashworth
• [asterisk-users] TDM2400P wiring.   C F
• [asterisk-users] Caller ID on Zap not always working   Nick Ellson
• [asterisk-users] asterisk to asterisk DID extentions   Matt
• [asterisk-users] Re: Defining sip users through mysql   Arkaitz
• [asterisk-users] o extension for voicemail app   Wing Wong
• [asterisk-users] Spying a channel in a meetme   Michiel van Baak
• [asterisk-users] Bandwidth usage   Lee Howard
• [asterisk-users] Asterisk+Panasonic KX-TDA100+zaphfc NT link problem   MC
• [Asterisk-Users] problem configuring a digium quad E1 card   Paul Philpott
• [asterisk-users] Which IP Phone is good to use at reception desk?   Mojo with Horan & Company, LLC
• R: [asterisk-users] Zaphfc woth florz patch   Henrik Woffinden
• [Asterisk-Users] problem configuring a digium quad E1 card   Jay R. Ashworth
• [asterisk-users] TDM2400P wiring.   Mailing List
• [asterisk-users] asterisk to asterisk DID extentions   Steve Glaus
• [asterisk-users] TDM2400P wiring.   C F
• [asterisk-users] CALEA support within asterisk?   Rich Adamson
• [asterisk-users] CALEA support within asterisk?   J. Oquendo
• [asterisk-users] CALEA support within asterisk?   Jay R. Ashworth
• [asterisk-users] CALEA support within asterisk?   J. Oquendo
• [asterisk-users] uniden uip200 phone hangs any ideas?   Jerry Geis
• [asterisk-users] Problems with automon   Clif Jones
• [asterisk-users] Extremely choppy sound on some of our POTS network calls; goes away with mute   sdgesa gaeharth
• [asterisk-users] newbie: intelligent handling of two ISDN lines (4 B-channels)   Ekkard Gerlach
• [asterisk-users] CALEA support within asterisk?   Jay R. Ashworth
• [asterisk-users] CALEA support within asterisk?   Colin Anderson
• [asterisk-users] CALEA support within asterisk?   Rich Adamson
• [asterisk-users] CALEA support within asterisk?   Matthew Thompson
• [asterisk-users] CALEA support within asterisk?   Brandon Galbraith
• [asterisk-users] CALEA support within asterisk?   Rich Adamson
• [asterisk-users] CALEA support within asterisk?   J. Oquendo
• [asterisk-users] Extremely choppy sound on some of our POTS network calls; goes away with mute   Andres
• [asterisk-users] Caller ID on Zap not always working   phil.dawson at marnock.com
• [asterisk-users] CALEA support within asterisk?   Jay R. Ashworth
• [asterisk-users] CALEA support within asterisk?   Patrick
• [asterisk-users] CALEA support within asterisk?   Jay R. Ashworth
• [asterisk-users] Building the Perfect Box   Conrad Wood
• [asterisk-users] Building a terrorist-friendly telephone network (Was: CALEA support)   Henry J. Cobb
• [asterisk-users] CALEA support within asterisk?   Brandon Galbraith
• [asterisk-users] Building a terrorist-friendly telephone network (Was: CALEA support)   Jay R. Ashworth
• [asterisk-users] Polycom Call Parking   Paul Dugas
• [asterisk-users] Digium TDM or SPA-3000?   R.R. Libera
• [asterisk-users] Building a terrorist-friendly telephone network (Was: CALEA support)   J. Oquendo
• [asterisk-users] CDR stats to one mysql database, multiple webstats packages   Chris Earle
• [asterisk-users] newbie: intelligent handling of two ISDN lines (4 B-channels)   Brian Candler
• [asterisk-users] AstmanProxy Not Collating Manager Info   Douglas Garstang
• [asterisk-users] authenticating forwarded calls   Mark Price
• [asterisk-users] Problems with automon   Michael Neuhauser
• [asterisk-users] Asterisk Directory listing   asterisk-user
• [asterisk-users] Digium TDM or SPA-3000?   Jay R. Ashworth
• [asterisk-users] Digium Interfaces in Tampa?   R.R. Libera
• [asterisk-users] authenticating forwarded calls   Andres
• [asterisk-users] Asterisk Directory listing   James Texter
• [asterisk-users] Polycom Buddy Watch Setup help request   Eric Bishop
• [asterisk-users] Digium TDM or SPA-3000?   Shawn Kelley
• [asterisk-users] Digium Interfaces in Tampa?   Matt Florell
• [asterisk-users] Re: Digium Interfaces in Tampa?   Steven
• [asterisk-users] Polycom Call Parking   Paul Dugas
• [asterisk-users] (no subject)   Jordan Novak
• [asterisk-users] Digium TDM or SPA-3000?   Joseph
• [asterisk-users] (no subject)   Alexander Lopez
• [asterisk-users] Strange problem(Munin-node-1.2.4-7)   Crazy Boy
• [asterisk-users] Call Forwarding not working for extension in queue, why?   Zeeshan Zakaria
• [asterisk-users] Screen pop based on incoming DID   Wolfgang Lumpp
• [asterisk-users] Re: extensions.conf strangeness   Martin Joseph
• [asterisk-users] DISA and legacy PBX   James Harper
• [asterisk-users] IVR for the called part (IVR inside out)   Daniel Cyt
• [asterisk-users] Call Forwarding not working for extension in queue, why?   Dovid B
• [asterisk-users] t1 voip to failover pri   adebayo omo-dare
• [asterisk-users] Call Forwarding not working for extension in queue, why?   Zeeshan Zakaria
• [asterisk-users] Asterisk and Attachment   Giedrius Augys
• [asterisk-users] Digium TDM or SPA-3000?   Thomas Kenyon
• [asterisk-users] Rejecting call   Eugeniy Khvastunov
• [asterisk-users] cisco 2600   Tijl Van den Broeck
• [asterisk-users] Re: Passing Arguments to FastAGI   Tony Mountifield
• [asterisk-users] verbose logging to file in 1.4   Benko
• [asterisk-users] IVR for the called part (IVR inside out)   Mark Phillips
• [asterisk-users] Help in MySQL + Asterisk.   raviprakash sunkara
• [asterisk-users] Help in MySQL + Asterisk.   Brian Rogan
• [asterisk-users] voicemail maintenance questions   Jordan Novak
• [asterisk-users] Re: CDR stats to one mysql database, multiple webstats packages   Tomislav Parčina
• [asterisk-users] Polycom Buddy Watch Setup help request   Scott Higginbotham
• [asterisk-users] Unknown signalling method 'pri_cpe'   Forrest Beck
• [asterisk-users] Polycom Call Parking   Noah Miller
• [asterisk-users] Unknown signalling method 'pri_cpe'   Tzafrir Cohen
• [asterisk-users] voicemail maintenance questions   Mark Phillips
• [asterisk-users] Spandsp and tif   Giedrius Augys
• [asterisk-users] Where is the PlayDTMF command?   Jan du Toit
• [asterisk-users] Where is the PlayDTMF command?   Moises Silva
• [asterisk-users] Strange problem(Munin-node-1.2.4-7)   Steve Glaus
• [asterisk-users] Polycom Call Parking   Kevin Kiely
• [asterisk-users] DISA and legacy PBX   Colin Anderson
• [asterisk-users] Polycom Call Parking   Noah Miller
• [asterisk-users] [Asterisk-Java] SipShowPeerAction   richard Coco
• [asterisk-users] Which IP Phone is good to use at reception desk?   Noah Miller
• [asterisk-users] FOP v.27 IAX trunks not "ringing"   David Cook
• [asterisk-users] Re: Passing Arguments to FastAGI   Douglas Garstang
• [asterisk-users] IVR for the called part (IVR inside out)   whois wes
• [asterisk-users] Spandsp and tif   Steve Underwood
• [asterisk-users] Zaptel problems   Shea, Matt
• [asterisk-users] Call Interception   Delca
• [asterisk-users] Zaptel problems   Colin Anderson
• [asterisk-users] Asterisk 1.4 moh - mohsuggest   Douglas Garstang
• [asterisk-users] Zaptel problems   Bernardo Vieira
• [asterisk-users] Call Interception   Bernardo Vieira
• [asterisk-users] Oneway audio   Giordano Grandis
• [asterisk-users] Digium TDM or SPA-3000?   Jay R. Ashworth
• [asterisk-users] digium compatibility notes   marek cervenka
• [asterisk-users] Call Interception   Jay R. Ashworth
• [asterisk-users] Transfer feature - howto?   Mike
• [asterisk-users] Call Interception   Time Bandit
• [asterisk-users] New tutorial - peering two * servers using IAX   lenz
• [asterisk-users] Call Interception   Steve Edwards
• [asterisk-users] snom 360: how to make record button working ?   noro kamen
• [asterisk-users] Call Interception   Don
• [asterisk-users] SIP client that runs on Linux or Solaris through X Windows?   Joe
• [asterisk-users] Wouldn't Tri-tone detection in Dial() be cool?   Steve Murphy
• [asterisk-users] Where is the PlayDTMF command?   Frank Church
• [asterisk-users] T1 incoming connects, but no sound   Nathan Bell
• [asterisk-users] Intel Chipset 945p compatible?   R.R. Libera
• [asterisk-users] New tutorial - peering two * servers using IAX   Douglas Garstang
• [asterisk-users] Need USA DID + trunk provider   R.R. Libera
• [asterisk-users] Need USA DID + trunk provider   R.R. Libera
• [asterisk-users] Where is the PlayDTMF command?   Mojo with Horan & Company, LLC
• [asterisk-users] digium compatibility notes   Noah Miller
• [asterisk-users] voicemail maintenance questions   Noah Miller
• [asterisk-users] Dialplan Syslog   Douglas Garstang
• [asterisk-users] Newbie question about meetme   omar parihuana
• [asterisk-users] [Asterisk-Java] SipShowPeerAction   Zack Kneisley
• [asterisk-users] Zaptel problems   Shea, Matt
• [asterisk-users] Wouldn't Tri-tone detection in Dial() be cool?   Bernardo Vieira
• [asterisk-users] New tutorial - peering two * servers using IAX   Alex Robar
• [asterisk-users] Need USA DID + trunk provider   Steve Glaus
• [asterisk-users] Transfer feature - howto?   Steve Glaus
• [asterisk-users] Intrado V9-1-1   Marnus van Niekerk
• [asterisk-users] Dialplan Syslog   Kristian Kielhofner
• [asterisk-users] Dialplan Syslog   Colin Anderson
• [asterisk-users] Dialplan Syslog   Time Bandit
• [asterisk-users] unable to call AT&T audio conference bridge   asterisk-user
• [asterisk-users] New tutorial - peering two * servers using IAX   Douglas Garstang
• [asterisk-users] Transfer feature - howto?   Colin Anderson
• [asterisk-users] Spandsp and tif   Giedrius Augys
• [asterisk-users] Voicemail maintenance   Jordan Novak
• [asterisk-users] TNT Max Password reset   Natambu Obleton
• [asterisk-users] Transfer feature - howto?   Mike
• [asterisk-users] UPDATE: Zaptel problems   Shea, Matt
• [asterisk-users] Zaptel problems   Watkins, Bradley
• [asterisk-users] IP Phones   bilal ghayyad
• [asterisk-users] Polycom Call Parking   Kevin Kiely
• [asterisk-users] unable to call AT&T audio conference bridge   BJ Weschke
• [asterisk-users] Newbie question about meetme   Moises Silva
• [asterisk-users] Where is the PlayDTMF command?   Moises Silva
• [asterisk-users] Newbie question about meetme   Mojo with Horan & Company, LLC
• [asterisk-users] Dialplan Syslog   Douglas Garstang
• [asterisk-users] Transfer feature - howto?   Colin Anderson
• [asterisk-users] Transfer feature - howto?   Steve Glaus
• [asterisk-users] Video Conference   bilal ghayyad
• [asterisk-users] IP Phones   Doug
• [asterisk-users] SIP client that runs on Linux or Solaris through X Windows?   Tim Panton
• [asterisk-users] How to make RTP does not go thru asterisk server   Anuj Jain
• [asterisk-users] IP Phones   Steve Glaus
• [asterisk-users] MODEM (data) througt asterisk ?   ABC- Florent BARBIER
• [asterisk-users] [Asterisk-Java] SipShowPeerAction   Richard OSS
• [asterisk-users] Calling Functions from AEL2   Douglas Garstang
• [asterisk-users] Dialplan Syslog   Steve Edwards
• [asterisk-users] no callerid from PSTN using TDM2400P   Naija Man
• [asterisk-users] TNT Max Password reset   Jay R. Ashworth
• [asterisk-users] [Asterisk-Java] SipShowPeerAction   Tim Panton
• [asterisk-users] help   Derek Kruger
• [asterisk-users] Re: [asteisk-users]USA DID + trunk   Steve Glaus
• [asterisk-users] snom 360: how to make record button working ?   Joel Hill
• [asterisk-users] snom 360 - how to make record button working ?   noro kamen
• [asterisk-users] How to make RTP does not go thru asterisk server   Anuj Jain
• [asterisk-users] T38Passthrough and 1.4 Beta   David Hindmarsh
• [asterisk-users] Re: [asteisk-users]USA DID + trunk   Steve Glaus
• [asterisk-users] Polycom Call Parking   Noah Miller
• [asterisk-users] Bandwidth requirements   hugolivude
• [asterisk-users] How to make RTP does not go thru asterisk server   Mojo with Horan & Company, LLC
• [asterisk-users] Where is the PlayDTMF command?   Frank Church
• [asterisk-users] Possible to set max voicemail mesasge limit per user   Christopher Aloi
• [asterisk-users] no callerid from PSTN using TDM2400P   Eric "ManxPower" Wieling
• [asterisk-users] Transfer feature - howto?   Eric "ManxPower" Wieling
• [asterisk-users] ZAP chanel doesn't reset if external caller hangs up in menu   Devraj Mukherjee
• [asterisk-users] Unknown signalling method 'pri_cpe'   Eric "ManxPower" Wieling
• [asterisk-users] Clipcomm CG 410 / FXO Gateway   Barry Fawthrop
• [asterisk-users] Bandwidth requirements   hugolivude
• [asterisk-users] asterisk to asterisk DID extentions   Paul Hales
• [asterisk-users] Bandwidth requirements   Steve Glaus
• [asterisk-users] IP Phones   Devraj Mukherjee
• [asterisk-users] Which IP Phone is good to use at reception desk?   Zeeshan Zakaria
• [asterisk-users] Clipcomm CG 410 / FXO Gateway   Cory Andrews
• [asterisk-users] Wouldn't Tri-tone detection in Dial() be cool?   Jay R. Ashworth
• [asterisk-users] asterisk to asterisk DID extentions   Avi Miller
• [asterisk-users] MODEM (data) througt asterisk ?   Jay R. Ashworth
• [asterisk-users] Polycom Buddy Watch Broken with 2.0.1 Firmware?   Anthony Rodgers
• [asterisk-users] Re: Calling Functions from AEL2   Steve Murphy
• [asterisk-users] New tutorial - peering two * servers using IAX   Crazy Boy
• [asterisk-users] AEL2 #include madness in Asterisk 1.4 - Murf?   Douglas Garstang
• [asterisk-users] no callerid from PSTN using TDM2400P   Crazy Boy
• [asterisk-users] Re: Calling Functions from AEL2   Douglas Garstang
• [asterisk-users] snom 360: how to make record button working ?   Remco Barendse
• [asterisk-users] echo cancellation on hard phones   Rosli Sukri
• [asterisk-users] snom 360: how to make record button working ?   Joel Hill
• [asterisk-users] CDR problem with call transfer   Kamran Ahmad
• [asterisk-users] asterisk-addons-1.2.4 Installation Problem   Abdul
• [asterisk-users] asterisk-addons-1.2.4 Installation Problem   yusuf
• [asterisk-users] asterisk-addons-1.2.4 Installation Problem   Avi Miller
• [asterisk-users] fonality acquires trixbox (asterisk@home) ?   Zoa
• [asterisk-users] New tutorial - peering two * servers using IAX   lenz
• [asterisk-users] Spandsp and tif   DRi at b-w-computer.de
• [asterisk-users] T38Passthrough and 1.4 Beta   Brian Candler
• [asterisk-users] Bandwidth requirements   Erik
• [asterisk-users] snom 360: how to make record button working ?   Remco Barendse
• [asterisk-users] "set verbose 4" in SVN trunk?   Brian Candler
• [asterisk-users] "set verbose 4" in SVN trunk?   Pryakhin Dimitry
• [asterisk-users] "set verbose 4" in SVN trunk?   Pryakhin Dimitry
• [asterisk-users] answering machine detection   Raj
• [asterisk-users] [Asterisk-Java] SipShowPeerAction   richard Coco
• [asterisk-users] Bandwidth requirements   Brian Candler
• [asterisk-users] spandsp logs?   Francesco Francesconi
• [asterisk-users] Re: Bandwidth requirements   Benny Amorsen
• [asterisk-users] "set verbose 4" in SVN trunk?   Brian Candler
• [asterisk-users] Silience on random calls   Pryakhin Dimitry
• [asterisk-users] Bandwidth requirements   raphael Jacquot
• [asterisk-users] Problems with Dial In - Dial Out via SIP - no voice   Christian Peter
• [asterisk-users] Problems with Dial In - Dial Out via SIP - no voice   Christian Peter
• [asterisk-users] TNT Max Password reset   Steve Kennedy
• [asterisk-users] Message count requested for mailbox 9002@from-sip but voicemail not loaded.   raviprakash sunkara
• [asterisk-users] Message count requested for mailbox 9002@from-sip but voicemail not loaded.   Francesco Francesconi
• [asterisk-users] Re: extensions.conf strangeness   Brian Candler
• [asterisk-users] answering machine detection   Matt Florell
• [asterisk-users] snom 360: how to make record button working ?   sip
• [asterisk-users] no callerid from PSTN using TDM2400P   Bob Chiodini
• [asterisk-users] New Version of "Tycho" Voicemail Manager released   Arnd Vehling
• [asterisk-users] Extremely choppy sound on some of our POTS network calls; goes away with mute   sdgesa gaeharth
• [asterisk-users] Extremely choppy sound on some of our POTSnetwork calls; goes away with mute   Andrew Shelton
• [asterisk-users] India:Reliance - E1configuration using TE110P   Rajkumar S
• [asterisk-users] Call Center requirements   Todd Houle Asterisk
• [asterisk-users] GXP - 2000 BLF   Andrew Shelton
• [asterisk-users] Re: AEL2 #include madness in Asterisk 1.4 - Murf?   Steve Murphy
• [asterisk-users] Re: Bandwidth requirements   Benny Amorsen
• [asterisk-users] Re: Bandwidth requirements   J. Oquendo
• [asterisk-users] Extremely choppy sound on some of our POTSnetwork calls; goes away with mute   sdgesa gaeharth
• [asterisk-users] snom 360: how to make record button working ?   Joe Pukepail
• [asterisk-users] TNT Max Password reset   James
• [asterisk-users] two asterisk and one NBX system   jose diaz
• [asterisk-users] Video Conference   Noah Miller
• [asterisk-users] TNT Max Password reset   Don
• [asterisk-users] TNT Max Password reset   asterisk
• [asterisk-users] Re: extensions.conf strangeness   Michael Neuhauser
• [asterisk-users] Extremely choppy sound on some of ourPOTSnetwork calls; goes away with mute   Andrew Shelton
• [asterisk-users] snom 360: how to make record button working ?   noro kamen
• [asterisk-users] two asterisk and one NBX system   Andrew Shelton
• [asterisk-users] AW: asterisk-users Digest, Vol 27, Issue 23   Daniel Hikel
• [asterisk-users] for god's sake somebody help me! ANSWEREDTIME=0 in astcc!!   Ali
• [asterisk-users] Where is the PlayDTMF command?   Moises Silva
• [asterisk-users] New tutorial - peering two * servers using IAX   lenz
• [asterisk-users] Re: Bandwidth requirements   Matthew Crocker
• [asterisk-users] two asterisk and one NBX system   Moises Silva
• [asterisk-users] Detecting Busy AGI Extensions?   Matthew Rubenstein
• [asterisk-users]FIX FOUND: Zaptel problems   Shea, Matt
• [asterisk-users] MODEM (data) througt asterisk ?   Time Bandit
• [asterisk-users] Asterisk integration   Brian Becker
• [asterisk-users] Video Conference   Dean Collins
• [asterisk-users] TNT Max Password reset   Jay R. Ashworth
• [asterisk-users] TNT Max Password reset   Natambu Obleton
• [asterisk-users] Re:UPDATE: Zaptel problems   mavince at optonline.net
• [asterisk-users] TNT Max Password reset   Jay R. Ashworth
• [asterisk-users] Asterisk manager   Michel Vaillancourt
• [asterisk-users] no callerid from PSTN using TDM2400P   Jay R. Ashworth
• [asterisk-users] snom 360: how to make record button working ?   Jay R. Ashworth
• [asterisk-users] cisco 2600   Gary Richardson
• [asterisk-users] Call Center requirements   BJ Weschke
• [asterisk-users] Dial out trhough a FXS channel on a TDM card   Robson Ribeiro
• [asterisk-users] Extremely choppy sound on some of ourPOTSnetwork calls; goes away with mute   Noah Miller
• [asterisk-users] Bandwidth requirements   Jay R. Ashworth
• [asterisk-users] Calling Functions from AEL2   Rushowr
• [asterisk-users] AEL2 Catching on?   Rushowr
• [asterisk-users] two asterisk and one NBX system   jose diaz
• [asterisk-users] TNT Max Password reset   Natambu Obleton
• [asterisk-users] Wouldn't Tri-tone detection in Dial() be cool?   Richard Lyman
• [asterisk-users] snom 360: how to make record button working ?   noro kamen
• [asterisk-users] sdsl   adebayo omo-dare
• [asterisk-users] How to make RTP does not go thru asterisk server   Mojo with Horan & Company, LLC
• [asterisk-users] Asterisk Server : IDE HDD frequent crash   Dushyanth
• [asterisk-users] CDR problem with call transfer   Mojo with Horan & Company, LLC
• [asterisk-users] fonality acquires trixbox (asterisk@home) ?   shadowym
• [asterisk-users] Getting Asterisk to work with GoogleTalk   Alvin Austin
• [asterisk-users] TNT Max Password reset   Ejay Hire
• [asterisk-users] we are having trouble detecting the # for making a transfer from an E1, usually under some load, please help   MF
• [asterisk-users] TNT Max Password reset   James
• [asterisk-users] Bandwidth requirements   omar parihuana
• [asterisk-users] pop a web page with DID in url   Michael Sampson
• [asterisk-users] Bandwidth requirements   Brian Candler
• [asterisk-users] Call Center requirements   John Novack
• [asterisk-users] "set verbose 4" in SVN trunk?   John Novack
• [asterisk-users] Video Conference   Noah Miller
• [asterisk-users] OT: Polycom time sync - sorta   Dave Fullerton
• [asterisk-users] PoE IP Phone   bilal ghayyad
• [asterisk-users] Help with gdb bt full results   Scott Higginbotham
• [asterisk-users] Problem with 2 machines connected with IAX   Matt
• [asterisk-users] Transfer feature - howto?   Steve Glaus
• [asterisk-users] No voice for when using Playback and background   Rajkumar S
• [asterisk-users] Bandwidth requirements   hugolivude
• [asterisk-users] pop a web page with DID in url   Time Bandit
• [asterisk-users] Video Conference   Dean Collins
• [asterisk-users] PoE IP Phone   J. Oquendo
• [asterisk-users] PoE IP Phone   Dave Cotton
• [asterisk-users] Polycom Call Parking   Mojo with Horan & Company, LLC
• [asterisk-users] PoE IP Phone   Joe Dennick
• [asterisk-users] SIP client that runs on Linux or Solaris through X Windows?   Joe
• [asterisk-users] Dial without phone   Mir
• [asterisk-users] Problem with 2 machines connected with IAX   Lenz
• [asterisk-users] Codes negotiation problems between Asterisk 1.4beta2 and Aastra 480i   Morten Isaksen
• [asterisk-users] PoE IP Phone   Steve Glaus
• [asterisk-users] PoE IP Phone   Andrew Joakimsen
• [asterisk-users] RE: Getting Asterisk to work with GoogleTalk   Bromont -
• [asterisk-users] ACK when using SIP Proxy   H Quintana
• [asterisk-users] AGI PHP   Thiago Maluf
• [asterisk-users] Re: PoE IP Phone   Benny Amorsen
• [asterisk-users] PoE IP Phone   Ejay Hire
• [asterisk-users] Re: Bandwidth requirements   Dan Austin
• [asterisk-users] Transfer feature - howto?   Eric "ManxPower" Wieling
• [asterisk-users] OT: Polycom time sync - sorta   Bill Gibbs
• [Asterisk-Users] How to create echo for learning purpose   Olivier
• [Asterisk-Users] How to create echo for learning purpose   Olivier
• [asterisk-users] No voice for when using Playback and background   Mojo with Horan & Company, LLC
• AW: [asterisk-users] PoE IP Phone   Christian Stredicke
• [asterisk-users] PoE IP Phone   wendys at tiscali.de
• [asterisk-users] DOA IAXy?   Erik Anderson
• [asterisk-users] PRI TON/pridialplan digit prefixing   Jesse Peterson
• [asterisk-users] Asterisk@Home problems   Edward0219 at aol.com
• [asterisk-users] IVR menu system external database information collection   Jordan Novak
• [asterisk-users] OT: Polycom time sync - sorta   Noah Miller
• [asterisk-users] PoE IP Phone   Josh Reineke
• [asterisk-users] failed registration   stan ford
• [asterisk-users] Codes negotiation problems between Asterisk1.4beta2 and Aastra 480i   Gareth Owen
• [asterisk-users] IAX or SIP termination provider that reaches6421xxxxxxx?   Mojo with Horan & Company, LLC
• [asterisk-users] IAX or SIP termination provider that reaches6421xxxxxxx?   Mojo with Horan & Company, LLC
• [asterisk-users] snom 360: how to make record button working ?   Mojo with Horan & Company, LLC
• [asterisk-users] Re: Bandwidth requirements   omar parihuana
• [asterisk-users] No Dialtone   Eddie Johnson Jr
• [asterisk-users] odd muting issue   Jason
• [asterisk-users] Spelling   Jay R. Ashworth
• [asterisk-users] odd muting issue   Jason
• [asterisk-users] OT: Polycom time sync - sorta   Jay R. Ashworth
• [asterisk-users] Asterisk@Home problems   Lacy Moore - Aspendora
• [asterisk-users] Re: PoE IP Phone   Jay R. Ashworth
• [asterisk-users] OT: Polycom time sync - sorta   Kristian Kielhofner
• [asterisk-users] Problem with 2 machines connected with IAX   Matt
• [asterisk-users] Problem with 2 machines connected with IAX   Matt
• [asterisk-users] Optus PRI via DSL   mattf at rapidnet.net.au
• [asterisk-users] RE: Getting Asterisk to work with GoogleTalk   Alvin Austin
• [asterisk-users] Optus PRI via DSL   mattf at rapidnet.net.au
• [asterisk-users] Newbie h/w Q, and confirming basic concepts   Mike Morris
• [asterisk-users] Asterisk@Home problems   William Piper
• [asterisk-users] Re: failed registration   stan ford
• [asterisk-users] Newbie h/w Q, and confirming basic concepts   Dovid B
• [asterisk-users] different dialtones for DISA   James Harper
• [asterisk-users] Re: failed registration   Dovid B
• Fwd: [asterisk-users] No voice for when using Playback and background   Rajkumar S
• [asterisk-users] Asterisk Server : IDE HDD frequent crash   Stuart Sheldon
• [asterisk-users] Asterisk Server : IDE HDD frequent crash   Sam Norris
• [asterisk-users] Asterisk CDR   Rizwan Hisham
• [asterisk-users] Re: PoE IP Phone   Lacy Moore - Aspendora
• [asterisk-users] Re: Where is the PlayDTMF command?   Jan du Toit
• [asterisk-users] Where is the PlayDTMF command?   Jan du Toit
• [asterisk-users] DOA IAXy?   bails
• [asterisk-users] Asterisk Configuration Complete Newbie question   K Y Iyer
• [asterisk-users] asterisk gui sans live cd   Patrick Aljord
• [asterisk-users] Re: OT: Polycom time sync - sorta   Benny Amorsen
• [asterisk-users] snom 360: how to make record button working ?   Remco Barendse
• [asterisk-users] New Asterisk StumbleUpon Group   Matt Riddell (IT)
• [asterisk-users] Asterisk Configuration Complete Newbie question   Lacy Moore - Aspendora
• [asterisk-users] Asterisk Configuration Complete Newbie question   K Y Iyer
• [asterisk-users] Re: extensions.conf strangeness   Brian Candler
• [asterisk-users] Newbie h/w Q, and confirming basic concepts   Brian Candler
• [asterisk-users] 2x* and realtime   Marnus van Niekerk
• [asterisk-users] How to forward DID to another Server   ram
• [asterisk-users] Asterisk act as a proxy ?   ram
• [asterisk-users] snom 360: how to make record button working ?   Terry Wade
• [asterisk-users] Re: PoE IP Phone   Tech Support
• [asterisk-users] Asterisk act as a proxy ?   Peter Bowyer
• [asterisk-users] Newbie h/w Q, and confirming basic concepts   John Novack
• [asterisk-users] Re: verbose logging to file in 1.4   Benko
• [asterisk-users] no callerid from PSTN using TDM2400P   Naija Man
• [asterisk-users] IP Phones   Forrest Beck
• [asterisk-users] IP Phones   jose diaz
• [asterisk-users] How to forward DID to another Server   Noah Miller
• [asterisk-users] Asterisk Server : IDE HDD frequent crash   Jay R. Ashworth
• [asterisk-users] Re: Where is the PlayDTMF command?   Jay R. Ashworth
• [asterisk-users] Asterisk Server : IDE HDD frequent crash   Matthew Rubenstein
• [asterisk-users] No Dialtone   Noah Miller
• [asterisk-users] Call Center requirements   Stephen Wingfield
• [asterisk-users] No Dialtone   Francesco Francesconi
• [asterisk-users] IP Phones   K Y Iyer
• [asterisk-users] Codes negotiation problems between Asterisk1.4beta2 and Aastra 480i   Morten Isaksen
• [asterisk-users] Asterisk Hangups on PRI Interface   Giorgio Incantalupo
• [asterisk-users] Tutorial - avoiding queue_log file rotation   Lenz
• [asterisk-users] defining trunks in sip.conf   Joao Pereira
• [asterisk-users] Tutorial - avoiding queue_log file rotation   Lenz
• [asterisk-users] asterisk gui sans live cd   Arnd Vehling
• [asterisk-users] Voicemail and Forwarding   Forrest Beck
• [asterisk-users] Asterisk@Home problems   Edward0219 at aol.com
• [asterisk-users] Asterisk@Home problems   Edward0219 at aol.com
• [asterisk-users] Re: Voicemail and Forwarding   Forrest Beck
• [asterisk-users] Voicemail and Forwarding   Noah Miller
• [asterisk-users] No Dialtone   Eddie Johnson Jr
• [asterisk-users] No Dialtone   Eddie Johnson Jr
• [asterisk-users] No Dialtone   Noah Miller
• [asterisk-users] Codes negotiation problems between Asterisk1.4beta2 and Aastra 480i   Marco Mouta
• [asterisk-users] Asterisk@Home problems   Alex Robar
• [asterisk-users] snom 360: how to make record button working ?   Mojo with Horan & Company, LLC
• [asterisk-users] How to forward DID to another Server   Douglas Garstang
• AW: [asterisk-users] PoE IP Phone   Michiel van Baak
• [asterisk-users] RE: Getting Asterisk to work with GoogleTalk   Robert LaPoint
• [asterisk-users] Asterisk RT on Disk On Module Performance and Durability   Erick Perez
• [asterisk-users] Problem with 2 machines connected with IAX   Matt
• [asterisk-users] pop a web page with DID in url   Michael Sampson
• [asterisk-users] Extremely choppy sound on some of ourPOTSnetwork calls; goes away with mute   sdgesa gaeharth
• [asterisk-users] Match & Chat Author?   Bart Fisher
• [asterisk-users] Voicemail MWI   Douglas Garstang
• [asterisk-users] SIP client that runs on Linux or Solaris through X Windows?   Tim Panton
• [asterisk-users] pop a web page with DID in url   Tim Panton
• [asterisk-users] Re: failed registration   stan ford
• [asterisk-users] Asterisk RT on Disk On Module Performance and Durability   Jeremy McNamara
• [asterisk-users] Re: verbose logging to file in 1.4   Brian Candler
• [asterisk-users] Asterisk RT on Disk On Module Performance and Durability   Kristian Kielhofner
• [asterisk-users] Voicemail MWI   Aaron Daniel
• [asterisk-users] Re: Asterisk Server : IDE HDD frequent crash   Martin Joseph
• [asterisk-users] Asterisk Postgres Native support   John Miloo
• [asterisk-users] Asterisk RT on Disk On Module Performance and Durability   Kristian Kielhofner
• [asterisk-users] Extremely choppy sound on some of ourPOTSnetwork calls; goes away with mute   Noah Miller
• [asterisk-users] AGI() in 1.2 and 1.4   Douglas Garstang
• [asterisk-users] AGI() in 1.2 and 1.4   Steve Totaro
• [asterisk-users] Re: Where is the PlayDTMF command?   Benny Amorsen
• [asterisk-users] swap CID with DID   Michael Sampson
• [asterisk-users] ChanIsAvail() in 1.2.12.1   Douglas Garstang
• [asterisk-users] swap CID with DID   Alex Robar
• [asterisk-users] ChanIsAvail() in 1.2.12.1   Julian Lyndon-Smith
• [asterisk-users] ChanIsAvail() in 1.2.12.1   Douglas Garstang
• [asterisk-users] Re: Asterisk Server : IDE HDD frequent crash   Jay R. Ashworth
• [asterisk-users] snom 360: how to make record button working ?   Jay R. Ashworth
• [asterisk-users] Asterisk RT on Disk On Module Performance and Durability   Erick Perez
• [asterisk-users] Odd echo issue with speaker phone   BerkHolz, Steven
• [asterisk-users] Does a "HST Saphir III ML PCI" work with Asterisk?   James Harper
• [asterisk-users] Codes negotiation problems betweenAsterisk1.4beta2 and Aastra 480i   Gareth Owen
• [asterisk-users] commercial asterisk   stan ford
• [asterisk-users] regexten & regcontext broken for SIP?   Philipp von Klitzing
• [asterisk-users] Asterisk access Postgres for Realtime Configuration   John Miloo
• [asterisk-users] Asterisk RT on Disk On Module Performance and Durability   Michael Graves
• [asterisk-users] A Call centre module on Asterisk   Imed Imed
• [asterisk-users] HTTP Connection Closed on 7960 SIP   Robert Goodyear
• [asterisk-users] HTTP Connection Closed on 7960 SIP   Aaron Daniel
• [asterisk-users] Options for moving to * friendly Business VSP   Al Stery
• [asterisk-users] Options for moving to * friendly Business VSP   Al Stery
• [asterisk-users] A Call centre module on Asterisk   Marnus van Niekerk
• [asterisk-users] astcc help-pleasssssseeee   Ali
• [asterisk-users] Re: AGI() in 1.2 and 1.4   Tony Mountifield
• [asterisk-users] SIP stuck channel soft hangup?   Martin Joseph
• [asterisk-users] pop a web page with DID in url   Lenz
• [asterisk-users] AEL2 Catching on?   Doug Lytle
• [asterisk-users] Outbound FXO call, getting "You must first dial..."   Nick Ellson
• [asterisk-users] astcc help-pleasssssseeee   Tzafrir Cohen
• [asterisk-users] Extremely choppy sound on some of ourPOTSnetwork calls; goes away with mute   Rich Adamson
• [asterisk-users] Outbound FXO call, getting "You must first dial..."   Rich Adamson
• [asterisk-users] Outbound FXO call, getting "You must first dial..."   Nick Ellson
• [asterisk-users] No Dialtone   Rich Adamson
• [asterisk-users] G729 Licence Consumption Problem   Alvaro Parres
• [asterisk-users] No Dialtone   Jay R. Ashworth
• [asterisk-users] No Dialtone   Eddie Johnson Jr
• [asterisk-users] No Dialtone   Eddie Johnson Jr
• [asterisk-users] SIP client that runs on Linux or Solaris through X Windows?   Tzafrir Cohen
• [asterisk-users] HTTP Connection Closed on 7960 SIP   Robert Goodyear
• [asterisk-users] Requirements for Asterisk & SER integration   Crazy Boy
• [asterisk-users] Voicemail maintenance   Tzafrir Cohen
• [asterisk-users] Requirements for Asterisk & SER integratio   Vamsi Pottangi
• [asterisk-users] ChanIsAvail() in 1.2.12.1   Thomas Winter
• [asterisk-users] Xorcom Astribank and 64 bit linux   Carlos Chavez
• [asterisk-users] Asterisk CDR   Tzafrir Cohen
• [asterisk-users] disabling hardware echo can on tdm2400p   Sean Kennedy
• [asterisk-users] Asterisk: Can anybody forward anybody's extension?   Zeeshan Zakaria
• [asterisk-users] pop a web page with DID in url   Nicolás Gudiño
• [asterisk-users] No Dialtone   Rich Adamson
• [asterisk-users] Real-time and priority "n"   Ronald Wiplinger
• [asterisk-users] No Dialtone   Eddie Johnson Jr
• [asterisk-users] ftp server   Dean Collins
• [asterisk-users] polycom auto cfg file   Dean Collins
• [asterisk-users] astcc help-pleasssssseeee   Ali
• [asterisk-users] New Asterisk StumbleUpon Group   f6hqz-m at hamwlan.net
• [asterisk-users] Re: Real-time and priority "n"   Tony Mountifield
• [asterisk-users] RE: ftp server   Dean Collins
• [asterisk-users] How can i store PAP2 or any device config in Asterisk   ram
• [asterisk-users] Xorcom Astribank and 64 bit linux   Tzafrir Cohen
• Fwd: [asterisk-users] How can i store PAP2 or any device config in Asterisk   Yair Hakak
• [asterisk-users] Asterisk RT on Disk On Module Performance and Durability   Brian Candler
• [asterisk-users] Transfer app and DTMF via SIP info   Michael Konietzny
• [asterisk-users] G729 Licence Consumption Problem   Thomas Kenyon
• [asterisk-users] How can i store PAP2 or any device config in Asterisk   ram
• [asterisk-users] ftp server   Avi Miller
• [asterisk-users] Sun Cluster and Asterisk   Giedrius Augys
• [asterisk-users] Optus PRI via DSL   Paul Hales
• [asterisk-users] ZAP chanel doesn't reset if external caller hangs up in menu   Tzafrir Cohen
• [asterisk-users] astcc help-pleasssssseeee   Brian Rogan
• [asterisk-users] OH323 Fake ring   Abdul
• [asterisk-users] DID is not working (call is not routing)   Crazy Boy
• [asterisk-users] Xorcom Astribank and 64 bit linux   Carlos Chavez
• [asterisk-users] Configuration / dialplan problem   Mark Muffett
• [asterisk-users] USA Origination recommended service?   R.R. Libera
• [asterisk-users] PRI issues   Doug Lytle
• [asterisk-users] Tellabs and a PRI   Doug Lytle
• [asterisk-users] ftp server   Gary Eck
• [asterisk-users] ftp server   Dean Collins
• [asterisk-users] ftp server   Gary Eck
• [asterisk-users] ftp server   Dean Collins
• [asterisk-users] No Dialtone   Jay R. Ashworth
• [asterisk-users] ftp server   Gary Eck
• [asterisk-users] Re: Real-time and priority "n"   Brian Capouch
• [asterisk-users] No Dialtone   Eddie Johnson Jr
• [asterisk-users] ftp server   Robert Goodyear
• [asterisk-users] ftp server   Noah Miller
• [asterisk-users] 2x* and realtime   Rushowr
• [asterisk-users] disabling hardware echo can on tdm2400p   Eric "ManxPower" Wieling
• [asterisk-users] Tellabs and a PRI   Eric "ManxPower" Wieling
• [asterisk-users] Tellabs and a PRI   Doug Lytle
• [asterisk-users] ftp server   Tzafrir Cohen
• [asterisk-users] New Asterisk StumbleUpon Group   Matt Riddell (IT)
• [asterisk-users] ftp server   Avi Miller
• [asterisk-users] Optus PRI via DSL   Avi Miller
• [asterisk-users] ftp server   Robert Goodyear
• [asterisk-users] Asterisk RT on Disk On Module Performance and Durability   Tzafrir Cohen
• [asterisk-users] ftp server   Dovid B
• [asterisk-users] 2x* and realtime   Dovid B
• [asterisk-users] Asterisk RT on Disk On Module Performance and Durability   Jeremy McNamara
• [asterisk-users] Blacklist to check http://whocalled.us   Noah Swint
• [asterisk-users] ftp server   Dean Collins
• [asterisk-users] Re: Real-time and priority "n"   Ronald Wiplinger
• [asterisk-users] SIP vs. SIP-B   Dovid B
• [asterisk-users] polycom reboot script   Dean Collins
• [asterisk-users] Asterisk RT on Disk On Module Performance and Durability   Kristian Kielhofner
• [asterisk-users] DID is not working (call is not routing)   William Piper
• [asterisk-users] ftp server   Avi Miller
• [asterisk-users] polycom reboot script   Avi Miller
• [asterisk-users] TDM22B   Jack Morgan
• [asterisk-users] How to make this easier   Tom Vile
• [asterisk-users] Asterisk Load balancing   jk
• [asterisk-users] How to make this easier   James Jones
• [asterisk-users] Asterisk Load balancing   James Jones
• [asterisk-users] question about astdb   unplug
• [asterisk-users] CDR - mysql with asterisk 1.2.12 not working   Joseph
• [asterisk-users] CDR - mysql with asterisk 1.2.12 not working   Hermann Wecke
• [asterisk-users] SIP stuck channel soft hangup?   Nic Bellamy
• [asterisk-users] Incoming sip line with INX (internationalnumber.com)   Daniel Cyt
• [asterisk-users] CDR - mysql with asterisk 1.2.12 not working   Joseph
• [asterisk-users] CDR - mysql with asterisk 1.2.12 not working   Hermann Wecke
• [asterisk-users] CDR - mysql with asterisk 1.2.12 not working   Joseph
• [asterisk-users] Polycom 601 & Expansion Module: Light the LEDs???   Doug
• [asterisk-users] password for vm users   stan ford
• [asterisk-users] DID is not working (call is not routing)   Crazy Boy
• [asterisk-users] Incoming sip line with INX (internationalnumber.com)   Peter Bowyer
• [asterisk-users] Asterisk 1.2.12 - Can NOT make call out / Asterisk terminate   Joseph
• [asterisk-users] Asterisk 1.2.12 - Can NOT make call out / Asterisk terminate   Peter Bowyer
• [asterisk-users] sip provider not working   sdcharly at gmail.com
• [asterisk-users] Asterisk 1.2.12 - Can NOT make call out / Asterisk terminate   Dave Cotton
• [asterisk-users] Redefinition of transfer   john
• [asterisk-users] Asterisk 1.2.12 - Can NOT make call out / Asterisk terminate   Joseph
• [asterisk-users] Beronet card strange log messages   Giorgio Incantalupo
• [asterisk-users] Asterisk 1.2.12 - Can NOT make call out / Asterisk terminate   Peter Bowyer
• [asterisk-users] Incoming sip line with INX(internationalnumber.com)   Daniel Cyt
• [asterisk-users] Incoming sip line with INX(internationalnumber.com)   Peter Bowyer
• [asterisk-users] Incoming sip line withINX(internationalnumber.com)   Daniel Cyt
• [asterisk-users] Re: Where is the PlayDTMF command?   Jan du Toit
• [asterisk-users] password for vm users   Marco Mouta
• [asterisk-users] SIP vz IAX...   raviprakash sunkara
• [asterisk-users] Re: Where is the PlayDTMF command?   Jan du Toit
• [asterisk-users] Odd echo issue with speaker phone   BerkHolz, Steven
• [asterisk-users] Re: PRI issues   Steven
• [asterisk-users] Odd echo issue with speaker phone   Doug Lytle
• [asterisk-users] Re: Outbound FXO call, getting "You must firstdial..."   Steven
• [asterisk-users] Get user context from dialplan.   José Luis Gómez
• [asterisk-users] Re: Odd echo issue with speaker phone   Steven
• [asterisk-users] Odd echo issue with speaker phone   Doug Lytle
• [asterisk-users] Polycom 601 & Expansion Module: Light the LEDs???   Jerry Jones
• [asterisk-users] Re: Asterisk 1.2.12 - Can NOT make call out / Asterisk terminate   Benny Amorsen
• [asterisk-users] Options for moving to * friendly Business VSP   Gleim, Jason
• [asterisk-users] DID is not working (call is not routing)   William Piper
• [asterisk-users] Blacklist to check http://whocalled.us   Michelle Dupuis
• [asterisk-users] H323 <-> SIP   tlott at gmx.net
• [asterisk-users] AstriCon Dallas in Two Weeks   Steven Sokol
• [asterisk-users] How to make this easier   Tom Vile
• [asterisk-users] AstriCon Dallas in Two Weeks   Dean Collins
• [asterisk-users] Lots and lots of log files   Ejay Hire
• [asterisk-users] how to play background music   ram
• [asterisk-users] G729 Licence Consumption Problem   Alvaro Parres
• [asterisk-users] Lots and lots of log files   J. Oquendo
• [asterisk-users] Lots and lots of log files   J. Oquendo
• [asterisk-users] Re: PRI issues   mavince at optonline.net
• [asterisk-users] isdn cross-over ...   Cesc
• [asterisk-users] VOIP with PSTN backup   Brian Candler
• [asterisk-users] disabling hardware echo can on tdm2400p   Ejay Hire
• [asterisk-users] isdn cross-over ...   Ejay Hire
• [asterisk-users] Re: PRI issues   Doug Lytle
• [asterisk-users] VOIP with PSTN backup   Peter Lindquist
• [asterisk-users] polycom reboot script   Noah Miller
• [asterisk-users] connecting multiple servers with iax - authentication fails   Benko
• [asterisk-users] ftp server   Noah Miller
• [asterisk-users] Re: PRI issues   Jay R. Ashworth
• [asterisk-users] Voicemail MWI   Race Vanderdecken
• [asterisk-users] isdn cross-over ...   Jay R. Ashworth
• [asterisk-users] isdn cross-over ...   Zoa
• [asterisk-users] Range Operator   Douglas Garstang
• [asterisk-users] snom 360: how to make record button working ?   Mojo with Horan & Company, LLC
• [asterisk-users] Polycom 601 & Expansion Module: Light the LEDs???   Faris Raouf
• [asterisk-users] Re: connecting multiple servers with iax - authentication fails   Benko
• [asterisk-users] Number Range   Douglas Garstang
• [asterisk-users] Re: PRI issues   Doug Lytle
• [asterisk-users] Lots and lots of log files   Tzafrir Cohen
• [asterisk-users] Asterisk@Home problems   Edward0219 at aol.com
• [asterisk-users] Range Operator   Doug Lytle
• [asterisk-users] External domain   Josemar Lohn
• [asterisk-users] GotoIfTime - much slowdown with 90 conditions?   Mike Dent
• [asterisk-users] Number Range   Ira
• [asterisk-users] Lots and lots of log files   J. Oquendo
• [asterisk-users] Number Range   Doug Lytle
• [asterisk-users] Asterisk@Home problems   Alex Robar
• [asterisk-users] ftp server   Jay R. Ashworth
• [asterisk-users] Cisco 7970 Unbootable After FW Upgrade   Jeremiah Millay
• [asterisk-users] VOIP with PSTN backup   Brian Candler
• [asterisk-users] snom 360: how to make record button working ?   Jay R. Ashworth
• [asterisk-users] isdn cross-over ...   Brian Candler
• [asterisk-users] Cisco 7970 Unbootable After FW Upgrade   Greg Oliver
• [asterisk-users] Range Operator   Eric "ManxPower" Wieling
• [asterisk-users] problem in ooh323   Andre Luiz Martins Rodrigues
• [asterisk-users] Asterisk RT on Disk On Module Performance and Durability   Erick Perez
• [asterisk-users] Asterisk@Home problems   Edward0219 at aol.com
• [asterisk-users] Asterisk RT on Disk On Module Performance andDurability   Douglas Garstang
• [asterisk-users] how to play pre-recorded file in meetme conference   Barry D. Hassler
• [asterisk-users] Asterisk RT on Disk On Module Performance andDurability   Aaron Daniel
• [asterisk-users] password for vm users   stan ford
• [asterisk-users] Asterisk RT on Disk On Module PerformanceandDurability   Douglas Garstang
• [asterisk-users] password for vm users   stan ford
• [asterisk-users] Function ENUMLOOKUP   Douglas Garstang
• [asterisk-users] isdn cross-over ...   Cesc
• [asterisk-users] ftp server   Avi Miller
• [asterisk-users] Function ENUMLOOKUP   Watkins, Bradley
• [asterisk-users] Asterisk RT on Disk On Module Performance andDurability   Erick Perez
• [asterisk-users] password for vm users   Aaron Daniel
• [asterisk-users] Range Operator   Mojo with Horan & Company, LLC
• [asterisk-users] MINNESOTA: TwinCities Asterisk Users Group - Saturday October 14th 2006 - 10:30am   asterisk_help at iwishi.nu
• [asterisk-users] Problem compiling libmfcr2.0.0.2 on Fedora Core 5   Carlos Chavez
• [asterisk-users] Asterisk RT on Disk On Module PerformanceandDurability   Douglas Garstang
• [asterisk-users] GotoIfTime - much slowdown with 90 conditions?   Doug Lytle
• [asterisk-users] Re: PRI issues   Doug Lytle
• [asterisk-users] Home Hardware SIP Proxy with use of POTS in Emergency   Brandon Galbraith
• [asterisk-users] Home Hardware SIP Proxy with use of POTS in Emergency   John Millican
• [asterisk-users] ooh323 error   Andre Luiz Martins Rodrigues
• [asterisk-users] Is there a way to collect dtmf digits during a call? (inband)   MF
• [asterisk-users] Home Hardware SIP Proxy with use of POTS in Emergency   Matt
• [asterisk-users] Error loading Unicall   Carlos Chavez
• [asterisk-users] Monitor Current outgoing calls   George Masgras
• [asterisk-users] Problem compiling libmfcr2.0.0.2 on Fedora Core 5   Moises Silva
• [asterisk-users] Error loading Unicall   Moises Silva
• [asterisk-users] Echo Cancel Cards   Forrest Beck
• [asterisk-users] Error loading Unicall   Carlos Chavez
• [asterisk-users] Asterisk@Home problems   Alex Robar
• [asterisk-users] T1 Passthrough   Forrest Beck
• [asterisk-users] T1 Passthrough   Don Pobanz
• [asterisk-users] Home Hardware SIP Proxy with use of POTS in Emergency   Brandon Galbraith
• [asterisk-users] Monitor Current outgoing calls   William Piper
• [asterisk-users] DID is not working (call is not routing)   Crazy Boy
• [asterisk-users] Asterisk@Home problems   Sharon Lim
• [asterisk-users] Echo Cancel Cards   Joseph
• [asterisk-users] Echo Cancel Cards   Rob Lith
• [asterisk-users] CDR - mysql with asterisk 1.2.12 not working   Garth van Sittert
• [asterisk-users] Kind of OT : Europeans going to Astricon   Stelios Koroneos
• [asterisk-users] Kind of OT : Europeans going to Astricon   Zoa
• [asterisk-users] Re: Echo Cancel Cards   Martin Joseph
• [asterisk-users] Re: Home Hardware SIP Proxy with use of POTS in Emergency   Martin Joseph
• [asterisk-users] Re: Asterisk 1.2.12 - Can NOT make call out / Asterisk terminate   Martin Joseph
• [asterisk-users] Re: SIP stuck channel soft hangup?   Martin Joseph
• [asterisk-users] Asterisk@Home problems   Peter Bowyer
• [asterisk-users] Asterisk on 64bit xeon   Akpome Akpoguma
• [asterisk-users] SIP vz IAX...   Tim Panton
• [asterisk-users] Asterisk RT on Disk On Module Performance andDurability   Tim Panton
• [asterisk-users] connecting multiple servers with iax - authentication fails   Tim Panton
• [asterisk-users] Asterisk on 64bit xeon   Zoa
• [asterisk-users] OT: Hand free solution recommandation   Administrator TOOTAI
• [asterisk-users] Asterisk on 64bit xeon   Akpome Akpoguma
• [asterisk-users] Lots and lots of log files   Tzafrir Cohen
• [asterisk-users] Asterisk on 64bit xeon   Tzafrir Cohen
• [asterisk-users] Echo Cancel Cards   Thomas Kenyon
• [asterisk-users] single conference, multiple numbers   Mike Williams
• [asterisk-users] single conference, multiple numbers   Dovid B
• [asterisk-users] Echo Cancel Cards   Dovid B
• [asterisk-users] help this....   raviprakash sunkara
• [asterisk-users] single conference, multiple numbers   Tzafrir Cohen
• [asterisk-users] single conference, multiple numbers   Thomas Kenyon
• [asterisk-users] Tutorial: Simple queue and agent debug monitoring   lenz
• [asterisk-users] how to play pre-recorded file in meetme conference   Brian Rogan
• [asterisk-users] CDR - mysql with asterisk 1.2.12 not working   Brian Rogan
• [asterisk-users] single conference, multiple numbers   Brian Rogan
• [asterisk-users] alive check for HA constellation   Sebastian Reitenbach
• [asterisk-users] alive check for HA constellation   Leonardo Silva
• [asterisk-users] single conference, multiple numbers   Mike Williams
• [asterisk-users] whisper paging   Hall, Eric M.
• [asterisk-users] T1 Passthrough   Dennis Walker
• [asterisk-users] OT: Hand free solution recommandation   Time Bandit
• [asterisk-users] Re: Range Operator   Benny Amorsen
• [asterisk-users] Echo Cancel Cards   John Novack
• [asterisk-users] alive check for HA constellation   Sebastian Reitenbach
• [asterisk-users] Echo Cancel Cards   R.R. Libera
• [asterisk-users] Echo Cancel Cards   Bernardo Vieira
• [asterisk-users] Error loading Unicall   Moises Silva
• [asterisk-users] Re: PRI issues   Jay R. Ashworth
• [Asterisk-Users] External Custom Extension Timeout   JD Austin
• [asterisk-users] whisper paging   C F
• [asterisk-users] Echo Cancel Cards   Forrest Beck
• [asterisk-users] FYI - Polycom SoundPoint IP 301 Denial of Service]   Rich Adamson
• [asterisk-users] whisper paging   Douglas Garstang
• [asterisk-users] OT: Hand free solution recommandation   Michelle Dupuis
• [asterisk-users] help this....   Ejay Hire
• [asterisk-users] Echo Cancel Cards   Matthew Thompson
• [asterisk-users] Inbound Callcenter with multiple DIDs   Michael Sampson
• [asterisk-users] Voicemail Press '0'   Douglas Garstang
• [asterisk-users] Xorcom TS-1 and Digium TE110P or TE210P   Morten Isaksen
• [asterisk-users] Asterisk 1.2.12.1 and snom 360 6.2.3 no audio   Holger von Ameln
• [asterisk-users] whisper paging   C F
• [asterisk-users] whisper paging   C F
• [asterisk-users] whisper paging   Douglas Garstang
• [asterisk-users] whisper paging   C F
• [asterisk-users] whisper paging   Joshua Colp
• [asterisk-users] Voicemail Press '0'   Doug Lytle
• [asterisk-users] connecting multiple servers with iax - authentication fails   Joshua Colp
• [asterisk-users] Voicemail Press '0'   Savoy, Kevin - Williston, ND
• [asterisk-users] Voicemail Press '0'   Joshua Colp
• [asterisk-users] Voicemail Press '0'   Douglas Garstang
• [asterisk-users] bristuff problem?   Louis-David Mitterrand
• [asterisk-users] Re: Where is the PlayDTMF command?   Moises Silva
• [asterisk-users] Voicemail Press '0'   Savoy, Kevin - Williston, ND
• [asterisk-users] VOIP with PSTN backup   Rich Adamson
• [asterisk-users] Inbound Callcenter with multiple DIDs   Earl Terwilliger
• [asterisk-users] Voicemail Press '0'   Doug Lytle
• [asterisk-users] T1 Passthrough   Vlad B
• [asterisk-users] Voicemail Press '0'   Eric "ManxPower" Wieling
• [asterisk-users] Connection question...   Danko Miocevic
• [asterisk-users] Mitel 5224/SIP no MWI   Jesse Peterson
• [asterisk-users] sequential Dial() commands   Mark Price
• [asterisk-users] Cisco CCM - Asterisk   Alyed Tzompa
• [asterisk-users] Inbound Callcenter with multiple DIDs   Dovid B
• [asterisk-users] Echo Cancel Cards   Dovid B
• [asterisk-users] Voicemail Press '0'   Douglas Garstang
• [asterisk-users] Inbound Callcenter with multiple DIDs   lenz
• [asterisk-users] sequential Dial() commands   Dave Fullerton
• [asterisk-users] Cisco CCM - Asterisk   end1r at comcast.net
• [asterisk-users] Voicemail Press '0'   Eric "ManxPower" Wieling
• [asterisk-users] sequential Dial() commands   Luki
• [asterisk-users] Cisco CCM - Asterisk   Lacy Moore - Aspendora
• [asterisk-users] Re: Voicemail Press '0'   LJ
• [asterisk-users] 1.4 and slow sound playback   Bill Merriam
• [asterisk-users] transfer from VM to Cell Phone   Mr. Jones
• [asterisk-users] 1.4 and slow sound playback   Brian Rogan
• [asterisk-users] Re: Voicemail Press '0'   Douglas Garstang
• [asterisk-users] Re: Voicemail Press '0'   Douglas Garstang
• [asterisk-users] Increase VoiceMail Messages Recording Gain - Audio Calls are Ok   Marco Mouta
• [asterisk-users] Hangup or busy when the peer answer outgoing calls   Eloy Gomez
• [asterisk-users] sequential Dial() commands   Dovid B
• [asterisk-users] voicemail issue   stan ford
• [asterisk-users] RE: Welcome to the "asterisk-users" mailing list   Issac Simchayof
• [asterisk-users] How big is *your* dialplan??   Steve Murphy
• [asterisk-users] transfer from VM to Cell Phone   Dovid B
• [asterisk-users] How big is *your* dialplan??   Aaron Daniel
• [asterisk-users] How big is *your* dialplan??   Doug Lytle
• [asterisk-users] How big is *your* dialplan??   Dovid B
• [asterisk-users] How big is *your* dialplan??   Dean Collins
• [asterisk-users] Connection question...   Dovid B
• [asterisk-users] Monitor Current outgoing calls   George Masgras
• [asterisk-users] How big is *your* dialplan??   Anders Brander
• [asterisk-users] Looking for AudioCodes 2 port FXS gateway - Asterisk compatability info   LJ
• [asterisk-users] Destar 0.2.0 released   Alejandro Ríos P.
• [asterisk-users] How big is *your* dialplan??   Justin Tunney
• [asterisk-users] How big is *your* dialplan??   Doug Lytle
• [asterisk-users] How big is *your* dialplan??   Moises Silva
• [asterisk-users] RE: Welcome to the "asterisk-users" mailing list   Jessee J Holmes
• [asterisk-users] Monitor Current outgoing calls   Ralph Liebessohn
• [asterisk-users] Understanding NAT Traversal   hugolivude
• [asterisk-users] Re: Re: Voicemail Press '0'   LJ
• [asterisk-users] Cisco CCM - Asterisk   Phil Genera
• [asterisk-users] How big is *your* dialplan??   Steve Murphy
• [asterisk-users] Re: SIP stuck channel soft hangup?   Martin Joseph
• [asterisk-users] how to play pre-recorded file in meetme conference   Barry D. Hassler
• [asterisk-users] Cisco CCM - Asterisk   Alyed Tzompa
• [asterisk-users] How big is *your* dialplan??   Marc Rieber
• [asterisk-users] RE: Welcome to the "asterisk-users" mailing list   Eric "ManxPower" Wieling
• [asterisk-users] Change the background of a conversation   Jan Eirik Sandnes
• [asterisk-users] How big is *your* dialplan??   www.IPKall.com
• [asterisk-users] New Asterisk StumbleUpon Group   Chris Ramsey
• [asterisk-users] Free copy of "TrixBox Made Easy"   Chris Ramsey
• [asterisk-users] Cisco CCM - Asterisk   Alyed Tzompa
• [asterisk-users] How big is *your* dialplan??   Chris Ramsey
• [asterisk-users] transfer from VM to Cell Phone   Chris Ramsey
• [asterisk-users] How big is *your* dialplan??   Alejandro Kauffmann
• [asterisk-users] Understanding NAT Traversal   Mojo with Horan & Company, LLC
• [asterisk-users] Hangup or busy when the peer answer outgoing calls   Mojo with Horan & Company, LLC
• [asterisk-users] Strange FXS disconnection problem.   David Bath
• [asterisk-users] Re: Re: Voicemail Press '0'   Douglas Garstang
• [asterisk-users] RE: Welcome to the "asterisk-users" mailing list   Mike Clark
• [asterisk-users] Lots and lots of log files   Nic Bellamy
• [asterisk-users] How big is *your* dialplan??   Peter J Dean
• [asterisk-users] asterisk crash in res_features.c   Conrad Wood
• [asterisk-users] How big is *your* dialplan??   Doug Lytle
• [asterisk-users] Change the background of a conversation   Dovid B
• [asterisk-users] Re: PRI issues   Doug Lytle
• [asterisk-users] Cisco CCM - Asterisk   Lacy Moore - Aspendora
• [asterisk-users] How big is *your* dialplan??   C F
• [asterisk-users] RE: Welcome to the "asterisk-users" mailing list   Dean Collins
• [asterisk-users] Change the background of a conversation   Dean Collins
• [asterisk-users] How big is *your* dialplan??   Andrew Kohlsmith
• [asterisk-users] How big is *your* dialplan??   Andrew Kohlsmith
• [asterisk-users] Understanding NAT Traversal   Samy Kamkar
• [asterisk-users] Cisco CCM - Asterisk   Greg Oliver
• [asterisk-users] How big is *your* dialplan??   Jeremy McNamara
• [asterisk-users] RE: Welcome to the "asterisk-users" mailing list   C F
• [asterisk-users] how can I detect a DTMF tone while on a bridged call ? anyone knows?   MF
• [asterisk-users] Understanding NAT Traversal   hugolivude
• [asterisk-users] Increase VoiceMail Messages Recording Gain - AudioCalls are Ok   Shawn Kelley
• [asterisk-users] Re: Mitel 5224/SIP no MWI [RESOLVED]   Jesse Peterson
• [asterisk-users] E164 caller ID   sip
• [asterisk-users] Re: SIP stuck channel soft hangup?   Nic Bellamy
• [asterisk-users] Understanding NAT Traversal   Cullin J. Wible
• [asterisk-users] Re: channel variable   Leif Madsen
• [asterisk-users] How big is *your* dialplan??   Tom Vile
• [asterisk-users] Re: Re: Re: Voicemail Press '0'   LJ
• [asterisk-users] How big is *your* dialplan??   Ira
• [asterisk-users] How big is *your* dialplan??   George Pajari
• [asterisk-users] Increase VoiceMail Messages Recording Gain - AudioCalls are Ok   Aaron Daniel
• [asterisk-users] How big is *your* dialplan??   Douglas Garstang
• [asterisk-users] RE: Welcome to the "asterisk-users" mailing list   Douglas Garstang
• [asterisk-users] WRT54GP2 provisioning   Curt Shaffer
• [asterisk-users] Cubix / Firefly softphone and Asterisk   Garth van Sittert
• [asterisk-users] Re: SIP stuck channel soft hangup?   Martin Joseph
• [asterisk-users] Re: Understanding NAT Traversal   Martin Joseph
• [asterisk-users] Psst... Top secret information: Code