Mail Index

• Thread Index

• RE: [Asterisk-Users] Re: Grandstream Early Dial
  • From: Dave Cotton
• Re: [Asterisk-Users] Anyone, ideas for incoming call management for CRM system
  • From: Olle E. Johansson
• [Asterisk-Users] Video
  • From: jna
• Re: [Asterisk-Users] New to asterisk? RUN... don't walk.
  • From: Olle E. Johansson
• Re: AW: [Asterisk-Users] Snom 200 with two extns defined anyone?
  • From: Olle E. Johansson
• Re: [Asterisk-Users] Video
  • From: Olle E. Johansson
• [Asterisk-Users] asterisk reload for FWD to register
  • From: Miguel Cavazos
• RE: [Asterisk-Users] New to asterisk? RUN... don't walk.
  • From: JR Richardson
• Re: [Asterisk-Users] asterisk reload for FWD to register
  • From: Philipp von Klitzing
• RE: [Asterisk-Users] Java? --> Ming!
  • From: Ray Burkholder
• Re: [Asterisk-Users] New to asterisk? RUN... don't walk.
  • From: WipeOut
• Re: [Asterisk-Users] Java?
  • From: Philipp von Klitzing
• [Asterisk-Users] asterisk gateway to other gateways
  • From: lito lampitoc
• [Asterisk-Users] help
  • From: Shanon Swafford
• Re: [Asterisk-Users] after hours - is this logic ok ?
  • From: Tilghman Lesher
• Re: [Asterisk-Users] Programming an unlocked ADSI phone?
  • From: Tilghman Lesher
• [Asterisk-Dev] A proposal regarding NATing firewalls
  • From: Martin Kihlgren
• [Asterisk-Users] * crash when forward voicemail --Nicolas Gudino
  • From: JR Richardson
• Re: [Asterisk-Users] after hours - is this logic ok ?
  • From: Lance Arbuckle
• [Asterisk-Users] asterisk gateway to other gateways
  • From: SW
• Re: [Asterisk-Users] Programming an unlocked ADSI phone?
  • From: Darren Nickerson
• Re: [Asterisk-Users] after hours - is this logic ok ?
  • From: Philipp von Klitzing
• Re: [Asterisk-Users] after hours - is this logic ok ?
  • From: Tilghman Lesher
• RE: [Asterisk-Users] New to asterisk? RUN... don't walk.
  • From: Brian West
• [Asterisk-Users] Newbridge Mainstreet 3624 T1 channel bank now Alcatel
  • From: TC
• Re: [Asterisk-Users] after hours - is this logic ok ?
  • From: Lance Arbuckle
• Re: [Asterisk-Users] Programming an unlocked ADSI phone?
  • From: Ken Godee
• Re: [Asterisk-Users] Newbridge Mainstreet 3624 T1 channel bank now Alcatel
  • From: Lance Arbuckle
• Re: [Asterisk-Users] Programming an unlocked ADSI phone?
  • From: Darren Nickerson
• Re: [Asterisk-Users] Video
  • From: Siggi Langauf
• Re: [Asterisk-Users] keeping Wiki in sync (new subject)
  • From: Olle E. Johansson
• Re: [Asterisk-Users] after hours - is this logic ok ?
  • From: Steven Critchfield
• Re: [Asterisk-Users] after hours - is this logic ok ?
  • From: Lance Arbuckle
• Re: [Asterisk-Dev] Comedian Mail's ADSI implementation broken on phones needing unlock codes?
  • From: asterisk
• [Asterisk-Dev] Congratulations and happy new year!
  • From: Ahmad Faiz
• Re: [Asterisk-Users] Video
  • From: jna
• Re: [Asterisk-Dev] Crappy New Year - AGI is b0rked
  • From: Jared Smith
• [Asterisk-Users] Prediction for 2004
  • From: Philipp von Klitzing
• [Asterisk-Users] How to load the driver of TDM400P card!
  • From: Quan Le Trung
• Re: [Asterisk-Dev] Comedian Mail's ADSI implementation broken on phones needing unlock codes?
  • From: Darren Nickerson
• Re: [Asterisk-Users] How to load the driver of TDM400P card!
  • From: Tilghman Lesher
• [Asterisk-Users] Re: How to load the driver of TDM400P card!
  • From: Steve Murphy
• [Asterisk-Users] sound driver advise needed
  • From: Steve Murphy
• Re: [Asterisk-Users] sound driver advise needed
  • From: Steven Critchfield
• Re: [Asterisk-Users] * crash when forward voicemail --Nicolas Gudino
  • From: James H. Cloos Jr.
• RE: [Asterisk-Users] New to asterisk? RUN... don't walk.
  • From: Girish Gopinath
• Re: [Asterisk-Users] * crash when forward voicemail --Nicolas Gudino
  • From: Eric Wieling
• [Asterisk-Users] SQL Updater Down!!!
  • From: Chandra
• Re: [Asterisk-Users] sound driver advise needed
  • From: andrewg
• Re: [Asterisk-Users] Programming an unlocked ADSI phone?
  • From: Tilghman Lesher
• [Asterisk-Users] FW: [Asterisk-Dev] Strange bells in voicemail and voicemailmain ?
  • From: Florian Overkamp
• Re: [Asterisk-Users] sound driver advise needed
  • From: Steven Critchfield
• [Asterisk-Users] IAXy Release ?
  • From: Kannaiyan Natesan
• [Asterisk-Users] one way choppy sound problem !
  • From: Dawid Mielnik
• Re: [Asterisk-Users] one way choppy sound problem !
  • From: WipeOut
• Re: [Asterisk-Users] Programming an unlocked ADSI phone?
  • From: Jayson Vantuyl
• Re: [Asterisk-Users] Anyone, ideas for incoming call management for CRM system
  • From: Alastair Maw
• Re: [Asterisk-Users] FW: [Asterisk-Dev] Strange bells in voicemail and voicemailmain ?
  • From: Patrick
• [Asterisk-Users] Call recording
  • From: [fabbricadigitale]
• Re: [Asterisk-Users] FW: [Asterisk-Dev] Strange bells in voicemail and voicemailmain ?
  • From: Mark Spencer
• RE: [Asterisk-Users] Call recording
  • From: Sergio Serrano Revuelto
• [Asterisk-Users] License questioni supose ??
  • From: Michael Devenijn
• Re: [Asterisk-Users] Call recording
  • From: Chandra
• RE: [Asterisk-Users] SQL Updater Down!!!
  • From: mattf
• Re: [Asterisk-Users] License questioni supose ??
  • From: Nicolas Bougues
• [Asterisk-Users] SIP client not registering to *
  • From: Chandra
• RE: [Asterisk-Users] New to asterisk? RUN... don't walk.
  • From: Paulo Mannheimer
• [Asterisk-Users] Slow wiki?
  • From: Philipp von Klitzing
• Re: [Asterisk-Users] License questioni supose ??
  • From: Steven Critchfield
• [Asterisk-Users] unsubscribe
  • From: Ranga
• Re: [Asterisk-Users] Slow wiki?
  • From: Steven Critchfield
• Re: [Asterisk-Users] Prediction for 2004
  • From: TC
• Re: [Asterisk-Users] unsubscribe
  • From: Steven Critchfield
• [Asterisk-Users] * Stresstool Help required
  • From: Girish Gopinath
• Re: [Asterisk-Users] License questioni supose ??
  • From: Andrew Thompson
• Re: [Asterisk-Users] License questioni supose ??
  • From: Steven Critchfield
• Re: [Asterisk-Users] unsubscribe
  • From: Ranga
• Re: [Asterisk-Users] * Stresstool Help required
  • From: Alastair Maw
• Re: [Asterisk-Users] unsubscribe
  • From: Steven Critchfield
• Re: [Asterisk-Users] Prediction for 2004
  • From: TC
• Re: [Asterisk-Users] Call recording
  • From: CW_ASN
• [Asterisk-Users] asterisk dies while making calls
  • From: Asterisk List
• Re: [Asterisk-Users] asterisk dies while making calls
  • From: Steven Critchfield
• [Asterisk-Users] Malloc debug kills asterisk?
  • From: Scott Stingel
• Re: [Asterisk-Users] Call recording
  • From: Olle E. Johansson
• Re: [Asterisk-Users] Slow wiki?
  • From: Olle E. Johansson
• [Asterisk-Users] T400P & E400P second source
  • From: John Ternovas
• Re: [Asterisk-Users] Residential router w/ QoS support?
  • From: Thilo Salmon
• Re: [Asterisk-Users] one way choppy sound problem !
  • From: Nicolas Gudino
• [Asterisk-Dev] FW: Welcome to the "Asterisk-Dev" mailing list (Digest mode)
  • From: Fernando Berretta
• Re: [Asterisk-Users] * crash when forward voicemail --Nicolas Gudino
  • From: Nicolas Gudino
• Re: [Asterisk-Users] T400P & E400P second source
  • From: Steven Critchfield
• RE: [Asterisk-Users] T400P & E400P second source
  • From: Scott Stingel
• Re: [Asterisk-Users] one way choppy sound problem !
  • From: Steven Critchfield
• [Asterisk-Users] hangup detection
  • From: Sean Adams
• [Asterisk-Users] mini-ITX suggestions
  • From: Gary Gapinski
• Re: [Asterisk-Users] hangup detection
  • From: Steven Critchfield
• Re: [Asterisk-Users] one way choppy sound problem !
  • From: Nicolas Gudino
• Re: [Asterisk-Users] hangup detection
  • From: Martin Pycko
• Re: [Asterisk-Users] mini-ITX suggestions
  • From: Steven Critchfield
• Re: [Asterisk-Users] T400P & E400P second source
  • From: Linus Surguy
• Re: [Asterisk-Users] one way choppy sound problem !
  • From: Steven Critchfield
• Re: [Asterisk-Users] hangup detection
  • From: Sean Adams
• Re: [Asterisk-Users] hangup detection
  • From: Sean Adams
• RE: [Asterisk-Users] mini-ITX suggestions
  • From: Steven Sokol
• Re: [Asterisk-Users] Newbridge Mainstreet 3624 T1 channel bank now Alcatel
  • From: Jeff Roberts
• [Asterisk-Dev] AGI still falls over
  • From: Iain Stevenson
• RE: [Asterisk-Users] Newbridge Mainstreet 3624 T1 channel bank no w Alcatel
  • From: Kris Boutilier
• Re: [Asterisk-Users] hangup detection
  • From: Sean Adams
• Re: [Asterisk-Users] hangup detection
  • From: Steven Critchfield
• Re: [Asterisk-Dev] AGI still falls over
  • From: Steven Critchfield
• [Asterisk-Users] Re: Video
  • From: Matt Lawson
• Re: [Asterisk-Users] hangup detection
  • From: Martin Pycko
• [Asterisk-Users] AgentCallbackLogin.
  • From: Shad Mortazavi
• Re: [Asterisk-Dev] Digium TE410P Card
  • From: Mike Nugent
• Re: [Asterisk-Users] AgentCallbackLogin.
  • From: Philipp von Klitzing
• Re: [Asterisk-Users] Newbridge Mainstreet 3624 T1 channel bank no w Alcatel
  • From: TC
• Re: [Asterisk-Dev] AGI still falls over
  • From: Marc Olivier Chouinard
• [Asterisk-Users] Grandstream Flash Button
  • From: Stephen R. Besch
• Re: [Asterisk-Dev] AGI still falls over
  • From: Iain Stevenson
• Re: [Asterisk-Dev] AGI still falls over
  • From: Mark Spencer
• [Asterisk-Users] Grandstream Flash Button
  • From: Stephen R. Besch
• Re: [Asterisk-Dev] AGI still falls over
  • From: Iain Stevenson
• Re: [Asterisk-Users] one way choppy sound problem !
  • From: Andres
• Re: [Asterisk-Dev] AGI still falls over
  • From: Steven Critchfield
• Re: [Asterisk-Users] hangup detection
  • From: Sean Adams
• [Asterisk-Users] Newbie - getting two local phones to communicate would be a good start :)
  • From: John Coll
• Re: [Asterisk-Dev] AGI still falls over
  • From: Iain Stevenson
• RE: [Asterisk-Users] Newbridge Mainstreet 3624 T1 channel bank no w Alcatel
  • From: scheesman
• Re: [Asterisk-Users] Newbie - getting two local phones to communicate would be a good start :)
  • From: rnc Info Lists
• RE: [Asterisk-Users] Newbie - getting two local phones to communicate would be a good start :)
  • From: John Coll
• RE: [Asterisk-Users] Newbie - getting two local phones to communicate would be a good start :)
  • From: David J Carter
• [Asterisk-Users] echo
  • From: Sean Adams
• [Asterisk-Users] Cisco SIP license?
  • From: Josh Edwards
• [Asterisk-Dev] Re: A proposal regarding NATing firewalls
  • From: Martin Kihlgren
• Re: [Asterisk-Users] Newbie - getting two local phones to communicate would be a good start :)
  • From: Rich Adamson
• [Asterisk-Users] Newbridge Mainstreet 3624 Manual
  • From: scheesman
• Re: [Asterisk-Users] mini-ITX suggestions
  • From: Patrick Cantwell
• Re: [Asterisk-Users] Call recording/SIP not loggin IN
  • From: Chandra
• [Asterisk-Users] Asterisk Gotoif / last called
  • From: vocalvoip
• [Asterisk-Users] Re: Cisco SIP license?
  • From: Adthrawn
• [Asterisk-Users] Newbie - getting two local phones to communicate would be a good start :)
  • From: SW
• RE: [Asterisk-Users] Newbie - getting two local phones to communicate would be a good start :)
  • From: David Gomillion
• [Asterisk-Dev] Active call information
  • From: Venkat Venkataraju
• Re: [Asterisk-Users] Call recording/SIP not loggin IN
  • From: CW_ASN
• Re: [Asterisk-Dev] Active call information
  • From: Steven Critchfield
• Re: [Asterisk-Dev] Active call information
  • From: CW_ASN
• [Asterisk-Users] SIP/grandstream not registering
  • From: Chandra
• Re: [Asterisk-Users] SIP/grandstream not registering
  • From: Glenn Dalgliesh
• [Asterisk-Dev] My email
  • From: Nahid Hossain
• Re: [Asterisk-Users] Slow wiki?
  • From: James H. Thompson
• Re: [Asterisk-Dev] Active call information
  • From: Venkat Venkataraju
• Re: [Asterisk-Dev] Active call information
  • From: Venkat Venkataraju
• Re: [Asterisk-Dev] Active call information
  • From: Venkat Venkataraju
• Re: [Asterisk-Dev] Active call information
  • From: CW_ASN
• Re: [Asterisk-Users] Asterisk Gotoif / last called
  • From: Philipp von Klitzing
• Re: [Asterisk-Users] Call recording/SIP not loggin IN
  • From: Philipp von Klitzing
• RE: [Asterisk-Users] Newbie - getting two local phones to communicate would be a good start :)
  • From: John Coll
• [Asterisk-Users] Re: Cisco SIP license?
  • From: Adthrawn
• RE: [Asterisk-Users] Newbie - getting two local phones to communicate would be a good start :)
  • From: Steven Critchfield
• Re: [Asterisk-Users] * Stresstool Help required
  • From: Girish Gopinath
• AW: AW: [Asterisk-Users] Snom 200 with two extns defined anyone?
  • From: Christian Stredicke
• Re: [Asterisk-Users] * Stresstool Help required
  • From: Girish Gopinath
• Re: [Asterisk-Dev] Active call information
  • From: Jeremy McNamara
• RE: [Asterisk-Users] Newbie - getting two local phones tocommunicate would be a good start :)
  • From: John Coll
• Re: [Asterisk-Dev] Active call information
  • From: Tilghman Lesher
• RE: [Asterisk-Users] Newbie - getting two local phones tocommunicate would be a good start :)
  • From: Sean Cheesman
• RE: [Asterisk-Users] Newbie - getting two local phones tocommunicate would be a good start :)
  • From: Dave Cotton
• RE: [Asterisk-Users] Newbie - getting two local phones tocommunicate would be a good start :)
  • From: SW
• RE: [Asterisk-Users] Newbie - getting two local phones tocommunicate would be a good start :)
  • From: Philipp von Klitzing
• RE: [Asterisk-Users] Newbie - getting two local phonestocommunicate would be a good start :)
  • From: John Coll
• RE: [Asterisk-Users] Newbie - getting two local phonestocommunicate would be a good start :)
  • From: Dave Cotton
• RE: [Asterisk-Users] Newbie - getting two local phonestocommunicate would be a good start :)
  • From: John Coll
• Re: [Asterisk-Dev] Active call information
  • From: CW_ASN
• Re: [Asterisk-Users] SIP/grandstream not registering
  • From: CW_ASN
• [Asterisk-Dev] More AGI problems - fails to stream file
  • From: Iain Stevenson
• Re: [Asterisk-Users] Newbie - getting two local phones to communicate would be a good start :)
  • From: Balaji NJL
• [Asterisk-Users] expression parsing
  • From: ml
• Re: [Asterisk-Dev] More AGI problems - fails to stream file
  • From: Steven Critchfield
• Re: [Asterisk-Dev] Active call information
  • From: John Todd
• RE: [Asterisk-Users] New to asterisk? RUN... don't walk.
  • From: Me
• Re: [Asterisk-Dev] More AGI problems - fails to stream file
  • From: Iain Stevenson
• Re: [Asterisk-Dev] More AGI problems - fails to stream file
  • From: Steven Critchfield
• Re: [Asterisk-Dev] More AGI problems - fails to stream file
  • From: Mark Spencer
• Re: [Asterisk-Users] Grandstream Early Dial
  • From: Greg Boehnlein
• Re: [Asterisk-Users] New to asterisk? RUN... don't walk.
  • From: Tilghman Lesher
• [Asterisk-Users] Free PSTN calls
  • From: Isaac McDonald
• Re: [Asterisk-Users] New to asterisk? RUN... don't walk.
  • From: Me
• Re: [Asterisk-Users] New to asterisk? RUN... don't walk.
  • From: Tilghman Lesher
• Re: [Asterisk-Users] mini-ITX suggestions
  • From: Leo Ann Boon
• RE: [Asterisk-Users] New to asterisk? RUN... don't walk.
  • From: Steven Critchfield
• Re: [Asterisk-Users] Re: Cisco SIP license?
  • From: Dan Tucny
• RE: [Asterisk-Users] New to asterisk? RUN... don't walk.
  • From: Paul Mahler
• Re: [Asterisk-Users] STOP THIS THREAD New to asterisk? RUN... don't walk.
  • From: Balaji NJL
• [Asterisk-Users] New to asterisk? RUN... don't walk.
  • From: Justin Sinclair
• Re: [Asterisk-Users] New to asterisk? RUN... don't walk.
  • From: Steve Sobol
• [Asterisk-Users] TDM400P driver modprobe failed
  • From: Michael
• Re: [Asterisk-Users] Java?
  • From: Masakazu Nakano
• Re: [Asterisk-Users] New to asterisk? RUN... don't walk.
  • From: Brian Capouch
• [Asterisk-Dev] Re: [Asterisk-Users] Programming an unlocked ADSI phone?
  • From: Darren Nickerson
• Re: [Asterisk-Users] Residential router w/ QoS support?
  • From: Michael Graves
• Re: AW: AW: [Asterisk-Users] Snom 200 with two extns defined anyone?
  • From: Michael Graves
• Re: [Asterisk-Users] Programming an unlocked ADSI phone?
  • From: Tilghman Lesher
• [Asterisk-Dev] cdr_odbc.c deadlock
  • From: Brian West
• Re: [Asterisk-Dev] Active call information
  • From: Venkat Venkataraju
• Re: [Asterisk-Users] mini-ITX suggestions
  • From: Olle E. Johansson
• Re: AW: AW: [Asterisk-Users] Snom 200 with two extns defined anyone?
  • From: Olle E. Johansson
• [Asterisk-Users] CAPI, transfering thru a 2nd PBX - keep original CallerID
  • From: Peer Oliver schmidt
• [Asterisk-Users] TDM400P & X101P cards, echo issues?
  • From: Patrick Cantwell
• [Asterisk-Users] Hardware to build an Enterprise Asterisk Universal Gateway
  • From: EDWARD WILSON
• [Asterisk-Users] Modem Communications thru *
  • From: fred alexander
• RE: [Asterisk-Users] STOP THIS THREAD New to asterisk? RUN... don't walk.
  • From: Josh Roberson
• [Asterisk-Users] POTS interfacing recommendation
  • From: Matthew Bloch
• RE: [Asterisk-Users] STOP THIS THREAD New to asterisk? RUN... don't walk.
  • From: Josh Roberson
• Re: [Asterisk-Users] POTS interfacing recommendation
  • From: rnc Info Lists
• Re: [Asterisk-Users] POTS interfacing recommendation
  • From: Olle E. Johansson
• Re: [Asterisk-Users] New to asterisk? RUN... don't walk.
  • From: Steve Totaro
• Re: AW: AW: [Asterisk-Users] Snom 200 with two extns defined anyone?
  • From: Rich Adamson
• Re: [Asterisk-Users] POTS interfacing recommendation
  • From: Matthew Bloch
• Re: [Asterisk-Users] Newbie - getting two local phonestocommunicate would be a good start :)
  • From: Mike Jagdis
• RE: [Asterisk-Users] STOP THIS THREAD New to asterisk? RUN... don't walk.
  • From: Rich Adamson
• Re: [Asterisk-Users] Hardware to build an Enterprise Asterisk Universal Gateway
  • From: Steven Critchfield
• RE: [Asterisk-Users] New to asterisk? RUN... don't walk.
  • From: Sean Cheesman
• Re: [Asterisk-Users] Modem Communications thru *
  • From: Steven Critchfield
• Re: [Asterisk-Users] Hardware to build an Enterprise AsteriskUniversal Gateway
  • From: Doug Shubert
• Re: [Asterisk-Users] POTS interfacing recommendation
  • From: Steven Critchfield
• Re: [Asterisk-Users] Hardware to build an Enterprise AsteriskUniversal Gateway
  • From: Andrew Kohlsmith
• Re: [Asterisk-Users] Hardware to build an Enterprise AsteriskUniversal Gateway
  • From: WipeOut
• Re: AW: AW: [Asterisk-Users] Snom 200 with two extns defined anyone?
  • From: Michael Graves
• Re: [Asterisk-Users] Newbie - getting two local phonestocommunicate would be a good start :)
  • From: Olle E. Johansson
• Re: [Asterisk-Users] Java?
  • From: Philipp von Klitzing
• Re: [Asterisk-Users] Hardware to build an Enterprise Asterisk Universal Gateway
  • From: Olle E. Johansson
• [Asterisk-Users] Voicemail Out call
  • From: Kevin
• Re: [Asterisk-Users] CAPI, transfering thru a 2nd PBX - keep original CallerID
  • From: Philipp von Klitzing
• Re: [Asterisk-Users] Hardware to build an Enterprise AsteriskUniversal Gateway
  • From: Olle E. Johansson
• Re: [Asterisk-Users] Asterisk, enterprise edition (New subject)
  • From: Olle E. Johansson
• [Asterisk-Users] Two E100P boards - could not load zaptel module - Channel 63 - no such device
  • From: Daniel Bichara
• Re: [Asterisk-Users] Hardware to build an Enterprise AsteriskUniversal Gateway
  • From: Rich Adamson
• Re: [Asterisk-Users] Voicemail Out call
  • From: Iain Stevenson
• Re: [Asterisk-Users] Two E100P boards - could not load zaptel module - Channel 63 - no such device
  • From: Thilo Salmon
• RE: [Asterisk-Users] Two E100P boards - could not load zaptel module - Channel 63 - no such device
  • From: Scott Stingel
• [Asterisk-Users] help - recording both sides of a conversation
  • From: Paul Mahler
• [Asterisk-Users] OT: Anyone going to Open Source Telephony Summit in Geilenkirchen from North Germany?
  • From: Peer Oliver schmidt
• Re: [Asterisk-Users] Hardware to build an Enterprise Asterisk Universal Gateway
  • From: Steven Critchfield
• Re: [Asterisk-Users] CAPI, transfering thru a 2nd PBX - keep original CallerID
  • From: Peer Oliver schmidt
• Re: [Asterisk-Users] help - recording both sides of a conversation
  • From: Philipp von Klitzing
• Re: AW: AW: [Asterisk-Users] Snom 200 with two extns defined anyone?
  • From: Rich Adamson
• Re: [Asterisk-Users] Hardware to build an Enterprise AsteriskUniversal Gateway
  • From: Steven Critchfield
• Re: [Asterisk-Users] Voicemail Out call
  • From: Rich Adamson
• Re: [Asterisk-Users] help - recording both sides of a conversation
  • From: Iain Stevenson
• Re: [Asterisk-Users] Asterisk, enterprise edition (New subject)
  • From: Nick Bachmann
• Re: [Asterisk-Users] Hardware to build an Enterprise AsteriskUniversal Gateway
  • From: Nick Bachmann
• RE: [Asterisk-Users] OT: Anyone going to Open Source Telephony Summit in Geilenkirchen from North Germany?
  • From: Craig Waddington
• Re: [Asterisk-Users] Asterisk, enterprise edition (New subject)
  • From: Steven Critchfield
• Re: [Asterisk-Users] Hardware to build an Enterprise AsteriskUniversal Gateway
  • From: Rich Adamson
• Re: [Asterisk-Users] Asterisk, enterprise edition (New subject)
  • From: WipeOut
• Re: [Asterisk-Users] OT: Anyone going to Open Source Telephony Summit in Geilenkirchen from North Germany?
  • From: WipeOut
• Re: [Asterisk-Users] Hardware to build an Enterprise AsteriskUniversal Gateway
  • From: WipeOut
• Re: [Asterisk-Users] Hardware to build an Enterprise AsteriskUniversal Gateway
  • From: WipeOut
• Re: [Asterisk-Users] Asterisk, enterprise edition (New subject)
  • From: Nick Bachmann
• Re: [Asterisk-Users] OT: Anyone going to Open Source Telephony Summit in Geilenkirchen from North Germany?
  • From: Peer Oliver schmidt
• Re: [Asterisk-Users] Asterisk, enterprise edition (New subject)
  • From: Nick Bachmann
• Re: [Asterisk-Users] Hardware to build an Enterprise AsteriskUniversal Gateway
  • From: James Sharp
• Re: [Asterisk-Users] help - recording both sides of a conversation
  • From: John Baker
• Re: [Asterisk-Users] Hardware to build an Enterprise AsteriskUniversal Gateway
  • From: Steven Critchfield
• Re: [Asterisk-Users] Hardware to build an Enterprise AsteriskUniversalGateway
  • From: Doug Shubert
• Re: [Asterisk-Users] Hardware to build an Enterprise AsteriskUniversal Gateway
  • From: Nick Bachmann
• Re: [Asterisk-Users] help - recording both sides of a conversation
  • From: Brian West
• Re: [Asterisk-Users] New to asterisk? RUN... don't walk.
  • From: John Haigh
• [Asterisk-Dev] Privacy Code -- need reviews
  • From: Steve Murphy
• [Asterisk-Dev] Privacy upgrade-- need review
  • From: Steve Murphy
• RE: [Asterisk-Users] Newbie - getting two local phonestocommunicate would be a good start :)
  • From: John Coll
• RE: [Asterisk-Users] Newbie - getting two local phonestocommunicate would be a good start :)
  • From: John Coll
• [Asterisk-Users] Newbie - MWI
  • From: John Coll
• [Asterisk-Users] Newbie - MWI
  • From: John Coll
• Re: [Asterisk-Users] Asterisk, enterprise edition (New subject)
  • From: Philipp von Klitzing
• Re: [Asterisk-Users] Newbie - MWI
  • From: Olle E. Johansson
• Re: [Asterisk-Users] Hardware to build an Enterprise AsteriskUniversal Gateway
  • From: Rich Adamson
• Re: [Asterisk-Users] Newbie - MWI
  • From: Paul Liew
• Re: [Asterisk-Users] Re: Grandstream Early Dial
  • From: Aaron Martin
• RE: [Asterisk-Dev] FW: Welcome to the "Asterisk-Dev" mailing list (Digest mode)
  • From: woody+asterisk
• Re: [Asterisk-Users] Hardware to build an Enterprise AsteriskUniversal Gateway
  • From: James Sharp
• [Asterisk-Users] Cisco 12sp+ program update
  • From: Rohde
• [Asterisk-Users] Voicepulse DID fast busy
  • From: Steve Totaro
• RE: [Asterisk-Users] Newbie - MWI
  • From: John Coll
• [Asterisk-Dev] Segfault in chan_iax
  • From: Tilghman Lesher
• Re: [Asterisk-Dev] Privacy upgrade-- need review
  • From: Jeremy McNamara
• Re: [Asterisk-Dev] Segfault in chan_iax
  • From: Mark Spencer
• Re: [Asterisk-Users] Newbie - MWI
  • From: Andrew Thompson
• [Asterisk-Users] Multi-line help
  • From: Sean Garland
• RE: [Asterisk-Users] Newbie - MWI
  • From: Sean Cheesman
• [Asterisk-Users] Earpiece Connections
  • From: Michael
• [Asterisk-Users] Cisco to Cisco - poor quality
  • From: Terence Parker
• [Asterisk-Users] Dutch/DTMF Caller ID
  • From: Andy Powell
• Re: [Asterisk-Users] Cisco to Cisco - poor quality
  • From: Doug Shubert
• Re: [Asterisk-Users] Cisco to Cisco - poor quality
  • From: Jared Smith
• Re: [Asterisk-Users] Cisco to Cisco - poor quality
  • From: Terence Parker
• Re: [Asterisk-Users] Cisco 12sp+ program update
  • From: Steven Critchfield
• Re: [Asterisk-Users] Multi-line help
  • From: Steven Critchfield
• RE: [Asterisk-Users] Cisco to Cisco - poor quality
  • From: daryl
• Re: [Asterisk-Users] Cisco to Cisco - poor quality
  • From: Doug Shubert
• [Asterisk-Users] pager reminder script
  • From: firedude
• RE: [Asterisk-Users] Re: Grandstream Early Dial
  • From: ml
• [Asterisk-Users] Sun Servers with UltraSparc Processors
  • From: Adthrawn
• [Asterisk-Users] Sun Servers with UltraSparc Processors
  • From: Adthrawn
• [Asterisk-Users] 4 X100P Cards
  • From: Brent Franks
• Re: [Asterisk-Dev] Segfault in chan_iax
  • From: Rich Adamson
• [Asterisk-Users] Hold and transfer problem
  • From: Kevin Walsh
• [Asterisk-Users] RE: SIP + DTMF problem
  • From: Hamish Archer
• Re: [Asterisk-Users] Sun Servers with UltraSparc Processors
  • From: Tilghman Lesher
• Re: [Asterisk-Users] Multi-line help
  • From: Rich Adamson
• RE: [Asterisk-Users] Sun Servers with UltraSparc Processors
  • From: daryl
• Re: [Asterisk-Users] Hardware to build an Enterprise AsteriskUniversal Gateway
  • From: Rich Adamson
• Re: [Asterisk-Users] Multi-line help
  • From: Steven Critchfield
• [Asterisk-Users] Grandstream Handytone 286 RTP Problems
  • From: Mike Machado
• Re: [Asterisk-Users] Multi-line help
  • From: Nicholas Comanos
• [Asterisk-Users] Re: Sun Servers with UltraSparc Processors
  • From: Ish
• Re: [Asterisk-Dev] Segfault in chan_iax
  • From: Brian West
• Re: [Asterisk-Users] Cisco to Cisco - poor quality
  • From: Eric Wieling
• Re: [Asterisk-Users] Grandstream Handytone 286 RTP Problems
  • From: Masakazu Nakano
• Re: [Asterisk-Users] Grandstream Handytone 286 RTP Problems
  • From: John Baker
• Re: [Asterisk-Users] Hardware to build an Enterprise AsteriskUniversal Gateway
  • From: Steven Critchfield
• Re: [Asterisk-Users] Grandstream Handytone 286 RTP Problems
  • From: Mike Machado
• Re: [Asterisk-Users] Re: Sun Servers with UltraSparc Processors
  • From: James Sharp
• RE:[Asterisk-Users] Grandstream Handytone 286 RTP Problems
  • From: SW
• re: [Asterisk-Users] Cisco to Cisco - poor quality
  • From: Todd Taylor
• [Asterisk-Users] Codec Negotiation Does not seem to work as expected ?? Help Please !!
  • From: SamW
• [Asterisk-Users] RE: Inexpensive Analog Ports
  • From: Asterisk Newbie
• Re: [Asterisk-Users] Hardware to build an Enterprise AsteriskUniversal Gateway
  • From: Nicolas Bougues
• Re: [Asterisk-Users] Codec Negotiation Does not seem to work as expected ?? Help Please !!
  • From: Steven Critchfield
• Re: [Asterisk-Users] RE: Inexpensive Analog Ports
  • From: Jim Thompson
• Re: [Asterisk-Users] Newbie - getting two local phonestocommunicate would be a good start :)
  • From: Olle E. Johansson
• Re: [Asterisk-Users] RE: Inexpensive Analog Ports
  • From: Steven Critchfield
• Re: [Asterisk-Users] Voicepulse DID fast busy
  • From: Lion Templin
• [Asterisk-Users] "Internal" ISDN bus
  • From: Peer Oliver schmidt
• RE: [Asterisk-Users] "Internal" ISDN bus
  • From: Florian Overkamp
• Re: [Asterisk-Users] RE: Inexpensive Analog Ports
  • From: Eric Wieling
• RE: [Asterisk-Users] "Internal" ISDN bus
  • From: Eric Wieling
• Re: [Asterisk-Users] Hardware to build an Enterprise AsteriskUniversal Gateway
  • From: WipeOut
• RE: [Asterisk-Users] "Internal" ISDN bus
  • From: Florian Overkamp
• Re: [Asterisk-Users] CAPI, transfering thru a 2nd PBX - keep original CallerID
  • From: Philipp von Klitzing
• RE: [Asterisk-Users] RE: Inexpensive Analog Ports
  • From: Asterisk Newbie
• Re: [Asterisk-Users] Hold and transfer problem
  • From: Philipp von Klitzing
• FW: [Asterisk-Users] SIP to SIP redirect while ringing
  • From: Michael Devenijn
• [Asterisk-Users] CLIR and isdn4linux
  • From: Cristian Manoni
• Re: [Asterisk-Users] Multi-line help
  • From: Philipp von Klitzing
• RE: [Asterisk-Users] CLIR and isdn4linux
  • From: Florian Overkamp
• Re: [Asterisk-Dev] Current database abstraction effort ?
  • From: Nicolas Bougues
• Re: [Asterisk-Users] Hardware to build an Enterprise AsteriskUniversal Gateway
  • From: Doug Shubert
• RE: [Asterisk-Users] one way choppy sound problem !
  • From: Dawid Mielnik
• RE: [Asterisk-Dev] Current database abstraction effort ?
  • From: Florian Overkamp
• Re: [Asterisk-Users] Grandstream Handytone 286 RTP Problems
  • From: robert ivanc
• RE: [Asterisk-Users] Weirdness with CALLERID / CALLERIDNAME from incoming PRI
  • From: Adams, Gavin
• RE: [Asterisk-Users] Hold and transfer problem
  • From: Kevin Walsh
• Re: [Asterisk-Dev] Current database abstraction effort ?
  • From: Nicolas Bougues
• [Asterisk-Users] Open G.729(A) Initiative
  • From: robert ivanc
• [Asterisk-Dev] seeking advise on sigpipe in app_
  • From: Freddi Hansen
• RE: [Asterisk-Users] Open G.729(A) Initiative
  • From: Florian Overkamp
• Re: [Asterisk-Users] mailbox= wrong context. was: Newbie - MWI
  • From: Andrew Thompson
• Re: [Asterisk-Users] Hardware to build an Enterprise AsteriskUniversal Gateway
  • From: Rich Adamson
• Re: [Asterisk-Users] Earpiece Connections
  • From: Andrew Thompson
• RE: [Asterisk-Users] Codec Negotiation Does not seem to work as expected ?? Help Please !!
  • From: SW
• Re: [Asterisk-Dev] Current database abstraction effort ?
  • From: James H. Cloos Jr.
• Re: [Asterisk-Users] Multi-line help
  • From: Siggi Langauf
• RE: [Asterisk-Dev] Current database abstraction effort ?
  • From: Florian Overkamp
• [Asterisk-Users] DID Trunk Lines and Caller ID
  • From: Michael
• Re: [Asterisk-Users] 4 X100P Cards
  • From: Michael Graves
• [Asterisk-Users] Re: echo
  • From: Stephen R. Besch
• [Asterisk-Dev] Some patches posted to bugs.digium.com
  • From: Stephen Davies
• Re: [Asterisk-Users] one way choppy sound problem !
  • From: Michael Van Donselaar
• Re: [Asterisk-Users] mailbox= wrong context. was: Newbie - MWI
  • From: Jared Smith
• Re: [Asterisk-Users] DID Trunk Lines and Caller ID
  • From: Steven Critchfield
• Re: [Asterisk-Dev] Some patches posted to bugs.digium.com
  • From: Martin Kihlgren
• Re: [Asterisk-Users] Hardware to build an Enterprise AsteriskUniversal Gateway
  • From: Martin Bene
• Re: [Asterisk-Users] one way choppy sound problem !
  • From: WipeOut
• [Asterisk-Users] Sip Trunking
  • From: Eduardo Goncalves
• [Asterisk-Users] Question about MP3's
  • From: B. J. Bomar
• Re: [Asterisk-Users] DID Trunk Lines and Caller ID
  • From: Jared Smith
• RE: AW: AW: [Asterisk-Users] Snom 200 with two extns defined anyone?
  • From: VoiceLynx
• RE: [Asterisk-Users] Codec Negotiation Does not seem to work as expected ?? Help Please !!
  • From: Samath
• RE: [Asterisk-Users] one way choppy sound problem !
  • From: Dawid Mielnik
• Re: [Asterisk-Dev] Some patches posted to bugs.digium.com
  • From: Florian Overkamp
• Re: [Asterisk-Users] mailbox= wrong context. was: Newbie - MWI
  • From: Olle E. Johansson
• Re: [Asterisk-Users] Sip Trunking
  • From: Jared Smith
• Re: [Asterisk-Users] mailbox= wrong context. was: Newbie - MWI
  • From: Olle E. Johansson
• Re: [Asterisk-Users] Sip Trunking
  • From: Brian West
• Re: [Asterisk-Users] one way choppy sound problem !
  • From: Nicolas Gudino
• Re: [Asterisk-Users] DID Trunk Lines and Caller ID
  • From: Steven Critchfield
• RE: [Asterisk-Users] one way choppy sound problem !
  • From: Dawid Mielnik
• Re: [Asterisk-Users] Sip Trunking
  • From: Steven Critchfield
• RE: [Asterisk-Users] one way choppy sound problem !
  • From: Steve Dolloff
• Re: [Asterisk-Users] AGI - IVR - Time Clock
  • From: Jason A. Pattie
• RE: [Asterisk-Users] RE: Inexpensive Analog Ports
  • From: Paul Mahler
• Re: [Asterisk-Users] Question about MP3's
  • From: Steven Critchfield
• Re: [Asterisk-Dev] Current database abstraction effort ?
  • From: Chris Albertson
• [Asterisk-Users] Re: 486 Busy message - SNOM 200
  • From: Matt Lawson
• Re: [Asterisk-Users] one way choppy sound problem !
  • From: TeleSIP
• [Asterisk-Dev] Re: Privacy upgrade-- need review-- bug report 752
  • From: Steve Murphy
• [Asterisk-Users] RE: DID Trunk Lines and Caller ID
  • From: Kekin Dand
• AW: AW: AW: [Asterisk-Users] Snom 200 with two extns defined anyone?
  • From: Christian Stredicke
• RE: [Asterisk-Users] DID Trunk Lines and Caller ID
  • From: Tim Thompson
• RE: [Asterisk-Users] mailbox= wrong context. was: Newbie - MWI
  • From: Sean Cheesman
• Re: [Asterisk-Users] Sip Trunking
  • From: Eduardo Goncalves
• Re: [Asterisk-Users] Codec Negotiation Does not seem to work as e xpected ?? Help Please !!
  • From: SamW
• Re: [Asterisk-Users] mailbox= wrong context. was: Newbie - MWI
  • From: Steven Critchfield
• Re: [Asterisk-Users] mailbox= wrong context. was: Newbie - MWI
  • From: Steven Critchfield
• Re: [Asterisk-Users] Sip Trunking
  • From: Eduardo Goncalves
• Re: [Asterisk-Users] mailbox= wrong context. was: Newbie - MWI
  • From: Andrew Thompson
• [Asterisk-Users] queue questions: max time in queue; customer option to drop out of queue
  • From: Ken Alker
• Re: [Asterisk-Users] AGI - IVR - Time Clock
  • From: Michiel Betel
• RE: [Asterisk-Users] Grandstream Quality Survey.... :P
  • From: Rick Smith
• Re: [Asterisk-Dev] Current database abstraction effort ?
  • From: Steven Critchfield
• [Asterisk-Users] reject connect from iaxtel.com
  • From: dkwok@xxxxxxxxxxxx
• Re: [Asterisk-Users] one way choppy sound problem !
  • From: WipeOut
• Re: [Asterisk-Users] help - recording both sides of a conversation
  • From: zoa
• [Asterisk-Users] I stumbled on this list...
  • From: calvis
• [Asterisk-Users] Echo with polycom phones
  • From: Sean Garland
• [Asterisk-Users] question re voicemail
  • From: Jess Magnaye
• [Asterisk-Users] Queue only ringing one agent at a time
  • From: Ken Alker
• Re: [Asterisk-Users] Sip Trunking
  • From: Steven Critchfield
• [Asterisk-Users] HTML Stripping in mailing lists?
  • From: Kannaiyan Natesan
• [Asterisk-Users] FW: This newbie gives up for now - sadly (2)
  • From: John Coll
• [Asterisk-Users] This newbie gives up for now - sadly
  • From: John Coll
• [Asterisk-Users] Are messages censored on this board?
  • From: John Coll
• [Asterisk-Users] Lindows ?
  • From: Francisco Perez-Landaeta
• [Asterisk-Users] Need Help...
  • From: Jess Magnaye
• [Asterisk-Users] MeetMe problem
  • From: Michael Graves
• [Asterisk-Dev] waitpid in app_agi.c
  • From: marco . parisotto
• [Asterisk-Dev] Crash/backtrace in app_voicemail.c / res_adsi.c
  • From: Tilghman Lesher
• [Asterisk-Users] Re: Earpiece Connections
  • From: Michael
• RE: [Asterisk-Users] This newbie gives up for now - sadly
  • From: Tim Thompson
• RE: [Asterisk-Users] Multi-line help
  • From: Sean Garland
• [Asterisk-Dev] waitpid in app_agi.c
  • From: marco.parisotto@xxxxxxxxxxxxxxxx
• Re: [Asterisk-Users] This newbie gives up for now - sadly
  • From: Tilghman Lesher
• [Asterisk-Users] This is a test
  • From: Sean Garland
• [Asterisk-Users] Echo on polycom sip phone
  • From: Sean Garland
• Re: [Asterisk-Users] I stumbled on this list...
  • From: Doug Heckaman III
• Re: [Asterisk-Dev] Current database abstraction effort ?
  • From: Brian West
• Re: [Asterisk-Dev] Current database abstraction effort ?
  • From: Chris Albertson
• Re: [Asterisk-Dev] Current database abstraction effort ?
  • From: Brian West
• Re: [Asterisk-Users] Cisco to Cisco - poor quality
  • From: Terence Parker
• Re: [Asterisk-Users] I stumbled on this list...
  • From: Nick Bachmann
• Re: [Asterisk-Users] I stumbled on this list...
  • From: Brian West
• Re: [Asterisk-Users] This newbie gives up for now - sadly
  • From: TC
• [Asterisk-Users] How to monitor calls initiated by .call file using manager interface?
  • From: Serge Mankovski
• [Asterisk-Users] asterisk sccp support
  • From: Martin Bene
• [Asterisk-Users] Identifying the Originating Cisco SIP Gateway
  • From: Ray Burkholder
• Re: [Asterisk-Users] question re voicemail
  • From: Brian West
• RE: [Asterisk-Users] question re voicemail
  • From: Sean Cheesman
• Re: [Asterisk-Users] I stumbled on this list...
  • From: Youness El Andaloussi
• [Asterisk-Users] Message waiting indicator
  • From: Sean Garland
• Re: [Asterisk-Users] Cisco 12sp+ program update
  • From: Rohde
• Re: [Asterisk-Users] Are messages censored on this board?
  • From: Jess Magnaye
• RE: [Asterisk-Users] Are messages censored on this board?
  • From: Sean Cheesman
• RE: [Asterisk-Users] Multi-line help
  • From: Sean Garland
• Re: [Asterisk-Users] Cisco to Cisco - poor quality
  • From: Terence Parker
• Re: [Asterisk-Users] reject connect from iaxtel.com
  • From: Rich Adamson
• Re: [Asterisk-Users] Are messages censored on this board?
  • From: Brian Christie
• Re: [Asterisk-Users] Are messages censored on this board?
  • From: Adam Hart
• Re: [Asterisk-Users] This newbie gives up for now - sadly
  • From: Derek
• RE: [Asterisk-Users] Grandstream Quality Survey.... :P
  • From: Ken Alker
• Re: [Asterisk-Users] This is a test
  • From: Doug Heckaman III
• Re: [Asterisk-Users] Sip Trunking
  • From: Steven Critchfield
• Re: [Asterisk-Users] Are messages censored on this board?
  • From: Doug Heckaman III
• [Asterisk-Users] problems dialing area code
  • From: Dave Thomas
• RE: [Asterisk-Users] Multi-line help
  • From: Rich Adamson
• Re: [Asterisk-Users] Grandstream Handytone 286 RTP Problems - FIXED
  • From: Mike Machado
• Re: [Asterisk-Users] This newbie gives up for now - sadly
  • From: Robert Hajime Lanning
• Re: [Asterisk-Users] Are messages censored on this board?
  • From: Tilghman Lesher
• RE: [Asterisk-Users] Are messages censored on this board?
  • From: woody+asterisk
• [Asterisk-Users] How to monitor calls initiated by .call file using manager interface?
  • From: Serge Mankovski
• RE: [Asterisk-Users] This is a test
  • From: woody+asterisk
• [Asterisk-Users] Special variable for AGENT
  • From: Eris Riswanto
• Re: [Asterisk-Users] Identifying the Originating Cisco SIP Gateway
  • From: Andres
• RE: [Asterisk-Users] I stumbled on this list...
  • From: Birlasekaran Dinesh
• Re: [Asterisk-Dev] Current database abstraction effort ?
  • From: Chris Albertson
• [Asterisk-Users] [Fwd: reject connect from iaxtel.com]
  • From: dkwok
• [Asterisk-Users] Interfacing Asterisk with PSTN network (Nortel SL100 PBX)
  • From: tony banks
• [Asterisk-Users] "Everyone is busy at this time" message ?
  • From: tony banks
• RE: [Asterisk-Users] FW: This newbie gives up for now - sadly (2)
  • From: Adam Goryachev
• [Asterisk-Users] IVR Question
  • From: tony banks
• [Asterisk-Dev] aes* files in CVS have DOS line endings
  • From: Stephen Davies
• Re: [Asterisk-Users] I stumbled on this list...
  • From: Balaji NJL
• Re: [Asterisk-Users] This newbie gives up for now - sadly
  • From: Olle E. Johansson
• Re: [Asterisk-Users] "Everyone is busy at this time" message ?
  • From: Olle E. Johansson
• Re: [Asterisk-Users] IVR Question
  • From: Olle E. Johansson
• [Asterisk-Dev] [Fwd: Re: [openss7]Project: Asterisk PBX]
  • From: Olle E. Johansson
• Re: [Asterisk-Dev] Current database abstraction effort ?
  • From: Nicolas Bougues
• Re: [Asterisk-Users] How to monitor calls initiated by .call file using manager interface?
  • From: Iain Stevenson
• Re: [Asterisk-Dev] Current database abstraction effort ?
  • From: Steven Critchfield
• RE: [Asterisk-Dev] [Fwd: Re: [openss7]Project: Asterisk PBX] and line-signalling ?
  • From: Florian Overkamp
• Re: [Asterisk-Users] SUBSCRIBE in chan_sip - anyone?
  • From: Olle E. Johansson
• RE: [Asterisk-Users] SUBSCRIBE in chan_sip - anyone?
  • From: Florian Overkamp
• Re: [Asterisk-Users] SUBSCRIBE in chan_sip - anyone?
  • From: Olle E. Johansson
• Re: [Asterisk-Users] Dial via sip gateway?
  • From: Rich Adamson
• Re: [Asterisk-Users] SUBSCRIBE in chan_sip - anyone?
  • From: Olle E. Johansson
• Re: [Asterisk-Users] Dial via sip gateway?
  • From: Rich Adamson
• Re: [Asterisk-Users] Dial via sip gateway?
  • From: Olle E. Johansson
• Re: [Asterisk-Dev] Using an additional modem to get CallerID information
  • From: Conroy, Lawrence (SMTP)
• [Asterisk-Dev] SUBSCRIBE support
  • From: Olle E. Johansson