Mail Index

• Thread Index

• [Asterisk-Users] Video Conferencing
  • From: Dakota
• Re: [Asterisk-Users] Sip man in the middle
  • From: Mike Bernson
• Re: [Asterisk-Users] voicemail/privacy system
  • From: Leif Neland
• Re: [Asterisk-Users] GXP-2000 fw 1.0.1.13 and NTP
  • From: Leif Neland
• RE: [Asterisk-Users] Video Conferencing
  • From: Nir Simionovich
• RE: [Asterisk-Users] Video Conferencing
  • From: Nir Simionovich
• [Asterisk-Users] Recommendations on web interface for IT staff
  • From: Chris Mason (Lists)
• [Asterisk-Users] CrystalFontz LCD display
  • From: Mike Hammett
• Re: [Asterisk-Users] How to check Queue Statistics
  • From: Lenz
• RE: [Asterisk-Users] Recommendations on web interface for IT staff
  • From: Senad Jordanovic
• [Asterisk-Users] Re: TDM2400 wierdness
  • From: LJ
• Re: [Asterisk-Users] Affordable IP Phones for Asterisk
  • From: VoIP Newbie
• [Asterisk-Users] Snom 190 occasionally NR, SIP 401
  • From: Stefan Tichy
• RE: [Asterisk-Users] Affordable IP Phones for Asterisk
  • From: Kerry Garrison
• Re: [Asterisk-Users] Asterisk & FXO & Panasonic PBX
  • From: VoIP Newbie
• RE: [Asterisk-Users] Need HT488 FXO example for both inbound andoutbound.
  • From: Bjorn Asmul
• [Asterisk-Users] Re: What is the best Dell Machine for Asterisk?
  • From: Louis-David Mitterrand
• [Asterisk-Users] Got 200 OK on REGISTER that isn't a register
  • From: Robert La Ferla
• Re: [Asterisk-Users] Got 200 OK on REGISTER that isn't a register
  • From: BJ Weschke
• [Asterisk-Users] Re: GXP-2000 fw 1.0.1.13 and NTP
  • From: Wolfgang S. Rupprecht
• Re: [Asterisk-Users] Need HT488 FXO example for both inboundandoutbound.
  • From: James Ronald
• Re: [Asterisk-Dev] Question on using system(find args -exec rm {} \; )
  • From: Rusty Dekema
• RE: [Asterisk-Users] Having major issues with TDM2400
  • From: gw
• RE: [Asterisk-Users] Having major issues with TDM2400
  • From: Kerry Garrison
• Re: [Asterisk-Users] voicemail/privacy system
  • From: Roy Kidder
• Re: [Asterisk-Users] Snom 190 occasionally NR, SIP 401
  • From: Michiel van Baak
• Re: [Asterisk-Users] Snom 190 occasionally NR, SIP 401
  • From: Michiel van Baak
• Re: [Asterisk-Dev] Problem on ZAP channel
  • From: Richard Lyman
• [Asterisk-Users] Cell phone dock/switch as Asterisk FXO source
  • From: Brian McEntire
• Re: [Asterisk-Users] Cell phone dock/switch as Asterisk FXO source
  • From: John Novack
• Re: [Asterisk-Users] Cell phone dock/switch as Asterisk FXO source
  • From: Morel Mosolff
• [Asterisk-Users] Asterisk 1.2.1 segmentation faulting!...
  • From: Francesco Peeters (Asterisk)
• RE: [Asterisk-Users] Asterisk 1.2.1 segmentation faulting!...
  • From: Carlos Alperin
• Re: [Asterisk-Users] Having major issues with TDM2400
  • From: C F
• Re: [Asterisk-Users] Asterisk & FXO & Panasonic PBX
  • From: C F
• RE: [Asterisk-Users] Having major issues with TDM2400
  • From: Kerry Garrison
• Re: [Asterisk-Users] Asterisk & FXO & Panasonic PBX
  • From: Hermann Wecke
• Re: [Asterisk-Users] Having major issues with TDM2400
  • From: BJ Weschke
• Re: [Asterisk-Users] Having major issues with TDM2400
  • From: C F
• RE: [Asterisk-Users] Having major issues with TDM2400
  • From: gw
• RE: [Asterisk-Users] Having major issues with TDM2400
  • From: gw
• RE: [Asterisk-Users] Having major issues with TDM2400
  • From: Kerry Garrison
• Re: [Asterisk-Users] Having major issues with TDM2400
  • From: BJ Weschke
• RE: [Asterisk-Users] Having major issues with TDM2400
  • From: Kerry Garrison
• Re: [Asterisk-Users] Having major issues with TDM2400
  • From: C F
• Re: [Asterisk-Users] Re: What is the best Dell Machine for Asterisk?
  • From: Craig Guy
• RE: [Asterisk-Users] Having major issues with TDM2400
  • From: Kerry Garrison
• RE: [Asterisk-Users] Having major issues with TDM2400
  • From: gw
• RE: [Asterisk-Users] Having major issues with TDM2400
  • From: Kerry Garrison
• Re: [Asterisk-Users] name that vendor...
  • From: Hermann Wecke
• [Asterisk-Users] (Fwd) hi there
  • From: Rehan AllahWala
• Re: [Asterisk-Users] (Fwd) hi there
  • From: Morel Mosolff
• Re: [Asterisk-Users] What is the best Dell Machine for Asterisk?
  • From: Mike Fedyk
• Re: [Asterisk-Users] Recommendations on web interface for IT staff
  • From: Mike Fedyk
• [Asterisk-Users] Re: [Asterisk-biz] (Fwd) hi there
  • From: Sahil Gupta
• [Asterisk-Users] Re: [Asterisk-biz] (Fwd) hi there
  • From: Rehan AllahWala
• RE: [Asterisk-Users] Having major issues with TDM2400
  • From: steve
• [Asterisk-Users] Codec
  • From: hrishikesh shrivastaw
• Re: [Asterisk-Users] Re: Asterisk Christmas Help request
  • From: Roman Volf
• [Asterisk-Users] Re: linux soft raid (was: What is the best Dell Machine for Asterisk?)
  • From: Louis-David Mitterrand
• Re: [Asterisk-Dev] Race issue in channel.c involving uniqueint on Asterisk 1.2.1
  • From: Dinesh Nair
• [Asterisk-Users] test
  • From: RdBSD
• Re: [Asterisk-Dev] chan_sip.c : ignoring domain part for incomingINVITE's causes conflicts between domains?
  • From: Olle E Johansson
• Re: [Asterisk-Users] GXP-2000 any good with * ?
  • From: Sven Fischer (support)
• [Asterisk-Users] Newbie Problem With Agents
  • From: vivek
• Re: [Asterisk-Users] test
  • From: pdhales
• Re: [Asterisk-Users] Newbie Problem With Agents
  • From: Morel Mosolff
• RE: [Asterisk-Users] Re: [Asterisk-biz] (Fwd) hi there
  • From: kevin ling
• [Asterisk-Users] Festival issues
  • From: Andrew Nowrot
• RE: [Asterisk-Users] Asterisk & FXO & Panasonic PBX
  • From: kevin ling
• [Asterisk-Dev] zaptel path
  • From: Igor Neves
• [Asterisk-Users] Happy New Asterisk Year!
  • From: Olle E Johansson
• Re: [Asterisk-Users] Video Conferencing
  • From: Adnan Ahmed
• [Asterisk-Users] unable to execute call file
  • From: vicky sarathy
• Re: [Asterisk-Users] unable to execute call file
  • From: Morel Mosolff
• [Asterisk-Users] connect more the one phone to ONE sip Acoount
  • From: Andreas Koch
• Re: [Asterisk-Users] unable to execute call file
  • From: Karsten Wemheuer
• Re: [Asterisk-Users] Realtime Multiple Asterisk boxes and rtcachefriends MWI
  • From: Kristian Larsson
• [Asterisk-Users] Asterisk PRI problems.
  • From: Kristian Larsson
• [Asterisk-Users] CAPI unable to handle busy()
  • From: Karsten Wemheuer
• [Asterisk-Users] Is it possible to get caller and called number with Asterisk Manager
  • From: amaury BOSSE
• Re: [Asterisk-Users] snom Firmware 5.0.
  • From: Remco Barende
• Re: [Asterisk-Users] CAPI unable to handle busy()
  • From: Armin Schindler
• Re: [Asterisk-Users] Asterisk PRI problems.
  • From: steve
• Re: [Asterisk-Users] Asterisk PRI problems.
  • From: Kristian Larsson
• Re: [Asterisk-Users] connect more the one phone to ONE sip Acoount
  • From: Leif Neland
• Re: [Asterisk-Users] CAPI unable to handle busy()
  • From: Karsten Wemheuer
• Re: [Asterisk-Users] Semi-OT: porting numbers away
  • From: Leif Neland
• RE: [Asterisk-Users] Regular Crashes
  • From: Andrew Gough
• Re: [Asterisk-Users] What is the best Dell Machine for Asterisk?
  • From: Simone Cittadini
• Re: [Asterisk-Users] Regular Crashes
  • From: Paradise Dove
• Re: [Asterisk-Users] Semi-OT: porting numbers away
  • From: Leif Neland
• Re: [Asterisk-Users] CAPI unable to handle busy()
  • From: Armin Schindler
• Re: [Asterisk-Users] Re: What is the best Dell Machine for Asterisk?
  • From: Administrator TOOTAI
• [Asterisk-Users] Q: How to dial out / transfer calls with manager
  • From: Don Fanning
• [Asterisk-Users] Echo after asterisk has been running for several days
  • From: Matt
• [Asterisk-Users] Translating between different codes
  • From: Bartosz Wegrzyn - asterisk
• Re: [Asterisk-Users] CrystalFontz LCD display
  • From: Matt Riddell
• Re: [Asterisk-Users] Q: How to dial out / transfer calls with manager
  • From: Moises Silva
• [Asterisk-Users] sip realtime and setvar
  • From: Morten Isaksen
• Re: [Asterisk-Users] Translating between different codes
  • From: Moises Silva
• Re: [Asterisk-Users] Translating between different codes
  • From: Bartosz Wegrzyn - asterisk
• Re: [Asterisk-Dev] Problem on ZAP channel
  • From: Steve McMahon
• [Asterisk-Users] E911 And Routing
  • From: Matt
• RE: [Asterisk-Users] Echo after asterisk has been running for severaldays
  • From: Carlos Alperin
• Re: [Asterisk-Users] Translating between different codes
  • From: Julio Arruda
• RE: [Asterisk-Users] Problem getting D channel up on Sangoma A102
  • From: Schochet, Wes
• Re: [Asterisk-Users] E911 And Routing
  • From: trixter aka Bret McDanel
• Re: [Asterisk-Users] E911 And Routing
  • From: Tom Vile
• Re: [Asterisk-Users] E911 And Routing
  • From: Morel Mosolff
• Re: [Asterisk-Users] Cell phone dock/switch as Asterisk FXO source
  • From: Jonathan Attwood
• RE: [Asterisk-Users] E911 And Routing
  • From: Alexander Lopez
• [Asterisk-Users] USB phone
  • From: VoIP Newbie
• Re: [Asterisk-Users] E911 And Routing
  • From: trixter aka Bret McDanel
• [Asterisk-Users] Client SIP fo Windows Mobile
  • From: Giordano Grandis
• RE: [Asterisk-Users] snom Firmware 5.0.
  • From: Usman Tahir
• Re: [Asterisk-Users] E911 And Routing
  • From: Leonard Burton
• Re: [Asterisk-Users] Client SIP fo Windows Mobile
  • From: Sparcmanmlist1@xxxxxxx
• [Asterisk-Users] RE: simulator for asterisk gateway
  • From: Tejas Shah
• RE: [Asterisk-Users] GXP-2000 any good with * ?
  • From: Ross C
• R: [Asterisk-Users] Client SIP fo Windows Mobile
  • From: Giordano Grandis
• RE: [Asterisk-Users] USB phone
  • From: Kerry Garrison
• [Asterisk-Users] Re: SPA-3000 + call waiting
  • From: Ugo Bellavance
• Re: [Asterisk-Users] Cell phone dock/switch as Asterisk FXO source
  • From: Matt Riddell
• Re: [Asterisk-Users] connect more the one phone to ONE sip Acoount
  • From: VoIP Newbie
• Re: [Asterisk-Users] GXP-2000 any good with * ?
  • From: Sven Fischer (support)
• RE: [Asterisk-Users] connect more the one phone to ONE sip Acoount
  • From: Kerry Garrison
• RE: [Asterisk-Users] Client SIP fo Windows Mobile
  • From: asterisk-users
• RE: [Asterisk-Users] snom Firmware 5.0.
  • From: Remco Barende
• RE: [Asterisk-Users] GXP-2000 any good with * ?
  • From: Ross C
• Re: [Asterisk-Users] name that vendor...
  • From: Carlos Chavez
• Re: [Asterisk-Users] USB phone
  • From: Matt Riddell
• RE: [Asterisk-Users] Q: How to dial out / transfer calls with manager
  • From: Don Fanning
• Re: [Asterisk-Users] NOOB: Need Help Learning How to Debug PRI (U.S.)
  • From: Don Pobanz
• Re: [Asterisk-Users] connect more the one phone to ONE sip Acoount
  • From: Olle E Johansson
• Re: [Asterisk-Users] connect more the one phone to ONE sip Acoount
  • From: C F
• Re: [Asterisk-Users] connect more the one phone to ONE sip Acoount
  • From: Dualcall.com
• Re: [Asterisk-Users] Having major issues with TDM2400
  • From: C F
• Re: [Asterisk-Users] connect more the one phone to ONE sip Acoount
  • From: Olle E Johansson
• Re: [Asterisk-Users] Having major issues with TDM2400
  • From: C F
• Re: [Asterisk-Users] connect more the one phone to ONE sip Acoount
  • From: C F
• re: [Asterisk-Users] Asterisk PRI problems.
  • From: Alyed Tzompa
• Re: [Asterisk-Users] Asterisk PRI problems.
  • From: Morel Mosolff
• [Asterisk-Users] SIP through freeBSD NAT
  • From: Alyed Tzompa
• Re: [Asterisk-Users] Newbie Problem With Agents
  • From: Michiel van Baak
• [Asterisk-Dev] test
  • From: Roy Sigurd Karlsbakk
• [Asterisk-Dev] moderation on asterisk-dev
  • From: Roy Sigurd Karlsbakk
• Re: [Asterisk-Dev] test
  • From: janvb@xxxxxxxxxxxxxxxxxxxx
• Re: [Asterisk-Dev] moderation on asterisk-dev
  • From: Kevin Hanson
• Re: [Asterisk-Dev] moderation on asterisk-dev
  • From: Zoa
• Re: [Asterisk-Dev] moderation on asterisk-dev
  • From: janvb@xxxxxxxxxxxxxxxxxxxx
• [Asterisk-Users] Re: Asterisk-Users Digest, Vol 18, Issue 6
  • From: Mike Hammett
• RE: [Asterisk-Users] GXP-2000 any good with * ?
  • From: Christian Stredicke
• [Asterisk-Users] @Home overwrites configs on startup
  • From: Schochet, Wes
• Re: [Asterisk-Users] Echo after asterisk has been running for severaldays
  • From: Matt
• Re: [Asterisk-Users] @Home overwrites configs on startup
  • From: Tom Vile
• RE: [Asterisk-Users] @Home overwrites configs on startup
  • From: Kerry Garrison
• RE: [Asterisk-Users] @Home overwrites configs on startup
  • From: Ross C
• [Asterisk-Users] Ignoring too old SIP packet packet problems
  • From: Tom Vile
• RE: [Asterisk-Users] GXP-2000 any good with * ?
  • From: Ross C
• [Asterisk-Users] option "r" in Dial command seems not to work
  • From: Franz Wu
• RE: [Asterisk-Users] @Home overwrites configs on startup
  • From: Schochet, Wes
• [Asterisk-Users] Festival clicks instead of sound and disconnects.
  • From: Robert La Ferla
• Re: [Asterisk-Users] Re: What is the best Dell Machine for Asterisk?
  • From: Mike Fedyk
• [Asterisk-Users] Re: What is the best Dell Machine for Asterisk?
  • From: Louis-David Mitterrand
• Re: [Asterisk-Users] What is the best Dell Machine for Asterisk?
  • From: Mike Fedyk
• Re: [Asterisk-Users] What is the best Dell Machine for Asterisk?
  • From: Zoa
• Re: [Asterisk-Users] Re: What is the best Dell Machine for Asterisk?
  • From: Mike Fedyk
• [Asterisk-Users] Re: sip realtime and setvar
  • From: Morten Isaksen
• RE: [Asterisk-Users] @Home overwrites configs on startup
  • From: Dean Collins
• RE: [Asterisk-Users] @Home overwrites configs on startup
  • From: Dean Collins
• [Asterisk-Users] Memory PB.
  • From: Fabrice
• Re: [Asterisk-Dev] test
  • From: Rod Dorman
• Re: [Asterisk-Users] What is the best Dell Machine for Asterisk?
  • From: Mike Fedyk
• Re: [Asterisk-Users] Is it possible to get caller and called numberwith Asterisk Manager
  • From: pdhales
• Re: [Asterisk-Dev] Race issue in channel.c involving uniqueint on Asterisk 1.2.1
  • From: Tilghman Lesher
• [Asterisk-Users] Asterisk Upgrade to 1.2
  • From: Adam Vocks
• Re: [Asterisk-Dev] Race issue in channel.c involving uniqueint on Asterisk 1.2.1
  • From: Luigi Rizzo
• [Asterisk-Users] Re: Ignoring too old SIP packet packet problems
  • From: Tom Vile
• Re: [Asterisk-Users] Re: What is the best Dell Machine for Asterisk?
  • From: Administrator TOOTAI
• RE: [Asterisk-Users] Asterisk Upgrade to 1.2
  • From: Adam Vocks
• [Asterisk-Users] Limit concurent calls per MSN on BRI (bristuff/zaphfc)?
  • From: Pisac
• Re: [Asterisk-Dev] Race issue in channel.c involving uniqueint on Asterisk 1.2.1
  • From: Tilghman Lesher
• Re: [Asterisk-Users] Re: What is the best Dell Machine for Asterisk?
  • From: Mike Fedyk
• Re: [Asterisk-Dev] Race issue in channel.c involving uniqueint on Asterisk 1.2.1
  • From: Luigi Rizzo
• Re: [Asterisk-Users] Re: What is the best Dell Machine for Asterisk?
  • From: Mike McMullen
• Re: [Asterisk-Users] Festival clicks instead of sound and disconnects.
  • From: Robert La Ferla
• Fwd: [Asterisk-Users] Having major issues with TDM2400
  • From: C F
• [Asterisk-Users] Phantom Call on hangup SPA 2000
  • From: Rene Nelson
• RE: [Asterisk-Dev] Race issue in channel.c involving uniqueint onAsterisk 1.2.1
  • From: Steve Totaro
• [Asterisk-Users] offline g.729 transcoding
  • From: Tim Harrison
• Re: [Asterisk-Users] offline g.729 transcoding
  • From: Morel Mosolff
• Re: [Asterisk-Users] To write Sphinx Interface in EAGI or app_xxx.c?
  • From: Robert La Ferla
• Re: [Asterisk-Users] Phantom Call on hangup SPA 2000
  • From: C F
• Re: [Asterisk-Dev] Race issue in channel.c involving uniqueint onAsterisk 1.2.1
  • From: Luigi Rizzo
• RE: [Asterisk-Users] offline g.729 transcoding
  • From: kevin ling
• Re: [Asterisk-Users] Cell phone dock/switch as Asterisk FXO source
  • From: Paul Dugas
• Re: [Asterisk-Dev] Race issue in channel.c involving uniqueint on Asterisk 1.2.1
  • From: David Woodhouse
• Re: [Asterisk-Users] Phantom Call on hangup SPA 2000
  • From: Eric \"ManxPower\" Wieling
• Re: [Asterisk-Users] SIP through freeBSD NAT
  • From: Eric \"ManxPower\" Wieling
• RE: [Asterisk-Users] offline g.729 transcoding
  • From: Bogdan Moldovan
• [Asterisk-Users] Mute a given channel.
  • From: Jan du Toit
• RE: [Asterisk-Users] offline g.729 transcoding
  • From: kevin ling
• RE: [Asterisk-Users] offline g.729 transcoding
  • From: Bogdan Moldovan
• Re: [Asterisk-Users] Limit concurent calls per MSN on BRI (bristuff/zaphfc)?
  • From: Karsten Wemheuer
• Re: Re: [Asterisk-Users] unable to execute call file
  • From: vicky sarathy
• Re: [Asterisk-Users] Asterisk PRI problems.
  • From: Kristian Larsson
• [Asterisk-Users] How to get To-header and From-header display name
  • From: Somesh S Shanbhag
• Re: [Asterisk-Users] How to get To-header and From-header display name
  • From: Morel Mosolff
• Re: [Asterisk-Users] Echo after asterisk has been running for severaldays
  • From: Kristof Hardy
• Re: [Asterisk-Users] Cell phone dock/switch as Asterisk FXO source
  • From: bbench
• [Asterisk-Users] call-limit kills hints
  • From: Joseph Rothstein
• [Asterisk-Dev] Handling SIP response 480
  • From: Papadopoulos Georgios
• Re: [Asterisk-Users] Is it possible to get caller and called numberwith Asterisk Manager
  • From: Giovanni Miano
• Re: [Asterisk-Users] CrystalFontz LCD display
  • From: Giovanni Miano
• Re: [Asterisk-Dev] Race issue in channel.c involving uniqueint on Asterisk 1.2.1
  • From: Tilghman Lesher
• [Asterisk-Users] dhcp auto-provision spa-3000 like hardphones?
  • From: asterisk
• [Asterisk-Users] AgentCallbackLogin pre-# announcement?
  • From: iris
• [Asterisk-Users] Meetme user join/leave
  • From: Diyanat Ali
• Re: [Asterisk-Users] CrystalFontz LCD display
  • From: Andrea Cristofanini - Gedam Europe Srl
• RE: [Asterisk-Users] AgentCallbackLogin pre-# announcement?
  • From: Steve Hanselman
• [Asterisk-Users] cisco 7960 registration fails
  • From: Ben Fitzgerald
• [Asterisk-Users] SetCallerPres
  • From: Kristian Larsson
• [Asterisk-Dev] asterisk-dev censorship
  • From: Roy Sigurd Karlsbakk
• [Asterisk-Dev] How to adapt call termination keys in Asterisk.
  • From: Obelix
• SV: [Asterisk-Dev] asterisk-dev censorship
  • From: Håkon Nessjøen
• Re: [Asterisk-Users] cisco 7960 registration fails
  • From: Tzafrir Cohen
• [Asterisk-Users] Txgain & Rxgain
  • From: Humberto Aicardi
• Re: SV: [Asterisk-Dev] asterisk-dev censorship
  • From: Tzafrir Cohen
• RE: [Asterisk-Users] How to set features.conf to change thehangup key.
  • From: Obelix
• Re: [Asterisk-Users] USB phone
  • From: Dushyanth Harinath
• [Asterisk-Users] Re: Re: Asterisk Christmas Help request
  • From: Tomislav Parcina
• Re: [Asterisk-Users] Re: Re: Asterisk Christmas Help request
  • From: Morel Mosolff
• [Asterisk-Users] Re: Re: Asterisk Christmas Help request
  • From: Tomislav Parcina
• RE: [Asterisk-Users] How to set features.conf to change thehangup key.
  • From: Bogdan Moldovan
• Re: [Asterisk-Users] call-limit kills hints
  • From: Eric \"ManxPower\" Wieling
• Re: [Asterisk-Users] cisco 7960 registration fails
  • From: miguel saravia
• Re: [Asterisk-Users] call-limit kills hints
  • From: Paradise Dove
• [Asterisk-Users] outbound sip calls on asterisk
  • From: James Burke
• Re: [Asterisk-Dev] Handling SIP response 480
  • From: Mikael Magnusson
• Re: [Asterisk-Users] cisco 7960 registration fails
  • From: Ben Fitzgerald
• Re: [Asterisk-Users] cisco 7960 registration fails
  • From: Ben Fitzgerald
• Re: [Asterisk-Users] USB phone
  • From: Leo Ann Boon
• [Asterisk-Users] Re: Re: Congestion problem
  • From: Tomislav Parcina
• [Asterisk-Users] Three Way Calling with HFC PCI Card
  • From: Henry Margies
• [Asterisk-Users] Where is the Prefix() application in Asterisk 1.2.1 ?
  • From: Bukoka Budoka
• [Asterisk-Users] Re: voicemail storage over odbc and postgres
  • From: Tomislav Parcina
• Re: [Asterisk-Users] Cell phone dock/switch as Asterisk FXO source
  • From: Noah Swint
• Re: [Asterisk-Users] cisco 7960 registration fails
  • From: Rich Adamson
• Re: [Asterisk-Users] Where is the Prefix() application in Asterisk 1.2.1 ?
  • From: Pete Barnwell
• Re: [Asterisk-Users] Where is the Prefix() application in Asterisk 1.2.1 ?
  • From: Morel Mosolff
• [Asterisk-Users] Howto compile chan_h323 on macosx 10.3?
  • From: Antonio Marquez
• Re: [Asterisk-Users] Realtime Multiple Asterisk boxes, iaxusers
  • From: Kevin P. Fleming
• RE: [Asterisk-Users] Problem with date & time on Aastra 480i sincerelease 1.3
  • From: Lee Archer
• [Asterisk-Users] Re: connect more the one phone to ONE sip Acoount
  • From: Mikael Magnusson
• RE: [Asterisk-Users] Problem with date & time on Aastra 480i sincerelease 1.3
  • From: Dave Cotton
• Re: [Asterisk-Users] Realtime Multiple Asterisk boxes, iaxusers
  • From: Rich Adamson
• RE: [Asterisk-Users] Problem with date & time on Aastra 480i sincerelease 1.3
  • From: Pete Barnwell
• RE: [Asterisk-Users] Re: connect more the one phone to ONE sip Acoount
  • From: Juan Janczuk
• Re: [Asterisk-Users] ACD with polycom ip phones
  • From: BJ Weschke
• RE: [Asterisk-Users] Problem with date & time on Aastra 480isincerelease 1.3
  • From: Lee Archer
• [Asterisk-Users] E1 with CAS but no call signalling?
  • From: Tony Mountifield
• Re: [Asterisk-Users] Realtime Multiple Asterisk boxes, iaxusers
  • From: Kevin P. Fleming
• Re: [Asterisk-Users] Cell phone dock/switch as Asterisk FXO source
  • From: Kevin P. Fleming
• RE: [Asterisk-Dev] Question on using system(find args -exec rm {} \; )
  • From: bday
• [Asterisk-Users] machine load (was "best dell" a long time ago)
  • From: Simone Cittadini
• RE: [Asterisk-Users] Problem with date & time on Aastra 480isincerelease 1.3
  • From: Lee Archer
• Re: [Asterisk-Dev] zaptel path
  • From: Kevin P. Fleming
• Re: [Asterisk-Users] E1 with CAS but no call signalling?
  • From: Kevin P. Fleming
• [Asterisk-Dev] RE: [Asterisk-Users] ACD with polycom ip phones
  • From: Alexander Lopez
• [Asterisk-Users] Question on SPA-2002
  • From: Matt
• Re: [Asterisk-Users] Cell phone dock/switch as Asterisk FXO source
  • From: bbench
• Re: [Asterisk-Users] Problem with date & time on Aastra 480isincerelease 1.3
  • From: Tzafrir Cohen
• RE: [Asterisk-Users] GXP-2000 fw 1.0.1.13 and NTP
  • From: Lee Archer
• RE: [Asterisk-Users] Problem with date & time on Aastra480isincerelease 1.3
  • From: Lee Archer
• [Asterisk-Users] Asterisk realtime mysql connection
  • From: Sig Lange
• RE: [Asterisk-Dev] asterisk-dev censorship
  • From: Steve Totaro
• Re: [Asterisk-Users] Question on SPA-2002
  • From: C F
• Re: [Asterisk-Users] Where is the Prefix() application in Asterisk 1.2.1 ?
  • From: C F
• [Asterisk-Users] Where is the Prefix() application in Asterisk 1.2.1 ?
  • From: Bukoka Budoka
• Re: [Asterisk-Users] Three Way Calling with HFC PCI Card
  • From: Giovanni Miano
• Re: [Asterisk-Dev] zaptel path
  • From: Igor Neves
• [Asterisk-Users] IAXTEL??
  • From: Kerry Garrison
• [Asterisk-Users] Anyone heard of this company? http://www.affinityvoiptelecom.com/
  • From: Matt
• Re: [Asterisk-Users] Question on SPA-2002
  • From: Matt
• RE: [Asterisk-Users] IAXTEL??
  • From: Bogdan Moldovan
• Re: [Asterisk-Users] Howto compile chan_h323 on macosx 10.3?
  • From: Moises Silva
• Re: [Asterisk-Users] snom Firmware 5.0.
  • From: Joe Pukepail
• [Asterisk-Users] Re: Where is the Prefix() application in Asterisk1.2.1 ?
  • From: Steven
• Re: [Asterisk-Dev] zaptel path
  • From: Kevin P. Fleming
• RE: [Asterisk-Dev] what might corrupt...? (fixed, but have a question)
  • From: Goldfinger, Todd A
• Re: [Asterisk-Users] IAXTEL??
  • From: Tom Vile
• [Asterisk-Users] AEL - Using a Macro in the Dial Command in AEL
  • From: John Melody
• RE: [Asterisk-Users] IAXTEL??
  • From: Kerry Garrison
• RE: [Asterisk-Users] Question on SPA-2002
  • From: Damon Estep
• Re: [Asterisk-Dev] zaptel path
  • From: Tzafrir Cohen
• RE: [Asterisk-Users] IAXTEL??
  • From: Bogdan Moldovan
• [Asterisk-Users] FC3 or FC1 (or something else?)
  • From: Brett, Gary
• Re: [Asterisk-Users] IAXTEL??
  • From: Ariel Batista
• [Asterisk-Users] Sipura SPA-1001 question
  • From: burke
• Re: [Asterisk-Users] IAXTEL??
  • From: Rich Adamson
• Re: [Asterisk-Dev] zaptel path
  • From: Igor Neves
• RE: [Asterisk-Users] FC3 or FC1 (or something else?)
  • From: Bogdan Moldovan
• [Asterisk-Users] Re IAXTEL
  • From: Dave Cotton
• Using *RT for HA purposes was: [Asterisk-Users] Realtime Multiple Asterisk boxes, iaxusers
  • From: Mike Fedyk
• Re: [Asterisk-Dev] zaptel path
  • From: Igor Neves
• Re: [Asterisk-Users] Asterisk PRI problems.
  • From: William Lloyd
• [Asterisk-Users] GSM Gateway / Terminal for sale
  • From: Sam Tam
• Re: [Asterisk-Users] asterisk AVM C2 again
  • From: stéphane plichon
• RE: [Asterisk-Users] Regular Crashes
  • From: Andrew Gough
• Re: [Asterisk-Users] FC3 or FC1 (or something else?)
  • From: Robert La Ferla
• [Asterisk-Users] Asterisk on Dell blade servers
  • From: Alistair Cunningham
• Re: [Asterisk-Users] Regular Crashes
  • From: Robert La Ferla
• Re: [Asterisk-Users] AEL - Using a Macro in the Dial Command in AEL
  • From: Kevin P. Fleming
• Re: Using *RT for HA purposes was: [Asterisk-Users] Realtime Multiple Asterisk boxes, iaxusers
  • From: Kevin P. Fleming
• [Asterisk-Users] IAX termination services
  • From: Jason D. Wolfe
• [Asterisk-Users] DTMF dialing
  • From: VoIP Newbie
• [Asterisk-Users] Re: [Asterisk-biz] Asterisk on Dell blade servers
  • From: Linus Surguy
• [Asterisk-Users] Re: FC3 or FC1 (or something else?)
  • From: Steven
• RE: [Asterisk-Users] Having major issues with TDM2400
  • From: Kerry Garrison
• [Asterisk-Users] Recording Agent Calls
  • From: Douglas Garstang
• Re: Using *RT for HA purposes was: [Asterisk-Users] Realtime Multiple Asterisk boxes, iaxusers
  • From: Mike Fedyk
• Re: [Asterisk-Users] RPID Issue
  • From: Ray Van Dolson
• Re: [Asterisk-Users] FC3 or FC1 (or something else?)
  • From: Mike Fedyk
• Re: [Asterisk-Users] IAX termination services
  • From: Jean-Michel Hiver
• Re: [Asterisk-Users] IAX termination services
  • From: Jean-Michel Hiver
• Re: [Asterisk-Users] Re: FC3 or FC1 (or something else?)
  • From: Ray Van Dolson
• Re: [Asterisk-Users] Re: FC3 or FC1 (or something else?)
  • From: burke
• Re: [Asterisk-Users] SIP through freeBSD NAT
  • From: Alyed Tzompa
• [Asterisk-Users] cannot register whit sip client when i'm outside the PBX LAN
  • From: Antonio Gallo
• Re: [Asterisk-Users] Re: FC3 or FC1 (or something else?)
  • From: Mojo with Horan & Company, LLC
• [Asterisk-Users] Heavy Static on incoming calls
  • From: Jason Adams
• Re: [Asterisk-Users] Cell phone dock/switch as Asterisk FXO source
  • From: Mojo with Horan & Company, LLC
• Re: [Asterisk-Users] CAPI unable to handle busy()
  • From: Karsten Wemheuer
• [Asterisk-Users] Looping Problem With Call Forwards - Do you have comments on my solution?
  • From: Brent Torrenga
• Re: [Asterisk-Dev] zaptel path
  • From: Tzafrir Cohen
• Re: [Asterisk-Users] SIP through freeBSD NAT
  • From: Eric \"ManxPower\" Wieling
• RE: [Asterisk-Users] Asterisk Upgrade to 1.2
  • From: Adam Vocks
• Re: [Asterisk-Users] call-limit kills hints
  • From: Olle E Johansson
• Re: [Asterisk-Users] Re: connect more the one phone to ONE sip Acoount
  • From: Olle E Johansson
• Re: [Asterisk-Users] Asterisk PRI problems.
  • From: Alyed Tzompa
• [Asterisk-Users] iax2 wireless and Multicast
  • From: Francisco Pérez Botella
• Re: [Asterisk-Users] FC3 or FC1 (or something else?)
  • From: Michael Stearne
• Re: [Asterisk-Users] SIP through freeBSD NAT
  • From: Alyed Tzompa
• Re: Using *RT for HA purposes was: [Asterisk-Users] Realtime Multiple Asterisk boxes, iaxusers
  • From: Kevin P. Fleming
• [Asterisk-Users] Experience with SetTransferCapability
  • From: McGhee, Stefano
• Re: [Asterisk-Users] SIP through freeBSD NAT
  • From: Julio Arruda
• Re: [Asterisk-Users] SIP through freeBSD NAT
  • From: Alyed Tzompa
• [Asterisk-Users] How do you check whether a channel is active and the number of calls
  • From: Obelix
• RE: [Asterisk-Users] call-limit kills hints
  • From: Senad Jordanovic
• [Asterisk-Users] OT: XML Content Manager for Cisco 79XX Phones
  • From: Corey S. McFadden
• Re: Using *RT for HA purposes was: [Asterisk-Users] Realtime Multiple Asterisk boxes, iaxusers
  • From: Rich Adamson
• [Asterisk-Users] Asterisk 1.2.1 Type of Service
  • From: Steve Beaumont
• [Asterisk-Users] Re: Asterisk-Users Digest, Vol 17, Issue 109
  • From: Mike Hammett
• Re: [Asterisk-Users] Heavy Static on incoming calls
  • From: Tom Vile
• Re: [Asterisk-Users] SIP through freeBSD NAT
  • From: Alyed Tzompa
• Re: [Asterisk-Users] Asterisk 1.2.1 Type of Service
  • From: Armin Schindler
• Re: [Asterisk-Users] FC3 or FC1 (or something else?)
  • From: Tzafrir Cohen
• Re: [Asterisk-Users] How do you check whether a channel is active and the number of calls
  • From: Matt
• Re: [Asterisk-Users] How do you check whether a channel is active and the number of calls
  • From: Obelix
• RE: [Asterisk-Users] Heavy Static on incoming calls
  • From: Jason Adams
• Re: [Asterisk-Users] Cell phone dock/switch as Asterisk FXO source
  • From: Jonathan Attwood
• re: [Asterisk-Users] How do you check whether a channel is active and the number of calls
  • From: Alyed Tzompa
• Re: [Asterisk-Users] FC3 or FC1 (or something else?)
  • From: Technical Support
• RE: [Asterisk-Users] Asterisk PRI problems.
  • From: O'Connor, Jonathan
• Re: [Asterisk-Users] Cell phone dock/switch as Asterisk FXO source
  • From: Erick Perez
• [Asterisk-Users] Sipbroker?
  • From: Matt
• RE: Using *RT for HA purposes was: [Asterisk-Users] Realtime MultipleAsterisk boxes, iaxusers
  • From: Asterisk
• Re: [Asterisk-Users] Cell phone dock/switch as Asterisk FXO source
  • From: Erick Perez
• Re: [Asterisk-Users] Sipbroker?
  • From: Ben Higley
• [Asterisk-Users] Where is the Prefix() application in Asterisk 1.2.1 ?
  • From: Bukoka Budoka
• Re: [Asterisk-Users] iax2 wireless and Multicast
  • From: Rich Adamson
• [Asterisk-Users] Uvox streams
  • From: Alexander Lopez
• Re: [Asterisk-Users] iax2 wireless and Multicast
  • From: pdhales
• [Asterisk-Users] Echo cancellation
  • From: Aaron Daniel
• Re: [Asterisk-Dev] Echo cancellation
  • From: Kevin P. Fleming
• Re: [Asterisk-Dev] Echo cancellation
  • From: Aaron Daniel
• Re: [Asterisk-Users] IAXTEL??
  • From: Kevin Bockman
• [Asterisk-Users] Resolving timing issues with dual PRIs in a TE411P
  • From: Anthony Rodgers
• Re: [Asterisk-Dev] Echo cancellation
  • From: Kevin P. Fleming
• Re: [Asterisk-Users] FC3 or FC1 (or something else?)
  • From: Michael Stearne
• [Asterisk-Users] confusion about contexts
  • From: Aisling O'Driscoll
• [Asterisk-Users] Dialer
  • From: Wiley Siler
• Re: [Asterisk-Users] Resolving timing issues with dual PRIs in a TE411P
  • From: Casey Boone
• RE: [Asterisk-Users] Dialer
  • From: Kerry Garrison
• [Asterisk-Users] Problems Upgrading to 1.2.1 on Fedora 3
  • From: Michael Stearne
• Re: Using *RT for HA purposes was: [Asterisk-Users] Realtime MultipleAsterisk boxes, iaxusers
  • From: Kevin P. Fleming
• Re: [Asterisk-Users] Dialer
  • From: Darren Wiebe
• Re: [Asterisk-Users] Resolving timing issues with dual PRIs in a TE411P
  • From: Anthony Rodgers
• [Asterisk-Users] bridging two active calls
  • From: Jason D. Wolfe
• Re: [Asterisk-Users] Problems Upgrading to 1.2.1 on Fedora 3
  • From: Mojo with Horan & Company, LLC
• [Asterisk-Users] Re: Cell phone dock/switch as Asterisk FXO source
  • From: GeekSpeed
• re: [Asterisk-Users] confusion about contexts
  • From: Alyed Tzompa
• RE: (Fwd) [Asterisk-Users] bridging two active calls
  • From: Jason D. Wolfe
• RE: [Asterisk-Users] Having major issues with TDM2400
  • From: Kerry Garrison
• Re: [Asterisk-Users] Problems Upgrading to 1.2.1 on Fedora 3
  • From: Michael Stearne
• RE: [Asterisk-Users] How do you check whether a channel is active andthe number of calls
  • From: Alexander Lopez
• RE: [Asterisk-Users] bridging two active calls
  • From: Alexander Lopez
• [Asterisk-Users] integration with Meridian/Norstar ATA2
  • From: Darrick Hartman
• Re: [Asterisk-Users] Problems Upgrading to 1.2.1 on Fedora 3
  • From: Ray Van Dolson
• Re: [Asterisk-Users] bridging two active calls
  • From: Matt Florell
• Re: [Asterisk-Users] voip-info: Asterisk record calls
  • From: Mojo with Horan & Company, LLC
• Re: [Asterisk-Users] Re: voip-info: Asterisk record calls
  • From: Mojo with Horan & Company, LLC
• Re: [Asterisk-Users] Problem with date & time on Aastra480isincerelease 1.3
  • From: Jacques Leisy
• [Asterisk-Users] Start recording after call started
  • From: Technical Support
• [Asterisk-Users] Re: FC3 or FC1 (or something else?)
  • From: Steven
• Re: [Asterisk-Users] FC3 or FC1 (or something else?)
  • From: L
• Re: [Asterisk-Users] FC3 or FC1 (or something else?)
  • From: Robert La Ferla
• Re: [Asterisk-Users] Having major issues with TDM2400
  • From: BJ Weschke
• Re: [Asterisk-Users] integration with Meridian/Norstar ATA2
  • From: Darrick Hartman
• RE: [Asterisk-Users] Having major issues with TDM2400
  • From: Kerry Garrison
• [Asterisk-Users] Re: Start recording after call started
  • From: LJ
• [Asterisk-Users] IAX2 channels denoted as '(None)'
  • From: Pete
• Re: [Asterisk-Users] Having major issues with TDM2400
  • From: C F
• Re: [Asterisk-Users] IAX2 channels denoted as '(None)'
  • From: Pete
• Re: [Asterisk-Users] voip-info: Asterisk record calls
  • From: Tim Litwiller
• [Asterisk-Users] Raw Hangup messages with IAX2?
  • From: Mike McMullen
• Re: [Asterisk-Users] Raw Hangup messages with IAX2?
  • From: Mike McMullen
• Re: [Asterisk-Users] Echo cancellation
  • From: Erick Baum
• [Asterisk-Users] Asterisk::LDAP
  • From: Derek Whitten
• Re: [Asterisk-Users] Problems Upgrading to 1.2.1 on Fedora 3
  • From: Michael Stearne
• Re: [Asterisk-Users] Echo cancellation
  • From: Aaron Daniel
• Re: [Asterisk-Users] Asterisk Christmas Help request
  • From: Asterisk@Home
• [Asterisk-Users] Detect a forwarded incoming call?
  • From: Frank Liu
• Re: [Asterisk-Users] Having major issues with TDM2400
  • From: Rob Lith
• Re: [Asterisk-Users] voip-info: Asterisk record calls
  • From: Tim Litwiller
• RE: [Asterisk-Users] Having major issues with TDM2400
  • From: Kerry Garrison
• [Asterisk-Users] Ominiis Asterisk TAPI driver
  • From: Tomislav Parcina
• [Asterisk-Users] Re: Re: voip-info: Asterisk record calls
  • From: Tomislav Parcina
• RE: [Asterisk-Users] Ominiis Asterisk TAPI driver
  • From: Kerry Garrison
• [Asterisk-Dev] How to get the transfer number
  • From: Chih-Wei Huang
• [Asterisk-Users]MusicOnHold don't start at begin
  • From: asterisk183
• Re: [Asterisk-Users] RPID Issue
  • From: Olle E Johansson
• Re: [Asterisk-Users] call-limit kills hints
  • From: Olle E Johansson
• RE: [Asterisk-Users] Regular Crashes
  • From: Andrew Gough
• RE: [Asterisk-Users] Is it possible to get caller and callednumberwith Asterisk Manager
  • From: amaury BOSSE
• [Asterisk-Dev] How to recover caller and called numbers with Asterisk Manager
  • From: amaury BOSSE
• [Asterisk-Users] SIP security
  • From: Tomislav Parcina
• Re: [Asterisk-Dev] Question on using system(find args -exec rm {} \; )
  • From: Noel Sharpe
• Re: [Asterisk-Users] SIP security
  • From: Olle E Johansson
• Re: Using *RT for HA purposes was: [Asterisk-Users] Realtime MultipleAsterisk boxes, iaxusers
  • From: tijmen van den brink
• Re: [Asterisk-Users] SIP security
  • From: trixter aka Bret McDanel
• [Asterisk-Users] Can I call another S0 bus device (BRI) locally without taking 2 channels through Telco provider ?
  • From: Robert Rozman
• [Asterisk-Users] call monitoring from 3th phone
  • From: turby
• Re: [Asterisk-Users] Ominiis Asterisk TAPI driver
  • From: Peer Oliver Schmidt
• [Asterisk-Dev] Reading sound and recognizing DTMF sounds at the same time in eagi script ?
  • From: Robert Rozman
• Re: [Asterisk-Users] Raw Hangup messages with IAX2?
  • From: Matt Riddell
• Re: Using *RT for HA purposes was: [Asterisk-Users] Realtime Multiple Asterisk boxes, iaxusers
  • From: Mike Fedyk
• Re: [Asterisk-Users] Asterisk on Dell blade servers
  • From: Matt Riddell
• [Asterisk-Users] remote sip client fail to register
  • From: Antonio Gallo
• [Asterisk-Users] RBT enable/disable
  • From: Code Lover
• Re: [Asterisk-Users] Asterisk 1.2.1 Type of Service
  • From: Steve Beaumont
• [Asterisk-Users] USB fxo/fxs devices
  • From: Zoltan Szecsei
• RE: [Asterisk-Users] confusion about contexts - SER
  • From: Aisling
• [Asterisk-Users] Re: Re: connect more the one phone to ONE sip Acoount
  • From: Tomislav Parcina
• [Asterisk-Users] RE: Echo after asterisk has been running for severaldays
  • From: Tomislav Parcina
• Re: [Asterisk-Users] Asterisk 1.2.1 Type of Service
  • From: Armin Schindler
• Re: [Asterisk-Users] Looping Problem With Call Forwards - Do you have comments on my solution?
  • From: Adam Goryachev
• [Asterisk-Users] Anybody successfully using vISDN on A@H?
  • From: Francesco Peeters (Asterisk)
• RE: [Asterisk-Users] Q: How to dial out / transfer calls with manager
  • From: Adam Goryachev
• Re: [Asterisk-Users] Raw Hangup messages with IAX2?
  • From: Francesco Peeters (Asterisk)
• Re: [Asterisk-Users] Looping Problem With Call Forwards - Do you have comments on my solution?
  • From: Peter Bowyer
• Re: [Asterisk-Users] Asterisk 1.2.1 Type of Service
  • From: Pete Barnwell
• Re: [Asterisk-Users] iax2 wireless and Multicast
  • From: tim panton
• RE: [Asterisk-Users] Can we dial agents from extensions.conf
  • From: Adam Goryachev
• [Asterisk-Dev] snom programmable buttons
  • From: cfh
• Re: [Asterisk-Users] Re: Re: connect more the one phone to ONE sip Acoount
  • From: Olle E Johansson
• Re: [Asterisk-Dev] snom programmable buttons
  • From: Olle E Johansson
• Re: [Asterisk-Users] Asterisk 1.2.1 Type of Service
  • From: Armin Schindler
• Re: Using *RT for HA purposes was: [Asterisk-Users] Realtime MultipleAsterisk boxes, iaxusers
  • From: Alistair Cunningham
• Re: [Asterisk-Users] Re: [Asterisk-biz] Asterisk on Dell blade servers
  • From: Alistair Cunningham
• Re: Using *RT for HA purposes was: [Asterisk-Users] Realtime MultipleAsterisk boxes, iaxusers
  • From: Peter Bowyer
• [Asterisk-Users] RxFax : Change FAX Resolution
  • From: Dushyanth Harinath
• Re: [Asterisk-Users] Asterisk on Dell blade servers
  • From: Michael J. Tubby B.Sc (Hons) G8TIC
• Re: [Asterisk-Users] iax2 wireless and Multicast
  • From: Francisco Pérez Botella
• Re: [Asterisk-Users] Problems Upgrading to 1.2.1 on Fedora 3
  • From: Tzafrir Cohen
• [Asterisk-Users] Compilation of OpenH323 libraries under CYGWIN...
  • From: Mauro Zanin
• Re: [Asterisk-Users] RxFax : Change FAX Resolution
  • From: Steve Underwood
• [Asterisk-Users] Re: OT: XML Content Manager for Cisco 79XX Phones
  • From: Tomislav Parcina
• [Asterisk-Users] Can i compile Asterik on Fedora 4 x86 64 and which hardware could you support ?
  • From: Mehmet Kürşat Gürbüz
• [Asterisk-Users] Re: call monitoring from 3th phone
  • From: LJ
• [Asterisk-Users] Re: OT: XML Content Manager for Cisco 79XX Phones
  • From: Tomislav Parcina
• Re: [Asterisk-Users] Raw Hangup messages with IAX2?
  • From: Mike McMullen
• [Asterisk-Users] Re: Asterisk on Dell blade servers
  • From: LJ
• [Asterisk-Users] suddenly iax calls don't work anymore
  • From: Gerald Dachs
• Re: [Asterisk-Users] Raw Hangup messages with IAX2?
  • From: Francesco Peeters (Asterisk)
• [Asterisk-Users] Entry level IP phone
  • From: Richard Smith
• RE: Using *RT for HA purposes was: [Asterisk-Users] RealtimeMultipleAsterisk boxes, iaxusers
  • From: Doug G
• [Asterisk-Users] Re: suddenly iax calls don't work anymore
  • From: Tony Mountifield
• Re: [Asterisk-Users] suddenly iax calls don't work anymore
  • From: Faris Raouf
• [Asterisk-Users] Unknown digits
  • From: Steven
• [Asterisk-Users] FYI new aricle on asteisk
  • From: Tony Nichols
• Re: Using *RT for HA purposes was: [Asterisk-Users] Realtime Multiple Asterisk boxes, iaxusers
  • From: Kevin P. Fleming
• Re: [Asterisk-Dev] Question on using system(find args -exec rm {} \; )
  • From: Tilghman Lesher
• Re: [Asterisk-Users] Re: Asterisk on Dell blade servers
  • From: Kevin P. Fleming
• [Asterisk-Dev] res/res_agi.c set accountcode
  • From: Cristian Draghici
• Re: [Asterisk-Users] suddenly iax calls don't work anymore
  • From: Gerald Dachs
• Re: Using *RT for HA purposes was: [Asterisk-Users] Realtime MultipleAsterisk boxes, iaxusers
  • From: Alistair Cunningham
• [Asterisk-Users] Re: Start recording after call started
  • From: Tomislav Parcina
• Re: Using *RT for HA purposes was: [Asterisk-Users] Realtime MultipleAsterisk boxes, iaxusers
  • From: Peter Bowyer