April 2005 Archives by date

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Starting: Fri Apr 1 00:00:03 MST 2005
Ending: Sat Apr 30 23:53:15 MST 2005
Messages: 6063

• [Asterisk-Users] Are there online forums instead of thisemailforum??   Francesco Peeters (signed)
• [Asterisk-Users] Re: Are there online forums instead of this   Greg
• [Asterisk-Users] User Regerstation, allowing non-registered users on *   Etienne Pretorius
• [Asterisk-Users] snom220   Altus Snyman
• [Asterisk-Users] Re: Are there online forums instead of this   Tim Bass
• [Asterisk-Users] Re: Are there online forums instead of this   Brian Capouch
• [Asterisk-Users] Ztdummy is Loaded but Asterisk is not using it   Mario Spendier
• [Asterisk-Users] Preserve g729 registration over reinstall??   Adam Goryachev
• [Asterisk-Users] Parking no   Luca Maccarini
• [Asterisk-Users] Are there online forums instead of thisemailforum??   Jean-Michel Hiver
• [Asterisk-Users] *** Asterisk 2.0 Stable release out now   Olle E. Johansson
• [Asterisk-Users] VoIP Provider problems   Johnathan Corgan
• [Asterisk-Users] register => with realtime   G.Marshall
• [Asterisk-Users] NuFone, VoIPJet, circuit (fast) busy question   Mike Benoit
• [Asterisk-Users] *** Asterisk 2.0 Stable release out now   Michael Manousos
• [Asterisk-Users] Playback starts before call answer   Chris Blake
• [Asterisk-Users] setting SIP to dial PSTN with TDM400P   Muhammad Haris
• [Asterisk-Users] Installing CAPI   Craig Guy
• [Asterisk-Users] *** Asterisk 2.0 Stable release out now   Dave Cotton
• [Asterisk-Users] sip.conf match   Pepe Aracil
• [Asterisk-Users] really small box   Irakli Natsvlishvili
• [Asterisk-Users] really small box   Jean-Michel Hiver
• [Asterisk-Users] Problems getting FXO channel working - Unable to create channel of type 'Zap' (cause 0)   Mark van Kerkwyk
• [Asterisk-Users] *** Asterisk 2.0 Stable release out now   Chris Hills
• [Asterisk-Users] using unixODBC   Kamran Ahmad
• [Asterisk-Users] RE: Asterisk Realtime - configuration help   Kristian Nielsen
• [Asterisk-Users] *** Asterisk 2.0 Stable release out now   Remco Barende
• [Asterisk-Users] setting SIP to dial PSTN with TDM400P   Martijn van Oosterhout
• [Asterisk-Users] chan_capi looking for missing channel_pvt.h   Jason Williams
• [Asterisk-Users] Problem with dial out via chan_capi   Kib Eki
• [Asterisk-Users] Sangoma VS. Digium   Andrew Thrift
• [Asterisk-Users] Re: Are there online forums instead of this email forum??   Tony Mountifield
• [Asterisk-Users] Re: Are there online forums instead of thisemailforum??   Tony Mountifield
• [Asterisk-Users] Re: Are there online forums instead of this emailforum??   Tony Mountifield
• [Asterisk-Users] *** Asterisk 2.0 Stable release out now   Matthew Boehm
• [Asterisk-Users] RE: Asterisk Realtime - configuration help   Matthew Boehm
• [Asterisk-Users] Re: Are there online forums instead of thisemailforum??   Tony Mountifield
• [Asterisk-Users] patlooptest: Usage, setup?   Eric Wieling aka ManxPower
• [Asterisk-Users] Are there online forums instead of this   Martijn van Oosterhout
• [Asterisk-Users] *** Asterisk 2.0 Stable release out now   Irakli Natsvlishvili
• [Asterisk-Users] H.323 call '...' cleared, reason 15 (Call ended due to security checks)   Cenk Yabas
• [Asterisk-Users] using unixODBC   Thierry Wehr
• [Asterisk-Users] Re: Are there online forums instead of this email forum??   Francesco Peeters
• [Asterisk-Users] Re: Are there online forums instead of thisemailforum??   Francesco Peeters (signed)
• [Asterisk-Users] [Fwd: Problem with dial out via chan_capi]   Kib Eki
• [Asterisk-Users] Eicon Diva Server BRI Setup   dorian logan
• [Asterisk-Users] H.323 call '...' cleared, reason 15 (Call ended due to security checks)   Michael Manousos
• [Asterisk-Users] SuperMicro X5DE8-GG Motherboard Goes Kaput afterInstalling TE410P Card - Yikes!   Steven Critchfield
• [Asterisk-Users] Re: Problem with dial out via chan_capi   Sergio
• [Asterisk-Users] queue.conf config   Obihuan
• [Asterisk-Users] Re: Eicon Diva Server BRI Setup   Kib Eki
• [Asterisk-Users] Eicon Diva Server BRI Setup   Kib Eki
• [Asterisk-Users] Re: Problem with dial out via chan_capi   Kib Eki
• [Asterisk-Users] Re: Problem with dial out via chan_capi   Eric Wieling aka ManxPower
• [Asterisk-Users] Are there online forums instead of this   Rich Adamson
• [Asterisk-Users] ${DIALSTATUS}   Manuel Schroeder
• [Asterisk-Users] Are there online forums instead of thisemailforum??   Andrew Kohlsmith
• [Asterisk-Users] [OT] Announcing MidwestTea.com   Art Zemon
• [Asterisk-Users] LDAP and Asterisk   Rob Scott
• [Asterisk-Users] "Unable to create channel of type Zap" Message   Andres Tello Abrego
• [Asterisk-Users] really small box   Matt Ryanczak
• [Asterisk-Users] Are there online forums instead of this email   Bruno Hertz
• Optimizing speex (was Re: [Asterisk-Users] Erratic CPU load )   Eric Giesselbach
• [Asterisk-Users] Livevoip still no DTMF?   Rich Adamson
• [Asterisk-Users] Are there online forums instead of this email   Francesco Peeters
• [Asterisk-Users] Xten-lite for linux   Dana Olson
• [Asterisk-Users] Phones "Callwaiting" enable by default?   Matt
• [Asterisk-Users] Re: Are there online forums instead of this   Bruno Hertz
• [Asterisk-Users] Problem with livevoip dial out   Rich Adamson
• [Asterisk-Users] Re: Polycom sound quality problems   Noah Miller
• [Asterisk-Users] Maybe an echo cancellation problem?   1 2
• [Asterisk-Users] Re: Are there online forums instead of this email forum??   Bruno Hertz
• [Asterisk-Users] Caller ID on voicemail messages   Matt Ryanczak
• [Asterisk-Users] Dial'ing multiple SIP devices impossible when forward activated   Louis-David Mitterrand
• [Asterisk-Users] Re: Are there online forums instead ofthisemailforum??   Tim Bass
• [Asterisk-Users] Re: Are there online forums insteadofthisemailforum??   Mark Charlton
• [Asterisk-Users] Re: Are there online forums instead ofthisemailforum??   Andrew Kohlsmith
• [Asterisk-Users] Re: Are there online forums insteadofthisemailforum??   Tim Bass
• [Asterisk-Users] Looping messages   Chris Blake
• [Asterisk-Users] Maybe an echo cancellation problem?   Eric Giesselbach
• [Asterisk-Users] Are there online forums instead of this email   Bruno Hertz
• [Asterisk-Users] Re: Are there online forums instead ofthisemailforum??   Steve Underwood
• [Asterisk-Users] Q.931 to SIGTRAN interface   Mike Mueller
• [Asterisk-Users] Are there online forums instead of this email   Tim Bass
• [Asterisk-Users] *** Asterisk 2.0 Stable release out now   steve szmidt
• [Asterisk-Users] Xten-lite for linux   Bruno Hertz
• [Asterisk-Users] really small box   Loucas Gatzoulis
• [Asterisk-Users] Re: Phones "Callwaiting" enable by default?   Noah Miller
• [Asterisk-Users] Timecard application   Chuck Bunn
• [Asterisk-Users] Re: Are there online forumsinstead ofthisemailforum??   Tim Bass
• [Asterisk-Users] VOIP to the PBX   Time Bandit
• [Asterisk-Users] Re: Are there online forumsinstead ofthisemailforum??   Steve Underwood
• [Asterisk-Users] *** Asterisk 2.0 Stable release out now   Michael Devenijn
• [Asterisk-Users] Are there online forums instead of this email forum??   Chuck Bunn
• [Asterisk-Users] Re: Are there online forums instead of this   Walt Reed
• [Asterisk-Users] Eicon Diva Server BRI Setup   dorian logan
• [Asterisk-Users] *** Asterisk 2.0 Stable release out now   Dave Cotton
• [Asterisk-Users] Sangoma VS. Digium   David Brodbeck
• [Asterisk-Users] *** Asterisk 2.0 Stable release out now   Nir Siminovich
• [Asterisk-Users] new release of chan_misdn !   Thomas Häger
• [Asterisk-Users] Zyxel Prestige 2002 (ATA)   Thore
• [Asterisk-Users] blind transfer question   Cirelle Internet Products
• [Asterisk-Users] Snom and Multiple calls   Josh Dady
• [Asterisk-Users] Are there online forums instead of this email   Incoming
• [Asterisk-Users] Erratic CPU load   Eric Giesselbach
• [Asterisk-Users] VOIP to the PBX   tmassey at obscorp.com
• [Asterisk-Users] Voicemail Email Bouncing   Hadar Pedhazur
• [Asterisk-Users] Specify Codec In Outbount Calls?   Linn Boyd
• [Asterisk-Users] Re: Snom and Multiple calls   Noah Miller
• [Asterisk-Users] ${DIALSTATUS}   Cirelle Internet Products
• [Asterisk-Users] Does asterisk@home support Dual-Processor installations?   Roger Hanson
• [Asterisk-Users] queue.conf config   Sean A. Newton
• [Asterisk-Users] Looping messages   MF Hulber
• [Asterisk-Users] Caller ID on voicemail messages   tmassey at obscorp.com
• [Asterisk-Users] Soyo G668 + Asterisk   Leonardo Gomes Figueira
• [Asterisk-Users] really small box   Matt Ryanczak
• [Asterisk-Users] Are there online forums instead of this email   Tim Bass
• [Asterisk-Users] Re: Are there online forumsinstead ofthisemailforum??   John Novack
• [Asterisk-Users] Re: LDAP and Asterisk   Iassen Hristov
• [Asterisk-Users] Specify Codec In Outbount Calls?   Rich Adamson
• [Asterisk-Users] Does asterisk@home support Dual-Processorinstallations?   Brent DeShazer
• [Asterisk-Users] Does asterisk@home support Dual-Processor installations?   John Novack
• [Asterisk-Users] queue.conf config   lenz
• [Asterisk-Users] Re: Polycom sound quality problems   Eric Mason
• [Asterisk-Users] MF Trunk Signaling   Michael Baird
• [Asterisk-Users] Re: Snom and Multiple calls   Josh Dady
• [Asterisk-Users] Sending DTMF back in a dialed/answered channel before bridging a call   BJ Weschke
• [Asterisk-Users] Re: Polycom sound quality problems   Wiley Siler
• [Asterisk-Users] Re: Dial'ing multiple SIP devices impossible when forward activated   Louis-David Mitterrand
• [Asterisk-Users] ADSI Input from 480 keypad?   Jimmy
• [Asterisk-Users] really small box   Wiley Siler
• [Asterisk-Users] What's the use of a multi line phone?   Andres Tello Abrego
• [Asterisk-Users] Are there online forums instead of this email   Andrew Kohlsmith
• [Asterisk-Users] Integrating door intercom?   Jeff Noxon
• [Asterisk-Users] Must be configuring something wrong spanDSP rxfax   Derrick Knight
• [Asterisk-Users] Re: Polycom sound quality problems   Wiley Siler
• [Asterisk-Users] Re: Livevoip still no DTMF?   Joel Jn-Francois
• [Asterisk-Users] "Unable to create channel of type Zap" Message   eng.yousri at egypt.com
• [Asterisk-Users] Re: Enhanced Queue App Revisited   Miguel Ruiz Velasco Sobrino
• [Asterisk-Users] really small box   Mike Hammett
• [Asterisk-Users] Voicemail Email Bouncing - SOLVED   Hadar Pedhazur
• [Asterisk-Users] *** Asterisk 2.0 Stable release out now   Mike Hammett
• [Asterisk-Users] Is this possible?   Paul
• [Asterisk-Users] Re: Snom and Multiple calls   Noah Miller
• [Asterisk-Users] Re: Polycom sound quality problems   Noah Miller
• [Asterisk-Users] Re: Snom and Multiple calls   Josh Dady
• [Asterisk-Users] Re: Are there online forums instead of this email forum??   Tom Ivar Helbekkmo
• [Asterisk-Users] Sending DTMF back in a dialed/answered channel before bridging a call   Ed Greenberg
• [Asterisk-Users] MF Trunk Signaling   James Taylor
• [Asterisk-Users] MF Trunk Signaling   Michael Baird
• [Asterisk-Users] Are there online forums instead of thisemailforum??   John Millican
• [Asterisk-Users] Livevoip still no DTMF?   Rich Adamson
• [Asterisk-Users] *** Asterisk 2.0 Stable release out now   Walt Reed
• [Asterisk-Users] *** Asterisk 2.0 Stable release out now   Brian C. Fertig
• [Asterisk-Users] Faxing through Broadvoice - HT286   Scott Wolfe
• RES: [Asterisk-Users] *** Asterisk 2.0 Stable release out now   Diego Aguirre
• [Asterisk-Users] Are there online forums instead of this email   Francesco Peeters
• [Asterisk-Users] *** Asterisk 2.0 Stable release out now   Paul
• [Asterisk-Users] Re: Livevoip still no DTMF?   Eric Wieling aka ManxPower
• [Asterisk-Users] Re: Snom and Multiple calls   Noah Miller
• [Asterisk-Users] *** Asterisk 2.0 Stable release out now   Paul
• [Asterisk-Users] Is this possible?   Cirelle Internet Products
• [Asterisk-Users] Re: Livevoip still no DTMF?   Andrew Kohlsmith
• [Asterisk-Users] *** Asterisk 2.0 Stable release out now   Karl J. Vesterling
• [Asterisk-Users] Re: Livevoip still no DTMF?   Eric Wieling aka ManxPower
• [Asterisk-Users] Issues with ringing on FXS ports   Ian Pattison
• [Asterisk-Users] Re: Snom and Multiple calls   Josh Dady
• [Asterisk-Users] Are there online forums instead of this email   Kenneth Porter
• [Asterisk-Users] Are there online forums instead of this email   Kenneth Porter
• [Asterisk-Users] Re: Snom and Multiple calls   Noah Miller
• [Asterisk-Users] Re: Snom and Multiple calls   Josh Dady
• [Asterisk-Users] Re: Livevoip still no DTMF?   Brandon Patterson
• [Asterisk-Users] Re: Livevoip still no DTMF?   Eric Wieling aka ManxPower
• [Asterisk-Users] Re: Livevoip still no DTMF?   Andrew Kohlsmith
• [Asterisk-Users] Ztdummy is Loaded but Asterisk is not using it   Paul Dugas
• [Asterisk-Users] *** Asterisk 2.0 Stable release out now   Kenneth Porter
• [Asterisk-Users] Re: Livevoip still no DTMF?   geek at mnwifi.net
• [Asterisk-Users] Make voicemail use Maildir...   Tim Howell
• [Asterisk-Users] Are there online forums instead of this email   Tim Bass
• [Asterisk-Users] Re: Livevoip still no DTMF?   Brandon Patterson
• [Asterisk-Users] Is this possible?   Paul
• [Asterisk-Users] Issues with ringing on FXS ports   Richard Scobie
• [Asterisk-Users] Codec not negotiating   Clay Reiche
• [Asterisk-Users] *** Asterisk 2.0 Stable release out now   Angel Diaz
• [Asterisk-Users] Maybe an echo cancellation problem?   1 2
• [Asterisk-Users] Is this possible?   tmassey at obscorp.com
• [Asterisk-Users] Are there online forums instead of this email   Francesco Peeters
• [Asterisk-Users] Re: Phones "Callwaiting" enable by default?   Matt
• [Asterisk-Users] Sangoma VS. Digium   Richard Scobie
• [Asterisk-Users] MF Trunk Signaling   James Taylor
• [Asterisk-Users] How to determine the voicemail file name for an AGI script   Bryan Boatright
• [Asterisk-Users] Re: Livevoip still no DTMF?   Moody
• [Asterisk-Users] "Unable to create channel of type Zap" Message   eng.yousri at egypt.com
• [Asterisk-Users] Re: Phones "Callwaiting" enable by default?   Noah Miller
• [Asterisk-Users] Integrating door intercom?   C F
• [Asterisk-Users] "Unable to create channel of type Zap" Message   eng.yousri at egypt.com
• [Asterisk-Users] Parking no   C F
• [Asterisk-Users] Re: Livevoip still no DTMF?   Rich Adamson
• [Asterisk-Users] What's the use of a multi line phone?   Remco Barende
• [Asterisk-Users] Re: Are there online forums instead of this email   Tom Ivar Helbekkmo
• [Asterisk-Users] What's the use of a multi line phone?   Wiley Siler
• [Asterisk-Users] Squeaking / chirping on ZAP Digium TDM400P   Kellner, Peter
• [Asterisk-Users] Are there online forums instead of this email   John Novack
• [Asterisk-Users] *** Asterisk 2.0 Stable release out now   Wiley Siler
• [Asterisk-Users] Re: Are there online forums instead of this email   Tom Ivar Helbekkmo
• [Asterisk-Users] Re: Are there online forums instead of this email   Francesco Peeters
• [Asterisk-Users] Squeaking / chirping on ZAP Digium TDM400P   Robert Webb
• [Asterisk-Users] Re: What's the use of a multi line phone?   Noah Miller
• [Asterisk-Users] Re: Are there online forums instead of this email   Tim Bass
• [Asterisk-Users] Re: Are there online forums instead of this email   Tim Bass
• [Asterisk-Users] Squeaking / chirping on ZAP Digium TDM400P   Kellner, Peter
• [Asterisk-Users] Re: Are there online forums instead of this email forum??   Tony Mountifield
• [Asterisk-Users] Datafire 2977   Tyler
• [Asterisk-Users] Codec not negotiating   Alex Vishnev
• [Asterisk-Users] What's the use of a multi line phone?   Race Vanderdecken
• [Asterisk-Users] Are there online forums instead of this email   Francesco Peeters
• [Asterisk-Users] Q.931 to SIGTRAN interface   Race Vanderdecken
• [Asterisk-Users] Re: Are there online forums instead of this email   JD
• [Asterisk-Users] Call bridging   Paul
• [Asterisk-Users] What's the use of a multi line phone?   Wiley Siler
• [Asterisk-Users] Preserving CallerID when forwarding to cellphone   Robert Goodyear
• [Asterisk-Users] Issues with ringing on FXS ports   tmassey at obscorp.com
• [Asterisk-Users] Re: Polycom sound quality problems   Max W Blackmer Jr
• [Asterisk-Users] sangoma S508/FT1 ISA   Michael Bielicki
• [Asterisk-Users] Preserving CallerID when forwarding to cellphone   Paul
• [Asterisk-Users] Re: Polycom sound quality problems   Wiley Siler
• [Asterisk-Users] Squeaking / chirping on ZAP Digium TDM400P   Kristian Kielhofner
• [Asterisk-Users] Strange DTA problem   Kerry Garrison
• [Asterisk-Users] Re: Are there online forums instead of this email forum??   Bruno Hertz
• [Asterisk-Users] Preserving CallerID when forwarding to cellphone   Robert Goodyear
• [Asterisk-Users] echo paid support   Damon Estep
• [Asterisk-Users] ACD Functionality   Steve Mann
• [Asterisk-Users] Random outbound:   Tim Connolly
• [Asterisk-Users] Random outbound:   Andres
• [Asterisk-Users] Squeaking / chirping on ZAP Digium TDM400P   Brian McSpadden
• [Asterisk-Users] *** Asterisk 2.0 Stable release out now   Carlos Chavez
• [Asterisk-Users] Random outbound:   Ronald Wiplinger
• [Asterisk-Users] *** Asterisk 2.0 Stable release out now   Kris Stark
• [Asterisk-Users] OT(?) Your subject line   Ronald Wiplinger
• [Asterisk-Users] OT(?) Your subject line   Don Murray
• [Asterisk-Users] Squeaking / chirping on ZAP Digium TDM400P   Kellner, Peter
• [Asterisk-Users] Display agent logged into a queue on the phone   Justin Zimmerman
• [Asterisk-Users] Mitel Ip phone ?   Pedro
• [Asterisk-Users] Q.931 to SIGTRAN interface   Mike Mueller
• [Asterisk-Users] Datafire 2977   Kevin P. Fleming
• [Asterisk-Users] IAX messages   Rob Roschewsk
• [Asterisk-Users] sangoma S508/FT1 ISA   Francois Menard (Mailing List Account)
• [Asterisk-Users] Implant GIPS's codec to Asterisk   Francois Menard (Mailing List Account)
• [Asterisk-Users] Are there online forums instead of this emailforum??   Jim Van Meggelen
• [Asterisk-Users] Is this possible?   Ryan Burke
• [Asterisk-Users] Queues   David Choo
• [Asterisk-Users] Is this possible?   Andrew Kohlsmith
• [Asterisk-Users] Realtime   David Choo
• [Asterisk-Users] Is this possible?   Robert Keller
• [Asterisk-Users] Is this possible?   Kerry Garrison
• [Asterisk-Users] Squeaking / chirping on ZAP Digium TDM400P   Sascha Ferley
• [Asterisk-Users] Queues   Henry Devito
• [Asterisk-Users] Squeaking / chirping on ZAP Digium TDM400P   Sascha Ferley
• [Asterisk-Users] Squeaking / chirping on ZAP Digium TDM400P   Kellner, Peter
• [Asterisk-Users] at-320 phone configuration difficulty   I put the Who? in Mishehu
• [Asterisk-Users] Re: Are there online forums instead of this email forum??   Tony Mountifield
• [Asterisk-Users] Re: Are there online forums instead of this email   Tony Mountifield
• [Asterisk-Users] Queues   lenz
• [Asterisk-Users] Shorewall firewall rules   Remco Barende
• [Asterisk-Users] Shorewall firewall rules   Mikael Magnusson
• [Asterisk-Users] call forwarding   Thore
• [Asterisk-Users] Shorewall firewall rules   Paul Hardacre
• [Asterisk-Users] Little question   flavio patria
• [Asterisk-Users] VAD In Asterisk with Zaptel   parijat
• [Asterisk-Users] Version 0.72 of IPSwitchBoard Released   Thorben Jensen
• [Asterisk-Users] VAD In Asterisk with Zaptel   parijat
• [Asterisk-Users] (no subject)   parijat
• [Asterisk-Users] Delaying answer of incoming calls   Justin Ridge
• [Asterisk-Users] :: Strange way of receiving calls ::   Reuben Grech
• [Asterisk-Users] :: Strange way of receiving calls ::   Peter Svensson
• [Asterisk-Users] Re: Are there online forums instead of this email   Bob Goddard
• [Asterisk-Users] H.323 call '.....' cleared, reason 8 (Transport failure)   Cenk Yabas
• [Asterisk-Users] :: Strange way of receiving calls ::   Erwin de Raad
• [Asterisk-Users] Issues with ringing on FXS ports   Wilson Pickett
• [Asterisk-Users] Sangoma VS. Digium   Chris Modesitt
• [Asterisk-Users] Registration to multiple GKs   VoIP Newbie
• [Asterisk-Users] Call bridging   Wilson Pickett
• [Asterisk-Users] :: Strange way of receiving calls ::   Wilson Pickett
• [Asterisk-Users] SIP register more then 1 with same username   Bashir Ullah - www.Lamsre.Com
• [Asterisk-Users] {extensions.conf} Dialing plans with queues....   Etienne Pretorius
• [Asterisk-Users] Are there online forums instead of this emailforum??   Bob Goddard
• [Asterisk-Users] astcc problems   wassim darwish
• [Asterisk-Users] Little question   flavio patria
• [Asterisk-Users] {extensions.conf} Dialing plans with queues....   Etienne Pretorius
• [Asterisk-Users] VoIP Provider problems   Bob Goddard
• [Asterisk-Users] Queues   David Choo
• [Asterisk-Users] Preserve g729 registration over reinstall??   Martijn van Oosterhout
• [Asterisk-Users] Starting with Asterisk-SIP   Mike
• [Asterisk-Users] Shorewall firewall rules   Rich Adamson
• [Asterisk-Users] VoIP Provider problems   Johnathan Corgan
• [Asterisk-Users] H.323 call '.....' cleared,reason 8 (Transport failure)   Alex Vishnev
• [Asterisk-Users] VoIP Provider problems   Rich Adamson
• [Asterisk-Users] astcc problems   Karl H. Putz
• [Asterisk-Users] Registration to multiple GKs   Alex Vishnev
• [Asterisk-Users] Two accounts at one provider and a 302 redirect problem   Christian Peter
• [Asterisk-Users] Two accounts at one provider and a 302 redirect problem   Rich Adamson
• [Asterisk-Users] Are there online forums instead of this email   Jean-Michel Hiver
• [Asterisk-Users] Asterisk Voice mail with CCM   Nathan Reeves
• [Asterisk-Users] TDM04B - TDM PCI MASTER ABORT   JOAO CARLOS MOURA
• [Asterisk-Users] VoIP Provider problems   Johnathan Corgan
• [Asterisk-Users] Starting with Asterisk-SIP   ruben cuevas rumin
• [Asterisk-Users] Dynamic Zap/{channel} allocation for out going possible?   Etienne Pretorius
• [Asterisk-Users] Starting with Asterisk-SIP   Jean-Michel Hiver
• [Asterisk-Users] Starting with Asterisk-SIP   Kerry Garrison
• [Asterisk-Users] Dynamic Zap/{channel} allocation for out going possible?   Etienne Pretorius
• [Asterisk-Users] Dynamic Zap/{channel} allocation for out going possible?   Etienne Pretorius
• [Asterisk-Users] Starting with Asterisk-SIP   Jeff Heath
• [Asterisk-Users] Dynamic Zap/{channel} allocation for out going possible?   Steven Critchfield
• [Asterisk-Users] Dynamic Zap/{channel} allocation for out going possible?   Ron Wellsted
• [Asterisk-Users] Dynamic Zap/{channel} allocation for out going possible?   Etienne Pretorius
• [Asterisk-Users] Preserve g729 registration over reinstall??   Mike Matthews
• [Asterisk-Users] Re: Are there online forums instead of this email   Jean-Michel Hiver
• [Asterisk-Users] Re: Starting with Asterisk-SIP   flavio patria
• [Asterisk-Users] Starting with Asterisk-SIP   Damon Estep
• [Asterisk-Users] SjPhone&H323   flavio patria
• [Asterisk-Users] Dynamic Zap/{channel} allocation for out going possible?   Etienne Pretorius
• [Asterisk-Users] Snom and Multiple calls   Karl Brose
• [Asterisk-Users] SjPhone&H323   flavio patria
• [Asterisk-Users] Passing varibles *out* of macros   Joe Presto
• [Asterisk-Users] Asterisk Voice mail with CCM   Nathan Alberti
• [Asterisk-Users] Re: Are there online forums instead of this email   Joseph
• [Asterisk-Users] Outbound calls with xlite and Xpro PocketPC   Robert Keller
• [Asterisk-Users] problem detecting answer on pri card   Richard
• [Asterisk-Users] Asterisk Auto-Startup on Ubuntu/Debian   Josh Alberts
• [Asterisk-Users] how to tell what ${DIALSTATUS} is being set   Randy Bush
• [Asterisk-Users] Asterisk Auto-Startup on Ubuntu/Debian   Ron Wellsted
• [Asterisk-Users] *** Asterisk 2.0 Stable release out now   James Gardiner
• [Asterisk-Users] *** Asterisk 2.0 Stable release out now   Brian Capouch
• [Asterisk-Users] echo paid support   James Taylor
• [Asterisk-Users] problem detecting answer on pri card   Alex Vishnev
• [Asterisk-Users] problem detecting answer on pri card   Richard
• [Asterisk-Users] Book Review: VoIP Telephony with Asterisk   Kerry Garrison
• [Asterisk-Users] Zaptel Anti-MMX Optimizations   Trevor Peirce
• [Asterisk-Users] Integrating door intercom?   Jeff Noxon
• [Asterisk-Users] Zaptel Anti-MMX Optimizations   Michael D Schelin
• [Asterisk-Users] Zaptel Anti-MMX Optimizations   Trevor Peirce
• [Asterisk-Users] how to tell what ${DIALSTATUS} is being set   jbryan at productiveconcepts.com
• [Asterisk-Users] Re: how to tell what ${DIALSTATUS} is being set   Tony Mountifield
• [Asterisk-Users] Asterisk Auto-Startup on Ubuntu/Debian   Josh Alberts
• [Asterisk-Users] Cisco Skinny Call Control Protocol on Asterisk   Alexandre Otto Durr
• [Asterisk-Users] wctdm module parameters (Was: Issues with ringing on FXS ports)   Richard Scobie
• [Asterisk-Users] problem detecting answer on pri card   Trevor Peirce
• [Asterisk-Users] Book Review: VoIP Telephony with Asterisk   Jean-Michel Hiver
• [Asterisk-Users] Re: Polycom sound quality problems   Eric Mason
• [Asterisk-Users] Shorewall firewall rules   Remco Barende
• [Asterisk-Users] Book Review: VoIP Telephony with Asterisk   John Novack
• [Asterisk-Users] *** Asterisk 2.0 Stable release out now   John Novack
• [Asterisk-Users] OH323 core dump   Dipole Moment
• [Asterisk-Users] Re: Polycom sound quality problems   Noah Miller
• [Asterisk-Users] Sipura - GSM or iLBC?   Matt
• [Asterisk-Users] OH323 core dump   Yves
• [Asterisk-Users] xlite regestration fails but calls to thru   Scott Wolfe
• [Asterisk-Users] xlite registration fails but calls to thru   Robert Keller
• [Asterisk-Users] Buying some Polycom IP300s   Dan Morin
• [Asterisk-Users] xlite registration fails but calls to thru   Scott Wolfe
• [Asterisk-Users] Buying some Polycom IP300s   Dan Morin
• [Asterisk-Users] Re: Are there online forums instead of, this email   Tore Hansen
• [Asterisk-Users] problem detecting answer on pri card   Richard
• [Asterisk-Users] Re: Are there online forums instead of, this email   Andrew Kohlsmith
• [Asterisk-Users] Buying some Polycom IP300s   Chris Scott
• [Asterisk-Users] Long Distance Acces Code   kritikus Araklidas
• [Asterisk-Users] Sangoma VS. Digium   Cory Andrews
• [Asterisk-Users] sangoma S508/FT1 ISA   Cory Andrews
• [Asterisk-Users] Realtime   Matthew Boehm
• [Asterisk-Users] SIP register more then 1 with same username   Matthew Boehm
• [Asterisk-Users] Buying some Polycom IP300s   Jim Sturtevant
• [Asterisk-Users] Packetization   Matt
• [Asterisk-Users] Re: Buying some Polycom IP300s   Noah Miller
• [Asterisk-Users] *** Asterisk 2.0 Stable release out now   James Gardiner
• [Asterisk-Users] D-Link router/Voip gateway locked to Lingo?   Jason Brown
• [Asterisk-Users] Dialing w/analog phone via FXS port.   Ken D'Ambrosio
• [Asterisk-Users] Asterisk Discussion Form   voipnewb at voipnewbie.com
• [Asterisk-Users] Asterisk Discussion Form * Should Read Asterisk Discussion Forum!*   voipnewb at voipnewbie.com
• [Asterisk-Users] Problem with asterisk -> ohphone   cmisip
• [Asterisk-Users] TE410P and Fax Server   Ma Zhiyong
• [Asterisk-Users] Asterisk Discussion Form   Ty Carter
• [Asterisk-Users] How to reset IAXy?   Lam H. Nguyen
• [Asterisk-Users] Macro Extension with Realtime and Mysql DB   kritikus Araklidas
• [Asterisk-Users] Asterisk Discussion Form   gsmith at voipsupply.com
• [Asterisk-Users] Sip registration Problems With Zyxel P2000W   Ugur GUNCER
• [Asterisk-Users] Sip registration Problems With Zyxel P2000W   Paul Dracevich
• [Asterisk-Users] Replace Adtran 608 With Asterisk   David
• [Asterisk-Users] Macro Extension with Realtime and Mysql DB   Matthew Boehm
• [Asterisk-Users] Integrating door intercom?   C F
• [Asterisk-Users] Delaying answer of incoming calls   C F
• Fwd: NDN: Re: [Asterisk-Users] Delaying answer of incoming calls   C F
• [Asterisk-Users] Packetization   Steven Critchfield
• [Asterisk-Users] sipcall.ch configuration problem   gramels
• [Asterisk-Users] How does asterisk know the did called on?   Courtney Couch
• [Asterisk-Users] New to asterisk.   Paul Lacatus
• [Asterisk-Users] Re: Are there online forums instead of, this email   Jean-Michel Hiver
• [Asterisk-Users] How does asterisk know the did called on?   Damian Funnell
• [Asterisk-Users] Asterisk Discussion Form   Francesco Peeters
• [Asterisk-Users] VoIP Provider problems   Joseph Gutowski
• [Asterisk-Users] Sip registration Problems With Zyxel P2000W   Thore
• [Asterisk-Users] Cisco Skinny Call Control Protocol on Asterisk   asterisk_on_oelf
• [Asterisk-Users] Passing varibles *out* of macros   Wilson Pickett
• [Asterisk-Users] New to asterisk.   Wilson Pickett
• [Asterisk-Users] Cisco Skinny Call Control Protocol on Asterisk   Remco Barende
• [Asterisk-Users] Where to post my impovements to ASTCC?   Ronald Wiplinger
• [Asterisk-Users] Re: Asterisk Discussion Forum   Tore Hansen
• [Asterisk-Users] Router with QoS recommendations   asterisk-Users at zybercity.dk
• [Asterisk-Users] Cisco Skinny Call Control Protocol on Asterisk   asterisk_on_oelf
• [Asterisk-Users] IPSwitchboard Version 0.73 Released   Thorben Jensen
• [Asterisk-Users] Registration to multiple GKs   Charles Wang
• [Asterisk-Users] Router with QoS recommendations   Tim Pushor
• [Asterisk-Users] VoIP Provider problems   Rich Adamson
• [Asterisk-Users] Sip registration Problems With Zyxel P2000W   Eric Rees
• [Asterisk-Users] How does asterisk know the did called on?   Rich Adamson
• [Asterisk-Users] Dialing w/analog phone via FXS port.   Rich Adamson
• [Asterisk-Users] xlite regestration fails but calls to thru   Rich Adamson
• [Asterisk-Users] Asterisk Voice mail with CCM   João Amaro
• [Asterisk-Users] Re: How does asterisk know the did called on?   Noah Miller
• [Asterisk-Users] Re: New to asterisk.   Noah Miller
• [Asterisk-Users] Re: Asterisk Discussion Forum   Noah Miller
• [Asterisk-Users] xlite regestration fails but calls to thru   Alex Vishnev
• [Asterisk-Users] Open Source Billing Software   Andrew Latham
• [Asterisk-Users] Registration to multiple GKs   Alex Vishnev
• [Asterisk-Users] Router with QoS recommendations   Tim Pushor
• [Asterisk-Users] Asterisk Auto-Startup on Ubuntu/Debian   Tzafrir Cohen
• [Asterisk-Users] Macro Extension with Realtime and Mysql DB   kritikus Araklidas
• [Asterisk-Users] Looping messages   Ezabi
• [Asterisk-Users] Asterisk with Jasomi Peerpoing   dhananjay sarnaik
• [Asterisk-Users] Asterisk Voice mail with CCM   Andrea Riela
• [Asterisk-Users] Are there online forums instead of this email forum??   Tzafrir Cohen
• [Asterisk-Users] SIP dialing in two extensions   Jozeph Brasil
• [Asterisk-Users] * INSTRUCTIONS FOR THE ASTERISK COMMUNITY - PLEASE READ NOW *   Olle E. Johansson
• [Asterisk-Users] SIP dialing in two extensions   administrator tootai
• [Asterisk-Users] SET & CHECK group   Mark Halverson
• [Asterisk-Users] Packetization   Matt
• [Asterisk-Users] Router with QoS recommendations   Philipp von Klitzing
• [Asterisk-Users] Passing varibles *out* of macros   Joe Presto
• [Asterisk-Users] Router with QoS recommendations   NVC List Manager
• [Asterisk-Users] Router with QoS recommendations   Andrew Kohlsmith
• [Asterisk-Users] Passing varibles *out* of macros   Gary Reuter
• [Asterisk-Users] Authenticating username   Martijn van Oosterhout
• [Asterisk-Users] Snom and Multiple calls   Philipp von Klitzing
• [Asterisk-Users] Sipura SPA 2000 - Miltiple Ring Tones   Trevor Peirce
• [Asterisk-Users] Detecting when a called mobile is not reachable?   Ian Hailey
• [Asterisk-Users] Buying some Polycom IP300s   Jim Van Meggelen
• [Asterisk-Users] Packetization   Bruce Komito
• [Asterisk-Users] Looking for res_config_pgsql   Martijn van Oosterhout
• [Asterisk-Users] Detecting when a called mobile is not reachable?   David John Walsh
• [Asterisk-Users] Buying some Polycom IP300s   Courtney Couch
• [Asterisk-Users] Asterisk on Suse minimal installation based on Suse Rescue - what to add to be bootable on HD partition ?   Robert Rozman
• [Asterisk-Users] Packetization   Matt
• [Asterisk-Users] Are there online forums instead of this email forum??   Leif Madsen - Certified Asterisk Consultant
• [Asterisk-Users] Packetization   Matt
• [Asterisk-Users] Packetization   Matt
• [Asterisk-Users] Asterisk Discussion Form   John Novack
• [Asterisk-Users] really small box   Irakli Natsvlishvili
• Fwd: NDN: Re: [Asterisk-Users] Delaying answer of incoming calls   John Novack
• [Asterisk-Users] Router with QoS recommendations   Irakli Natsvlishvili
• [Asterisk-Users] CAPI/Dialing out   Philip Hofstetter
• [Asterisk-Users] Zaptel Anti-MMX Optimizations   I put the Who? in Mishehu
• Fwd: NDN: Re: [Asterisk-Users] Delaying answer of incoming calls   C F
• [Asterisk-Users] IAX messages   pabut
• [Asterisk-Users] SET & CHECK group   C F
• [Asterisk-Users] How to reset IAXy?   I put the Who? in Mishehu
• [Asterisk-Users] VG248 and Asterisk   Steve Blair
• [Asterisk-Users] Zaptel Anti-MMX Optimizations   Trevor Peirce
• [Asterisk-Users] Authenticating username   Nabeel Jafferali
• [Asterisk-Users] problems with call-forward from ccme to * on sip trunk   Andrea Riela
• [Asterisk-Users] AS5300+SIP+ASTERISK or AS5300+MGCP   jafar mohammed
• [Asterisk-Users] AS5300+SIP+ASTERISK or AS5300+MGCP   jafar mohammed
• [Asterisk-Users] *** Asterisk 2.0 Stable release out now   Matt Riddell
• [Asterisk-Users] Asterisk Realtime Capabilities   Rod Bacon
• [Asterisk-Users] Buying some Polycom IP300s   Kong
• [Asterisk-Users] How does asterisk know the did called on?   Eric Wieling aka ManxPower
• [Asterisk-Users] problems with call-forward from ccme to * on sip trunk   Nathan Alberti
• [Asterisk-Users] Re: using unixODBC   Kamran Ahmad
• [Asterisk-Users] Detecting when a called mobile is not reachable?   Eric Wieling aka ManxPower
• [Asterisk-Users] How to reset IAXy?   Ronald Wiplinger
• [Asterisk-Users] SET & CHECK group   Eric Wieling aka ManxPower
• [Asterisk-Users] Detecting when a called mobile is not reachable?   Rod Bacon
• [Asterisk-Users] Joshua Chessman   Rod Bacon
• [Asterisk-Users] Re: Asterisk-Users Digest, Vol 9, Issue 21   Kamran Ahmad
• [Asterisk-Users] Detecting when a called mobile is not reachable?   Eric Wieling aka ManxPower
• [Asterisk-Users] AS5300+SIP+ASTERISK or AS5300+MGCP   Leandro Tenorio
• [Asterisk-Users] Asterisk <-> Altigen   Dan Perik
• [Asterisk-Users] Asterisk <-> Altigen   Rusty Shackleford
• [Asterisk-Users] creating conference call   Keiron Liddle
• [Asterisk-Users] AGI Dial Plan   Lee Lee
• [Asterisk-Users] Re: X100P interrupt load   Dinesh Nair
• [Asterisk-Users] Asterisk@Home Question   * KAPIL *
• [Asterisk-Users] Asterisk@Home Question   * KAPIL *
• [Asterisk-Users] Asterisk Realtime Capabilities   Matthew Boehm
• [Asterisk-Users] Asterisk@Home Question   Nabeel Jafferali
• [Asterisk-Users] Music On Hold and ATA-186 w/Silence Supression   Alejandro G
• [Asterisk-Users] V92 modem with asterisk   Alexandre Charles
• [Asterisk-Users] how to configure groups using a sip phone   deepak.dhiman at uf4.net
• [Asterisk-Users] Music On Hold and ATA-186 w/Silence Supression   Robert Lawrence
• [Asterisk-Users] how to configure groups using a sip phone   Rod Bacon
• [Asterisk-Users] Previous sip reload not yet done   administrator tootai
• [Asterisk-Users] Asterisk Realtime Capabilities   Rod Bacon
• [Asterisk-Users] RE: AS5300+SIP+ASTERISK or AS5300+MGCP   jafar mohammed
• [Asterisk-Users] V92 modem with asterisk   Rod Bacon
• [Asterisk-Users] Wellgate 3701   Asterisk user list
• [Asterisk-Users] Buying some Polycom IP300s   Rod Bacon
• [Asterisk-Users] AGI Dial Plan   Jean-Michel Hiver
• [Asterisk-Users] Buying some Polycom IP300s   Paul Hales
• [Asterisk-Users] Asterisk@Home Question   Kerry Garrison
• [Asterisk-Users] Previous sip reload not yet done   Nabeel Jafferali
• [Asterisk-Users] How to send email from the dial plan?   Ronald Wiplinger
• [Asterisk-Users] Sangoma VS. Digium   Dinesh Nair
• [Asterisk-Users] error while compiling asterisk-1.0.7   Kamran Ahmad
• [Asterisk-Users] Time sync on PRI   Tobias Jönsson
• [Asterisk-Users] How to send email from the dial plan?   Olle E. Johansson
• [Asterisk-Users] at-320 phone configuration difficulty   clive at sentechsa.com
• [Asterisk-Users] Zaptel group members - dial out on a availible port via trial and error?   Etienne Pretorius
• [Asterisk-Users] Q.931 to SIGTRAN interface   Dinesh Nair
• [Asterisk-Users] Previous sip reload not yet done   administrator tootai
• [Asterisk-Users] problems with call-forward from ccme to * on sip trunk   Andrea Riela
• [Asterisk-Users] how to configure groups using a sip phone   Deepak Dhiman
• [Asterisk-Users] Router with QoS recommendations   Shaoul Jacobson - TELLINK
• [Asterisk-Users] Previous sip reload not yet done   Olle E. Johansson
• [Asterisk-Users] problems with call-forward from ccme to * on sip trunk   Andrea Riela
• [Asterisk-Users] Previous sip reload not yet done   administrator tootai
• [Asterisk-Users] Sangoma VS. Digium   Michael Bielicki
• [Asterisk-Users] Asterisk+Sipgate - just one step away..   Razvan Cosma
• [Asterisk-Users] Router with QoS recommendations   tim panton
• [Asterisk-Users] Planet VIP 450   Altus Snyman
• [Asterisk-Users] Detecting when a called mobile is not reachable?   David John Walsh
• [Asterisk-Users] Asterisk+Sipgate - just one step away..   administrator tootai
• [Asterisk-Users] Manipulation based on SIP extension   Irakli Natsvlishvili
• [Asterisk-Users] Difficulty in configuring Asterisk to ensure that Call Transfers (SIP phone) are properly recorded and billable   Peter Dean
• [Asterisk-Users] Authenticating username   David John Walsh
• [Asterisk-Users] Realtime & voicemail   Ronald Wiplinger
• [Asterisk-Users] Music On Hold and ATA-186 w/Silence Supression   Irakli Natsvlishvili
• [Asterisk-Users] Difficulty in configuring Asterisk to ensure that Call Transfers (SIP phone) are properly recorded and billable   Peter Dean
• [Asterisk-Users] Asterisk+Sipgate - just one step away..   Razvan Cosma
• [Asterisk-Users] Router with QoS recommendations   Shaoul Jacobson - TELLINK
• [Asterisk-Users] Supervised transfer problems   Daniele Gallina - 3P System S.r.l.
• [Asterisk-Users] SIP Jitter buffer   1 2
• [Asterisk-Users] How do you do Line Hunting in Asterisk?   Etienne Pretorius
• [Asterisk-Users] Time sync on PRI   Peter Svensson
• [Asterisk-Users] Re: error while compiling asterisk-1.0.7   Tony Mountifield
• [Asterisk-Users] Zap - What is going on?   Etienne Pretorius
• [Asterisk-Users] Asterisk@Home Question   Tony Davidson
• [Asterisk-Users] Manipulation based on SIP extension   Henry Devito
• [Asterisk-Users] Zap - What is going on?   Rob Scott
• [Asterisk-Users] Previous sip reload not yet done   Cirelle Internet Products
• [Asterisk-Users] Sending faxes and call accounting   Chris Mason (Lists)
• [Asterisk-Users] Zap - What is going on?   Etienne Pretorius
• [Asterisk-Users] Best way for nated sip peers thru a database   Laurent FOULONNEAU
• [Asterisk-Users] Previous sip reload not yet done   administrator tootai
• [Asterisk-Users] Supervised transfer problems   Josiah Bryan
• [Asterisk-Users] Re: ASTCC question: Trunk LOCAL   Ronald Wiplinger
• [Asterisk-Users] Realtime mysql problem?   Matt Schulte
• [Asterisk-Users] SIP Jitter buffer   Rich Adamson
• [Asterisk-Users] Router with QoS recommendations   Bruno Hertz
• [Asterisk-Users] Realtime mysql problem?   Wojciech Tryc
• [Asterisk-Users] Asterisk@Home Question   Jeff R Glassman
• [Asterisk-Users] Asterisk@Home Question   Giles Coochey
• [Asterisk-Users] Realtime mysql problem?   Matt Schulte
• [Asterisk-Users] Router with QoS recommendations   David McNett
• [Asterisk-Users] PRI: received SETUP message for call that is not a new call, wicked!   Mark Elkins
• [Asterisk-Users] Router with QoS recommendations   Tim Pushor
• [Asterisk-Users] V92 modem with asterisk   Glenn
• [Asterisk-Users] X-Lite to Zap, no Voice on other phone!   Etienne Pretorius
• [Asterisk-Users] Asterisk Realtime Capabilities   Matthew Boehm
• [Asterisk-Users] Realtime mysql problem?   Matthew Boehm
• [Asterisk-Users] IAXy audio troubles (only on INCOMING calls)   niels at wxn.nl
• [Asterisk-Users] Best way for nated sip peers thru a database   Matthew Boehm
• [Asterisk-Users] Router with QoS recommendations   Arnaud PIGNARD
• [Asterisk-Users] Browser based configuration of Asterisk   Chuck Bunn
• [Asterisk-Users] SIP Absolute Timeout   kaiser
• [Asterisk-Users] Detecting when a called mobile is not reachable?   Ian Hailey
• [Asterisk-Users] problems with call-forward from ccme to * on sip trunk   Andrea Riela
• [Asterisk-Users] Realtime mysql problem?   Wojciech Tryc
• [Asterisk-Users] SIP phones to Asterisk using MAC address instead of IP address   Chuck Bunn
• [Asterisk-Users] Sending faxes and call accounting   Glenn
• [Asterisk-Users] Distributed services such as voicemail using Asterisk   Chuck Bunn
• [Asterisk-Users] Asterisk Realtime - extensions configuration help   Shaoul Jacobson - TELLINK
• [Asterisk-Users] SIP phones to Asterisk using MAC address insteadof IP address   Giles Coochey
• [Asterisk-Users] configuring md5 authentication   Dov Bigio
• [Asterisk-Users] Re: X100P interrupt load   Jesse D. Guardiani
• [Asterisk-Users] Realtime mysql problem?   Matt Schulte
• [Asterisk-Users] newbie - want to use asterisk as an internal PBX   mak kwak
• [Asterisk-Users] configuring md5 authentication   Maik Schmitt
• [Asterisk-Users] Asterisk Realtime - extensions configuration help   Matthew Boehm
• [Asterisk-Users] Just a test   Rick Baranowski
• [Asterisk-Users] Re: Snom and Multiple calls   Josh Dady
• [Asterisk-Users] Does the agent queue app support Aftercall and AUX agent status?   Steve Mann
• [Asterisk-Users] Asterisk@Home Question   Dalon Westergreen
• [Asterisk-Users] Problem registering 'SJPhone'?   Chuck Bunn
• [Asterisk-Users] Asterisk-1.0.7 Build - Serious issues   Dana Olson
• [Asterisk-Users] Router with QoS recommendations   James H. Thompson
• [Asterisk-Users] Sangoma VS. Digium   Dana Olson
• [Asterisk-Users] Realtime mysql problem?   Matthew Boehm
• [Asterisk-Users] D-Link router/Voip gateway locked to Lingo?   Joseph
• [Asterisk-Users] Router with QoS recommendations   Tim Pushor
• [Asterisk-Users] X-Lite to Zap, no Voice on other phone!   Wilson Pickett
• [Asterisk-Users] Realtime mysql problem?   Matt Schulte
• [Asterisk-Users] Asterisk and Ingate registration   c waddy
• [Asterisk-Users] Asterisk-1.0.7 Build - Serious issues   Kanuri, Seshu (Company IT)
• [Asterisk-Users] Realtime mysql problem?   Matthew Boehm
• [Asterisk-Users] Asterisk Realtime - extensions configurationhelp   Shaoul Jacobson - TELLINK
• [Asterisk-Users] Browser based configuration of Asterisk   Time Bandit
• [Asterisk-Users] IAXy audio troubles (only on INCOMING calls)   Time Bandit
• [Asterisk-Users] X-Lite to Zap, no Voice on other phone!   Etienne Pretorius
• [Asterisk-Users] Asterisk Realtime - extensions configurationhelp   Matthew Boehm
• [Asterisk-Users] Realtime mysql problem?   Ronald Wiplinger
• [Asterisk-Users] Collect Calls   John Spearle
• [Asterisk-Users] Collect Calls   Nabeel Jafferali
• [Asterisk-Users] Collect Calls   Kerry Garrison
• [Asterisk-Users] Sending faxes and call accounting   brian at txshirts.com
• [Asterisk-Users] FGD Support   James Taylor
• [Asterisk-Users] X-Lite to Zap, no Voice on other phone!   Etienne Pretorius
• [Asterisk-Users] rookie getting started question   Randy Paries
• [Asterisk-Users] Asterisk and clarent   Jorge Alayon
• [Asterisk-Users] Can I set queue not to hangup?   Steve Edwards
• [Asterisk-Users] Can't see ANI2 (aka info digits) from PRI t1   Steve Edwards
• [Asterisk-Users] Ring Twice   VoIP Newbie
• [Asterisk-Users] SIP phones to Asterisk using MAC addressinsteadof IP address   Alex Vishnev
• [Asterisk-Users] Can I set queue not to hangup?   Etienne Pretorius
• [Asterisk-Users] Time sync on PRI   Eric Wieling aka ManxPower
• [Asterisk-Users] IPSwitchBoard speaks other languages   Thorben Jensen
• [Asterisk-Users] SIP phones to Asterisk using MAC addressinsteadofIP address   Kanuri, Seshu (Company IT)
• [Asterisk-Users] How to control codecs when originating a call?   Tony Mountifield
• [Asterisk-Users] Agent logoff   James Murray
• [Asterisk-Users] mISDN + chan_misdn and DTMF   Alex
• [Asterisk-Users] Re: Polycom sound quality problems   Max W Blackmer Jr
• [Asterisk-Users] Can I set queue not to hangup?   Richard Lyman
• [Asterisk-Users] monmp3thread: Request to schedule in the past?!?!   Glenn Powers
• [Asterisk-Users] vmail.cgi - can't forward messages   David Brodbeck
• [Asterisk-Users] compilation of asterisk   Alex
• [Asterisk-Users] monmp3thread: Request to schedule in the past?!?!   Kanuri, Seshu (Company IT)
• [Asterisk-Users] hardware requirements   Antonio Airoso
• [Asterisk-Users] compilation of asterisk   Daniel Bruce Lynes
• [Asterisk-Users] broadvoice   Matt
• [Asterisk-Users] Asterisk on WRT54GS   Dalon Westergreen
• [Asterisk-Users] ZAP problem (No channel type registered for 'Zap')   Maik Hassel
• [Asterisk-Users] ZAP problem (No channel type registered for 'Zap')   Jason Miller
• [Asterisk-Users] Asterisk and clarent   Jay Ray
• [Asterisk-Users] ZAP problem (No channel type registered for 'Zap')   Eric Wieling aka ManxPower
• [Asterisk-Users] broadvoice   JD Austin
• [Asterisk-Users] broadvoice   Kerry Garrison
• [Asterisk-Users] hardware requirements   Steve Mann
• [Asterisk-Users] broadvoice   Paul
• [Asterisk-Users] Asterisk on WRT54GS   Kanuri, Seshu (Company IT)
• [Asterisk-Users] Asterisk on WRT54GS   Juergen K. Zick
• [Asterisk-Users] broadvoice   Matt
• [Asterisk-Users] Router with QoS recommendations   Cameron Beattie
• [Asterisk-Users] Asterisk on WRT54GS   Brancaleoni Matteo
• [Asterisk-Users] monmp3thread: Request to schedule in the past?!?!   Steve Mann
• [Asterisk-Users] broadvoice   Kevin Kiely
• [Asterisk-Users] Mitel Ip phone ?   Kris Edwards
• [Asterisk-Users] broadvoice   Dalon Westergreen
• [Asterisk-Users] Concurrent calls: best provider?   Scott Wolfe
• [Asterisk-Users] broadvoice   Matt
• [Asterisk-Users] monmp3thread: Request to schedule in the past?!?!   Eric Wieling aka ManxPower
• [Asterisk-Users] Re: Collect Calls   Mark Halverson
• [Asterisk-Users] broadvoice   Steve Mann
• [Asterisk-Users] compilation of asterisk   Sean A. Newton
• [Asterisk-Users] broadvoice   Mike Matthews
• [Asterisk-Users] distinctive ringing in a queue?   Louis-David Mitterrand
• [Asterisk-Users] Asterisk Realtime Capabilities   Cameron Beattie
• [Asterisk-Users] Asterisk@Home Question   Mike Matthews
• [Asterisk-Users] monmp3thread: Request to schedule in the past?!?!   Joseph Finley
• [Asterisk-Users] rookie getting started question   Sean Kennedy
• [Asterisk-Users] Livevoip DTMF via IAX almost   Rich Adamson
• [Asterisk-Users] monmp3thread: Request to schedule in the past?!?!   Eric Wieling aka ManxPower
• [Asterisk-Users] Concurrent calls: best provider?   Kevin Kiely
• [Asterisk-Users] call redirection from outside line?   Jacob Cazzell
• [Asterisk-Users] SIPTone II and PoE   George Gardiner
• [Asterisk-Users] broadvoice   Johnathan Corgan
• [Asterisk-Users] broadvoice   hugolivude
• [Asterisk-Users] broadvoice   Matt
• [Asterisk-Users] broadvoice   hugolivude
• [Asterisk-Users] SIP microphone not working after ZAP and sounddriver installation   Maik Hassel
• [Asterisk-Users] Zaptel group members - dial out on a availible port via trial and error?   Cameron Beattie
• [Asterisk-Users] Asterisk queue crash   Øystein Tusvik
• [Asterisk-Users] Realtime mysql problem?   Cameron Beattie
• [Asterisk-Users] Concurrent calls: best provider?   John Millican
• [Asterisk-Users] IP Address of caller variable?   William M. Sandiford
• [Asterisk-Users] newbie - want to use asterisk as an internal PBX   end1r
• [Asterisk-Users] newbie - want to use asterisk as an internal PBX   Cameron Beattie
• [Asterisk-Users] CVS-HEAD compile? Was: Checkgroup and transfers   Sean A. Newton
• [Asterisk-Users] broadvoice   Steve Mann
• [Asterisk-Users] FGD Support   Jason Miller
• [Asterisk-Users] call redirection from outside line?   Wiley Siler
• [Asterisk-Users] Concurrent calls: best provider?   Brian McSpadden
• [Asterisk-Users] broadvoice   Matt
• [Asterisk-Users] L2 QoS switch   Bob Knight
• [Asterisk-Users] Operators guide   Tony Davidson
• [Asterisk-Users] DTMF Caller ID in Brazil   Han van Hulst
• [Asterisk-Users] Asterisk Realtime Capabilities   Matthew Boehm
• [Asterisk-Users] Concurrent calls: best provider?   John Millican
• [Asterisk-Users] Authentication with DB Support   kritikus Araklidas
• [Asterisk-Users] call redirection from outside line?   Scott Nelson
• [Asterisk-Users] how to configure groups using a sip phone   Rod Bacon
• [Asterisk-Users] TE405P takes ~5mins to load.   Paul Belanger
• [Asterisk-Users] rookie getting started question   Randy Paries
• [Asterisk-Users] Can't see ANI2 (aka info digits) from PRI t1   C F
• [Asterisk-Users] broadvoice   Matt
• [Asterisk-Users] chan_sccp compile error   Rob Wise
• [Asterisk-Users] rookie getting started question   Eric Wieling aka ManxPower
• [Asterisk-Users] SIP and firewall   Maik Hassel
• [Asterisk-Users] Problem registering 'SJPhone'?   Chuck Bunn
• [Asterisk-Users] Re: chan_sccp compile error   Rob Wise
• [Asterisk-Users] broadvoice   Rich Adamson
• [Asterisk-Users] Channel bank question   Sean Kennedy
• [Asterisk-Users] TE405P takes ~5mins to load.   Andrew Kohlsmith
• [Asterisk-Users] Livevoip DTMF via IAX almost   Rich Adamson
• [Asterisk-Users] Livevoip DTMF via IAX almost   Rich Adamson
• [Asterisk-Users] Channel bank question   Alexander Lopez
• [Asterisk-Users] Re: Polycom sound quality problems   Eric Mason
• [Asterisk-Users] Channel bank question   Andrew Kohlsmith
• [Asterisk-Users] SIP and firewall   Robert Jackson
• [Asterisk-Users] rookie getting started question   Alexander Lopez
• [Asterisk-Users] rookie getting started question   J P Edmund
• [Asterisk-Users] zapata.conf parameter order - feature or bug?   Rod Bacon
• [Asterisk-Users] Channel bank question   Damon Estep
• [Asterisk-Users] ASTCC - not saving configuration   Nabeel Jafferali
• [Asterisk-Users] SIP/SDP packaged in Multipart/Mixed mime type   Ed Greenberg
• [Asterisk-Users] bandwidth   Bernie
• [Asterisk-Users] broadvoice   Dalon Westergreen
• [Asterisk-Users] call redirection from outside line?   Adam Goryachev
• [Asterisk-Users] Set system time over the phone   Mike Sander
• [Asterisk-Users] bandwidth   William Boehlke
• [Asterisk-Users] AAH 0.6 - Change Network Gateway   Wiley Siler
• [Asterisk-Users] Asterisk Realtime Capabilities   Cameron Beattie
• [Asterisk-Users] TE405P takes ~5mins to load.   Derrick Knight
• [Asterisk-Users] bandwidth   Bernie
• [Asterisk-Users] ASTCC - not saving configuration   Nabeel Jafferali
• [Asterisk-Users] Digium Hires Kevin Flemming   Mark Spencer
• [Asterisk-Users] bandwidth   Bruno Hertz
• [Asterisk-Users] TE405P takes ~5mins to load.   Andrew Kohlsmith
• [Asterisk-Users] register => with realtime   Matthew Boehm
• [Asterisk-Users] Best way for nated sip peers thru a database   Laurent Foulonneau
• [Asterisk-Users] Authentication with DB Support   El Flynn
• [Asterisk-Users] Detecting Downed SIP Phone   John Goerzen
• [Asterisk-Users] Re: X100P interrupt load   Dinesh Nair
• [Asterisk-Users] broadvoice   hugolivude
• [Asterisk-Users] bandwidth   Paul Hales
• [Asterisk-Users] Channel bank question   Ariel Batista
• [Asterisk-Users] TE405P takes ~5mins to load.   Derrick Knight
• [Asterisk-Users] Transient SIP Registration Issues   Richard J. Sears
• [Asterisk-Users] rookie getting started question   Randy Paries
• [Asterisk-Users] bandwidth   Richard J. Sears
• [Asterisk-Users] AAH 0.6 - Change Network Gateway   W. Kevin Hunt
• [Asterisk-Users] AAH 0.6 - Change Network Gateway   Henry Devito
• [Asterisk-Users] Re: V92 modem with asterisk   Tore Hansen
• [Asterisk-Users] Sending faxes and call accounting   Chris Mason
• [Asterisk-Users] Voicemailbox detection:   Tim Connolly
• [Asterisk-Users] bandwidth   Eric Wieling aka ManxPower
• [Asterisk-Users] Voicemailbox detection:   Nabeel Jafferali
• [Asterisk-Users] bandwidth   Eric Wieling aka ManxPower
• [Asterisk-Users] Detecting Downed SIP Phone   Eric Wieling aka ManxPower
• [Asterisk-Users] Transient SIP Registration Issues   Eric Wieling aka ManxPower
• [Asterisk-Users] monmp3thread: Request to schedule in the past?!?!   Glenn
• [Asterisk-Users] Compatability with AudioCodes MP-108   LJ
• [Asterisk-Users] Voicemailbox detection:   Tim Connolly
• [Asterisk-Users] Detecting Downed SIP Phone   Rusty Shackleford
• [Asterisk-Users] Set system time over the phone   Asterisk at Home
• [Asterisk-Users] Livevoip DTMF via IAX almost   Gary Reuter
• [Asterisk-Users] Asterisk on WRT54GS   Greg Boehnlein
• [Asterisk-Users] Weird Errors with Realtime and MySQL   William M. Sandiford
• [Asterisk-Users] Installation problem   Jerry Hobbs
• To all chan_sccp users! (Was: [Asterisk-Users] chan_sccp compile error)   Julien Goodwin
• [Asterisk-Users] Installation problem   Alexander Lopez
• [Asterisk-Users] Re: Detecting Downed SIP Phone   John Goerzen
• [Asterisk-Users] Buying some Polycom IP300s   brian at txshirts.com
• [Asterisk-Users] asterisk on UML   snacktime
• [Asterisk-Users] Re: Re: X100P interrupt load   Jesse Guardiani
• [Asterisk-Users] Asterisk Discussion Forums provided by Digium   Kevin P. Fleming
• [Asterisk-Users] help regading outbound calls   Muhammad Haris
• [Asterisk-Users] Outgoing faxes with chan_capi?   Andrew Furey
• [Asterisk-Users] Asterisk Discussion Forums provided by Digium   Matt Klein
• [Asterisk-Users] Operators guide   Wilson Pickett
• [Asterisk-Users] Set system time over the phone   Peter Bowyer
• [Asterisk-Users] rookie getting started question   Wilson Pickett
• [Asterisk-Users] asterisk on UML   Kristian Kielhofner
• [Asterisk-Users] Set system time over the phone   Matt Riddell
• [Asterisk-Users] Set system time over the phone   Roman Volf
• [Asterisk-Users] livevoip callerid   Cameron Schaus
• [Asterisk-Users] Asterisk Discussion Forums provided by Digium   Matt Riddell
• [Asterisk-Users] Freesoftswitch.com   Jaime Martelino
• [Asterisk-Users] Asterisk on WRT54GS - Slightly OT   Juergen K. Zick
• [Asterisk-Users] rookie getting started question   Shaoul Jacobson - TELLINK
• [Asterisk-Users] Outgoing calls on PRI   cereal killer
• [Asterisk-Users] SIP Phone binary   Klaus Peras
• [Asterisk-Users] Set system time over the phone   Peter Bowyer
• [Asterisk-Users] Outgoing calls on PRI   Peter Svensson
• [Asterisk-Users] SIP Phone binary   Nardis Dome
• [Asterisk-Users] Outgoing calls on PRI   cereal killer
• [Asterisk-Users] Incomming Call issues   Sascha Ferley
• [Asterisk-Users] E100p zapata errors   Ugur GUNCER
• [Asterisk-Users] sniffing bridged video call on zap channels   Klaus Darilion
• [Asterisk-Users] Asterisk Realtime - extensions configurationhelp / solved   Shaoul Jacobson - TELLINK
• [Asterisk-Users] asterisk FWD intelligent routing   trixter http://www.0xdecafbad.com
• [Asterisk-Users] Incomming Call issues   Martijn van Oosterhout
• [Asterisk-Users] zaptel not starting issues   Terry Wade
• [Asterisk-Users] "Multiplexing" (or what ever the term is) FXO ports into a "Trunk"   David John Walsh
• [Asterisk-Users] No Voice to POTS.   Etienne Pretorius
• [Asterisk-Users] SIP / PTT over Cellular   Klaus Darilion
• [Asterisk-Users] Set system time over the phone   Tzafrir Cohen
• [Asterisk-Users] Realtime queues?   Kristian Nielsen
• [Asterisk-Users] Transient SIP Registration Issues   Cirelle Internet Products
• [Asterisk-Users] i3micro VTA111 and Asterisk   Rikard Westlund
• [Asterisk-Users] WRT54GP2A-AT   Andre Normandin
• [Asterisk-Users] busy line status on CISCO 7940/7960   Sergio
• [Asterisk-Users] ISDN-30 in UK   Timothy Potter
• [Asterisk-Users] livevoip callerid   MF Hulber
• [Asterisk-Users] livevoip callerid   Rich Adamson
• [Asterisk-Users] how to make asterisk only for SIP and direct RTP   Kamran Ahmad
• [Asterisk-Users] ISDN-30 in UK   asterisk at sipex.co.uk
• [Asterisk-Users] MOH and OptiPoint400 std SIP   Nardis Dome
• [Asterisk-Users] Sending faxes and call accounting   Rich Adamson
• [Asterisk-Users] fault tolerant asterisk system   Infocus
• [Asterisk-Users] fault tolerant asterisk system   Andres Tello Abrego
• [Asterisk-Users] MOH and OptiPoint400 std SIP   Chris Hills
• [Asterisk-Users] SIP / PTT over Cellular   Joseph
• [Asterisk-Users] Checking for the in-use conference   Vyom A
• [Asterisk-Users] Re: busy line status on CISCO 7940/7960   Sergio
• [Asterisk-Users] asterisk on UML   Robert Jackson
• [Asterisk-Users] Problems registering Asterisk with Vocal   Daniel Mizrachi (Globalsip)
• [Asterisk-Users] busy line status on CISCO 7940/7960   Rich Adamson
• [Asterisk-Users] Operators guide   Tony Davidson
• [Asterisk-Users] fault tolerant asterisk system   Matt
• [Asterisk-Users] fault tolerant asterisk system   Matt
• [Asterisk-Users] livevoip callerid   Rich Adamson
• [Asterisk-Users] Asterisk - SS7 or ISDN   Roy Sigurd Karlsbakk
• [Asterisk-Users] Asterisk and HylaFAX integration   Gavin Hamill
• [Asterisk-Users] Re: busy line status on CISCO 7940/7960   Eric Wieling aka ManxPower
• [Asterisk-Users] D Channel Becoming "CORRUPTED"?   Matt
• [Asterisk-Users] multiple PBXs on one server.   Andres Tello Abrego
• [Asterisk-Users] fault tolerant asterisk system   Infocus
• [Asterisk-Users] D Channel Becoming "CORRUPTED"?   Gatewood, Andy
• [Asterisk-Users] Tracing dial plan branching   Chris Mason (Lists)
• [Asterisk-Users] D Channel Becoming "CORRUPTED"?   Damon Estep
• [Asterisk-Users] Asterisk and HylaFAX integration   Lee Howard
• [Asterisk-Users] AAH 0.6 - Change Network Gateway   Wiley Siler
• [Asterisk-Users] Buying some Polycom IP300s   Wiley Siler
• [Asterisk-Users] Asterisk and HylaFAX integration   Gavin Hamill
• [Asterisk-Users] VoIP network configuration using Asterisk and SIP   Mariña Varela Senín
• [Asterisk-Users] Asterisk and C# (Dotnet)   Maxime Renaud
• [Asterisk-Users] prevent callerid spoofing between asterisks   elkur
• [Asterisk-Users] Buying some Polycom IP300s   Philip Trauring
• [Asterisk-Users] OT: CRTC mandates 911/E911 for VoIP in Canada   William M. Sandiford
• [Asterisk-Users] Asterisk and C# (Dotnet)   Springer Christophe
• [Asterisk-Users] VoIP network configuration using Asterisk and SIP   Moises Silva
• [Asterisk-Users] AAH 0.6 - Change Network Gateway   Wiley Siler
• [Asterisk-Users] Asterisk and clarent   Jorge Alayon
• [Asterisk-Users] re: Problem: Compiling error for SpanDSP   Steve Hanselman
• [Asterisk-Users] SIP -> PSTN: mISDN DTMF tones generation   Alex
• [Asterisk-Users] call redirection from outside line?   Josiah Bryan
• [Asterisk-Users] mysql and confs   Scott
• [Asterisk-Users] Authenticating username   Nabeel Jafferali
• [Asterisk-Users] Operators guide   Wilson Pickett
• [Asterisk-Users] Can't see ANI2 (aka info digits) from PRI t1   Steve Edwards
• [Asterisk-Users] WRT54GP2A-AT Morality   Race Vanderdecken
• [Asterisk-Users] Best Performance   Ugur GUNCER
• [Asterisk-Users] Voicemailbox detection:   David Brodbeck
• [Asterisk-Users] prevent callerid spoofing between asterisks   Sean Kennedy
• [Asterisk-Users] OT: CRTC mandates 911/E911 for VoIP in Canada   Damon Estep
• [Asterisk-Users] WRT54GP2A-AT Morality   Andrew Kohlsmith
• [Asterisk-Users] WRT54GP2A-AT   Damon Estep
• [Asterisk-Users] MOH and OptiPoint400 std SIP   Nardis Dome
• [Asterisk-Users] Snom and Multiple calls   Josh Dady
• [Asterisk-Users] VOIP 911 Mandatory in Canada   Damon Estep
• [Asterisk-Users] Voicemailbox detection:   Shaun Ewing
• [Asterisk-Users] VOIP 911 Mandatory in Canada   James Taylor
• [Asterisk-Users] How do I retrieve voice mail in Asterisk   Chuck Bunn
• [Asterisk-Users] WRT54GP2A-AT   Wiley Siler
• [Asterisk-Users] Awaiting Ack   Flavio Correa
• [Asterisk-Users] VOIP 911 Mandatory in Canada   Ronald Wiplinger
• [Asterisk-Users] How do I retrieve voice mail in Asterisk   JD Austin
• [Asterisk-Users] Buying some Polycom IP300s   Adam Robins
• [Asterisk-Users] How do I retrieve voice mail in Asterisk   Josiah Bryan
• [Asterisk-Users] How do I retrieve voice mail in Asterisk   Wiley Siler
• [Asterisk-Users] Can't see ANI2 (aka info digits) from PRI t1   Karl H. Putz
• [Asterisk-Users] Operators guide   Andrew Latham
• [Asterisk-Users] Can't see ANI2 (aka info digits) from PRI t1   James Taylor
• [Asterisk-Users] how to configure groups using a sip phone   Jason Quigley
• [Asterisk-Users] Agents   James Murray
• [Asterisk-Users] How do I retrieve voice mail in Asterisk   Nathan Alberti
• [Asterisk-Users] missing ring-tone   Steve Hanselman
• [Asterisk-Users] Asterisk Voice mail with CCM   Nathan Alberti
• [Asterisk-Users] TE110P/Hipath3750 - Yellow Alarm   Henry Jensen
• [Asterisk-Users] mysql and confs   James Taylor
• [Asterisk-Users] problems with call-forward from ccme to * on sip trunk   Nathan Alberti
• [Asterisk-Users] sip <-> oh323 / real-time / g729 - one way audio   Shaoul Jacobson - TELLINK
• [Asterisk-Users] How do I retrieve voice mail in Asterisk   Jay Ray
• [Asterisk-Users] missing ring-tone   Wiley Siler
• [Asterisk-Users] Asterisk Voice mail with CCM   ht at phonitel.com
• [Asterisk-Users] sip <-> oh323 / real-time / g729 - one way audio   ht at phonitel.com
• [Asterisk-Users] TE110P/Hipath3750 - Yellow Alarm   Steve Totaro
• [Asterisk-Users] fault tolerant asterisk system   Nathan Alberti
• [Asterisk-Users] Digtial Receptionist Recorded Greeting Location Problem   mr. barker
• [Asterisk-Users] Command Reference   Chris
• [Asterisk-Users] Concurrent calls: best provider?   Scott Wolfe
• [Asterisk-Users] Realtime mysql problem?   Matt Schulte
• [Asterisk-Users] Asterisk Voice mail with CCM   Nathan Alberti
• [Asterisk-Users] TE110P/Hipath3750 - Yellow Alarm   Jens Kübler
• [Asterisk-Users] Cisco 7940/60 failed to take SIP image from tftp server   Davin O'Neill
• [Asterisk-Users] realtime management for sip with mysql   Bioz
• [Asterisk-Users] using asterisk as a gateway for residential IP telephony clients   Video Dery / Internet du Royaume
• [Asterisk-Users] Channel bank question   Sean Kennedy
• [Asterisk-Users] Re: busy line status on CISCO 7940/7960   Sergio
• [Asterisk-Users] Monitor Command   Daniel Burget
• [Asterisk-Users] Buying some Polycom IP300s   Steve Hsieh
• [Asterisk-Users] Digtial Receptionist Recorded Greeting LocationProblem   mr. barker
• [Asterisk-Users] missing ring-tone   Lars L. Christensen
• [Asterisk-Users] Command Reference   Wiley Siler
• [Asterisk-Users] Question about sending Caller-ID Name over PRI NI2 bug 3554   Chris Modesitt
• [Asterisk-Users] Channel bank question   Alexander Lopez
• [Asterisk-Users] TE110P/Hipath3750 - Yellow Alarm   Henry Jensen
• [Asterisk-Users] Concurrent calls: best provider?   Wiley Siler
• [Asterisk-Users] realtime management for sip with mysql   William M. Sandiford
• [Asterisk-Users] voicemail access   richard Coco
• [Asterisk-Users] SIPTone II and PoE   George Gardiner
• [Asterisk-Users] asterisk sounds   Dov Bigio
• [Asterisk-Users] multiple PBXs on one server.   Scott
• [Asterisk-Users] AAH 0.6 to 0.8 Upgrade   Wiley Siler
• [Asterisk-Users] multiple PBXs on one server.   Josiah Bryan
• [Asterisk-Users] Cisco 7960 SIP images   Glenn Powers
• [Asterisk-Users] Sound quality with Xten Xlite softphones...   Maik Hassel
• [Asterisk-Users] WRT54GP2A-AT   Dalon Westergreen
• [Asterisk-Users] multiple PBXs on one server.   Wiley Siler
• [Asterisk-Users] multiple PBXs on one server.   William Boehlke
• [Asterisk-Users] Command Reference   Steven Critchfield
• [Asterisk-Users] Transient SIP Registration Issues   Richard Sears
• [Asterisk-Users] multiple PBXs on one server.   Eric Wieling aka ManxPower
• [Asterisk-Users] asterisk sounds   Josiah Bryan
• Fwd: [Asterisk-Users] CheckGroup and transfers   Sean A. Newton
• [Asterisk-Users] Cisco 7940/60 failed to take SIP image from tftp server   John Breeden
• [Asterisk-Users] "Multiplexing" (or what ever the term is) FXO ports into a "Trunk"   David John Walsh
• [Asterisk-Users] missing ring-tone   Steve Hanselman
• [Asterisk-Users] Accessing Conferencing Bridges   Robert Spangler
• [Asterisk-Users] multiple PBXs on one server.   Race Vanderdecken
• [Asterisk-Users] PortaSIP/PortaBilling incompatibility (provider: sipcall.ch)   Andreas Greulich
• [Asterisk-Users] Queue works, but the caller hears silence instead of ring tone   Jon Califf
• [Asterisk-Users] "Multiplexing" (or what ever the term is) FXOports into a "Trunk"   Steve Mann
• [Asterisk-Users] asterisk sounds   Eric Wieling aka ManxPower
• [Asterisk-Users] asterisk sounds   MF Hulber
• [Asterisk-Users] Transient SIP Registration Issues   Kanuri, Seshu (Company IT)
• [Asterisk-Users] asterisk sounds   Leandro Morgado
• [Asterisk-Users] "Multiplexing" (or what ever the term is) FXO ports into a "Trunk"   Josiah Bryan
• [Asterisk-Users] asterisk on UML   Michael Greb
• [Asterisk-Users] TE405P takes ~5mins to load.   Paul Belanger
• [Asterisk-Users] using asterisk as a gateway for residential IP telephony clients   Matt
• [Asterisk-Users] asterisk sounds   Josiah Bryan
• [Asterisk-Users] "Multiplexing" (or what ever the term is) FXOports into a "Trunk"   David John Walsh
• [Asterisk-Users] TE110P/Hipath3750 - Yellow Alarm   Peter Svensson
• [Asterisk-Users] Sound quality with Xten Xlite softphones...   Bruno Hertz
• [Asterisk-Users] TE405P takes ~5mins to load.   Andrew Kohlsmith
• [Asterisk-Users] chan_iax2 stops listening to packets   Martijn van Oosterhout
• [Asterisk-Users] AGI call problem   Adam Robins
• [Asterisk-Users] Operators guide   Cameron Beattie
• [Asterisk-Users] TE405P takes ~5mins to load.   Derrick Knight
• [Asterisk-Users] asterisk sounds   David John Walsh
• [Asterisk-Users] Polycom IP500 does not show elapsed call time on LCD.   Kanuri, Seshu (Company IT)
• [Asterisk-Users] Sound quality with Xten Xlite softphones...   Wiley Siler
• [Asterisk-Users] AGI call problem   Josiah Bryan
• [Asterisk-Users] Should PRI running over t100p be able to survive short yellow alarms?   Kris Boutilier
• [Asterisk-Users] help regading outbound calls   Cameron Beattie
• [Asterisk-Users] Cisco 7940/60 failed to take SIP image from tftp server   Ron Wellsted
• [Asterisk-Users] asterisk sounds   Carlos Rojas
• [Asterisk-Users] asterisk sounds   Josiah Bryan
• [Asterisk-Users] Sound quality with Xten Xlite softphones...   Sascha Ferley
• [Asterisk-Users] Weird Errors with Realtime and MySQL   William M. Sandiford
• [Asterisk-Users] contracting   CJ Toma
• [Asterisk-Users] Queue works, but the caller hears silence instead of ring tone   MF Hulber
• [Asterisk-Users] Queue works, but the caller hears silence instead of ring tone   Jon Califf
• [Asterisk-Users] asterisk on UML   Cameron Beattie
• [Asterisk-Users] Should PRI running over t100p be able to survive short yellow alarms?   Peter Svensson
• [Asterisk-Users] Dialogic D/300SC-1E1 and D/600SC-2E1 with *   Richard Dutton
• [Asterisk-Users] using asterisk as a gateway for residential IP telephony clients   Andrew Latham
• [Asterisk-Users] AGI call problem   Adam Robins
• [Asterisk-Users] asterisk sounds   Steve Mann
• [Asterisk-Users] asterisk on UML   Steve Mann
• [Asterisk-Users] asterisk sounds   Dov Bigio
• [Asterisk-Users] Dialogic D/300SC-1E1 and D/600SC-2E1 with *   tim panton
• [Asterisk-Users] Queue works, but the caller hears silence instead of ring tone   Jon Califf
• [Asterisk-Users] Queue works, but the caller hears silence instead of ring tone   Jon Califf
• [Asterisk-Users] Set system time over the phone   Mike Sander
• [Asterisk-Users] Set system time over the phone   Mike Sander
• [Asterisk-Users] realtime management for sip with mysql   Cirelle Internet Products
• [Asterisk-Users] Multiple CDR Locations   Aaron Daniel
• [Asterisk-Users] Best Performance   tim panton
• [Asterisk-Users] Re: Outgoing faxes with chan_capi?   Stefan Tichy
• [Asterisk-Users] Cisco 7940/60 failed to take SIP image from tftp server   Davin O'Neill
• [Asterisk-Users] WRT54GP2A-AT   Damon Estep
• [Asterisk-Users] Multiple CDR Locations   Fares Gianluca
• [Asterisk-Users] Re: mISDN + chan_misdn and DTMF   Stefan Tichy
• [Asterisk-Users] Multiple CDR Locations   Kristof Hardy
• [Asterisk-Users] Cisco 7940/60 failed to take SIP image from tftp server   Ron Wellsted
• [Asterisk-Users] Weird Errors with Realtime and MySQL   William M. Sandiford
• [Asterisk-Users] new user TDM400P and T1 card problems   Ben Davidson
• [Asterisk-Users] WRT54GP2A-AT   Wiley Siler
• [Asterisk-Users] new user TDM400P and T1 card problems   Ben Davidson
• [Asterisk-Users] new user TDM400P and T1 card problems   Tim Pushor
• [Asterisk-Users] zaptel not starting issues   Cameron Beattie
• [Asterisk-Users] Dialogic D/300SC-1E1 and D/600SC-2E1 with *   Michael Bielicki
• [Asterisk-Users] TE405P and Dell Poweredge 6450 Incompatible?   mboehm at cytelcom.com
• [Asterisk-Users] realtime management for sip with mysql   William M. Sandiford
• [Asterisk-Users] Set system time over the phone   Kenneth Porter
• [Asterisk-Users] Atcom AT-320 multiple lines?   Gavin Hamill
• [Asterisk-Users] WRT54GP2A-AT   William M. Sandiford
• [Asterisk-Users] how to make asterisk only for SIP and direct RTP   Cameron Beattie
• [Asterisk-Users] TE405P and Dell Poweredge 6450 Incompatible?   Brian Chrystal
• [Asterisk-Users] Operators guide   Tony Davidson
• [Asterisk-Users] AGI call problem   Mike Roberts
• [Asterisk-Users] Multiple CDR Locations   Chris HARIGA
• [Asterisk-Users] Multiple CDR Locations   Aaron Daniel
• [Asterisk-Users] how to make asterisk only for SIP and direct RTP   Cameron Beattie
• [Asterisk-Users] livevoip callerid   Paul Fielding
• [Asterisk-Users] Multiple CDR Locations   Chris HARIGA
• [Asterisk-Users] Asterisk and C# (Dotnet)   Cameron Beattie
• [Asterisk-Users] Should PRI running over t100p be able to survive short yellow alarms?   Eric Wieling aka ManxPower
• [Asterisk-Users] Cell phone friendly MOH   Iassen Hristov
• [Asterisk-Users] TE405P and Dell Poweredge 6450 Incompatible?   Ben Hencke
• [Asterisk-Users] Asterisk and C# (Dotnet)   Michel Brabants
• [Asterisk-Users] TE405P and Dell Poweredge 6450 Incompatible?   Scott Stingel
• [Asterisk-Users] Petition for IAX firmware   denon
• [Asterisk-Users] Dialogic D/300SC-1E1 and D/600SC-2E1 with *   James Taylor
• [Asterisk-Users] realtime management for sip with mysql   Laurent Foulonneau
• [Asterisk-Users] Petition for IAX firmware   Sean Kennedy
• [Asterisk-Users] Automatic Start   J Hobbs
• [Asterisk-Users] Cell phone friendly MOH   Adam Holt
• [Asterisk-Users] asterisk on UML   Abraham WEI
• [Asterisk-Users] Asterisk and Nortel Meridian DTMF   Carlos Chavez
• [Asterisk-Users] asterisk on UML   Scott
• [Asterisk-Users] Cisco 7940/60 failed to take SIP image from tftp server   John Breeden
• [Asterisk-Users] RE: Digium ISDN card   Scott Stingel
• [Asterisk-Users] Question   Gonchi Mateos
• [Asterisk-Users] AAH 0.6 to 0.8 Upgrade   dean collins
• [Asterisk-Users] Realtime mysql problem?   Matt Schulte
• [Asterisk-Users] Stopping Retransmission Found: 102 Error with Polycom IP300   Min Hwan Chang
• [Asterisk-Users] .conf to realtime conversion script?   Rod Bacon
• [Asterisk-Users] Automatic Start   David Choo
• [Asterisk-Users] Re: Asterisk-Users Digest, Vol 9, Issue 44   Noah Miller
• [Asterisk-Users] Best Performance   Ugur GUNCER
• [Asterisk-Users] Grandstream HandyTone-488, * -> FXO problems   Dan Perik
• [Asterisk-Users] Automatic Start   J Hobbs
• [Asterisk-Users] About Audio Latency from PSTN to SIP   Qiao Yuansong
• [Asterisk-Users] SRV and Asterisk   CM Rahman Jr.
• [Asterisk-Users] really small box   Paul Hales
• [Asterisk-Users] Help with simple callback application from newbie   Herbert Chan
• [Asterisk-Users] Asterisk @ Home 0.8 Question   Listacc
• [Asterisk-Users] query about cdr configuration   deepak.dhiman at uf4.net
• [Asterisk-Users] Asterisk Voice mail with CCM   Nathan Alberti
• [Asterisk-Users] TE110P/Hipath3750 - Yellow Alarm   Henry Jensen
• [Asterisk-Users] query about cdr configuration   Yair Hakak
• [Asterisk-Users] SCCP   Bellows, Jared
• [Asterisk-Users] SCCP   Florian Overkamp
• [Asterisk-Users] Voicemail and SJphone   Kib Eki
• [Asterisk-Users] Fritz Card ISDN in UK - Unable to dial. 0x3301/0x3302 errors.   Paul Redstone
• [Asterisk-Users] TE110P/Hipath3750 - Yellow Alarm   Peter Svensson
• [Asterisk-Users] How can I make base calls with X-Lite via Asterisk?   Abraham WEI
• [Asterisk-Users] Asterisk - SS7 or ISDN   Brian Capouch
• [Asterisk-Users] IAX2 and NATs that increment ports   CuPoTKa
• [Asterisk-Users] How can I make base calls with X-Lite via Asterisk?   Martijn van Oosterhout
• [Asterisk-Users] really small box   Sascha Ferley
• [Asterisk-Users] Petition for IAX firmware   clive at sentechsa.com
• [Asterisk-Users] asterisk on UML   Danny Froberg
• [Asterisk-Users] Re: mISDN + chan_misdn and DTMF   Alex
• [Asterisk-Users] How can I add entry for a UA into asterisk when asterisk is running?   Abraham WEI
• [Asterisk-Users] How can I add entry for a UA into asterisk when asterisk is running?   Danny Froberg
• [Asterisk-Users] X-Lite codec related...   Etienne Pretorius
• [Asterisk-Users] Web interface for realtime Mysql friends/peer   Laurent Foulonneau
• [Asterisk-Users] FXO-FXS parameters   Asterisk user list
• [Asterisk-Users] Web interface for realtime Mysql friends/peer   Ronald Wiplinger
• [Asterisk-Users] Web interface for realtime Mysql friends/peer   Matteo Brancaleoni
• [Asterisk-Users] asterisk sounds   Chris Blake
• [Asterisk-Users] Petition for IAX firmware   Adam Goryachev
• [Asterisk-Users] (no subject)   Paul Redstone
• [Asterisk-Users] fedora 3   Altus Snyman
• [Asterisk-Users] Re: How can I make base calls with X-Lite via Asterisk?   Aldo Bergamini
• [Asterisk-Users] Fritz Card ISDN in UK - Unable to dial. 0x3301/0x3302 errors   Paul Redstone
• [Asterisk-Users] fedora 3   Roger Hanson
• [Asterisk-Users] Fritz Card ISDN in UK - Unable to dial. 0x3301/0x3302 errors   Dave Cotton
• [Asterisk-Users] Asterisk & Windows Messenger 5: Which is the correct/preferred DTMFmode setting?   Evert Meulie
• [Asterisk-Users] wcte11xp works only after cold reboot   Alessio Focardi
• [Asterisk-Users] asterisk is giving error- unable to write audio data codec_speex.so   deepak.dhiman at uf4.net
• [Asterisk-Users] Got S-frame while link down   Henry Jensen
• [Asterisk-Users] Got S-frame while link down   Henry Jensen
• [Asterisk-Users] Problem compiling 2nd AVM Fritz   Shane Dalgleish
• [Asterisk-Users] Petition for IAX firmware   mattf
• [Asterisk-Users] Voicemail and SJphone   Rikard Westlund
• [Asterisk-Users] IPSwitchBoard - Now in Spanish   Thorben Jensen
• [Asterisk-Users] Fritz Card ISDN in UK - Unable to dial. 0x3301/0x3302 errors   Andrew Furey
• [Asterisk-Users] Voicemail and SJphone   Kib Eki
• [Asterisk-Users] Zaptel Compile on a virtual dedicated host.   vgrskovic at optonline.net
• [Asterisk-Users] Snom 190 + NAT   Simon
• [Asterisk-Users] IAX2 and NATs that increment ports   Eric Wieling aka ManxPower
• [Asterisk-Users] Stopping Retransmission Found: 102 Error with Polycom IP300   Eric Wieling aka ManxPower
• [Asterisk-Users] Snom 190 + NAT   Gustavo Russo
• [Asterisk-Users] IAX2 and NATs that increment ports   CuPoTKa
• [Asterisk-Users] fedora 3   Altus Snyman
• [Asterisk-Users] fedora 3   iMRAN
• [Asterisk-Users] Fritz Card ISDN in UK - Unable to dial.   Paul Redstone
• [Asterisk-Users] Zaptel Compile on a virtual dedicated host.   Giles Coochey
• [Asterisk-Users] IAX2 and NATs that increment ports   Eric Wieling aka ManxPower
• [Asterisk-Users] Snom 190 + NAT   Simon
• [Asterisk-Users] Snom 190 + NAT   Gustavo Russo
• [Asterisk-Users] wcte11xp works only after cold reboot   Stuart Hirst
• [Asterisk-Users] how can i connect a cost display on asterisk   Han van Hulst
• [Asterisk-Users] spandsp-0.0.2 configure problem   Parker, Blake (MIS)
• [Asterisk-Users] Syntax checker for Asterisk config files   Chuck Bunn
• [Asterisk-Users] IPTABLES Firewall   Koa CG
• [Asterisk-Users] IPTABLES Firewall   Matt
• [Asterisk-Users] Re: IPTABLES Firewall   Matt
• [Asterisk-Users] how can i connect a cost display on asterisk   ht at phonitel.com
• [Asterisk-Users] Re: asterisk on UML   John Goerzen
• [Asterisk-Users] Cisco 7960 forgets VLAN setting   Ron Wellsted
• [Asterisk-Users] how can i connect a cost display on asterisk   Han van Hulst
• [Asterisk-Users] IPTABLES Firewall   Sean Kennedy
• [Asterisk-Users] SMS with VOIP phone WIP 5000 from hitachi   Jerry Geis
• [Asterisk-Users] Grandstream HandyTone-488, * -> FXO problems   Wai Wu
• [Asterisk-Users] how can i connect a cost display on asterisk   ht at phonitel.com
• [Asterisk-Users] how can i connect a cost display on asterisk   Steve Totaro
• [Asterisk-Users] Stopping Retransmission Found: 102 Error with Polycom IP300   Julian J. M.
• [Asterisk-Users] Multiple BroadVoice Accounts Problem with Incoming calls   David Shaw
• [Asterisk-Users] IPTABLES Firewall   Matt
• [Asterisk-Users] IPTABLES Firewall   Matt
• [Asterisk-Users] how can i connect a cost display on asterisk   Peter Corlett
• [Asterisk-Users] Dialogic D/300SC-1E1 and D/600SC-2E1 with *   Richard Dutton
• [Asterisk-Users] TE405P and Dell Poweredge 6450 Incompatible?   David Brodbeck
• [Asterisk-Users] Re: IPTABLES Firewall   CuPoTKa
• [Asterisk-Users] Dialogic D/300SC-1E1 and D/600SC-2E1 with *   Eric Wieling aka ManxPower
• [Asterisk-Users] Atcom AT-320 multiple lines?   Kanuri, Seshu (Company IT)
• [Asterisk-Users] Re: IPTABLES Firewall   Matt
• [Asterisk-Users] Atcom AT-320 multiple lines?   Gavin Hamill
• [Asterisk-Users] Latest Bristuff crashes on modprobe -r qozap ?   Robert Rozman
• [Asterisk-Users] Comedian Voicemail Issues   Carlos Lenz
• [Asterisk-Users] Fritz Card ISDN in UK - Unable to dial.0x3301/0x3302 errors   Craig Guy
• [Asterisk-Users] TE405P and Dell Poweredge 6450 Incompatible?   mboehm at cytelcom.com
• [Asterisk-Users] Asterisk and phone system   Jeffrey Sharpe
• [Asterisk-Users] Syntax checker for Asterisk config files   Mark Charlton
• [Asterisk-Users] Multiple BroadVoice Accounts Problem with Incoming calls   Randy Johnson
• [Asterisk-Users] looking for some draft (sip - iax2 mapping)   Atif Rasheed
• [Asterisk-Users] Asterisk and phone system   Wiley Siler
• [Asterisk-Users] dial out and "all circuits are busy"   J. Arnaud
• [Asterisk-Users] V-9970 Paging Setup   Mike Flynn
• [Asterisk-Users] Latest Bristuff crashes on modprobe -r qozap ?   Michael Bielicki
• [Asterisk-Users] dial out and "all circuits are busy"   Eric Wieling aka ManxPower
• [Asterisk-Users] ser <-> asterisk configs anyone?   G.Marshall
• [Asterisk-Users] Multiple BroadVoice Accounts Problem with Incoming calls {Scanned}   David Shaw
• [Asterisk-Users] Call gets cut off after 5 minute   Wai Wu
• [Asterisk-Users] how can i connect a cost display on asterisk   Jorge Alayon
• [Asterisk-Users] ser <-> asterisk configs anyone?   Steve Mann
• [Asterisk-Users] RE: Asterisk and phone system   Jason Kawakami
• [Asterisk-Users] RE: Asterisk and phone system   Eric Wieling aka ManxPower
• [Asterisk-Users] Cant Hear Any Sound   Ugur GUNCER
• [Asterisk-Users] Stopping Retransmission Found: 102 Error with Polycom IP300   Min Hwan Chang
• [Asterisk-Users] Any success with BRI in the US?   Patrick Conroy
• [Asterisk-Users] Web interface for realtime Mysql friends/peer   Laurent Foulonneau
• [Asterisk-Users] Fritz Card ISDN in UK - Unable to   Paul Redstone
• [Asterisk-Users] Script Perl Autodialer   Maximiliano J. Goldsmid
• [Asterisk-Users] Keypad disabled on AriaVoice SIP phone   Ian Pattison
• [Asterisk-Users] Script Perl Autodialer   Adam Goryachev
• [Asterisk-Users] I want to call another pc with TDM11B Card   Yousri Farouk
• [Asterisk-Users] RE: Asterisk and phone system   Jeffrey Sharpe
• [Asterisk-Users] Keypad disabled on AriaVoice SIP phone   Jon Califf
• [Asterisk-Users] Script Perl Autodialer   Brancaleoni Matteo
• [Asterisk-Users] Keypad disabled on AriaVoice SIP phone   Kanuri, Seshu (Company IT)
• [Asterisk-Users] Keypad disabled on AriaVoice SIP phone   Ian Pattison
• [Asterisk-Users] NOTICE: chan_sip:7654 handle_request: Registeration failed HELP !!   Manjit Riat
• [Asterisk-Users] Web interface for realtime Mysql friends/peer   G.Marshall
• [Asterisk-Users] ser <-> asterisk configs anyone?   G.Marshall
• [Asterisk-Users] Keypad disabled on AriaVoice SIP phone   Ian Pattison
• [Asterisk-Users] Web interface for realtime Mysql friends/peer   Kanuri, Seshu (Company IT)
• [Asterisk-Users] Automatic Start   Cameron Beattie
• [Asterisk-Users] SIP - SIP Problems   Ian Pattison
• [Asterisk-Users] Connecting asterisk to existing PBX - newbie question   Dan Reagan
• [Asterisk-Users] asterisk is giving error- unable to write audiodata codec_speex.so   Cameron Beattie
• [Asterisk-Users] SRV Bounty   Matt Schulte
• [Asterisk-Users] Asterisk and phone system   Cameron Beattie
• [Asterisk-Users] Problems using Asterisk 1.0.3 with Vocal 1.4   Daniel Mizrachi (Globalsip)
• [Asterisk-Users] Ingate Firewall and Asterisk Integration   c waddy
• [Asterisk-Users] wcte11xp works only after cold reboot   Ron Wellsted
• [Asterisk-Users] Keypad disabled on AriaVoice SIP phone   Kanuri, Seshu (Company IT)
• [Asterisk-Users] Asterisk .call files   Kanuri, Seshu (Company IT)
• [Asterisk-Users] Asterisk .call files   Alexander Lopez
• [Asterisk-Users] Asterisk, ACD, Queues and Call Transfer Issue   Dov Bigio
• [Asterisk-Users] Asterisk .call files   Andrew Kohlsmith
• [Asterisk-Users] Asterisk .call files   João Amaro
• [Asterisk-Users] Asterisk .call files   Gilbert Abboud
• [Asterisk-Users] RE:RE: RE: Asterisk and phone system   Jason Kawakami
• [Asterisk-Users] Keypad disabled on AriaVoice SIP phone   Ian Pattison
• [Asterisk-Users] Grandstream HandyTone-488, * -> FXO problems   Andrejus Stavickis
• [Asterisk-Users] Cisco 7960 forgets VLAN setting   Ron Wellsted
• [Asterisk-Users] SIP messages truncated to 256 characters   Hendrik Magilsen
• [Asterisk-Users] Liveviop problem   Andrejus Stavickis
• [Asterisk-Users] Cisco 7940 Outgoing Audio   Bellows, Jared
• [Asterisk-Users] "Choppy" sounds after transferring to ISDN client or after a time   Michiel van Baak
• [Asterisk-Users] Cisco 7940 Outgoing Audio   Alexander Lopez
• [Asterisk-Users] Liveviop problem   Wiley Siler
• [Asterisk-Users] Liveviop problem   geek at mnwifi.net
• [Asterisk-Users] SIP messages truncated to 256 characters   Mikael Magnusson
• [Asterisk-Users] wcte11xp works only after cold reboot   Stuart Hirst
• [Asterisk-Users] Cisco 7940 Outgoing Audio   Andy Hamilton
• [Asterisk-Users] AMP & Handset Provisioning   Stuart Hirst
• [Asterisk-Users] Liveviop problem   Wiley Siler
• [Asterisk-Users] wcte11xp works only after cold reboot   Rich Adamson
• [Asterisk-Users] wcte11xp works only after cold reboot   Ron Wellsted
• [Asterisk-Users] AMP & Handset Provisioning   Jason Becker
• [Asterisk-Users] Liveviop problem   Rich Adamson
• [Asterisk-Users] How to avoid that certain calls come into the voicemail (e.g. wakeup calls)?   Ronald Wiplinger
• [Asterisk-Users] Context overlap?   Dylan VanHerpen
• [Asterisk-Users] How to avoid that certain calls come into the voicemail (e.g. wakeup calls)?   Andy Hamilton
• [Asterisk-Users] Cisco 7940 SIP and No compatible codecs!   Ron Joffe
• [Asterisk-Users] Context overlap?   Moises Silva
• [Asterisk-Users] Realtime UPDATE   Rod Bacon
• [Asterisk-Users] Realtime UPDATE   Rod Bacon
• [Asterisk-Users] DIDs in 510, 408, 916 415 area code   M. Ehsanul Karim
• [Asterisk-Users] how can i connect a cost display on asterisk   Hakem Taourchi
• [Asterisk-Users] Realtime UPDATE   Thierry Wehr
• [Asterisk-Users] DIDs in 510, 408, 916 415 area code   Vern Norman
• [Asterisk-Users] Realtime UPDATE   Rod Bacon
• [Asterisk-Users] How to avoid that certain calls come into thevoicemail (e.g. wakeup calls)?   Rod Bacon
• [Asterisk-Users] rookie question   Randy Paries
• [Asterisk-Users] Polycom IP 600 not ringing   Alex
• [Asterisk-Users] Polycom IP 600 not ringing   Alex
• [Asterisk-Users] Context overlap?   Dylan VanHerpen
• [Asterisk-Users] No more updates of IP address and port in CVS HEAD   Rod Bacon
• [Asterisk-Users] Web interface for realtime Mysql friends/peer   Matt Gibson
• [Asterisk-Users] Asterisk and broadsoft   Digital Support Technologies
• [Asterisk-Users] DIDs in 510, 408, 916 415 area code   Digital Support Technologies
• [Asterisk-Users] SIP messages truncated to 256 characters   Hendrik A. Magilsen
• [Asterisk-Users] SIP - SIP Problems   Joseph
• [Asterisk-Users] SIP - SIP Problems   Rod Bacon
• [Asterisk-Users] how can i connect a cost display on asterisk   Eric Wieling
• [Asterisk-Users] Asterisk on Slack 10.0   Gary Guthary
• [Asterisk-Users] Polycom IP 600 not ringing   Eric Wieling
• [Asterisk-Users] Asterisk on Slack 10.0   Andrew Kohlsmith
• [Asterisk-Users] Asterisk on Slack 10.0   snacktime
• [Asterisk-Users] Receiving calls from and to H323 devices   Guillermo Salas M.
• [Asterisk-Users] Direct Channel Answering   David Choo
• [Asterisk-Users] Account Codes with SIP   Matt Darnell
• [Asterisk-Users] Problems with new Asterisk@Home install and Broadvoice, no incoming calls   Craig Simon
• [Asterisk-Users] Asterisk on Slack 10.0   mattf
• [Asterisk-Users] Help with simple callback application from newbie   CM Rahman Jr.
• [Asterisk-Users] MWI for SER and Asterisk - ast_data vs "realtime"   Cameron Beattie
• [Asterisk-Users] SIP Softphone for testing with Asterisk   raymond
• [Asterisk-Users] SIP Softphone for testing with Asterisk   Brian Capouch
• [Asterisk-Users] rout call from ser to asterisk   Kamran Ahmad
• [Asterisk-Users] SIP Softphone for testing with Asterisk   Rod Bacon
• [Asterisk-Users] Any gsm -> g7231 codec translator?   cmisip
• [Asterisk-Users] Polycom IP 600 not ringing   Alex
• [Asterisk-Users] Realtime UPDATE   Arnaud PIGNARD
• [Asterisk-Users] Beeps during Sip to Sip phone calls   Me
• [Asterisk-Users] Help using wav files for IVR   beonice
• [Asterisk-Users] Configuring the Sipura for static IP and registering with Asterisk.   Bharat M. Sarvan
• [Asterisk-Users] Account Codes with SIP   Nabeel Jafferali
• [Asterisk-Users] Re: Broadvoice   Craig Simon
• [Asterisk-Users] Help using wav files for IVR   BJ Weschke
• [Asterisk-Users] Beeps during Sip to Sip phone calls   Me
• [Asterisk-Users] Curry and Asterisk   Paul Hales
• [Asterisk-Users] Account Codes with SIP   Matt Darnell
• [Asterisk-Users] busy line status on CISCO 7940/7960   Julien Goodwin
• [Asterisk-Users] Curry and Asterisk   Kimble Young
• [Asterisk-Users] About Audio Latency from PSTN to SIP   Qiao Yuansong
• [Asterisk-Users] Realtime UPDATE   Rod Bacon
• [Asterisk-Users] Curry and Asterisk   Jay Milk
• [Asterisk-Users] Direct Channel Answering   Cameron Beattie
• [Asterisk-Users] Beeps during Sip to Sip phone calls   Rich Adamson
• [Asterisk-Users] Curry and Asterisk   Cameron Beattie
• [Asterisk-Users] Configuring the Sipura for static IP and registering with Asterisk.   Rich Adamson
• [Asterisk-Users] rout call from ser to asterisk   Cameron Beattie
• [Asterisk-Users] Beeps during Sip to Sip phone calls   Cameron Beattie
• [Asterisk-Users] Beeps during Sip to Sip phone calls   Rod Bacon
• [Asterisk-Users] Broadvoice not ringing the phone   Craig Simon
• [Asterisk-Users] Liveviop problem   Paul Fielding
• [Asterisk-Users] X100P call keeps ringing and ringing   asterisk at mazone.info
• [Asterisk-Users] Beeps during Sip to Sip phone calls   Daryll Strauss
• [Asterisk-Users] Latest CVS chokes Sipura SPA-841   Brian Capouch
• [Asterisk-Users] Cant Hear Any Sounds   Ugur GUNCER
• [Asterisk-Users] Digium TDM400 Failover on Power Loss   Mike Sander
• [Asterisk-Users] Help using wav files for IVR   Bill Ford
• [Asterisk-Users] Measure the Signal of Zap   Damian Minkov
• [Asterisk-Users] Acccess Voice Mail From Outside Line   Bill Ford
• [Asterisk-Users] Re: X100P interrupt load   parijat
• [Asterisk-Users] Connecting asterisk to existing PBX - newbie question   Sascha Ferley
• [Asterisk-Users] Call Interception   cereal killer
• [Asterisk-Users] Access Voicemail From Outside   Bill Ford
• [Asterisk-Users] Digium TDM400 Failover on Power Loss   Bill Ford
• [Asterisk-Users] Cisco 7940 Outgoing Audio   Julien Goodwin
• [Asterisk-Users] Access Voicemail From Outside   ht at phonitel.com
• [Asterisk-Users] Access Voicemail From Outside   Bill Ford
• [Asterisk-Users] capi segfault when incoming call is answered   Thomas Andrews
• [Asterisk-Users] about mpg123   Andrea Riela
• [Asterisk-Users] Manually adjusting volume in an IAX channel   Caleb
• [Asterisk-Users] rookie question   Kris Edwards
• [Asterisk-Users] Call Interception   Wilson Pickett
• [Asterisk-Users] Fax - Capi   Dennie Verstrepen
• [Asterisk-Users] Direct Channel Answering   David Choo
• [Asterisk-Users] Access Voicemail From Outside   David Choo
• [Asterisk-Users] How Can i phone traditional PSTN Phone using TDM11B   Yousri Farouk
• [Asterisk-Users] RE: Sangoma VS. Digium   cmould
• [Asterisk-Users] Linux & Asterisk   Asterisk Pbx
• [Asterisk-Users] capi segfault when incoming call is answered   Thomas Andrews
• [Asterisk-Users] SIP Softphone for testing with Asterisk   Esben Stien
• [Asterisk-Users] Linux & Asterisk   Kristof Hardy
• [Asterisk-Users] Help using wav files for IVR   trixter http://www.0xdecafbad.com
• [Asterisk-Users] Re: Broadvoice   trixter http://www.0xdecafbad.com
• [Asterisk-Users] Linux & Asterisk   Matteo Brancaleoni
• [Asterisk-Users] Sangoma VS. Digium   Roy Sigurd Karlsbakk
• [Asterisk-Users] SIP Softphone for testing with Asterisk   raymond
• [Asterisk-Users] MWI for SER and Asterisk - ast_data vs "realtime"   Steve Blair
• [Asterisk-Users] Integration of Alcatel 4400 with Asterisk   corroy steven
• [Asterisk-Users] rout call from ser to asterisk   Steve Blair
• [Asterisk-Users] Sangoma VS. Digium   Roy Sigurd Karlsbakk
• [Asterisk-Users] Sangoma VS. Digium   Roy Sigurd Karlsbakk
• [Asterisk-Users] [again] Sangoma PRI vs TE410?   Roy Sigurd Karlsbakk
• [Asterisk-Users] Help: Problem with X101P   Yusuf Iqbal
• [Asterisk-Users] MWI for SER and Asterisk - ast_data vs "realtime"   Cameron Beattie
• [Asterisk-Users] Sangoma VS. Digium   Nir Simionovich
• [Asterisk-Users] MWI for SER and Asterisk - ast_data vs "realtime"   Steve Blair
• [Asterisk-Users] Sangoma VS. Digium   Matteo Brancaleoni
• [Asterisk-Users] Astcc Problems   chawki hammoud
• [Asterisk-Users] busy line status on CISCO 7940/7960   Joseph
• [Asterisk-Users] keeping dynamic queue members over restart?   Roy Sigurd Karlsbakk
• [Asterisk-Users] SIP Softphone for testing with Asterisk   iMRAN
• [Asterisk-Users] MWI for SER and Asterisk - ast_data vs "realtime"   Steve Blair
• [Asterisk-Users] T.38 fax with SIP devices   Mark Dutton
• [Asterisk-Users] Sangoma VS. Digium   ht at phonitel.com
• [Asterisk-Users] Sangoma VS. Digium   Matteo Brancaleoni
• [Asterisk-Users] Micronet 128K TA Card   Ivan Meic (Vox Mundi)
• [Asterisk-Users] slightly-ot: Where to buy sip phones in Massachusetts   Mark Charlton
• [Asterisk-Users] "404 User Not Found" when calling between two SIP UA's   Abraham WEI
• [Asterisk-Users] Voice mail through Call Manager   Mohamed Farid
• [Asterisk-Users] Sangoma VS. Digium   ht at phonitel.com
• [Asterisk-Users] voip phone reviews   Jon Lawrence
• [Asterisk-Users] Sangoma VS. Digium   Peter Svensson
• [Asterisk-Users] Acccess Voice Mail From Outside Line   dean collins
• [Asterisk-Users] capi segfault when incoming call is answered   Craig Guy
• [Asterisk-Users] Sangoma VS. Digium   Adam Goryachev
• [Asterisk-Users] GSM Hardware Setup   Jan Kellerhoff
• [Asterisk-Users] keeping dynamic queue members over restart?   Adam Goryachev
• [Asterisk-Users] [again] Sangoma PRI vs TE410?   Andrew Kohlsmith
• [Asterisk-Users] Sangoma VS. Digium   dean collins
• [Asterisk-Users] GSM Hardware Setup   ht at phonitel.com
• [Asterisk-Users] Micronet 128K TA Card   Ivan Meic (Vox Mundi)
• [Asterisk-Users] IAX2 trunk frames documentation.   Marcin Okraszewszki
• [Asterisk-Users] call behind NAT   Kamran Ahmad
• [Asterisk-Users] Re: [again] Sangoma PRI vs TE410?   Tony Mountifield
• [Asterisk-Users] Micronet 128K TA Card   Robson Ribeiro
• [Asterisk-Users] how can i connect a cost display on asterisk   Nicolás Gudiño
• [Asterisk-Users] T.38 fax with SIP devices   Clay Reiche
• [Asterisk-Users] Re: [again] Sangoma PRI vs TE410?   Andrew Kohlsmith
• [Asterisk-Users] Beeps during Sip to Sip phone calls   John Novack
• [Asterisk-Users] Call Interception   cereal killer
• [Asterisk-Users] Micronet 128K TA Card   Robson Ribeiro
• [Asterisk-Users] SIP - SIP Problems   Ian Pattison
• [Asterisk-Users] Beeps during Sip to Sip phone calls   Eric Wieling
• [Asterisk-Users] Cisco ATA 186: Only incoming - no outgoing call   Vyom A
• [Asterisk-Users] Re: [again] Sangoma PRI vs TE410?   Michael Bielicki
• [Asterisk-Users] Realtime UPDATE   Matthew Boehm
• [Asterisk-Users] MWI for SER and Asterisk - ast_data vs "realtime"   Matthew Boehm
• [Asterisk-Users] mangle + to 00   marek cervenka
• [Asterisk-Users] Call Interception   Matteo Brancaleoni
• [Asterisk-Users] Call Interception   Josiah Bryan
• [Asterisk-Users] Any success with BRI in the US?   Brian G
• [Asterisk-Users] QoS for Asterisk with ISA 2004 Server?   Zeno Lee
• [Asterisk-Users] SIP Softphone for testing with Asterisk   Guillermo Salas M
• [Asterisk-Users] Receiving calls from and to H323 devices   Guillermo Salas M
• [Asterisk-Users] Liveviop problem   Andrejus Stavickis
• [Asterisk-Users] mangle + to 00   Scott Nelson
• [Asterisk-Users] MWI for SER and Asterisk - ast_data vs "realtime"   Matteo Brancaleoni
• [Asterisk-Users] Sangoma VS. Digium   Dana Olson
• [Asterisk-Users] Call Interception   Ivan Meic (Vox Mundi)
• [Asterisk-Users] MWI for SER and Asterisk - ast_data vs "realtime"   Matteo Brancaleoni
• [Asterisk-Users] Sangoma VS. Digium   Scott Stingel
• [Asterisk-Users] Liveviop problem   Eric Wieling
• [Asterisk-Users] [OT]: Wiki Etiquette   Sean Kennedy
• [Asterisk-Users] SpanDSP HELP   Parker, Blake (MIS)
• [Asterisk-Users] open source Asterisk Application of the year?   ht at phonitel.com
• [Asterisk-Users] Call Interception   Matteo Brancaleoni
• [Asterisk-Users] Digium TDM400 Failover on Power Loss   Chris Mason
• [Asterisk-Users] My Sangoma Experience - Review   mattf
• [Asterisk-Users] fault tolerant asterisk system   Roy Sigurd Karlsbakk
• [Asterisk-Users] IAX2 trunk frames documentation.   Michael Bielicki
• [Asterisk-Users] Sangoma VS. Digium   Richard Lyman
• [Asterisk-Users] Account Codes with SIP   Nabeel Jafferali
• [Asterisk-Users] Call Interception   Rich Adamson
• [Asterisk-Users] MWI for SER and Asterisk - ast_data vs "realtime"   Matthew Boehm
• [Asterisk-Users] RE: Sangoma VS. Digium   Scott Stingel
• [Asterisk-Users] IVR - newbie question   Paul Hewlett
• [Asterisk-Users] X100p, IRQ and Noises   Antonio José dos Santos Brandão
• [Asterisk-Users] RE: Sangoma VS. Digium   Sergey Kuznetsov
• [Asterisk-Users] SetCdrUserField   Luca Maccarini
• [Asterisk-Users] Asterisk and clarent   Rafael J. Risco G.V.
• [Asterisk-Users] My Sangoma Experience - Review   Scott Stingel
• [Asterisk-Users] Liveviop problem   Andrejus Stavickis
• [Asterisk-Users] Fax to email problem   Chris Blake
• [Asterisk-Users] Voicemail localization   Guy Decarpentrie
• [Asterisk-Users] T.38 fax with SIP devices   Moody
• [Asterisk-Users] Liveviop problem   Mike Benoit
• [Asterisk-Users] Liveviop problem   Rich Adamson
• [Asterisk-Users] unlimited iax termination   joe at jsci.net
• [Asterisk-Users] [OT]: Wiki Etiquette   Scott Stingel
• [Asterisk-Users] [Serusers] No translator path exists for channel type H323 :-| :-| :-|   Carlos Maynard
• [Asterisk-Users] T.38 fax with SIP devices   Mark Dutton
• [Asterisk-Users] PRI Advice...   Matt Loretitsch
• [Asterisk-Users] My Sangoma Experience - Review   Danny Froberg
• [Asterisk-Users] patch to add distinctive ringing to queues   Louis-David Mitterrand
• [Asterisk-Users] Re: Ring Twice   VoIP Newbie
• [Asterisk-Users] Call Interception   Rich Adamson
• [Asterisk-Users] Call Interception   Josiah Bryan
• [Asterisk-Users] Latest CVS chokes Sipura SPA-841   Danny Froberg
• [Asterisk-Users] T.38 fax with SIP devices   Scott Wolfe
• [Asterisk-Users] open source Asterisk Application of the year ?   mattf
• [Asterisk-Users] Liveviop problem   Wiley Siler
• [Asterisk-Users] [OT]: Wiki Etiquette   Ariel Batista
• [Asterisk-Users] RE: Sangoma VS. Digium   ht at phonitel.com
• [Asterisk-Users] My Sangoma Experience - Review   Tom
• [Asterisk-Users] IVR - newbie question   Danny Froberg
• [Asterisk-Users] FGD Support   Michael Baird
• [Asterisk-Users] Call Interception   Florian Overkamp
• [Asterisk-Users] CDR mysql userfield column truncated at 239 characters   Troy Swaine
• [Asterisk-Users] Asterisk and clarent   Danny N
• [Asterisk-Users] unlimited iax termination   Damon Estep
• [Asterisk-Users] Fax to email problem   Guy Decarpentrie
• [Asterisk-Users] AES vs AEC   1 2
• [Asterisk-Users] IVR - newbie question   Wiley Siler
• [Asterisk-Users] "Choppy" sounds after transferring to ISDN client or after a time   Michiel van Baak
• [Asterisk-Users] PRI Advice...   Damon Estep
• [Asterisk-Users] Asterisk on Slack 10.0   Cameron Schaus
• [Asterisk-Users] SIP - SIP Problems   Ian Pattison
• [Asterisk-Users] Asterisk and clarent   Jorge Alayon
• [Asterisk-Users] Liveviop problem   Mike Benoit
• [Asterisk-Users] Digium TDM400 Failover on Power Loss   Gerard Marcel
• [Asterisk-Users] [OT]: Wiki Etiquette   Alexander Lopez
• [Asterisk-Users] Asterisk and clarent   Alexander Lopez
• [Asterisk-Users] My Sangoma Experience - Review   mattf
• [Asterisk-Users] Re: Ring Twice   Chris Modesitt
• [Asterisk-Users] Can asterisk get code for cmd Authenticate from Database   Bartosz Wegrzyn - asterisk
• [Asterisk-Users] Sangoma VS. Digium   Thorsten lockert
• [Asterisk-Users] voip phone reviews   Nathan Alberti
• [Asterisk-Users] IVR - newbie question   Paul Hewlett
• [Asterisk-Users] My Sangoma Experience - Review   Zoa
• [Asterisk-Users] Liveviop problem   Sergey Kuznetsov
• [Asterisk-Users] Help with simple callback application from newbie   Robert Goodyear
• [Asterisk-Users] AES vs AEC   Peter Svensson
• [Asterisk-Users] SIP - SIP Problems   Tim Pushor
• [Asterisk-Users] unlimited iax termination   Mike Matthews
• [Asterisk-Users] Access Voicemail From Outside   David Brodbeck
• [Asterisk-Users] Liveviop problem   Wiley Siler
• [Asterisk-Users] Lag in meetme   Kris Edwards
• [Asterisk-Users] Liveviop problem   Rich Adamson
• [Asterisk-Users] Liveviop problem   Wiley Siler
• [Asterisk-Users] Issues with ringing on FXS ports   Ian Pattison
• [Asterisk-Users] FGD Support   Jason Miller
• [Asterisk-Users] My Sangoma Experience - Review   mattf
• [Asterisk-Users] How Can I make 2 instances of the nufone H323 channel run?   Kido NOAGBODJI
• [Asterisk-Users] Sangoma VS. Digium   Craig Guy
• [Asterisk-Users] oh323 compilation   Gabriel Millerd
• [Asterisk-Users] Asterisk - SS7 or ISDN   Angel Diaz
• [Asterisk-Users] IPSwitchBoard Version 0.76 released   Thorben Jensen
• [Asterisk-Users] Issues with ringing on FXS ports   Ian Pattison
• [Asterisk-Users] Receiving calls from and to H323 devices   Brian Capouch
• [Asterisk-Users] Issues with ringing on FXS ports   Andrew Kohlsmith
• [Asterisk-Users] Lag in meetme   Matt Ryanczak
• [Asterisk-Users] Asterisk @ Home v0.8   Brian Watters
• [Asterisk-Users] oh323 compilation   Nardis Dome
• [Asterisk-Users] Fax to email problem   Craig Guy
• [Asterisk-Users] Sangoma VS. Digium   Andrew Kohlsmith
• [Asterisk-Users] oh323 compilation   Guillermo Salas M
• [Asterisk-Users] Asterisk Vs. Cisco, et. al.?   Ken D'Ambrosio
• [Asterisk-Users] Zap (analog line) and volume   Maik Hassel
• [Asterisk-Users] Can asterisk get code for cmd Authenticate fromDatabase   Steve Mann
• [Asterisk-Users] Asterisk Vs. Cisco, et. al.?   Justin B Newman
• [Asterisk-Users] IVR - newbie question   Mark Charlton
• [Asterisk-Users] oh323 compilation   Race Vanderdecken
• [Asterisk-Users] Sangoma VS. Digium   Timothy Costello
• [Asterisk-Users] Lag in meetme   BJ Weschke
• [Asterisk-Users] Remembering State   jamesm
• [Asterisk-Users] Database lookups?   Jan Johansson
• [Asterisk-Users] Issues with ringing on FXS ports   Rich Adamson
• [Asterisk-Users] Latest CVS chokes Sipura SPA-841   Brian Capouch
• [Asterisk-Users] Ring Twice   Trevor Peirce
• [Asterisk-Users] Zap (analog line) and volume   Rich Adamson
• [Asterisk-Users] Sangoma VS. Digium   Rich Adamson
• [Asterisk-Users] SIP UA behind NAT and REINVITE ???   William M. Sandiford
• [Asterisk-Users] TE405P vs TE410P   Tony Mountifield
• [Asterisk-Users] Re: Lag in meetme   Tony Mountifield
• [Asterisk-Users] TE405P vs TE410P   Andrew Kohlsmith
• [Asterisk-Users] Database lookups?   Josiah Bryan
• [Asterisk-Users] Voice controlled calling?   magnus
• [Asterisk-Users] "Mic-To-Speaker-loop" on ZAP lines???   Maik Hassel
• [Asterisk-Users] Zap (analog line) and volume   David Brodbeck
• [Asterisk-Users] SIP REFER - find brick wall, bang head, repeat!   Hendrik Magilsen
• [Asterisk-Users] OT: pinout for"standard"telephoneheadsetrequired.?   Mike Dewey
• [Asterisk-Users] Re: TE405P vs TE410P   Tony Mountifield
• [Asterisk-Users] Voice controlled calling?   Andrejus Stavickis
• [Asterisk-Users] Voice controlled calling?   dean collins
• [Asterisk-Users] Voice controlled calling?   Josiah Bryan
• [Asterisk-Users] Database lookups?   Steve Mann
• [Asterisk-Users] Can asterisk get code for cmd AuthenticatefromDatabase   Bartosz Wegrzyn - asterisk
• [Asterisk-Users] SpanDSP HELP   Tzafrir Cohen
• [Asterisk-Users] OT: pinout for"standard"telephoneheadsetrequired.?   Nabeel Jafferali
• [Asterisk-Users] Re: TE405P vs TE410P   Andrew Kohlsmith
• [Asterisk-Users] Voice controlled calling?   Brandon Patterson
• FW: Out of Office AutoReply: [Asterisk-Users] Voice controlled calling?   dean collins
• [Asterisk-Users] Voice controlled calling?   dean collins
• [Asterisk-Users] Voice controlled calling?   Alex Vishnev
• [Asterisk-Users] Help using wav files for IVR   beonice
• [Asterisk-Users] Voice controlled calling?   Gavin Hamill
• [Asterisk-Users] Conferancing with different interface   Ugur GUNCER
• [Asterisk-Users] Zap (analog line) and volume   Steve Mann
• Out of Office AutoReply: [Asterisk-Users] Voice controlledcalling?   Steve Mann
• [Asterisk-Users] oh323 compilation   Tzafrir Cohen
• [Asterisk-Users] Cisco 7960 forgets VLAN setting   Ron Wellsted
• [Asterisk-Users] open source Asterisk Application of the year?   Matt Riddell
• [Asterisk-Users] Database lookups?   Steve Prior
• [Asterisk-Users] open source Asterisk Application of the year?   Michiel van Baak
• [Asterisk-Users] Cisco 7960 forgets VLAN setting   Michiel van Baak
• [Asterisk-Users] Voice controlled calling? Pt 2   magnus
• [Asterisk-Users] x100p disconnect on "D" tone   Brian Leyton
• [Asterisk-Users] Realtime UPDATE   Rod Bacon
• [Asterisk-Users] Using manager interface to play aanouncments in a MeetMe   Dan Austin
• [Asterisk-Users] How to turn off automatic pick up for Incoming calls A@H v0.6   Min Hwan Chang
• [Asterisk-Users] zaptel.conf digium and quadBri together (e1 and isdn together)   Victor Alvarez
• [Asterisk-Users] Sangoma VS. Digium   Matt Riddell
• [Asterisk-Users] Using manager interface to play aanouncments in a MeetMe   mattf
• [Asterisk-Users] fedora 3   C F
• Out of Office AutoReply: [Asterisk-Users] fedora 3   C F
• [Asterisk-Users] zaptel.conf digium and quadBri together (e1 and isdn together)   Scott Stingel
• [Asterisk-Users] open source Asterisk Application of the year?   dean collins
• [Asterisk-Users] Low volume in recorded messages   Chris Mason
• [Asterisk-Users] Local Number Ports   Damon Estep
• [Asterisk-Users] Voice controlled calling?   Steve Kennedy
• [Asterisk-Users] Measure the Signal of Zap   Leandro Morgado
• [Asterisk-Users] Local Number Ports   Matt Klein
• [Asterisk-Users] Using manager interface to play aanouncmentsin a MeetMe   Dan Austin
• [Asterisk-Users] Getting a good deal on a PRI   snacktime
• [Asterisk-Users] Getting a good deal on a PRI   Harry McGregor
• [Asterisk-Users] Measure the Signal of Zap   Leandro Morgado
• [Asterisk-Users] Getting a good deal on a PRI   Michael Welter
• [Asterisk-Users] Can somebody with HEAD please test MOH on agent calls? M3976   Nick Bachmann
• [Asterisk-Users] Re: Beeps during Sip to Sip phone calls   Doug Meredith
• [Asterisk-Users] Using manager interface to play aanouncments in a MeetMe   trixter http://www.0xdecafbad.com
• [Asterisk-Users] secret/username - what does it really do?   Don Murray
• [Asterisk-Users] Livevoip responds to DTMF via IAX issue   The Phone Guys
• [Asterisk-Users] Asterisk Max TNT   Michael Baird
• [Asterisk-Users] Help with simple callback application from newbie   Raf D'Halleweyn (list)
• [Asterisk-Users] Re: Beeps during Sip to Sip phone calls   Michael D Schelin
• [Asterisk-Users] Getting a good deal on a PRI   Michael D Schelin
• [Asterisk-Users] Answering without ringing from PRI   Ugur GUNCER
• [Asterisk-Users] Using manager interface to play aanouncments in a MeetMe   mattf
• [Asterisk-Users] Melbourne Asterisk Consultants   Callum McGillivray
• [Asterisk-Users] stand alone Voice Mail   Michael D Schelin
• [Asterisk-Users] Livevoip responds to DTMF via IAX issue   geek at mnwifi.net
• [Asterisk-Users] About Audio Latency from PSTN to SIP   Qiao Yuansong
• [Asterisk-Users] Answering without ringing from PRI   Andrew Kohlsmith
• [Asterisk-Users] failover outbound dialplan   Jason Brown
• [Asterisk-Users] Answering without ringing from PRI   Mathew McKernan
• [Asterisk-Users] RE: Asterisk-Users Digest, Vol 9, Issue 67   Jeff Glassman
• [Asterisk-Users] failover outbound dialplan   Steve Edwards
• [Asterisk-Users] RE: Asterisk-Users Digest, Vol 9, Issue 67   Jeff Glassman
• [Asterisk-Users] Getting a good deal on a PRI   snacktime
• [Asterisk-Users] "Mic-To-Speaker-loop" on ZAP lines???   Ian Pattison
• [Asterisk-Users] failover outbound dialplan   Ronald Wiplinger
• [Asterisk-Users] Re: Answering without ringing from PRI   LJ
• [Asterisk-Users] MWI for SER and Asterisk - ast_data vs "realtime"   Terry Wilson
• [Asterisk-Users] how to pass G723.1   Kamran Ahmad
• [Asterisk-Users] Help with simple callback application from newbie   CM Rahman Jr.
• [Asterisk-Users] Local Number Ports   Damon Estep
• [Asterisk-Users] Asterisk quit abnormally   Qiao Yuansong
• [Asterisk-Users] Canreinvite issue   kaiser
• [Asterisk-Users] Getting a good deal on a PRI   Michael D Schelin
• [Asterisk-Users] Asterisk Google Group?   Damon Estep
• [Asterisk-Users] APPRADIUS cdr_radius.so   Rod Bacon
• [Asterisk-Users] Re: Answering without ringing from PRI   Ugur GUNCER
• [Asterisk-Users] SRV Bounty   Matt Riddell
• [Asterisk-Users] Answering without ringing from PRI   Ugur GUNCER
• [Asterisk-Users] Asterisk Google Group?   Matt Riddell
• [Asterisk-Users] Asterisk Google Group?   Calvis
• [Asterisk-Users] T.38 fax with SIP devices   Michael D Schelin
• [Asterisk-Users] PRI Advice...   Eric Wieling
• [Asterisk-Users] Looking for feedback on IAX2 Phones from Netweb   Cory Andrews
• [Asterisk-Users] Asterisk Google Group?   Damon Estep
• [Asterisk-Users] Off Topic - Employment Opportunity - PERL, Melbourne, AU.   Rod Bacon
• [Asterisk-Users] Off Topic - Employment Opportunity - PERL, Melbourne, AU.   Jean-Michel Hiver
• [Asterisk-Users] SRV Bounty   Ronald Wiplinger
• [Asterisk-Users] Off Topic - Employment Opportunity - PERL, Melbourne, AU.   Jean-Michel Hiver
• [Asterisk-Users] Reply-To? (was: Off Topic - Employment Opportunity - PERL, Melbourne, AU.)   Jean-Michel Hiver
• [Asterisk-Users] SRV Bounty   Brian Capouch
• [Asterisk-Users] Reply-To?   Brian Capouch
• [Asterisk-Users] Off Topic - Employment Opportunity - PERL, Melbourne, AU.   Rod Bacon
• [Asterisk-Users] Re: Beeps during Sip to Sip phone calls   Luki
• [Asterisk-Users] Sangoma VS. Digium   Brancaleoni Matteo
• [Asterisk-Users] RE:Asterisk Voice mail with CCM   dakemp
• [Asterisk-Users] Re: Livevoip IAX DTMF troubles   Rich Adamson
• [Asterisk-Users] RE:Asterisk Voice mail with CCM   Nathan Alberti
• [Asterisk-Users] Using manager interface to play aanouncmentsin a MeetMe   Dan Austin
• [Asterisk-Users] "404 User Not Found" when calling between two X-Lites   Abraham WEI
• [Asterisk-Users] Canreinvite issue   snacktime
• [Asterisk-Users] iax / realtime problems   Paul P. Pongco
• [Asterisk-Users] failover outbound dialplan   Steve Edwards
• [Asterisk-Users] Fax to email problem   Chris Blake
• [Asterisk-Users] Zap Answer without ringing   Bashir Ullah - www.Lamsre.Com
• [Asterisk-Users] User Regerstation, allowing non-registered users on *   Etienne Pretorius
• [Asterisk-Users] Fax to email problem   Chris Blake
• [Asterisk-Users] external access to voicemail?   Mick Hastings
• [Asterisk-Users] Fax to email problem   Guy Decarpentrie
• [Asterisk-Users] Undefined symbol in res_features & Others   Ow Mun Heng
• [Asterisk-Users] Looking for feedback on IAX2 Phones from Netweb   Wilson Pickett
• [Asterisk-Users] Call Interception   Alex
• [Asterisk-Users] Asterisk based Call Accounting software - 1st release   San Singhania
• [Asterisk-Users] Delayed dial under Asterisk ?   Robert Rozman
• [Asterisk-Users] G723 call through GW   Kamran Ahmad
• [Asterisk-Users] Looking for feedback on IAX2 Phones from Netweb   Wing Hui
• [Asterisk-Users] Answering without ringing from PRI   Steve Hanselman
• [Asterisk-Users] Fw: Registration Problem with Firefly Softphone   raymond
• [Asterisk-Users] oh323 compilation   Michael Manousos
• [Asterisk-Users] Fw: Registration Problem with Firefly Softphone   raymond
• [Asterisk-Users] compiling oh323 Undefined symbol in res_features & Others   Shaoul Jacobson - TELLINK
• [Asterisk-Users] Asterisk and CAS   David Hajek
• [Asterisk-Users] oh323 DTMF bug   Kido NOAGBODJI
• [Asterisk-Users] SIP UA behind NAT and REINVITE ???   Martijn van Oosterhout
• [Asterisk-Users] failover outbound dialplan   Remco Barende
• [Asterisk-Users] Re: busy line status on CISCO 7940/7960   Sergio
• [Asterisk-Users] stand alone Voice Mail   Steve Blair
• [Asterisk-Users] Sangoma VS. Digium   Steve Underwood
• [Asterisk-Users] failover outbound dialplan   Wojciech Tryc
• [Asterisk-Users] Sangoma VS. Digium   Steve Underwood
• [Asterisk-Users] Dialogic D/300SC-1E1 and D/600SC-2E1 with *   Steve Underwood
• [Asterisk-Users] Asterisk and CAS   Steve Underwood
• [Asterisk-Users] Re: Delayed dial under Asterisk ?   Mick Hastings
• [Asterisk-Users] about mpg123   Vahan Yerkanian
• [Asterisk-Users] NVFaxEmail   Justin Newman
• [Asterisk-Users] User Regerstation, allowing non-registered users on *   Martijn van Oosterhout
• [Asterisk-Users] re: external access to voicemail?   Justin Newman
• [Asterisk-Users] RE: [Asterisk-Dev] Re: Livevoip IAX DTMF troubles   Robert Webb
• [Asterisk-Users] Difference Between NAT=yes and QUALIFY=yes and STUN...   Matt
• [Asterisk-Users] Call from publicIP to PrivateIP   Kamran Ahmad
• [Asterisk-Users] linejack and iax2 !   Jalal
• [Asterisk-Users] oh323 DTMF Bug   Kido NOAGBODJI
• [Asterisk-Users] SRV Bounty   Matt Schulte
• [Asterisk-Users] "404 User Not Found" when calling between two X-Lites   Rich Adamson
• [Asterisk-Users] Can a SIP Phone talk directly with anoyher SIP phone (ext a to Ext b)   cmould
• [Asterisk-Users] iax / realtime problems   Matt Schulte
• [Asterisk-Users] Answering without ringing from PRI   Ugur GUNCER
• [Asterisk-Users] RE: [Asterisk-Dev] Re: Livevoip IAX DTMF troubles   Rich Adamson
• [Asterisk-Users] Web interface for realtime Mysql friends/peer   G.Marshall
• [Asterisk-Users] Asterisk and HylaFAX integration   Kevin Brennan
• [Asterisk-Users] Getting a good deal on a PRI   Steve Mann
• [Asterisk-Users] Livevoip responds to DTMF via IAX issue   Rich Adamson
• [Asterisk-Users] Asterisk based Call Accounting software - 1st release   Guillermo Salas M
• [Asterisk-Users] Reply-To?   Bruno Hertz
• [Asterisk-Users] Web interface for realtime Mysql friends/peer   Douglas Conrad
• [Asterisk-Users] X100P doesn't check for dialtone   Malcolm Taylor
• [Asterisk-Users] Getting a good deal on a PRI   Damon Estep
• [Asterisk-Users] Asterisk Google Group?   Bruno Hertz
• [Asterisk-Users] Asterisk .call files   Gilbert Abboud
• [Asterisk-Users] snom and "hint" priority   Michael George
• [Asterisk-Users] Asterisk Google Group?   trixter http://www.0xdecafbad.com
• [Asterisk-Users] inquire about connected channel (show channels)   Jerry Geis
• [Asterisk-Users] Any opinions on quality/service of Teliax?   Jacob Cazzell
• [Asterisk-Users] Web interface for realtime Mysql friends/peer   Jesus Mogollon
• [Asterisk-Users] AMP 1.10.007 problem on cdr_mysql_table.sql   Colin Anderson
• [Asterisk-Users] inquire about connected channel (show channels)   Josiah Bryan
• [Asterisk-Users] Asterisk@Home .8 SPA-2000   David Shaw
• [Asterisk-Users] Web interface for realtime Mysql friends/peer   oguer
• [Asterisk-Users] Asterisk Google Group?   Bruno Hertz
• [Asterisk-Users] T.38 fax with SIP devices   Mark Dutton
• [Asterisk-Users] Any opinions on quality/service of Teliax?   Wiley Siler
• [Asterisk-Users] T.38 fax with SIP devices   Mark Dutton
• [Asterisk-Users] X100P doesn't check for dialtone   John Novack
• [Asterisk-Users] Asterisk based Call Accounting software - 1st release   Ronald Wiplinger
• [Asterisk-Users] AMP 1.10.007 problem on cdr_mysql_table.sql   Kristof Hardy
• [Asterisk-Users] Major issues with VoicePulse today   Zeno Lee
• [Asterisk-Users] Re: Beeps during Sip to Sip phone calls   Rich Adamson
• [Asterisk-Users] PRI card and TDM400P in same box   Jason Brown
• [Asterisk-Users] Any opinions on quality/service of Teliax?   Mark Willis
• [Asterisk-Users] Asterisk based Call Accounting software - 1st release   Jalal
• [Asterisk-Users] AMP 1.10.007 problem on cdr_mysql_table.sql   Colin Anderson
• [Asterisk-Users] Reply-To?   John Novack
• [Asterisk-Users] Test settings   Ronald Wiplinger
• [Asterisk-Users] Unable to open master device '/dev/zap/ctl'   Juan Luis Moyano
• [Asterisk-Users] Any opinions on quality/service of Teliax?   Rich Adamson
• [Asterisk-Users] Asterisk based Call Accounting software - 1st release   Giles Coochey
• [Asterisk-Users] PRI card and TDM400P in same box   Damon Estep
• [Asterisk-Users] Several INVITE messages sent by Asterisk   Marlène Beray
• [Asterisk-Users] Difference Between NAT=yes and QUALIFY=yes and STUN...   Eric Wieling
• [Asterisk-Users] PRI card and TDM400P in same box   Colin Anderson
• [Asterisk-Users] Channel bank replacement   Peter Hoppe
• [Asterisk-Users] Getting a good deal on a PRI   Scott Laird
• [Asterisk-Users] Channel bank replacement   ht at phonitel.com
• [Asterisk-Users] Channel bank replacement   Gavin Hamill
• [Asterisk-Users] Asterisk based Call Accounting software - 1strelease   Chris Mason (Lists)
• [Asterisk-Users] SNOM 190: Unknown SIP command 'PUBLISH'   Ronald Wiplinger
• [Asterisk-Users] Asterisk based Call Accounting software - 1strelease   Ronald Wiplinger
• [Asterisk-Users] Re: Reply-To?   Bruno Wolff III
• [Asterisk-Users] Asterisk based Call Accounting software - 1strelease   ht at phonitel.com
• [Asterisk-Users] Getting a good deal on a PRI   Tom Hayden
• [Asterisk-Users] PCI-PRI cards - what to buy????   Maik Hassel
• [Asterisk-Users] T.38 fax with SIP devices   Steve Underwood
• [Asterisk-Users] Asterisk 1.0.5-BRIstuffed-0.2.0-RC7e   Jalal
• [Asterisk-Users] Echo on analog line related to FXO card?   Maik Hassel
• [Asterisk-Users] Asterisk Google Group?   Damon Estep
• [Asterisk-Users] Reply-To?   Mark Charlton
• [Asterisk-Users] Re: Reply-To?   Bruno Wolff III
• [Asterisk-Users] Any opinions on quality/service of Teliax?   Brian McSpadden
• [Asterisk-Users] Asterisk based Call Accounting software- 1strelease   Chris Mason (Lists)
• [Asterisk-Users] iaxtel.com - low bandwidth codecs   Drew Einhorn
• [Asterisk-Users] asterisk missing dtmf what would cause that   Jerry Geis
• [Asterisk-Users] Long wait for ring   Drew Einhorn
• [Asterisk-Users] Several INVITE messages sent by Asterisk   Julian J. M.
• [Asterisk-Users] Asterisk based Call Accounting software- 1strelease   ht at phonitel.com
• [Asterisk-Users] Asterisk based CallAccounting software- 1strelease   Parker, Blake (MIS)
• [Asterisk-Users] Cannot access voicemail   Jeff Heath
• [Asterisk-Users] Re: PRI card and TDM400P in same box   Noah Miller
• [Asterisk-Users] inquire about connected channel (show channels)   Nicolás Gudiño
• [Asterisk-Users] Asterisk based Call Accounting software- 1strelease   Ronald Wiplinger
• [Asterisk-Users] x100p disconnect on "D" tone   Henry Devito
• [Asterisk-Users] Asterisk Google Group?   Bruno Hertz
• [Asterisk-Users] Reply-To?   Bruno Hertz
• [Asterisk-Users] Cannot access voicemail   Jeff Heath
• [Asterisk-Users] Channel bank replacement   Peter Hoppe
• [Asterisk-Users] Asterisk - SER - sharing single mySQL db and tables   Gatewood, Andy
• [Asterisk-Users] Asterisk Google Group?   Damon Estep
• [Asterisk-Users] Connecting asterisk to existing PBX - newbie question   GP
• [Asterisk-Users] Cannot access voicemail   Wiley Siler
• [Asterisk-Users] Asterisk based Call Accounting software - 1strelease   Gonzalo Servat
• [Asterisk-Users] Asterisk based Call Accounting software - 1strelease   trixter http://www.0xdecafbad.com
• [Asterisk-Users] Asterisk Max TNT   James Taylor
• [Asterisk-Users] Asterisk Google Group?   Bruno Hertz
• [Asterisk-Users] Channel bank replacement   Matt Schulte
• [Asterisk-Users] Unable to open master device '/dev/zap/ctl'   Ben Hencke
• [Asterisk-Users] Asterisk Google Group?   trixter http://www.0xdecafbad.com
• [Asterisk-Users] Long wait for ring   Wilson Pickett
• [Asterisk-Users] call forwardin and parking   Thore
• [Asterisk-Users] Asterisk based CallAccounting software- 1strelease   trixter http://www.0xdecafbad.com
• [Asterisk-Users] unlimited iax termination   iaxsales at jsci.net
• [Asterisk-Users] Asterisk Google Group?   Damon Estep
• [Asterisk-Users] Asterisk Google Group?   Damon Estep
• [Asterisk-Users] Unable to open master device '/dev/zap/ctl'   Juan Luis Moyano
• [Asterisk-Users] SIP peer doesn't report busy properly   Remco Barende
• [Asterisk-Users] PCI-PRI cards - what to buy????   Scott Laird
• [Asterisk-Users] PRI Advice...   Michael Crozier
• [Asterisk-Users] unlimited iax termination   Kerry Garrison
• [Asterisk-Users] Asterisk Google Group?   trixter http://www.0xdecafbad.com
• [Asterisk-Users] Dell suggestions for Quad T1 system   Ben Hencke
• [Asterisk-Users] Cannot access voicemail -- RESOLVED   Jeff Heath
• [Asterisk-Users] unlimited iax termination   dean collins
• [Asterisk-Users] Reply-To?   Josiah Bryan
• [Asterisk-Users] snom and "hint" priority   Karl Brose
• [Asterisk-Users] Any opinions on quality/service of Teliax?   Mohit Muthanna
• [Asterisk-Users] Unable to open master device '/dev/zap/ctl'   Juan Luis Moyano
• [Asterisk-Users] Call from publicIP to PrivateIP   Andy Hamilton
• [Asterisk-Users] Allison sounds in native format   Jon Bebeau
• [Asterisk-Users] Connecting Asterisk to a SIP Gateway   Karras, Darin
• [Asterisk-Users] Realtime & voicemail   Terry Wilson
• [Asterisk-Users] x100p disconnect on "D" tone   Brian Leyton
• [Asterisk-Users] Asterisk Google Group?   Bruno Hertz
• [Asterisk-Users] RE:: Connecting asterisk to existing PBX - newbie   Jason Kawakami
• [Asterisk-Users] SNOM 190: Unknown SIP command 'PUBLISH'   NVC List Manager
• [Asterisk-Users] codec translation hints   snacktime
• [Asterisk-Users] Re: Reply-To?   Bruno Wolff III
• [Asterisk-Users] Reply-To?   Bruno Hertz
• [Asterisk-Users] unlimited iax termination   NVC List Manager
• [Asterisk-Users] Re: codec translation hints   snacktime
• [Asterisk-Users] RE: X100P doesn't check for dialtone   Justin Newman
• [Asterisk-Users] Major issues with VoicePulse today   Matthew Marlowe
• [Asterisk-Users] RE:: Connecting asterisk to existing PBX - newbie   dean collins
• [Asterisk-Users] T.38 fax with SIP devices   Michael D Schelin
• [Asterisk-Users] Re: codec translation hints   snacktime
• [Asterisk-Users] unlimited iax termination   iaxsales at jsci.net
• [Asterisk-Users] Asterisk Google Group?   tim panton
• [Asterisk-Users] unlimited iax termination   Mike Matthews
• [Asterisk-Users] PRI Advice...   Matt Loretitsch
• [Asterisk-Users] Asterisk Google Group?   Bruno Hertz
• [Asterisk-Users] RE:: Connecting asterisk to existing PBX - newbie   Will McCown
• [Asterisk-Users] Re: Delayed dial under Asterisk ?   Robert Rozman
• [Asterisk-Users] Warning, flexible rate not heavily tested!   Ronald Wiplinger
• [Asterisk-Users] Re: codec translation hints   Rich Adamson
• [Asterisk-Users] Re: unlimited iax termination   John Goerzen
• [Asterisk-Users] Re: unlimited iax termination   John Goerzen
• [Asterisk-Users] Connecting Asterisk to a SIP Gateway   Ed Greenberg
• [Asterisk-Users] Re: unlimited iax termination   Kerry Garrison
• [Asterisk-Users] Re: unlimited iax termination   trixter http://www.0xdecafbad.com
• [Asterisk-Users] Asterisk Google Group?   Damon Estep
• [Asterisk-Users] Convertnig from Norstar to * to save money   Mike Robinson
• [Asterisk-Users] Asterisk Google Group?   Damon Estep
• [Asterisk-Users] Dell suggestions for Quad T1 system   Andres
• [Asterisk-Users] iaxtel.com - low bandwidth codecs   Drew Einhorn
• [Asterisk-Users] Web interface for realtime Mysql friends/peer   Matt G
• [Asterisk-Users] unlimited iax termination   Andres
• [Asterisk-Users] Sangoma VS. Digium   Max W Blackmer Jr
• [Asterisk-Users] iaxtel.com - low bandwidth codecs   Rich Adamson
• [Asterisk-Users] Marco Extensions   Chris
• [Asterisk-Users] Using manager interface to play aanouncmentsin a MeetMe   Dan Austin
• [Asterisk-Users] Asterisk Google Group?   Bruno Hertz
• [Asterisk-Users] Long wait for ring   Jorge Merlino
• [Asterisk-Users] UK ISDN with Asterisk   Henry Owens
• [Asterisk-Users] Long wait for ring   Drew Einhorn
• [Asterisk-Users] iaxtel.com - low bandwidth codecs   Rich Adamson
• [Asterisk-Users] asterisk missing dtmf what would cause that   Phill Wolf
• [Asterisk-Users] Re: Lag in meetme   BJ Weschke
• [Asterisk-Users] UK ISDN with Asterisk   Gavin Hamill
• [Asterisk-Users] Re: Lag in meetme   Kris Edwards
• [Asterisk-Users] debugging voice quality issues   snacktime
• [Asterisk-Users] unlimited iax termination   Chris Mason
• [Asterisk-Users] Asterisk and RT (Request Tracker) setup?   Mike Dent
• [Asterisk-Users] Asterisk Memory Requirements   Eric Rees
• [Asterisk-Users] Convertnig from Norstar to * to save money   Jeff Heath
• [Asterisk-Users] Convertnig from Norstar to * to save money   William Boehlke
• [Asterisk-Users] Running a Marco from the dial command   Chris
• [Asterisk-Users] Asterisk and RT (Request Tracker) setup?   Alexander Lopez
• [Asterisk-Users] How many FXS/FXO ports do I need?   Aaron O'Hara
• [Asterisk-Users] Asterisk Memory Requirements   Luki
• [Asterisk-Users] How many FXS/FXO ports do I need?   Kerry Garrison
• [Asterisk-Users] How many FXS/FXO ports do I need?   Andrew Kohlsmith
• [Asterisk-Users] Call from publicIP to PrivateIP   Eric Wieling
• [Asterisk-Users] codec translation hints   Eric Wieling
• [Asterisk-Users] Warning, flexible rate not heavily tested!   Eric Wieling
• [Asterisk-Users] "s" extension doesn't work with ata   Drew Einhorn
• [Asterisk-Users] Phone card implementation issues in IAX   Xu Wang
• [Asterisk-Users] Multiple BroadVoice Accounts Problem withIncoming calls {Scanned}   Kerry Garrison
• [Asterisk-Users] Frame slip on Wildcard TDM400P?   Trent Tuggle
• [Asterisk-Users] "s" extension doesn't work with ata   Carlos Rojas
• [Asterisk-Users] "s" extension doesn't work with ata   Andrew Kohlsmith
• [Asterisk-Users] Using manager interface to play aanouncmentsin aMeetMe   Dan Austin
• [Asterisk-Users] Asterisk and RT (Request Tracker) setup?   Kris Boutilier
• [Asterisk-Users] Asterisk Google Group?   Roman Volf
• [Asterisk-Users] Asterisk Google Group?   Roman Volf
• [Asterisk-Users] Convertnig from Norstar to * to save money   Kris Boutilier
• [Asterisk-Users] Asterisk Google Group?   Bruno Hertz
• [Asterisk-Users] Call from publicIP to PrivateIP   Kamran Ahmad
• [Asterisk-Users] Re: Ring Twice   VoIP Newbie
• [Asterisk-Users] SIP Softphone for testing with Asterisk   Christian Stredicke
• [Asterisk-Users] Re: Lag in meetme [SOLVED]   Kris Edwards
• [Asterisk-Users] Re: Lag in meetme (by the way)   Kris Edwards
• [Asterisk-Users] Database lookups?   Jan Johansson
• [Asterisk-Users] oh323 on @homeasterisk   CM Rahman Jr.
• [Asterisk-Users] zap to sip caller id "forwarding"   Tom Makulski
• [Asterisk-Users] Asterisk Memory Requirements   Cameron Schaus
• [Asterisk-Users] g726 > gsm not working with sipura   snacktime
• [Asterisk-Users] iax / realtime problems   Paul P. Pongco
• [Asterisk-Users] g726 > gsm not working with sipura   Brian Capouch
• [Asterisk-Users] g726 > gsm not working with sipura   snacktime
• [Asterisk-Users] Astcc Patch   chawki hammoud
• [Asterisk-Users] sip phone extensions at a remote site   cmould
• [Asterisk-Users] UK ISDN with Asterisk   Duncan Rogerson
• [Asterisk-Users] Dell suggestions for Quad T1 system   Kevin Brennan
• [Asterisk-Users] fax pass through on te410p   Kevin Brennan
• [Asterisk-Users] Re: Lag in meetme   Tony Mountifield
• [Asterisk-Users] Asterisk and RT (Request Tracker) setup?   Mike Dent
• [Asterisk-Users] UK ISDN with Asterisk   Henry Owens
• [Asterisk-Users] Sangoma VS. Digium   Craig Guy
• [Asterisk-Users] UK ISDN with Asterisk   Craig Guy
• [Asterisk-Users] Hardware dimesioning issues   David John Walsh
• [Asterisk-Users] UK ISDN with Asterisk   Henry Owens
• [Asterisk-Users] fax pass through on te410p   Steve Underwood
• [Asterisk-Users] HOW TO SET THE TIME TO DIAL AFTER astcc-accountnum and astcc-phonenum   wassim darwish
• [Asterisk-Users] UK ISDN with Asterisk   John Daragon
• [Asterisk-Users] UK ISDN with Asterisk   Henry Owens
• [Asterisk-Users] unlimited iax termination   Rich Adamson
• [Asterisk-Users] fax pass through on te410p   Kevin Brennan
• [Asterisk-Users] SIP peer doesn't report busy properly   Florian Overkamp
• [Asterisk-Users] Call rejected by XXX: No authority found   Alexander Fitterling
• [Asterisk-Users] How many FXS/FXO ports do I need?   Rich Adamson
• [Asterisk-Users] dyndns alias clients: needs to register in iax.conf as well?   Alexander Fitterling
• [Asterisk-Users] Shorewall settings?   Alexander Fitterling
• [Asterisk-Users] oh323 on @homeasterisk   Mike Sander
• [Asterisk-Users] Asterisk Memory Requirements   Eric Rees
• [Asterisk-Users] UK ISDN with Asterisk   John Daragon
• [Asterisk-Users] Asterisk Memory Requirements   Tony Hoyle
• [Asterisk-Users] Terrible crackling on analogue line and X100P card   Stuart Ford
• [Asterisk-Users] unlimited iax termination   Ronald Wiplinger
• [Asterisk-Users] fax pass through on te410p   Henry Devito
• [Asterisk-Users] unlimited iax termination   Chris Mason (Lists)
• [Asterisk-Users] unlimited iax termination   Kerry Garrison
• [Asterisk-Users] Terrible crackling on analogue line and X100P card   Bill Ford
• [Asterisk-Users] How to change language using manager interface?   Thorben Jensen
• [Asterisk-Users] Dialing With Backgound Music   Ugur GUNCER
• [Asterisk-Users] Terrible crackling on analogue line and X100P card   dean collins
• [Asterisk-Users] Dialing With Backgound Music   Giovanni Miano
• [Asterisk-Users] Terrible crackling on analogue line and X100Pcard   Stuart Ford
• [Asterisk-Users] Terrible crackling on analogue line and X100P card   Damian Funnell
• [Asterisk-Users] How to change language using manager interface?   Guy Decarpentrie
• [Asterisk-Users] Dialing With Backgound Music   Eric Wieling
• [Asterisk-Users] "s" extension doesn't work with ata   Eric Wieling
• [Asterisk-Users] Running a Marco from the dial command   Wilson Pickett
• [Asterisk-Users] Call rejected by XXX: No authority found   Wilson Pickett
• [Asterisk-Users] unlimited iax termination   Ronald Wiplinger
• [Asterisk-Users] Terrible crackling on analogue line and X100P card   Damian Funnell
• [Asterisk-Users] Running a Marco from the dial command   Chris
• [Asterisk-Users] Asterisk as protocol conventer beetwen SIP and H.323   Adam Rybak
• [Asterisk-Users] how to pass G723.1   Chetan Sarva
• [Asterisk-Users] Asterisk as protocol conventer beetwen SIP and H.323   Sahil Gupta
• [Asterisk-Users] SPA and NAT traversal   Jim Sturtevant
• [Asterisk-Users] Asterisk Dual Servers   Juan Luis Moyano
• [Asterisk-Users] DS3000P - 20 E1 capacity on single card   izo
• [Asterisk-Users] unlimited iax termination   Jeff Glassman
• [Asterisk-Users] DS3000P - 20 E1 capacity on single card   Eric Wieling
• [Asterisk-Users] Hardware dimesioning issues   David John Walsh
• [Asterisk-Users] SPA and NAT traversal   Eric Wieling
• [Asterisk-Users] Asterisk as protocol conventer beetwen SIP and H.323   Adam Rybak
• [Asterisk-Users] Terrible crackling on analogue line and X100P card   Sascha Ferley
• [Asterisk-Users] FWD no longer doing IAX?   Scott Wolfe
• [Asterisk-Users] CallerID name lookup AGI script   Jim Meehan
• [Asterisk-Users] AgentLogin to MeetMe conference?   Steve Edwards
• [Asterisk-Users] How to change language using manager interface?   Thorben Jensen
• [Asterisk-Users] FWD no longer doing IAX?   Carlos Chavez
• [Asterisk-Users] DS3000P - 20 E1 capacity on single card   Remco Barende
• [Asterisk-Users] SPA and NAT traversal   Jim Sturtevant
• [Asterisk-Users] FWD no longer doing IAX?   r00t
• [Asterisk-Users] SPA and NAT traversal   Eric Wieling
• [Asterisk-Users] SPA and NAT traversal   Nabeel Jafferali
• [Asterisk-Users] OT: ManxPower 2005 European Tour   Eric Wieling
• [Asterisk-Users] Syntax error near unexpected token 'fi'   Chuck Bunn
• [Asterisk-Users] Syntax error near unexpected token 'fi'   Luki
• [Asterisk-Users] SPA and NAT traversal   Jim Sturtevant
• [Asterisk-Users] SPA and NAT traversal   Nabeel Jafferali
• [Asterisk-Users] "s" extension doesn't work with ata   Scott Nelson
• [Asterisk-Users] Grandstream HandyTone-488, * -> FXO problems   Dan Perik
• [Asterisk-Users] Terrible crackling on analogue line and X100P card   Paul
• [Asterisk-Users] Netlogic inbound DID issue   James Taylor
• [Asterisk-Users] Any opinions on quality/service of Teliax?   Bellows, Jared
• [Asterisk-Users] Any opinions on quality/service of Teliax?   Brian Dingman
• [Asterisk-Users] unlimited iax termination   Paul
• [Asterisk-Users] Using manager interface to play aanouncments in aMeetMe   mattf
• [Asterisk-Users] unlimited iax termination   Rich Adamson
• [Asterisk-Users] unlimited iax termination   Rich Adamson
• [Asterisk-Users] DS3000P - 20 E1 capacity on single card   Andrew Kohlsmith
• [Asterisk-Users] Terrible crackling on analogue line and X100P card   Rich Adamson
• [Asterisk-Users] Any opinions on quality/service of Teliax?   Brian McSpadden
• [Asterisk-Users] Any opinions on quality/service of Teliax?   Eric Wieling
• [Asterisk-Users] Terrible crackling on analogue line and X100Pcard   Stuart Ford
• [Asterisk-Users] DS3000P - 20 E1 capacity on single card   Eric Wieling
• [Asterisk-Users] Configuring the Sipura for static IP and registering with Asterisk.   Jerry
• [Asterisk-Users] Can I set queue not to hangup?   Steve Edwards
• [Asterisk-Users] DS3000P - 20 E1 capacity on single card   Andrew Kohlsmith
• [Asterisk-Users] Channel bank replacement   Jerry
• [Asterisk-Users] DS3000P - 20 E1 capacity on single card   Eric Wieling
• [Asterisk-Users] Stanaphone - eureka   dean collins
• [Asterisk-Users] Terrible crackling on analogue line and X100Pcard   dean collins
• [Asterisk-Users] Confusion re; 407 Proxy Authentication Required   Kevin Mayer
• [Asterisk-Users] Terrible crackling on analogue line and X100Pcard   Rich Adamson
• [Asterisk-Users] Multiple Servers and One Central Voicemail   Anton Krall
• [Asterisk-Users] Terrible crackling on analogue line and X100Pcard   Rich Adamson
• [Asterisk-Users] Terrible crackling on analogue line and X100Pcard   dean collins
• [Asterisk-Users] Multiple Servers and One Central Voicemail   Anton Jackson-Smith
• [Asterisk-Users] Configuring the Sipura for static IP and registering with Asterisk.   Michael D Schelin
• [Asterisk-Users] SPA and NAT traversal   Michael D Schelin
• [Asterisk-Users] Multiple Servers and One Central Voicemail   Luki
• [Asterisk-Users] Multiple Servers and One Central Voicemail   Anton Krall
• [Asterisk-Users] Multiple Servers and One Central Voicemail   Anton Jackson-Smith
• [Asterisk-Users] Multiple Servers and One Central Voicemail   Anton Krall
• [Asterisk-Users] Using zap channels for fax   Chris Mason (Lists)
• [Asterisk-Users] question about oh323   Joe S
• [Asterisk-Users] How to upgrade safe?   Ronald Wiplinger
• [Asterisk-Users] Any free SIP softphone with IM capacity for Windows?   Thai Duong
• [Asterisk-Users] Can you comment on this Qos script? How does one shape RTP?   cmisip
• [Asterisk-Users] Fax, which one do I need?   Ronald Wiplinger
• [Asterisk-Users] Cann't get CallerID on Zap channel, Please Help!!   Dominic Lu
• [Asterisk-Users] Cann't get CallerID on Zap channel, Please Help!!   Remco Barende
• [Asterisk-Users] Terrible crackling on analogue line and X100Pcard   Stuart Ford
• [Asterisk-Users] IPSwitchBoard Version 0.77 Released   Thorben Jensen
• [Asterisk-Users] ast-rad-acc.pl problem   Ma Zhiyong
• [Asterisk-Users] CVS compile issue on res_odbc.o   David Shirley
• [Asterisk-Users] Snom only one way audio   Ronald Wiplinger
• [Asterisk-Users] Asterisk becomes after one month unstabled   Ronald Wiplinger
• [Asterisk-Users] Asterisk::LCR - Least Cost Routing for Asterisk   Jean-Michel Hiver
• [Asterisk-Users] Can you comment on this Qos script? How does one shape RTP?   Doug Lytle
• [Asterisk-Users] ignorepat changing the sound of dialtone   Thomas Andrews
• [Asterisk-Users] Re: Asterisk Dual Servers   Noah Miller
• [Asterisk-Users] Terrible crackling on analogue line and X100Pcard   Rich Adamson
• [Asterisk-Users] search the mailing list   Equipe du Royaume
• [Asterisk-Users] search the mailing list   David Shirley
• [Asterisk-Users] search the mailing list   Nabeel Jafferali
• [Asterisk-Users] search the mailing list   Time Bandit
• [Asterisk-Users] Terrible crackling on analogue line and X100Pcard   Rich Adamson
• [Asterisk-Users] Why do calls go silent after 10 minutes   Tony
• [Asterisk-Users] Asterisk::LCR - Least Cost Routing for Asterisk   Leif Madsen - Certified Asterisk Consultant
• [Asterisk-Users] decimal arithmetic operations in Asterisk   Rizwan Chaudhry
• [Asterisk-Users] Can you comment on this Qos script? How does one shape RTP?   cmisip
• [Asterisk-Users] How to upgrade safe?   Andrew Kohlsmith
• [Asterisk-Users] Re: Asterisk Dual Servers   Rich Adamson
• [Asterisk-Users] ignorepat changing the sound of dialtone   Andy Hamilton
• [Asterisk-Users] UK PSTN Calling From OH323 Problem   Iain Young
• [Asterisk-Users] How to upgrade safe?   Rich Adamson
• [Asterisk-Users] Can you comment on this Qos script? How does one shape RTP?   Andrew Kohlsmith
• [Asterisk-Users] How To conferance   Ugur GUNCER
• [Asterisk-Users] Asterisk::LCR - Least Cost Routing for Asterisk   Ronald Wiplinger
• [Asterisk-Users] Terrible crackling on analogue line and X100Pcard   John Novack
• [Asterisk-Users] DS3000P - 20 E1 capacity on single card   Matthew Boehm
• [Asterisk-Users] iax / realtime problems   Matthew Boehm
• [Asterisk-Users] Fax to Email   Bill Ford
• [Asterisk-Users] Can you comment on this Qos script? How does one shape RTP?   Doug Lytle
• [Asterisk-Users] CallerID name lookup AGI script   Brian Dingman
• [Asterisk-Users] S100I - competitive price?   Matthew Boehm
• [Asterisk-Users] Asterisk becomes after one month unstabled   Niksa Baldun
• [Asterisk-Users] SMS suddenly not sending out   Wilson Pickett
• [Asterisk-Users] Terrible crackling on analogue line and X100P card   Damian Funnell
• [Asterisk-Users] Can you comment on this Qos script? How does one shape RTP?   cmisip
• [Asterisk-Users] problem with unicall and asterisk   kiran
• [Asterisk-Users] Asterisk::LCR - Least Cost Routing for Asterisk   Jean-Michel Hiver
• [Asterisk-Users] Multiple Servers and 1 Central Voicemail   Jason Brown
• [Asterisk-Users] Asterisk::LCR - Least Cost Routing for Asterisk   Ronald Wiplinger
• [Asterisk-Users] Terrible crackling on analogue line and X100Pcard   Rich Adamson
• [Asterisk-Users] Yet another version of IPS Freeware   Thorben Jensen
• [Asterisk-Users] Sipura SPA-841 Phone Review   Kerry Garrison
• [Asterisk-Users] sipura 3000 - "Call Leg/Transaction Does Not Exist" - only happens sometimes   Chris Stenton
• [Asterisk-Users] Sipura SPA-841 Phone Review   Trevor Peirce
• [Asterisk-Users] Sipura SPA-841 Phone Review   JD
• [Asterisk-Users] SPA and NAT traversal   Jim Sturtevant
• [Asterisk-Users] Sipura SPA-841 Phone Review   Kerry Garrison
• [Asterisk-Users] binding virtual IP address   Xu Wang
• [Asterisk-Users] sipura 3000 - "Call Leg/Transaction Does Not Exist" - only happens sometimes   Mike Benoit
• [Asterisk-Users] Terrible crackling on analogue line and X100P card   Rich Adamson
• [Asterisk-Users] Re: Fax to Email   Justin Newman
• [Asterisk-Users] sipura 3000 - "Call Leg/Transaction Does Not Exist" - only happens sometimes   Rich Adamson
• [Asterisk-Users] Problems with meetme.   Xisco (Personal)
• [Asterisk-Users] Multiple Servers and 1 Central Voicemail   C F
• [Asterisk-Users] Can you comment on this Qos script? How does one shape RTP?   Julian J. M.
• [Asterisk-Users] Problems with meetme.   Adam Rybak
• [Asterisk-Users] polycom phones   Richard
• [Asterisk-Users] Fax detect/transfer problem?   Jim Meehan
• [Asterisk-Users] Sipura SPA-841 Phone Review   John Novack
• [Asterisk-Users] Snom only one way audio   Gareth J. Greenaway
• [Asterisk-Users] Asterisk @ Home AMP problem   Bernie Courtney
• [Asterisk-Users] International callback strategies   snacktime
• [Asterisk-Users] Asterisk @ Home AMP problem   Guillermo Salas M
• [Asterisk-Users] Need /etc/zaptel.conf for TE110P   Tim Connolly
• [Asterisk-Users] Sipura SPA-841 Phone Review   Craig
• [Asterisk-Users] Re: International callback strategies   snacktime
• [Asterisk-Users] Asterisk @ Home AMP problem   Kerry Garrison
• [Asterisk-Users] Problems trying to compile H323 from CVS-STABLE   Tony Mountifield
• [Asterisk-Users] snom360 & hint priority   Henry Devito
• [Asterisk-Users] Snom only one way audio   Karl Brose
• [Asterisk-Users] Re: International callback strategies   snacktime
• [Asterisk-Users] Re: International callback strategies   Hakem Taourchi
• [Asterisk-Users] Re: International callback strategies   snacktime
• [Asterisk-Users] Re: Fax to Email   Bill Ford
• [Asterisk-Users] Re: International callback strategies   Rich Adamson
• [Asterisk-Users] append # to dial string   John Breeden
• [Asterisk-Users] unexpected crash ......   Laurent Foulonneau
• [Asterisk-Users] Sipura SPA-841 Phone Review   Andrew Kohlsmith
• [Asterisk-Users] DS3000P - 20 E1 capacity on single card   Andrew Kohlsmith
• [Asterisk-Users] PTSN POTS Differences   Robert Keller
• [Asterisk-Users] SIP outgoing problem   Paul
• [Asterisk-Users] no ring on inbound SIP calls   snacktime
• [Asterisk-Users] Re: no ring on inbound SIP calls   snacktime
• [Asterisk-Users] SIP outgoing problem   snacktime
• [Asterisk-Users] SIP outgoing problem   Tim Connolly
• [Asterisk-Users] snom360 & hint priority   Henry Devito
• [Asterisk-Users] snom360 & hint priority   Karl Brose
• [Asterisk-Users] Re: no ring on inbound SIP calls   snacktime
• [Asterisk-Users] SIP outgoing problem   Eric Wieling
• [Asterisk-Users] Re: no ring on inbound SIP calls   Eric Wieling
• [Asterisk-Users] Re: no ring on inbound SIP calls   snacktime
• [Asterisk-Users] VM answer call after 20 sec...   Jamie Fargen
• [Asterisk-Users] Sipura SPA-841 Phone Review   Trevor Peirce
• [Asterisk-Users] Re: no ring on inbound SIP calls   snacktime
• [Asterisk-Users] Re: no ring on inbound SIP calls   Rich Adamson
• [Asterisk-Users] VM answer call after 20 sec...   Jamie Fargen
• [Asterisk-Users] PTSN POTS Differences   Rich Adamson
• [Asterisk-Users] VM answer call after 20 sec...   Nabeel Jafferali
• [Asterisk-Users] SIP outgoing problem   Tony Hoyle
• [Asterisk-Users] Re: no ring on inbound SIP calls   Rich Adamson
• [Asterisk-Users] unexpected crash ......   Luki
• [Asterisk-Users] append # to dial string   Rich Adamson
• [Asterisk-Users] Re: Fax to Email   Rich Adamson
• [Asterisk-Users] restructuring my dialplan   Rich Adamson
• [Asterisk-Users] Re: no ring on inbound SIP calls   snacktime
• [Asterisk-Users] Fax detect/transfer problem?   Jim Meehan
• [Asterisk-Users] Re: no ring on inbound SIP calls   Rich Adamson
• [Asterisk-Users] PTSN POTS Differences SOLVED   Robert Keller
• [Asterisk-Users] Broadvoice problem: Bad request!   Ronald Wiplinger
• [Asterisk-Users] VM answer call after 20 sec...   Rich Adamson
• [Asterisk-Users] problem with unicall   kiran
• [Asterisk-Users] Re: no ring on inbound SIP calls   Rich Adamson
• [Asterisk-Users] problem with astrisk on MFC R2   kiran
• [Asterisk-Users] SIP outgoing problem   Eric Wieling
• [Asterisk-Users] Linux & Asterisk   Shaun Dwyer
• [Asterisk-Users] Callback application   snacktime
• [Asterisk-Users] SIP outgoing problem   Eric Wieling
• [Asterisk-Users] Cannot open chan_zap:   Tim Connolly
• [Asterisk-Users] Re: International callback strategies   Adam Goryachev
• [Asterisk-Users] append # to dial string   John Breeden
• [Asterisk-Users] From OH323 to SIP or OH323 without gatekeeper   Joe S
• [Asterisk-Users] SIP outbound call audio quality change   Paul
• [Asterisk-Users] VAD/DTX implementation through zaptel cards   parijat at varaha.com
• [Asterisk-Users] Cannot open chan_zap:   Torsten Krueger
• [Asterisk-Users] 4 x ISDN2 hardware...?   Marc
• [Asterisk-Users] 4 x ISDN2 hardware...?   Rod Bacon
• [Asterisk-Users] Callback application   Rod Bacon
• [Asterisk-Users] Callback application   Adam Goryachev
• [Asterisk-Users] 4 x ISDN2 hardware...?   Marc
• [Asterisk-Users] Setgroup & Checkgroup   Ronald Wiplinger
• [Asterisk-Users] TDM400P power supply   Ricardo Peironcely
• [Asterisk-Users] 4 x ISDN2 hardware...?   Peer Oliver Schmidt
• [Asterisk-Users] ignorepat changing the sound of dialtone   Thomas Andrews
• [Asterisk-Users] UK ISDN with Asterisk   Tim Robinson
• [Asterisk-Users] Conferance DialPlan   Ugur GUNCER
• [Asterisk-Users] 4 x ISDN2 hardware...?   Gavin Hamill
• [Asterisk-Users] Re: PTSN POTS Differences SOLVED   Tony Mountifield
• [Asterisk-Users] IAX calls between asterisk boxes works 1 way only   James Bean
• [Asterisk-Users] iax / realtime problems   Paul P. Pongco
• [Asterisk-Users] Snom 'virtual' extension monitoring?   Remco Barende
• [Asterisk-Users] voicetronix dtmf   Altus Snyman
• [Asterisk-Users] "404 User Not Found" when calling between two X-Lites   Abraham WEI
• [Asterisk-Users] Supply ringing noise to IAX callers   James Bean
• [Asterisk-Users] Can I exit from asterisk console without stopping asterisk?   Abraham WEI
• [Asterisk-Users] IAX calls between asterisk boxes works 1 way only   El Flynn
• [Asterisk-Users] Re: International callback strategies   ht at phonitel.com
• [Asterisk-Users] UK CallerID patch with 1.0.7 / 1-0 CVS   Gavin Hamill
• [Asterisk-Users] SIP UA behind NAT and REINVITE ???   Cameron Beattie
• [Asterisk-Users] Aculab   Jochen Witte
• [Asterisk-Users] CDR and TDS   David Masure
• [Asterisk-Users] call forwarding and parking   Thore
• [Asterisk-Users] secret/username - what does it really do?   Cameron Beattie
• [Asterisk-Users] Aculab   Isamar Maia
• [Asterisk-Users] How to turn off automatic pick up for Incomingcalls A@H v0.6   Cameron Beattie
• [Asterisk-Users] stand alone Voice Mail   Cameron Beattie
• [Asterisk-Users] UK CallerID patch with 1.0.7 / 1-0 CVS   Gavin Hamill
• [Asterisk-Users] Can I exit from asterisk console without stopping asterisk?   Bill Ford
• [Asterisk-Users] Direct Broadband connection of ip phone to LiveVoip?   C W Nel
• [Asterisk-Users] Username containing an "@"   Christoph Beckmeyer
• [Asterisk-Users] Supply ringing noise to IAX callers   James Bean
• [Asterisk-Users] IAX calls between asterisk boxes works 1 way only   James Bean
• [Asterisk-Users] How to turn off automatic pick up for Incoming calls A@H v0.6   Time Bandit
• [Asterisk-Users] Re: CDR and TDS   Tony Mountifield
• [Asterisk-Users] Re: Fax to Email   Bartosz Jozwiak
• [Asterisk-Users] PTSN POTS Differences SOLVED   Rich Adamson
• [Asterisk-Users] Re: no ring on inbound SIP calls   Peter Svensson
• [Asterisk-Users] Terrible crackling on analogue line and X100Pcard   Stuart Ford
• [Asterisk-Users] TDM02B on 2 a/b ports of a PBX not working... help   Dimitris Kouimintzis
• [Asterisk-Users] TDM400P power supply   Rich Adamson
• [Asterisk-Users] SIP Attended/Supervised transfer & features.conf   Gonchi Mateos
• [Asterisk-Users] Multiple Servers and 1 Central Voicemail   Jason Brown
• [Asterisk-Users] "404 User Not Found" when calling between two X-Lites   Rich Adamson
• [Asterisk-Users] IPS version 0.79 released   Thorben Jensen
• [Asterisk-Users] Can I exit from asterisk console without stoppingasterisk?   Henry Devito
• [Asterisk-Users] Zaptel Compile on a virtual dedicated host.   vgrskovic at optonline.net
• [Asterisk-Users] Shared call appearances   Craig
• [Asterisk-Users] Zaptel Compile on a virtual dedicated host.   Henry
• [Asterisk-Users] Zaptel Compile on a virtual dedicated host.   Gonzalo Servat
• [Asterisk-Users] append # to dial string   Eric Wieling
• [Asterisk-Users] VAD/DTX implementation through zaptel cards   Eric Wieling
• [Asterisk-Users] CDR and TDS   Eric Wieling
• [Asterisk-Users] Zaptel Compile on a virtual dedicated host.   vgrskovic at optonline.net
• [Asterisk-Users] Sangoma A101 + Rhino channelbank   Felician CHELU
• [Asterisk-Users] TE110P/Hipath3750 - Yellow Alarm   Henry Jensen
• [Asterisk-Users] Re: PTSN POTS Differences SOLVED   Robert Keller
• [Asterisk-Users] Can you comment on this Qos script? How does one shape RTP?   Sean Kennedy
• [Asterisk-Users] Interface bonding + asterisk   Jesus Mogollon
• [Asterisk-Users] Can you comment on this Qos script? How doesone shape RTP?   Henry
• [Asterisk-Users] Can you comment on this Qos script? How does one shape RTP?   trixter http://www.0xdecafbad.com
• [Asterisk-Users] Can you comment on this Qos script? How does one shape RTP?   Andrew Kohlsmith
• [Asterisk-Users] Sangoma A101 + Rhino channelbank   mattf
• [Asterisk-Users] TDM400P power supply   Ricardo Peironcely
• [Asterisk-Users] wcfxo problem   Dave Weis
• [Asterisk-Users] Problem with X101P   Yusuf Iqbal
• [Asterisk-Users] Can you comment on this Qos script? How does one shape RTP?   trixter http://www.0xdecafbad.com
• [Asterisk-Users] (no subject)   Robert Webb
• [Asterisk-Users] Sipura SPA-841 Phone Review   Doug Millsaps
• [Asterisk-Users] From OH323 to SIP or OH323 without gatekeeper   Bruno Hertz
• [Asterisk-Users] wcfxo problem   Sahil Gupta
• [Asterisk-Users] TDM400P Revision question.   Robert Webb
• [Asterisk-Users] Intercom with Aastra 480e?   Bobby Lacey
• [Asterisk-Users] Manipulate Asterisk Database from manager?   Matt
• [Asterisk-Users] Why 's' doesn't take over unknown extension in context ?   Robert Rozman
• [Asterisk-Users] timed Loop   Chris
• [Asterisk-Users] Getting CVS HEAD   Guillermo Salas M
• [Asterisk-Users] Is it possible to PickupChan / Dial Pickup / Steal a call that has not been answered?   Lane
• [Asterisk-Users] Why 's' doesn't take over unknown extension incontext ?   Steve Mann
• [Asterisk-Users] Getting CVS HEAD   Jochen Witte
• [Asterisk-Users] Simulated dialtone like in other PBX   List Receiver
• [Asterisk-Users] SIP Attended/Supervised transfer & features.conf   Michiel van Baak
• [Asterisk-Users] Problem with X101P   Scott Stingel
• [Asterisk-Users] (no subject)   Rich Adamson
• [Asterisk-Users] Re: PTSN POTS Differences SOLVED   Tony Mountifield
• [Asterisk-Users] Linux & Asterisk   Race Vanderdecken
• [Asterisk-Users] Getting CVS HEAD   Andy Hamilton
• [Asterisk-Users] timed Loop   Race Vanderdecken
• [Asterisk-Users] Rebooting Asterisk box shows Asterisk failing to shutdown   Chuck Bunn
• [Asterisk-Users] Need to Reduce Latency   Art Zemon
• [Asterisk-Users] Asterisk - SS7 or ISDN   Markku Korpi
• [Asterisk-Users] Low cost box for hosting Asterisk and at least one TDM400p   Chuck Bunn
• [Asterisk-Users] iax / realtime problems   Matthew Boehm
• [Asterisk-Users] TDM400p reliability????   Chuck Bunn
• [Asterisk-Users] Is it possible to PickupChan / Dial Pickup / Steal a call that has not been answered?   Rich Adamson
• [Asterisk-Users] Play Sound File Without Answer Channel ??   Angel Diaz
• [Asterisk-Users] Outgoing faxes with chan_capi?   Rémi Letot
• [Asterisk-Users] append # to dial string   John Breeden
• [Asterisk-Users] Getting CVS HEAD   Rich Adamson
• [Asterisk-Users] Roadmap for Asterisk??   Chuck Bunn
• [Asterisk-Users] Setgroup & Checkgroup   Damian Funnell
• [Asterisk-Users] Is it possible to PickupChan / Dial Pickup / Steal a call that has not been answered?   Lane
• [Asterisk-Users] DID via SIP/IAX   Max Clark
• [Asterisk-Users] Re: PTSN POTS Differences RESOLVED...no really this time...   Robert Keller
• [Asterisk-Users] IPswitch Monitor Extension   Ronald Wiplinger
• [Asterisk-Users] timed Loop   Dylan VanHerpen
• [Asterisk-Users] timed Loop   Chris
• [Asterisk-Users] Setgroup & Checkgroup   Ronald Wiplinger
• [Asterisk-Users] timed Loop   Chris
• [Asterisk-Users] Line Noise HELP!   Paul
• [Asterisk-Users] IPswitch Monitor Extension   Thorben Jensen
• [Asterisk-Users] 499 Error on X-lite / asterisk setup   David John Walsh
• [Asterisk-Users] Setgroup & Checkgroup   Damian Funnell
• [Asterisk-Users] ADSI Input from 480 keypad?   Jimmy
• [Asterisk-Users] From OH323 to SIP or OH323 without gatekeeper   Joe S
• [Asterisk-Users] IPswitch Monitor Extension   Ronald Wiplinger
• [Asterisk-Users] polycom phones   Michael Crozier
• [Asterisk-Users] Bounty: Request for PRI Debug   Trevor Peirce
• [Asterisk-Users] Line Noise HELP!   Damian Funnell
• [Asterisk-Users] Asterisk T.38 framing   Warrick FitzGerald
• [Asterisk-Users] Connection to SIP Gateway   Karras, Darin
• [Asterisk-Users] Undefined symbol in res_features & Others   Matt Roth
• [Asterisk-Users] Can I exit from asterisk console without stopping asterisk?   Leif Madsen - Certified Asterisk Consultant
• [Asterisk-Users] Using manager interface to play aanouncmentsin aMeetMe   Dan Austin
• [Asterisk-Users] PortaSIP/PortaBilling incompatibility (provider: sipcall.ch)   gramels
• [Asterisk-Users] SIP outbound call audio quality change   Tony Hoyle
• [Asterisk-Users] Low cost box for hosting Asterisk and at least one TDM400p   Wilson Pickett
• [Asterisk-Users] debugging broken/distorted sound with SIP   snacktime
• [Asterisk-Users] 499 Error on X-lite / asterisk setup   Robert Keller
• [Asterisk-Users] 499 Error on X-lite / asterisk setup   David John Walsh
• [Asterisk-Users] Manipulate Asterisk Database from manager?   Brian Roy
• [Asterisk-Users] Manipulate Asterisk Database from manager?   Matt
• [Asterisk-Users] Suggestions about where to start from   francesco.pellegrini at frameweb.it
• [Asterisk-Users] AgentLogin to MeetMe conference?   Steve Edwards
• [Asterisk-Users] 4 x ISDN2 hardware...?   Rémi Letot
• [Asterisk-Users] SIP outgoing problem   Tony Hoyle
• [Asterisk-Users] Re: polycom phones   Noah Miller
• [Asterisk-Users] Play Sound File Without Answer Channel ??   Mikael Magnusson
• [Asterisk-Users] Line Noise HELP!   Paul
• [Asterisk-Users] From OH323 to SIP or OH323 without gatekeeper   Bruno Hertz
• [Asterisk-Users] Maximum retries exceeded on call   Walter Willis
• [Asterisk-Users] Low cost box for hosting Asterisk and at leastone TDM400p   Wiley Siler
• [Asterisk-Users] Manipulate Asterisk Database from manager?   Dan Austin
• [Asterisk-Users] Manipulate Asterisk Database from manager?   Josiah Bryan
• [Asterisk-Users] Suggestions about where to start from   Wiley Siler
• [Asterisk-Users] Re: Using manager interface to play aanouncmentsin aMeetMe   Tony Mountifield
• [Asterisk-Users] DS3000P - 20 E1 capacity on single card   Kevin P. Fleming
• [Asterisk-Users] Re: Asterisk-Users Digest, Vol 9, Issue 96   Angel Diaz
• [Asterisk-Users] Play Sound File Without Answer Channel   Angel Diaz
• [Asterisk-Users] IPswitch Monitor Extension   Thorben Jensen
• [Asterisk-Users] DS3000P - 20 E1 capacity on single card   Kevin P. Fleming
• [Asterisk-Users] PRI Advice...   Michael Crozier
• [Asterisk-Users] TDM400p reliability????   Rich Adamson
• [Asterisk-Users] Play Sound File Without Answer Channel   Angel Diaz
• [Asterisk-Users] Line Noise HELP!   Damian Funnell
• [Asterisk-Users] From OH323 to SIP or OH323 without gatekeeper   Joe S
• [Asterisk-Users] Low cost box for hosting Asterisk and at least oneTDM400p   Rusty Shackleford
• [Asterisk-Users] Low cost box for hosting Asterisk and at leastone TDM400p   Kerry Garrison
• [Asterisk-Users] DS3000P - 20 E1 capacity on single card   Andres
• [Asterisk-Users] Re: Problems trying to compile H323 from CVS-STABLE   Tony Mountifield
• [Asterisk-Users] "Refresh" asterisk internal database?   Matt
• [Asterisk-Users] From OH323 to SIP or OH323 without gatekeeper   Bruno Hertz
• [Asterisk-Users] BROADVOICE - Incomming calls are dropped after 1-2 min   Bartosz Wegrzyn - asterisk
• [Asterisk-Users] Why 's' doesn't take over unknown extension incontext ?   Eric Wieling aka ManxPower
• [Asterisk-Users] Asterisk becomes after one month unstabled   David Brodbeck
• [Asterisk-Users] timed Loop   Eric Wieling aka ManxPower
• [Asterisk-Users] x100p disconnect on "D" tone   David Brodbeck
• [Asterisk-Users] Low cost box for hosting Asterisk and at leastone TDM400p   Chuck Bunn
• [Asterisk-Users] Low cost box for hosting Asterisk and at leastoneTDM400p   Wiley Siler
• [Asterisk-Users] RE: Ebay listing selling Asterisk @ Home and AMP for over 1000 dollars   Charles Osstyn
• [Asterisk-Users] 499 Error on X-lite / asterisk setup   David John Walsh
• [Asterisk-Users] DS3000P - 20 E1 capacity on single card   Matthew Boehm
• [Asterisk-Users] IPswitch Monitor Extension   Joseph Gutowski
• [Asterisk-Users] DS3000P - 20 E1 capacity on single card   Andrew Kohlsmith
• [Asterisk-Users] RE: Ebay listing selling Asterisk @ Home and AM P for over 1000 dollars   Colin Anderson
• [Asterisk-Users] Play Sound File Without Answer Channel   Eric Wieling aka ManxPower
• [Asterisk-Users] Play Sound File Without Answer Channel   Angel Diaz
• [Asterisk-Users] Low cost box for hosting Asterisk and at leastone TDM400p   Ken Godee
• [Asterisk-Users] Sip transfer and redirect in a Company setting   Jeb Campbell
• [Asterisk-Users] DS3000P - 20 E1 capacity on single card   Andres
• [Asterisk-Users] Line Noise HELP!   Andre Normandin
• [Asterisk-Users] Changing DTMF mode depending on codec chosen   Andre Normandin
• [Asterisk-Users] Latest Bristuff crashes on modprobe -r qozap ?   Robert Rozman
• [Asterisk-Users] Low cost box for hosting Asterisk and at leastoneTDM400p   Wiley Siler
• [Asterisk-Users] RE: Ebay listing selling Asterisk @ Home and AMP for over 1000 dollars   Guillermo Salas M
• [Asterisk-Users] Line Noise HELP!   Andre Normandin
• [Asterisk-Users] RE: Ebay listing selling Asterisk @ Home and AMPfor over 1000 dollars   dean collins
• [Asterisk-Users] Low cost box for hosting Asterisk and at leastoneTDM400p   Henry Devito
• [Asterisk-Users] Cannot open chan_zap:   Tim Connolly
• [Asterisk-Users] Why 's' doesn't take over unknown extensionincontext ?   Robert Rozman
• [Asterisk-Users] Check for stutter dialtone on ZAP FXO channel   Cameron Beattie
• [Asterisk-Users] "Refresh" asterisk internal database?   Terry Wilson
• [Asterisk-Users] Line Noise HELP!   Damian Funnell
• [Asterisk-Users] Registration issues with Sipura SPA-841   GliTcH
• [Asterisk-Users] AstCC problems   Dave Kettmann
• [Asterisk-Users] Re: Using manager interface to play aanouncmentsinaMeetMe   Dan Austin
• [Asterisk-Users] timed Loop   Chris
• [Asterisk-Users] BROADVOICE - Incomming calls are dropped after 1-2 min   Craig Simon
• [Asterisk-Users] Low cost box for hosting Asterisk and at leastone TDM400p   GliTcH
• [Asterisk-Users] broadvoice config problem.   Craig Simon
• [Asterisk-Users] Asterisk becomes after one month unstabled   Rich Adamson
• [Asterisk-Users] UK ISDN with Asterisk   David Phelan
• [Asterisk-Users] timed Loop   Eric Wieling aka ManxPower
• [Asterisk-Users] Maximum amount of users on one asterisk server?   Matt
• [Asterisk-Users] Play Sound File Without Answer Channel   Eric Wieling aka ManxPower
• [Asterisk-Users] help   whminfo at hotlink.com.br
• [Asterisk-Users] RE: Ebay listing selling Asterisk @ Home and AMP for over 1000 dollars   Rich Adamson
• [Asterisk-Users] IPswitch Monitor Extension   Ronald Wiplinger
• [Asterisk-Users] Q931 Setup message   Eugenio De Vena
• [Asterisk-Users] Low cost box for hosting Asterisk and at leastone TDM400p   Rich Adamson
• [Asterisk-Users] IPswitch Monitor Extension   Ronald Wiplinger
• [Asterisk-Users] Linksys PAP2 Dual Incoming Calls   mmiranda at americatel.com.sv
• [Asterisk-Users] DS3000P - 20 E1 capacity on single card   Bicom Systems
• [Asterisk-Users] Pre-install questions   Mike Myers
• [Asterisk-Users] Line Noise HELP!   Rich Adamson
• [Asterisk-Users] broadvoice config problem.   Johnathan Corgan
• [Asterisk-Users] Dialogic cards compatibility   Leo Salas
• [Asterisk-Users] broadvoice config problem.   Craig Simon
• [Asterisk-Users] Cannot open chan_zap:   Rich Adamson
• [Asterisk-Users] Line Noise HELP!   Damian Funnell
• [Asterisk-Users] Check for stutter dialtone on ZAP FXO channel   Rich Adamson
• [Asterisk-Users] Linksys PAP2 Dual Incoming Calls   Rusty Shackleford
• [Asterisk-Users] BroadVoice Issues   Brian Watters
• [Asterisk-Users] Line Noise HELP!   Rich Adamson
• [Asterisk-Users] broadvoice config problem.   Craig Simon
• [Asterisk-Users] DS3000P - 20 E1 capacity on single card   Matthew Crocker
• [Asterisk-Users] Asterisk-Users] RE:Asterisk Voice mail with CCM   dakemp
• [Asterisk-Users] broadvoice config problem.   Rich Adamson
• [Asterisk-Users] Low cost box for hosting Asterisk and at leastoneTDM400p   David John Walsh
• [Asterisk-Users] Bounty: Request for PRI Debug   Matt Fredrickson
• [Asterisk-Users] DS3000P - 20 E1 capacity on single card   Andrew Kohlsmith
• [Asterisk-Users] ASTCC - IVR prompts   David John Walsh
• [Asterisk-Users] Linux & Asterisk   Paul Hales
• [Asterisk-Users] Best FXO Voip Gateway for Asterisk   Chad Brown
• [Asterisk-Users] RE: Ebay listing selling Asterisk @ Home and AMPfor over 1000 dollars   Robert Webb
• [Asterisk-Users] broadvoice config problem.   Johnathan Corgan
• [Asterisk-Users] Receiving calls from and to H323 devices   VOIP Consultant
• [Asterisk-Users] TDM400P Revision   Robert Webb
• [Asterisk-Users] RE: Ebay listing selling Asterisk @ Home and AMPfor over 1000 dollars   Ronald Wiplinger
• [Asterisk-Users] Can you comment on this Qos script? How does one shape RTP?   Kenneth Porter
• [Asterisk-Users] broadvoice config problem.   snacktime
• [Asterisk-Users] BROADVOICE - Incomming calls are dropped after1-2 min   Bartosz Wegrzyn - asterisk
• [Asterisk-Users] Advice in provider for business use   Bartosz Wegrzyn - asterisk
• [Asterisk-Users] Dialing Out   doug at simflex.com
• [Asterisk-Users] RE: Ebay listing selling Asterisk @ Home and AMPfor over 1000 dollars   Jon Gabrielson
• [Asterisk-Users] RE: Ebay listing selling Asterisk @ Home and AMPfor over 1000 dollars   Robert Webb
• [Asterisk-Users] OT: Thunderbird threading   Tony Davidson
• [Asterisk-Users] Linksys PAP2 Dual Incoming Calls   Matt Darnell
• [Asterisk-Users] Bounty: Request for PRI Debug   Trevor Peirce
• [Asterisk-Users] Monitor with Asterisk@Home   mr. barker
• [Asterisk-Users] Problem Detecting Answer on a PRI Outcall (sometimes)   jkeller at isecorp.com
• [Asterisk-Users] Asterisk@Home 0.8 released {Scanned}   David Shaw
• [Asterisk-Users] Monitor with Asterisk@Home   Kyle Hagan
• [Asterisk-Users] Problem Detecting Answer on a PRI Outcall (sometimes)   jkeller at isecorp.com
• [Asterisk-Users] broadvoice config problem.   Craig Simon
• [Asterisk-Users] BROADVOICE - Incomming calls are dropped after1-2 min   Kerry Garrison
• [Asterisk-Users] TDM400P Revision   Rich Adamson
• [Asterisk-Users] Asterisk did not play music when pressing hold button on SJPhone   Chee Foong
• [Asterisk-Users] E911?   Matthew S.Krawitz
• [Asterisk-Users] RE: Ebay listing selling Asterisk @ Home andAMPfor over 1000 dollars   Kerry Garrison
• [Asterisk-Users] Asterisk Realtime - can't see sip friend   Cameron Beattie
• [Asterisk-Users] broadvoice config problem.   Rich Adamson
• [Asterisk-Users] Remote phone often appears to be disconnected   Ronald Wiplinger
• [Asterisk-Users] broadvoice config problem.   Johnathan Corgan
• [Asterisk-Users] Low cost box for hosting Asterisk and at leastoneTDM400p   Rich Adamson
• [Asterisk-Users] Line Noise HELP!   Seth Remington
• [Asterisk-Users] ChanSpy -- Slow/garbled audio and console errors   James W.Brinkerhoff
• [Asterisk-Users] broadvoice config problem.   Rich Adamson
• [Asterisk-Users] Changing DTMF mode depending on codec chosen   Rich Adamson
• [Asterisk-Users] Asterisk management portal   Michael D Schelin
• [Asterisk-Users] H.323 General Questions   Rod Bacon
• [Asterisk-Users] Asterisk management portal   JD
• [Asterisk-Users] Asterisk management portal   Brian Watters
• [Asterisk-Users] Asterisk management portal   David Phelan
• [Asterisk-Users] Asterisk Dual Servers {Scanned}   David Shaw
• [Asterisk-Users] Trunk Seize - Line 1 - CO1: Does it exist in an Asterisk environment?   Ben Ryan
• [Asterisk-Users] Trunk Seize - Line 1 - CO1: Does it exist in an Asterisk environment?   Michael Loftis
• [Asterisk-Users] agent autologoff   Anton Krall
• [Asterisk-Users] Re: polycom phones   Greg Boehnlein
• [Asterisk-Users] Changing DTMF mode depending on codec chosen   Luki
• [Asterisk-Users] Cannot open chan_zap:   Tim Connolly
• [Asterisk-Users] Zyxel P2000W Finally (Almost) Working   Rod Bacon
• [Asterisk-Users] Cann't get CallerID on Zap channel, Please Help!!   Dominic Lu
• [Asterisk-Users] Zyxel P2000W Finally (Almost) Working   Rod Bacon
• [Asterisk-Users] Cannot open chan_zap:   Eric Wieling aka ManxPower
• [Asterisk-Users] Losing CallerName info if no CID sent   Matt Gibson
• [Asterisk-Users] Trunk Seize - Line 1 - CO1: Does it exist in an Asterisk environment?   Peter Svensson
• [Asterisk-Users] Zyxel P2000W Finally (Almost) Working   Michael D Schelin
• [Asterisk-Users] Cannot open chan_zap:   Tim Connolly
• [Asterisk-Users] (no subject)   Sascha Ferley
• [Asterisk-Users] Has anyone got Asterisk working behind a NAT connection to users within a NAT   Fats Neutron
• [Asterisk-Users] Problem with * transfer   Thorben Jensen
• [Asterisk-Users] Re: Problems trying to compile H323 from CVS-STABLE   Michael Manousos
• [Asterisk-Users] Asterisk quit abnormally   Qiao Yuansong
• [Asterisk-Users] About Audio Latency from PSTN to SIP   Qiao Yuansong
• [Asterisk-Users] Best FXO Voip Gateway for Asterisk   ht at phonitel.com
• [Asterisk-Users] Re: Problems trying to compile H323 from CVS-STABLE   Tony Mountifield
• [Asterisk-Users] Asterisk on HP DL380 G4 - chan_zap.so problems   Lukas Kaiser
• [Asterisk-Users] Voicemail quota   Chee Foong
• [Asterisk-Users] Supervisor monitor / barge in - automatically on call setup?   David John Walsh
• [Asterisk-Users] Supervisor monitor / barge in - automatically on call setup?   Ronald Wiplinger
• [Asterisk-Users] Re: Problems trying to compile H323 from CVS-STABLE   Michael Manousos
• [Asterisk-Users] Petition for IAX firmware   Wilson Pickett
• [Asterisk-Users] Asterisk quit abnormally   Ronald Wiplinger
• [Asterisk-Users] Has anyone got Asterisk working behind a NAT connection to users within a NAT   Wilson Pickett
• [Asterisk-Users] Asterisk Addons compile errors   lie ka
• [Asterisk-Users] H.323 Question   Daniel Eboa
• [Asterisk-Users] TE110P - NT-Mode ?   Henry Jensen
• [Asterisk-Users] Changing DTMF mode depending on codec chosen   Andre Normandin
• [Asterisk-Users] Changing DTMF mode depending on codec chosen   Andre Normandin
• [Asterisk-Users] Line Noise HELP!   Andre Normandin
• [Asterisk-Users] Problem with fxo   Julio Saura
• [Asterisk-Users] Version 0.80 of IPS released   Thorben Jensen
• [Asterisk-Users] Petition for IAX firmware   Sascha Ferley
• [Asterisk-Users] (no subject)   Sascha Ferley
• [Asterisk-Users] Getting CVS HEAD   Guillermo Salas M.
• [Asterisk-Users] TE410P and X101P problem   Lee Lee
• [Asterisk-Users] From OH323 to SIP or OH323 without gatekeeper   Guillermo Salas M.
• [Asterisk-Users] RE: Monitor with Asterisk@Home   mr. barker
• [Asterisk-Users] Petition for IAX firmware   Guillermo Salas M.
• [Asterisk-Users] Re: TE110P - NT-Mode ?   Tony Mountifield
• [Asterisk-Users] (no subject)   Rich Adamson
• [Asterisk-Users] Re: TE110P - NT-Mode ?   Henry Jensen
• [Asterisk-Users] Changing DTMF mode depending on codec chosen   Rich Adamson
• [Asterisk-Users] multiple asterisk boxes with "show channels"   Jerry Geis
• [Asterisk-Users] Re: From OH323 to SIP or OH323 without gatekeeper   Bruno Hertz
• [Asterisk-Users] Re: TE110P - NT-Mode ?   Tony Mountifield
• [Asterisk-Users] Remote phone often appears to be disconnected   Julian J. M.
• [Asterisk-Users] Interface bonding + asterisk   Bob Goddard
• [Asterisk-Users] How to get list of codecs   Pavel Siderov - Hostmates
• [Asterisk-Users] Low cost box for hosting Asterisk and atleastone TDM400p - THIN CLIENT MAYBE?   Chuck Bunn
• [Asterisk-Users] Asterisk Addons compile errors   I put the Who? in Mishehu
• [Asterisk-Users] Multiple Servers and 1 Central Voicemail   Anton Krall
• [Asterisk-Users] Version 0.80 of IPS released   Ivan Meic (Vox Mundi)
• [Asterisk-Users] Low cost box for hosting Asterisk and atleastoneTDM400p - THIN CLIENT MAYBE?   Wiley Siler
• [Asterisk-Users] NENA CAMA Trunks for 911 and *   Damon Estep
• [Asterisk-Users] Cannot open chan_zap:   Bob Goddard
• [Asterisk-Users] Meetme disconnecting clients that use VAD   Steven Langley
• [Asterisk-Users] Internet Conection Broken and asterisk can not route any calls   Obihuan
• [Asterisk-Users] VAD/DTX implementation through zaptel cards   Steve Kann
• [Asterisk-Users] Trunk Seize - Line 1 - CO1: Does it exist in an Asterisk environment?   I put the Who? in Mishehu
• [Asterisk-Users] multiple line usage on Polycom IP300   MobilPete
• [Asterisk-Users] TE410P and X101P problem   Scott Stingel
• [Asterisk-Users] Getting CVS HEAD   Jon Califf
• [Asterisk-Users] Problem with fxo   Moises Silva
• [Asterisk-Users] (no subject)   Rich Adamson
• [Asterisk-Users] TE410P and X101P problem   Adam Goryachev
• [Asterisk-Users] Power Consumption of a Digium Wildcard TE410P   Oliver Rath
• [Asterisk-Users] Version 0.80 of IPS released   Adam Goryachev
• [Asterisk-Users] VAD/DTX implementation through zaptel cards   Steve Underwood
• [Asterisk-Users] multiple line usage on Polycom IP300   Josiah Bryan
• [Asterisk-Users] Version 0.80 of IPS released   Ronald Wiplinger
• [Asterisk-Users] DS3000P - 20 E1 capacity on single card   Kevin P. Fleming
• [Asterisk-Users] RE: Ebay listing selling Asterisk @ Home andAMPfor over 1000 dollars   I put the Who? in Mishehu
• [Asterisk-Users] DS3000P - 20 E1 capacity on single card   Kevin P. Fleming
• [Asterisk-Users] Dialing Out   Dana Olson
• [Asterisk-Users] Acceptable voice time delay   chawki hammoud
• [Asterisk-Users] Has anyone got Asterisk working behind a NAT connection to users within a NAT   Michiel van Baak
• [Asterisk-Users] multiple line usage on Polycom IP300   MobilPete
• [Asterisk-Users] RE: Ebay listing selling Asterisk @ Home (Blah Blah)   Christopher Jacob
• [Asterisk-Users] multiple line usage on Polycom IP300   Wiley Siler
• [Asterisk-Users] Acceptable voice time delay   Sean Kennedy
• [Asterisk-Users] multiple line usage on Polycom IP300   Jerry
• [Asterisk-Users] Acceptable voice time delay   Rob Scott
• [Asterisk-Users] Petition for IAX firmware   Wilson Pickett
• [Asterisk-Users] Re: multiple line usage on Polycom IP300   Noah Miller
• [Asterisk-Users] DS3000P - 20 E1 capacity on single card   Bicom Systems
• [Asterisk-Users] Version 0.80 of IPS released   Ronald Wiplinger
• [Asterisk-Users] Remote phone often appears to be disconnected   Ronald Wiplinger
• [Asterisk-Users] Dialing Out (My mistake, here is the entire message)   doug at simflex.com
• [Asterisk-Users] RE: Ebay listing selling Asterisk @Home andAMPfor over 1000 dollars   Kerry Garrison
• [Asterisk-Users] Low cost box for hosting Asterisk and atleastone TDM400p - THIN CLIENT MAYBE?   Dana Olson
• [Asterisk-Users] Line Noise HELP!   Rich Adamson
• [Asterisk-Users] multiple line usage on Polycom IP300   Josiah Bryan
• [Asterisk-Users] QoS TOS numbers and Cisco IOS   Noah Miller
• [Asterisk-Users] Re: polycom phones   Trevor Harrison
• [Asterisk-Users] How to get list of codecs   Moises Silva
• [Asterisk-Users] Re: Asterisk-Users Digest, Vol 9, Issue 104   Aldo Bergamini
• [Asterisk-Users] Multiple Servers and 1 Central Voicemail   Luki
• [Asterisk-Users] Multiple TDM cards on the same box   Nir Simionovich
• [Asterisk-Users] How do I reduce echo on the Caller side   Joel Jn-Francois
• [Asterisk-Users] TE110P - NT-Mode ?   Matt Fredrickson
• [Asterisk-Users] Noises on ZAP Channels   Carsten Bock
• [Asterisk-Users] Agents   jamesm
• [Asterisk-Users] Low cost box for hosting Asterisk and atleastone TDM400p - THIN CLIENT MAYBE?   Andrew Latham
• [Asterisk-Users] DS3000P - 20 E1 capacity on single card   Kevin P. Fleming
• [Asterisk-Users] Multiple Servers and 1 Central Voicemail   Josiah Bryan
• [Asterisk-Users] Dumb question ?   mr. barker
• [Asterisk-Users] How many licenses of G729 do I need?   Ronald Wiplinger
• [Asterisk-Users] LCDial and default provider   Alex
• [Asterisk-Users] Acceptable voice time delay   chawki hammoud
• [Asterisk-Users] Multiple TDM cards on the same box   Wiley Siler
• [Asterisk-Users] Has anyone got Asterisk working behind a NAT connection to users within a NAT   Wilson Pickett
• [Asterisk-Users] How do I reduce echo on the Caller side   Andrew Kohlsmith
• [Asterisk-Users] Dumb question ?   Josiah Bryan
• [Asterisk-Users] Local Echo   Noah Silverman
• [Asterisk-Users] (no subject)   Jon Bebeau
• [Asterisk-Users] Meetme and billing   Dipen Gandhi
• [Asterisk-Users] QoS TOS numbers and Cisco IOS   Rich Adamson
• [Asterisk-Users] Re: Multiple Servers and 1 Central Voicemail   Noah Miller
• [Asterisk-Users] Multiple Servers and 1 Central Voicemail   Fares Gianluca
• [Asterisk-Users] Multiple Servers and 1 Central Voicemail   Johnathan Corgan
• [Asterisk-Users] Re: Dumb question ?   Noah Miller
• [Asterisk-Users] DS3000P - 20 E1 capacity on single card   Jon Bebeau
• [Asterisk-Users] Cisco 7960s and skinny   Morris, Simon
• [Asterisk-Users] Low cost box for hosting Asterisk and atleastoneTDM400p - THIN CLIENT MAYBE?   Wiley Siler
• [Asterisk-Users] Multiple Servers and 1 Central Voicemail   Jason Brown
• [Asterisk-Users] Dumb question ?   chawki hammoud
• SV: [Asterisk-Users] Version 0.80 of IPS released   Thorben Jensen
• [Asterisk-Users] Version 0.80 of IPS released   Thorben Jensen
• [Asterisk-Users] Iaxy, Transfer, & #   Tim Pushor
• [Asterisk-Users] Version 0.80 of IPS released   Thorben Jensen
• [Asterisk-Users] Version 0.80 of IPS released   Thorben Jensen
• [Asterisk-Users] DS3000P - 20 E1 capacity on single card   tmassey at obscorp.com
• [Asterisk-Users] Attempting native bridge of   Xu Wang
• [Asterisk-Users] RE: Ebay listing selling Asterisk @ Home andAMPfor over 1000 dollars   Jean-Michel Hiver
• [Asterisk-Users] Accountcode in SIP.CONF not set   Ed Greenberg
• [Asterisk-Users] g729 versus g711   snacktime
• [Asterisk-Users] Version 0.80 of IPS released   Ivan Meic (Vox Mundi)
• [Asterisk-Users] Music on hold not working between SIP clients   Tommaso Calosi
• [Asterisk-Users] Anybody who is using a live stream as MOH   Kris Edwards
• [Asterisk-Users] How many licenses of G729 do I need?   Jean-Michel Hiver
• [Asterisk-Users] Asterisk@Home - Newer Mobo - Memory   Wiley Siler
• [Asterisk-Users] Line Noise HELP!   Damian Funnell
• [Asterisk-Users] RE: Ebay listing selling Asterisk @ Home and AMPfor over 1000 dollars   David Brodbeck
• [Asterisk-Users] Local Echo   Rich Adamson
• [Asterisk-Users] overwriting config file problem   mr. barker
• [Asterisk-Users] DS3000P - 20 E1 capacity on single card   Bicom Systems
• [Asterisk-Users] DS3000P - 20 E1 capacity on single card   Matthew Boehm
• [Asterisk-Users] Local Echo   Noah Silverman
• [Asterisk-Users] RE: Ebay listing selling Asterisk @ Home and AMPfor over 1000 dollars   Robert Webb
• [Asterisk-Users] overwriting config file problem   Robert Webb
• [Asterisk-Users] RE: Ebay listing selling Asterisk @ Home and AMPfor over 1000 dollars   trixter http://www.0xdecafbad.com
• [Asterisk-Users] Re: How do I reduce echo on asterisk   Joel Jn-Francois
• [Asterisk-Users] RE: Ebay listing selling Asterisk @ Home and AMPfor over 1000 dollars   trixter http://www.0xdecafbad.com
• [Asterisk-Users] Looking for comments on robustness of SpanDSP / app-rxfax / mime-construct   Colin Anderson
• [Asterisk-Users] Attempting native bridge of   Robert Goodyear
• [Asterisk-Users] RE: Ebay listing selling Asterisk @ Home and AMPfor over 1000 dollars   Neal Walton
• [Asterisk-Users] CISCO 12SP+/30VIP + Asterisk   Juergen K. Zick
• [Asterisk-Users] 503 Service Unavailable   Michael D Schelin
• [Asterisk-Users] Attempting native bridge of   Xu Wang
• [Asterisk-Users] Multiple Servers and 1 Central Voicemail   Anton Krall
• [Asterisk-Users] agent autologoff   Anton Krall
• [Asterisk-Users] Re: Multiple Servers and 1 Central Voicemail   Luki
• [Asterisk-Users] OH323 and outgoing calls.   Adam Rybak
• [Asterisk-Users] IAX2 - Between two ASterisk Servers   Chris
• [Asterisk-Users] Local Echo   Jeff Heath
• [Asterisk-Users] RE: Sangoma VS. Digium   Jason Brown
• [Asterisk-Users] Line Noise HELP!   Daniel Bruce Lynes
• [Asterisk-Users] Changing DTMF mode depending on codec chosen   Andre Normandin
• [Asterisk-Users] Multiple Servers and 1 Central Voicemail   Johnathan Corgan
• [Asterisk-Users] Local Echo   Noah Silverman
• [Asterisk-Users] Re: How do I reduce echo on asterisk   Jeff Heath
• [Asterisk-Users] ztdummy   Daniel Bruce Lynes
• [Asterisk-Users] TDM02B on 2 a/b ports of a PBX not working.. . help   Brian Leyton
• [Asterisk-Users] Cisco 7960s and skinny   Andy Hamilton
• [Asterisk-Users] IAX2 - Between two ASterisk Servers   Colin Anderson
• [Asterisk-Users] multiple line usage on Polycom IP300   Eric Wieling
• [Asterisk-Users] QoS TOS numbers and Cisco IOS   Eric Wieling
• [Asterisk-Users] IAX2 - Between two ASterisk Servers   Tim Litwiller
• [Asterisk-Users] Line Noise HELP!   Paul
• [Asterisk-Users] Jitter problems IAX to Livevoip   John Goerzen
• [Asterisk-Users] Re: [Asterisk-Dev] Iax Trunking LD Service   Andrew Kohlsmith
• [Asterisk-Users] Local Echo   Jeff Heath
• [Asterisk-Users] Attempting native bridge of   Robert Goodyear
• [Asterisk-Users] Version 0.80 of IPS released   David John Walsh
• [Asterisk-Users] Attempting native bridge of   Xu Wang
• [Asterisk-Users] voice mail playback   Michael D Schelin
• [Asterisk-Users] binding Asterisk to virtual IP   Xu Wang
• [Asterisk-Users] Attempting native bridge of   Mike Price
• [Asterisk-Users] binding Asterisk to virtual IP   Eric Wieling
• [Asterisk-Users] PRI Errors with TE110P   Aaron Mathews
• [Asterisk-Users] binding Asterisk to virtual IP   Kerry Garrison
• [Asterisk-Users] Looking for comments on robustness of SpanDS P / app-rxfax / mime-construct <--More information, and I figured out wh y   Colin Anderson
• [Asterisk-Users] binding Asterisk to virtual IP   Xu Wang
• [Asterisk-Users] IAX2 - Between two ASterisk Servers   Chris
• [Asterisk-Users] binding Asterisk to virtual IP   Daniel Bruce Lynes
• [Asterisk-Users] binding Asterisk to virtual IP   Kerry Garrison
• [Asterisk-Users] IAX2 - Between two ASterisk Servers   Colin Anderson
• [Asterisk-Users] Documentation link,   Andres Paglayan
• [Asterisk-Users] binding Asterisk to virtual IP   Xu Wang
• [Asterisk-Users] Realtime Friends   Rod Bacon
• [Asterisk-Users] Local Echo   Rod Bacon
• [Asterisk-Users] Local Echo   Noah Silverman
• [Asterisk-Users] Local Echo   Matt Fredrickson
• [Asterisk-Users] IAX2 - Between two ASterisk Servers   Chris
• [Asterisk-Users] binding Asterisk to virtual IP   Kerry Garrison
• [Asterisk-Users] Local Echo   Rod Bacon
• [Asterisk-Users] binding Asterisk to virtual IP   Xu Wang
• [Asterisk-Users] Version 0.80 of IPS released   Adam Goryachev
• [Asterisk-Users] Local Echo   Jeff Heath
• [Asterisk-Users] Looking for comments on robustness of SpanDSP / app-rxfax / mime-construct   Boris Bakchiev
• [Asterisk-Users] Local Echo   Noah Silverman
• [Asterisk-Users] g729 versus g711   Steve Underwood
• [Asterisk-Users] overwriting config file problem   Asterisk at Home
• [Asterisk-Users] Local Echo   Noah Silverman
• [Asterisk-Users] Attempting native bridge of   Robert Goodyear
• [Asterisk-Users] OH323: Sending CallerID to H323 voip provider...   Robert Rozman
• [Asterisk-Users] Outbound calling stops working after configuring trunk lines   Ryan J. Hutchinson
• [Asterisk-Users] BroadVoice Inbound Not working ..   Brian Watters
• [Asterisk-Users] BroadVoice Inbound Not working ..   Kerry Garrison
• [Asterisk-Users] BroadVoice Inbound Not working ..   Johnathan Corgan
• [Asterisk-Users] New SNOM 190 Firmware   Rod Bacon
• [Asterisk-Users] Australian anologue callerid   Kiel R Stirling
• [Asterisk-Users] Dialogic cards compatibility   C F
• [Asterisk-Users] BroadVoice Inbound Not working ..   Dalon Westergreen
• [Asterisk-Users] 3-Way Calling in Asterisk   aram
• [Asterisk-Users] Local Echo   Bashir Ullah - www.Lamsre.Com
• [Asterisk-Users] Blank voicemails being sent to users   Matt Darnell
• [Asterisk-Users] New SNOM 190 Firmware   Shaun Dwyer
• [Asterisk-Users] Blank voicemails being sent to users   John Novack ( Mozilla - portable )
• [Asterisk-Users] Sip transfer and redirect in a Company setting   C F
• [Asterisk-Users] Setgroup & Checkgroup   C F
• [Asterisk-Users] Question about Macros   Joe S
• [Asterisk-Users] Setgroup & Checkgroup   C F
• [Asterisk-Users] voice mail playback   Michael D Schelin
• [Asterisk-Users] New SNOM 190 Firmware   Rod Bacon
• [Asterisk-Users] semantics terminology   Thomas Andrews
• [Asterisk-Users] SIP Attended/Supervised transfer & features.conf   C F
• [Asterisk-Users] Shared call appearances   C F
• [Asterisk-Users] Script routing Logic Question in .conf files   mr. barker
• [Asterisk-Users] Compile/modprobe issue   Steven P. Donegan
• [Asterisk-Users] Running asterisk without special hardware   Manish Sapariya
• [Asterisk-Users] DS3000P - 20 E1 capacity on single card   Kevin P. Fleming
• [Asterisk-Users] DS3000P - 20 E1 capacity on single card   Kevin P. Fleming
• [Asterisk-Users] Running asterisk without special hardware   Damian Funnell
• [Asterisk-Users] Testing call back to back mfcr2 using chan_unicall with TE400P   Eris Riswanto
• [Asterisk-Users] Verizon ISDN   Gregory Wiktor - ADCom Corp.
• [Asterisk-Users] Any success with BRI in the US?   Gregory Wiktor - ADCom Corp.
• [Asterisk-Users] Hangup after Transfer problem.   Karl J. Vesterling
• [Asterisk-Users] BroadVoice Inbound Not working ..   Brian Watters
• [Asterisk-Users] Compile/modprobe issue   Jeffrey C. Ollie
• [Asterisk-Users] Looking for comments on robustness of SpanDS P / app-rxfax / mime-construct <--solved   Colin Anderson
• [Asterisk-Users] Local Echo   Adam Goryachev
• [Asterisk-Users] Attempting native bridge of   snacktime
• [Asterisk-Users] invalid extension(need help)   amna saleem
• [Asterisk-Users] Attempting native bridge of   Xu Wang
• [Asterisk-Users] invalid extension (need help)   amna saleem
• [Asterisk-Users] Question about Macros   Joe S
• [Asterisk-Users] Attempting native bridge of   snacktime
• [Asterisk-Users] Article on IAX in Network World   Brian Capouch
• [Asterisk-Users] BroadVoice Inbound Not working ..   Dalon Westergreen
• [Asterisk-Users] weird call transfer problem   Anton Krall
• [Asterisk-Users] binding Asterisk to virtual IP   Xu Wang
• [Asterisk-Users] Problem reading digits from OH323 caller   Joe S
• [Asterisk-Users] DS3000P - 20 E1 capacity on single card   Matt Klein
• [Asterisk-Users] DS3000P - 20 E1 capacity on single card   Matt Klein
• [Asterisk-Users] Agents   Dylan VanHerpen
• [Asterisk-Users] Interesting Cisco 7960 issue, the phone picks up without me!   Craig Simon
• [Asterisk-Users] chan_sip.c:7215 handle_request   Xu Wang
• FW: [Asterisk-Users] binding Asterisk to virtual IP   Xu Wang
• [Asterisk-Users] voice mail playback   Rich Adamson
• [Asterisk-Users] invalid extension (need help)   El Flynn
• [Asterisk-Users] Documentation link,   Wilson Pickett
• [Asterisk-Users] weird call transfer problem   El Flynn
• [Asterisk-Users] VAD/DTX implementation through zaptel cards   parijat
• [Asterisk-Users] attension mark spencer   amna saleem
• [Asterisk-Users] Polycom V500 With Asterisk Setup   David Choo
• [Asterisk-Users] Version 0.80 of IPS released   Thorben Jensen
• [Asterisk-Users] Changing DTMF mode depending on codec chosen   Rich Adamson
• [Asterisk-Users] Problem with fxo   Julio Saura
• [Asterisk-Users] Attempting native bridge of   Robert Goodyear
• [Asterisk-Users] New PRI install with new te110p   Me
• [Asterisk-Users] attension mark spencer   Dave Cotton
• [Asterisk-Users] Codecs and * pass through...   Etienne Pretorius
• [Asterisk-Users] weird call transfer problem   Anton Krall
• [Asterisk-Users] Sipura SPA-841 and Asterisk 1.0.7 with chan_misdn   David Phelan
• [Asterisk-Users] Codecs and * pass through...   clive at sentechsa.com
• [Asterisk-Users] Codecs and * pass through...   Etienne Pretorius
• [Asterisk-Users] How to get list of codecs   Pavel Siderov - Hostmates
• [Asterisk-Users] Sangoma A101 + Rhino channelbank   Felician CHELU
• [Asterisk-Users] Re: Problem reading digits from OH323 caller   Tony Mountifield
• [Asterisk-Users] PortaSIP/PortaBilling incompatibility (provider: sipcall.ch)   Marc SCHAEFER
• [Asterisk-Users] Cisco 7960s and skinny   Julien Goodwin
• [Asterisk-Users] iaxcomm   amna saleem
• [Asterisk-Users] Turtle Firewall - Sip user   Michael Sanders
• [Asterisk-Users] ISDN Fritz and TDM400   Robson Ribeiro
• [Asterisk-Users] ISDN Fritz and TDM400   Elmar Haneke
• [Asterisk-Users] ISDN Fritz and TDM400   Robson Ribeiro
• [Asterisk-Users] codec quality   Steven Langley
• [Asterisk-Users] New PRI install with new te110p   Andrew Kohlsmith
• [Asterisk-Users] invalid extension (need help)   Andrew Kohlsmith
• [Asterisk-Users] DS3000P - 20 E1 capacity on single card   Andrew Kohlsmith
• [Asterisk-Users] Version 0.80 of IPS released   Chris Mason (Lists)
• [Asterisk-Users] Re: Cisco 7960s and skinny   Sergio
• [Asterisk-Users] Asterisk / Quintum CRSP codec problems   Pavel Siderov
• [Asterisk-Users] Compile/modprobe issue   Steven P. Donegan
• [Asterisk-Users] Cisco 7940G SIP Conversion   Michael West
• [Asterisk-Users] Question about Routing Order in .conf files   mr. barker
• [Asterisk-Users] Multiple TDM400x Cards on the same box   Nir Simionovich
• [Asterisk-Users] VAD/DTX implementation through zaptel cards   Eric Wieling
• [Asterisk-Users] Busy line status and chan_capi?   Kib Eki
• [Asterisk-Users] VAD/DTX implementation through zaptel cards   parijat
• [Asterisk-Users] x-ten lite error   asterisk at frameweb.it
• [Asterisk-Users] 3-Way Calling in Asterisk   Wiley Siler
• [Asterisk-Users] OH323 and Asterisk CVS-HEAD-03/21/05-15:32:10   Jose R. Ortiz Ubarri
• [Asterisk-Users] Cisco 7940G SIP Conversion   Boris Bakchiev
• [Asterisk-Users] Re: Cisco 7960s and skinny   Julien Goodwin
• [Asterisk-Users] Who is willing to help an Asterisk newby?   Wolf N. Paul
• [Asterisk-Users] Who is willing to help an Asterisk newby?   Simon Morris
• [Asterisk-Users] 3-Way Calling in Asterisk   Allen Niven
• [Asterisk-Users] Who is willing to help an Asterisk newby?   mr. barker
• [Asterisk-Users] Asterisk on debian sarge doesn't start with CAPI module errors   Simon Morris
• [Asterisk-Users] ni1 (ppp) and national isdn on te110p   Jason McAffee
• [Asterisk-Users] Re: Running asterisk without special hardware   Bruno Hertz
• [Asterisk-Users] DS3000P - 20 E1 capacity on single card   Steve Underwood
• [Asterisk-Users] Acceptable voice time delay   Sean Kennedy
• [Asterisk-Users] Question about Macros   Kanuri, Seshu (Company IT)
• [Asterisk-Users] PCI 1xE1, 2xE1 cards from Russia for MFC/R2 signaling for Asterisk IP-PBX   Maxim Litnitsky
• [Asterisk-Users] SNOM 220 with >7 "lines"   Michael Welter
• [Asterisk-Users] Zaptel and Fritz Card   Robson Ribeiro
• [Asterisk-Users] OH323 and Asterisk CVS-HEAD-03/21/05-15:32:10   Michael Manousos
• [Asterisk-Users] Transferring a call   Dennie Verstrepen
• [Asterisk-Users] SNOM 190: Unknown SIP command 'PUBLISH'   Ronald Wiplinger
• [Asterisk-Users] Local Echo   Matt Fredrickson
• [Asterisk-Users] Local Echo   Matt Fredrickson
• [Asterisk-Users] Zaptel and Fritz Card   Peer Oliver Schmidt
• [Asterisk-Users] i need help   eng.yousri at egypt.com
• [Asterisk-Users] OH323 and Asterisk CVS-HEAD-03/21/05-15:32:10   Guillermo Salas M
• [Asterisk-Users] Local Echo   Neal Walton
• [Asterisk-Users] IPSwitchBoard is now Event Driven   Thorben Jensen
• [Asterisk-Users] binding Asterisk to virtual IP   Leif Madsen - Certified Asterisk Consultant
• [Asterisk-Users] Version 0.80 of IPS released   Thorben Jensen
• [Asterisk-Users] IAX2 - Between two ASterisk Servers   Chris
• [Asterisk-Users] SIP registration fails   William Marks
• [Asterisk-Users] DS3000P - 20 E1 capacity on single card   Kevin P. Fleming
• [Asterisk-Users] DS3000P - 20 E1 capacity on single card   Kevin P. Fleming
• [Asterisk-Users] Acceptable voice time delay   chawki hammoud
• [Asterisk-Users] Acceptable voice time delay   chawki hammoud
• [Asterisk-Users] FRAME_CONTROL (5) dropping calls on PRI   Jeb Campbell
• [Asterisk-Users] i need help   Wiley Siler
• [Asterisk-Users] IAX2 - Between two ASterisk Servers   Colin Anderson
• [Asterisk-Users]Unable to register license for G729 codec   Mohammed Firdosh Nasim
• [Asterisk-Users] 3-Way Calling in Asterisk   Anton Krall
• [Asterisk-Users] Newbie Question on how to handle main office number   Nick Teagle
• [Asterisk-Users] ZAP channel hangs up with no apparent reason   Ezabi
• [Asterisk-Users] DS3000P - 20 E1 capacity on single card   Andrew Kohlsmith
• [Asterisk-Users] snom and "hint" priority   Josh Dady
• [Asterisk-Users] Zaptel and Fritz Card   Robson Ribeiro
• [Asterisk-Users] 3-Way Calling in Asterisk   Wiley Siler
• [Asterisk-Users]Unable to register license for G729 codec   Jon Lewis
• [Asterisk-Users] Newbie Question on how to handle main office number   Wiley Siler
• [Asterisk-Users] DS3000P - 20 E1 capacity on single card   Steve Underwood
• [Asterisk-Users] CVS-HEAD Zaptel with 1.0.x CVS Asterisk   Eric Wieling
• [Asterisk-Users] OH323 and Asterisk CVS-HEAD-03/21/05-15:32:10   Jose R. Ortiz Ubarri
• [Asterisk-Users]Unable to register license for G729 codec   Daniel Eboa
• [Asterisk-Users] Newbie Question on how to handle main office number   Kerry Garrison
• [Asterisk-Users] x-ten lite error   Robert Keller
• [Asterisk-Users] 3-Way Calling in Asterisk   Andre Normandin
• [Asterisk-Users] Pretty Voicemail Docs   Eric Wieling
• [Asterisk-Users] VAD/DTX implementation through zaptel cards   Eric Wieling
• [Asterisk-Users] ZAP channel hangs up with no apparent reason   Andrew Kohlsmith
• [Asterisk-Users] PRI Errors with TE110P   Eric Wieling
• [Asterisk-Users] IAXy Provision   Wiley Siler
• [Asterisk-Users] SNOM 220 with >7 "lines"   Nils Ohlmeier
• [Asterisk-Users] SIP ACD system for station to station calls   Colin Stefani
• [Asterisk-Users] Local Echo   Andrew Kohlsmith
• [Asterisk-Users] Zaptel and Fritz Card   Peer Oliver Schmidt
• [Asterisk-Users] Attempting native bridge of   snacktime
• [Asterisk-Users] Busy line status and chan_capi?   Damian Funnell
• [Asterisk-Users] 3-Way Calling in Asterisk   Eric Wieling
• [Asterisk-Users] Sipura SPA-841 auto-answer support [patch]   Sebastian Atala
• [Asterisk-Users] binding Asterisk to virtual IP   snacktime
• [Asterisk-Users] ISDN Fritz and TDM400   Damian Funnell
• [Asterisk-Users] OH323 and Asterisk CVS-HEAD-03/21/05-15:32:10   Jose R. Ortiz Ubarri
• [Asterisk-Users] Network Audit Tools   Brian Watters
• [Asterisk-Users] Newbie Question on how to handle main office number   Nick Teagle
• [Asterisk-Users] OH323 and Asterisk CVS-HEAD-03/21/05-15:32:10   Jose R. Ortiz Ubarri
• [Asterisk-Users] BroadVoice Inbound Not working ..   Brian Watters
• [Asterisk-Users] Asterisk on debian sarge doesn't start with CAPI module errors   Morris, Simon
• [Asterisk-Users] Pretty Voicemail Docs   Wiley Siler
• [Asterisk-Users] 3-Way Calling in Asterisk   Anton Krall
• [Asterisk-Users] 3-Way Calling in Asterisk   Wiley Siler
• [Asterisk-Users] sip reload or realtime   doug at simflex.com
• [Asterisk-Users] Verizon ISDN   Brian G
• [Asterisk-Users] Any success with BRI in the US?   Brian G
• [Asterisk-Users] polycom dial...rings 4 ever, but redial connects   Allen Niven
• [Asterisk-Users] Asterisk on debian sarge doesn't start with CAPImodule errors   Gregory Wiktor - ADCom Corp.
• [Asterisk-Users] Pretty Voicemail Docs   Jeffrey C. Ollie
• [Asterisk-Users] Oh323 & AAH 0.8   Oswaldo Arratia
• [Asterisk-Users] IAX introducing huge latency   chawki hammoud
• [Asterisk-Users] Re: Problem reading digits from OH323 caller   Joe S
• [Asterisk-Users] ISDN Fritz and TDM400   Michael Bielicki
• [Asterisk-Users] ZAP channel hangs up with no apparent reason   Moises Silva
• [Asterisk-Users] Asterisk crashing? (gdb trace included)   Mike Dent
• [Asterisk-Users] Line Noise HELP!   Rich Adamson
• [Asterisk-Users] IAXy Provision   Colin Anderson
• [Asterisk-Users] Clipcomm CG-410 with asterisk?   Joel Duffield
• [Asterisk-Users] need to ask you about your dlink and NAT/voip   mylist at lightwavetech.com
• [Asterisk-Users] SIP registration fails   Kanuri, Seshu (Company IT)
• [Asterisk-Users] SIP ACD system for station to station calls   dean collins
• [Asterisk-Users]Unable to register license for G729 codec   Kanuri, Seshu (Company IT)
• [Asterisk-Users] SPA-3000 and quiet voicemail   John Goerzen
• [Asterisk-Users] Pretty Voicemail Docs   Brian McSpadden
• [Asterisk-Users] IAXy Provision   Time Bandit
• [Asterisk-Users] asterisk from cvs head crashes on via samuel 2, kernel 2.6.11-gentoo-r4   jeff oconnell
• [Asterisk-Users] VOIP Regulations in INDIA   Vikram Rangnekar
• [Asterisk-Users] IAX introducing huge latency   steve at daviesfam.org
• [Asterisk-Users] RE: Ebay listing selling Asterisk @ Home andAMPfor over 1000 dollars   Bashir Ullah - www.Lamsre.Com
• [Asterisk-Users] Hardware dimesioning issues   Cameron Beattie
• [Asterisk-Users] sip phone extensions at a remote site   Cameron Beattie
• [Asterisk-Users] 3-Way Calling in Asterisk   Giovanni Powell
• [Asterisk-Users] AstCC Problems   Dave Kettmann
• [Asterisk-Users] sip reload or realtime   Matthew Boehm
• [Asterisk-Users] Grandstream Won't hangup like Polycom 600 will   George Burt
• [Asterisk-Users] IAX introducing huge latency   Andrew Kohlsmith
• [Asterisk-Users] S100I - competitive price?   Kevin P. Fleming
• [Asterisk-Users] DS3000P - 20 E1 capacity on single card   Andrew Latham
• [Asterisk-Users] Transferring a call   Brian Leyton
• AW: [Asterisk-Users] SIP registration fails   William Marks
• [Asterisk-Users] Loop Detection   Daniel Corbe
• [Asterisk-Users] PSTN VOIP integration not allowed in INDIA   Vikram Rangnekar
• [Asterisk-Users] asterisk from cvs head crashes on via samuel 2, kernel 2.6.11-gentoo-r4   Bob Goddard
• [Asterisk-Users] IAXy Provision   Wiley Siler
• [Asterisk-Users] sip reload or realtime   doug at simflex.com
• [Asterisk-Users] S100I - competitive price?   Kanuri, Seshu (Company IT)
• [Asterisk-Users] TDM card periodic buzz   Trent Tuggle
• [Asterisk-Users] asterisk from cvs head crashes on via samuel 2, kernel 2.6.11-gentoo-r4   jeff oconnell
• [Asterisk-Users] Why does this Macro Loop?   Mystery Glitch
• [Asterisk-Users] VOIP Regulations in INDIA   Kanuri, Seshu (Company IT)
• [Asterisk-Users] Using zap channels for fax   Cameron Beattie
• [Asterisk-Users] Choppy music on hold   Joel Duffield
• [Asterisk-Users] Cann't get CallerID on Zap channel, Please Help!!   Cameron Beattie
• [Asterisk-Users] Advice sought on how to automatically and safely reboot * box   mchapman2 at carolina.rr.com
• [Asterisk-Users] ISDN Fritz and TDM400   Robson Ribeiro
• [Asterisk-Users] TDM400P Revision question.   Robert Webb
• [Asterisk-Users] Asterisk@Home 0.9 released   Asterisk at Home
• [Asterisk-Users] Verizon ISDN   Gregory Wiktor - ADCom Corp.
• [Asterisk-Users] Grandstream Won't hangup like Polycom 600 will   Karl A. Krueger
• [Asterisk-Users] Asterisk@Home 0.9 released   Asterisk at Home
• [Asterisk-Users] S100I - competitive price?   Matthew Boehm
• [Asterisk-Users] UK PSTN Calling From OH323 Problem   Cameron Beattie
• [Asterisk-Users] sip reload or realtime   Matthew Boehm
• [Asterisk-Users] Advice sought on how to automatically and sa fely reboot * box   David Brodbeck
• [Asterisk-Users] Why does this Macro Loop?   Andrew Kohlsmith
• [Asterisk-Users] SIP Clients over Wan losing connection   Dan Levine
• [Asterisk-Users] TDM card periodic buzz   Andrew Kohlsmith
• [Asterisk-Users] Advice sought on how to automatically and sa fely reboot * box   Colin Anderson
• [Asterisk-Users] Local Echo   Johann
• [Asterisk-Users] DISA() and predefined ACCOUNTCODE variable   Michael George
• [Asterisk-Users] Re: Fax to Email   Cameron Beattie
• [Asterisk-Users] Re: Why does this Macro Loop?   Tony Mountifield
• [Asterisk-Users] weird call transfer problem   C F
• [Asterisk-Users] VOIP Regulations in INDIA   Dan Austin
• [Asterisk-Users] Help, Outbound Problems   Linn Boyd
• [Asterisk-Users] PSTN VOIP integration not allowed in INDIA   Dan Austin
• [Asterisk-Users] ztdummy   Brian Leyton
• [Asterisk-Users] (no subject)   Sascha Ferley
• [Asterisk-Users] Asterisk@Home 0.9 released   Wiley Siler
• [Asterisk-Users] Polycom Vendor Recommendation   Wiley Siler
• [Asterisk-Users] VOIP Regulations in INDIA   Brian Capouch
• [Asterisk-Users] Re: Why does this Macro Loop?   Tony Mountifield
• [Asterisk-Users] about sip and skinny   Andrea Riela
• [Asterisk-Users] BroadVoice Inbound Not working ..   MF Hulber
• [Asterisk-Users] Asterisk@Home 0.9 released   David John Walsh
• [Asterisk-Users] several intances of asterisk launched   Bertrand
• [Asterisk-Users] OH323 and Asterisk CVS-HEAD-03/21/05-15:32:10   Rod Bacon
• [Asterisk-Users] VOIP Regulations in INDIA   dean collins
• [Asterisk-Users] connecting Asterisk as SIP gateway to a Verso BHT1000   MobilPete
• [Asterisk-Users] DS3000P - 16 E1 capacity on single card   Kevin P. Fleming
• [Asterisk-Users] Polycom Vendor Recommendation   MobilPete
• [Asterisk-Users] Connection to SIP Gateway   Cameron Beattie
• [Asterisk-Users] Low cost box for hosting Asterisk and atleastoneTDM400p - THIN CLIENT MAYBE?   Cameron Beattie
• [Asterisk-Users] trying to figure out a few error messages in *   Brian Chrystal
• [Asterisk-Users] ztdummy   Asterisk at Home
• [Asterisk-Users] Asterisk@Home 0.9 released   Asterisk at Home
• [Asterisk-Users] Asterisk@Home 0.9 released   Asterisk at Home
• [Asterisk-Users] trying to figure out a few error messages in *   Brian Chrystal
• [Asterisk-Users] does meetme need ztdummy   Xu Wang
• [Asterisk-Users] does meetme need ztdummy   BJ Weschke
• [Asterisk-Users] does meetme need ztdummy   Rod Bacon
• [Asterisk-Users] OH323 and Asterisk CVS-HEAD-03/21/05-15:32:10   Rod Bacon
• [Asterisk-Users] Telephone line installation.   Manjit Riat
• [Asterisk-Users] Telephone line installation.   C F
• [Asterisk-Users] H.323 in CVS Head   Rod Bacon
• [Asterisk-Users] ZyXEL Router Terrible Voice Quality   aza
• [Asterisk-Users] SIP Deadlock problem.   Eric Rees
• [Asterisk-Users] show translation   Anton Krall
• [Asterisk-Users] S100I - competitive price?   Kevin P. Fleming
• [Asterisk-Users] Cannot dial two phones at the same time   Eddie
• [Asterisk-Users] Cannot dial two phones at the same time   Eddie
• [Asterisk-Users] ZyXEL Router Terrible Voice Quality   Rod Bacon
• [Asterisk-Users] bashing my head against broadvoice   Mark Phillips
• [Asterisk-Users] New Zealand Telco (TelstraClear) query   Chris Hodgetts
• [Asterisk-Users] Telephone line installation.   Manjit Riat
• [Asterisk-Users] Cannot dial two phones at the same time   Eddie
• [Asterisk-Users] bashing my head against broadvoice   Kevin Bockman
• [Asterisk-Users] Telephone line installation.   Scott Stingel
• [Asterisk-Users] DS3000P - 20 E1 capacity on single card   tmassey at obscorp.com
• [Asterisk-Users] Telephone line installation.   Brian Leyton
• [Asterisk-Users] Telephone line installation.   Gregory Junker
• [Asterisk-Users] ztdummy   Paul Fielding
• [Asterisk-Users] Low cost box for hosting Asterisk and atleastoneTDM400p - THIN CLIENT MAYBE?   Robert Augustyn
• [Asterisk-Users] New Zealand Telco (TelstraClear) query   Mike Price
• [Asterisk-Users] cannot dial two phones using zap   Eddie
• [Asterisk-Users] Re: PSTN VOIP integration not allowed in INDIA   Vikram Rangnekar
• [Asterisk-Users] Changing IRQ's on TDM   Damian Funnell
• [Asterisk-Users] Channel 0 on Zap ???   Tim Connolly
• [Asterisk-Users] Re: iaxcomm   amna saleem
• [Asterisk-Users] Channel 0 on Zap ???   Rod Bacon
• [Asterisk-Users] ztdummy   Rod Bacon
• [Asterisk-Users] ztdummy   Paul Hales
• [Asterisk-Users] oh-323 compilation error !   urban viking
• [Asterisk-Users] Re: Running asterisk without special hardwar e   Brian Leyton
• [Asterisk-Users] Re: Running asterisk without special hardware   Manish Sapariya
• [Asterisk-Users] Re: Fax to Email   Trent Tuggle
• [Asterisk-Users] Polycom Vendor Recommendation   Wiley Siler
• [Asterisk-Users] Asterisk@Home 0.9 released   Wiley Siler
• [Asterisk-Users] RTP not being sent by asterisk   trixter http://www.0xdecafbad.com
• [Asterisk-Users] RTP not being sent by asterisk   Rod Bacon
• [Asterisk-Users] RTP not being sent by asterisk   Rod Bacon
• [Asterisk-Users] RTP not being sent by asterisk   trixter http://www.0xdecafbad.com
• [Asterisk-Users] RTP not being sent by asterisk   trixter http://www.0xdecafbad.com
• [Asterisk-Users] RTP not being sent by asterisk   trixter http://www.0xdecafbad.com
• [Asterisk-Users] RTP not being sent by asterisk   Rod Bacon
• [Asterisk-Users] Strange intermittent NAT problem with BT100s   Tomas Florian
• [Asterisk-Users] trying the xc-ast queue_log analyzer   lenz
• [Asterisk-Users] RTP problem   trixter http://www.0xdecafbad.com
• [Asterisk-Users] Cisco 7940G SIP Conversion   Michael West
• [Asterisk-Users] oh-323 compilation error !   Nardis Dome
• [Asterisk-Users] Problem compiling 2nd AVM Fritz   Robson Ribeiro
• [Asterisk-Users] Cisco 7960 command-line dialer   Nabeel Jafferali
• [Asterisk-Users] Busy line status and chan_capi?   Kib Eki
• [Asterisk-Users] Busy line status and chan_capi?   Elmar Haneke
• [Asterisk-Users] Cisco 7940G SIP Conversion   Simon Morris
• [Asterisk-Users] need help   amna saleem
• [Asterisk-Users] pbx to asterisk   Altus Snyman
• [Asterisk-Users] Busy line status and chan_capi?   Kib Eki
• [Asterisk-Users] pbx to asterisk   steve at daviesfam.org
• [Asterisk-Users] Article on IAX in Network World   tim panton
• [Asterisk-Users] Hylafax and Asterisk   Kib Eki
• [Asterisk-Users] ztdummy   Tzafrir Cohen
• [Asterisk-Users] Re: Cisco 7960 command-line dialer   Mick Hastings
• [Asterisk-Users] need help   amna saleem
• [Asterisk-Users] Re: SIP Deadlock problem.   Mick Hastings
• [Asterisk-Users] need urgent help   amna saleem
• [Asterisk-Users] need help   Zoa
• [Asterisk-Users] Is there a SIP protocol stack inside asterisk?   Abraham WEI
• [Asterisk-Users] RealTime   Me
• [Asterisk-Users] IAX blind transfers   Paul Seymour
• [Asterisk-Users] voicetronix bri   Altus Snyman
• [Asterisk-Users] Registration   Rene Pavelko
• [Asterisk-Users] voicetronix bri   Michael Bielicki
• [Asterisk-Users] Is there a SIP protocol stack inside asterisk?   steve at daviesfam.org
• how to ask for help [was: Re: [Asterisk-Users] need urgent help]   Tzafrir Cohen
• [Asterisk-Users] RealTime   G.Marshall
• [Asterisk-Users] IAX blind transfers   steve at daviesfam.org
• [Asterisk-Users] voicetronix bri   Altus Snyman
• [Asterisk-Users] <register> syntax and limitation therewith   mailing
• [Asterisk-Users] IAX blind transfers   Michael Puchol
• [Asterisk-Users] voicetronix bri   steve at daviesfam.org
• [Asterisk-Users] lost DTMF digits   David Farrant
• [Asterisk-Users] Dropped calls from Junghans octo-bri card   Nelson
• [Asterisk-Users] Cannot dial two phones at the same time   Michael George
• [Asterisk-Users] Busy line status and chan_capi?   Elmar Haneke
• [Asterisk-Users] Please make sure there is subject in mails   ht at phonitel.com
• [Asterisk-Users] ISDN BRI + echo cancelling + Fax   Remco Barende
• [Asterisk-Users] ISDN BRI + echo cancelling + Fax   Michael Bielicki
• [Asterisk-Users] IAX blind transfers   Paul Seymour
• [Asterisk-Users] no voice tone   Diaa Fawzy
• [Asterisk-Users] Re: ISDN BRI + echo cancelling + Fax   Edwin Groothuis
• [Asterisk-Users] (no subject)   Rich Adamson
• [Asterisk-Users] Re: ISDN BRI + echo cancelling + Fax   Michael Bielicki
• [Asterisk-Users] IAX blind transfers   steve at daviesfam.org
• [Asterisk-Users] cannot dial two phones using zap   Rob Scott
• [Asterisk-Users] Re: ISDN BRI + echo cancelling + Fax   Remco Barende
• [Asterisk-Users] Dialing rules   Robert P. McKenzie
• [Asterisk-Users] Hylafax and Asterisk   Frank Becker
• [Asterisk-Users] IAX blind transfers   steve at daviesfam.org
• [Asterisk-Users] Re: Asterisk-Users Sip Reload or Realtime   yusuf
• [Asterisk-Users] Hylafax and Asterisk   Peer Oliver Schmidt
• [Asterisk-Users] Why does this Macro Loop?   Eric Wieling
• [Asterisk-Users] RTP not being sent by asterisk   Alex Vishnev
• [Asterisk-Users] Asterisk@home first experience   Bruno Quintas
• [Asterisk-Users] Strange intermittent NAT problem with BT100s   Alex Vishnev
• [Asterisk-Users] sip phones make connection but no-sound is heared   me me
• [Asterisk-Users] trying the xc-ast queue_log analyzer   Alex Vishnev
• [Asterisk-Users] Telephone line installation.   Rich Adamson
• [Asterisk-Users] RTP problem   Eric Wieling
• [Asterisk-Users] BOUNTY - ztdummy & modules   Eric Wieling
• [Asterisk-Users] BOUNTY: app_hangup from exten => h   Eric Wieling
• [Asterisk-Users] ACCOUNTCODE lost after DISA()   Michael George
• [Asterisk-Users] BOUNTY - ztdummy & modules   Tzafrir Cohen
• [Asterisk-Users] Re: Running asterisk without special hardware   Bruno Hertz
• [Asterisk-Users] Asterisk@Home 0.9 released   Time Bandit
• [Asterisk-Users] SIP Incoming Problem   Ben Price
• [Asterisk-Users] About Audio Latency from PSTN to SIP   chawki hammoud
• [Asterisk-Users] SIP Incoming Problem   dean collins
• [Asterisk-Users] SPA-3000 and quiet voicemail   Rich Adamson
• [Asterisk-Users] BOUNTY: app_hangup from exten => h   Andrew Kohlsmith
• [Asterisk-Users] IPSwitchBoard Version 0.86 Released   Thorben Jensen
• [Asterisk-Users] BOUNTY: app_hangup from exten => h   Eric Wieling
• [Asterisk-Users] delay problem in asterisk   wassim darwish
• [Asterisk-Users] TEST   Julio Tejera
• [Asterisk-Users] BOUNTY: app_hangup from exten => h   Andrew Kohlsmith
• [Asterisk-Users] Changing IRQ's on TDM   Andrew Kohlsmith
• [Asterisk-Users] MoH stopped working with cisco 7912/7960   Simone Cittadini
• [Asterisk-Users] BOUNTY: app_hangup from exten => h   Eric Wieling
• [Asterisk-Users] Ring two extensions at the same time   G.Marshall
• [Asterisk-Users] Re: BOUNTY: app_hangup from exten => h   Tony Mountifield
• [Asterisk-Users] TDM400P Revision question.   Rich Adamson
• [Asterisk-Users] Cisco 7960s and skinny   asterisk_on_oelf
• [Asterisk-Users] Asterisk@Home 0.9 released   Asterisk at Home
• [Asterisk-Users] Ring two extensions at the same time   Sean Kennedy
• [Asterisk-Users] Steal a call from a SIP extension   Sean Kennedy
• [Asterisk-Users] cisco 79xx and SIP call statistics   Jeremy Hinton
• [Asterisk-Users] Segregating a test version of asterisk - libpri/zaptel locations   Julien Levi
• [Asterisk-Users] Zap won't dial out?   Tim Connolly
• [Asterisk-Users] Who is a QUALITY IAX Termination Provider for 800 DID's?   Linn Boyd
• [Asterisk-Users] TDM400P Revision question.   Robert Webb
• [Asterisk-Users] Ring two extensions at the same time   Julio Tejera
• [Asterisk-Users] Ring two extensions at the same time   Robert Webb
• [Asterisk-Users] Changing IRQ's on TDM   Rich Adamson
• [Asterisk-Users] Re: Fax to Email   Rich Adamson
• [Asterisk-Users] Steal a call from a SIP extension   Josiah Bryan
• [Asterisk-Users] Zap won't dial out?   Tim Connolly
• [Asterisk-Users] Zap won't dial out?   Eric Wieling
• [Asterisk-Users] Voicemail Email   Chris
• [Asterisk-Users] Who is a QUALITY IAX Termination Provider for 800 DID's?   Paul
• [Asterisk-Users] Polycom IP500 phones and Presence feature   jjara at e-contact.cl
• [Asterisk-Users] Re: Why does this Macro Loop?   steve at daviesfam.org
• [Asterisk-Users] Toshiba CTX100 integration with PABX for two site   Stephen
• [Asterisk-Users] Voicemail Email   JD Austin
• [Asterisk-Users] Zap won't dial out?   Tim Connolly
• [Asterisk-Users] Hylafax and Asterisk   Frank Becker
• [Asterisk-Users] Re: Why does this Macro Loop?   Tony Mountifield
• [Asterisk-Users] Ring two extensions at the same time   Damian Funnell
• [Asterisk-Users] RTP problem   trixter http://www.0xdecafbad.com
• [Asterisk-Users] G.729A codec amd64/intel x86-64 optimisation?   Roy Sigurd Karlsbakk
• [Asterisk-Users] Matching on '@' in extensions   Ronan Mullally
• [Asterisk-Users] Changing IRQ's on TDM   Andrew Kohlsmith
• [Asterisk-Users] Siemens optiPoint 420 phone and Asterisk   Franz Knipp
• [Asterisk-Users] Zap won't dial out?   Tim Connolly
• [Asterisk-Users] Polycom IP500 phones do not update time from time server   Kanuri, Seshu (Company IT)
• [Asterisk-Users] Registration   Kanuri, Seshu (Company IT)
• [Asterisk-Users] TDM400P Revision question.   Robert Webb
• [Asterisk-Users] asterisk hosting   snacktime
• [Asterisk-Users] Wall Mount PC Case   Wiley Siler
• [Asterisk-Users] TDM400P Revision question.   Ian Pattison
• [Asterisk-Users] S100I - competitive price?   Matthew Boehm
• [Asterisk-Users] Re: Polycom IP500 phones and Presence feature   Noah Miller
• [Asterisk-Users] Changing IRQ's on TDM   Rich Adamson
• [Asterisk-Users] Re: Asterisk-Users Sip Reload or Realtime   Matthew Boehm
• [Asterisk-Users] Call Files to Terminate a call to the dialplan not directly to a channel   Mystery Glitch
• [Asterisk-Users] Polycom IP500 phones do not update time from time server   Steve Clark
• [Asterisk-Users] TDM400P Revision question.   Rich Adamson
• [Asterisk-Users] TDM400P Revision question.   Robert Webb
• [Asterisk-Users] RealTime   Matthew Boehm
• [Asterisk-Users] Voicemail name (greet.wav) is not played   bam
• [Asterisk-Users] Re: Polycom IP500 phones do not update time from   Noah Miller
• [Asterisk-Users] TDM400P Revision question.   Andrew Kohlsmith
• [Asterisk-Users] Voicemail name (greet.wav) is not played   Roderick A. Anderson
• [Asterisk-Users] trying the xc-ast queue_log analyzer   lenz
• [Asterisk-Users] Re: Fax to Email   Richard Lyman
• [Asterisk-Users] trying the xc-ast queue_log analyzer   Richard Lyman
• [Asterisk-Users] SPA-3000 and quiet voicemail   Shadow Roldan
• [Asterisk-Users] Re: Voicemail name (greet.wav) is not played   bam
• [Asterisk-Users] Telephone line installation.   Manjit Riat
• [Asterisk-Users] Re: Voicemail name (greet.wav) is not played   Joel Duffield
• [Asterisk-Users] Voicemail name (greet.wav) is not played   Trevor Peirce
• [Asterisk-Users] Who is a QUALITY IAX Termination Provider for 800 DID's?   snacktime
• [Asterisk-Users] Ring two extensions at the same time   G.Marshall
• [Asterisk-Users] Toshiba CTX100 integration with PABX for two site   Henry Devito
• [Asterisk-Users] Siemens optiPoint 420 phone and Asterisk   Chris
• [Asterisk-Users] Polycom IP500 phones do not update time from time server   Sean Kennedy
• [Asterisk-Users] does meetme need ztdummy   Daniel Bruce Lynes
• [Asterisk-Users] making an action based on the status of multiple extensions   Jon Gabrielson
• [Asterisk-Users] Line Presence:   Sean Kennedy
• [Asterisk-Users] Telephone line installation.   Daniel Bruce Lynes
• [Asterisk-Users] Line Presence:   Henry Devito
• [Asterisk-Users] Line Presence:   Robert Webb
• [Asterisk-Users] making an action based on the status of multiple extensions   C F
• [Asterisk-Users] delay problem in asterisk   Andy Hamilton
• [Asterisk-Users] Invalid extension handling   Adam Robins
• [Asterisk-Users] Hylafax and Asterisk   Michiel van Baak
• [Asterisk-Users] Overheard conversation. Comments please !   Asterisk
• [Asterisk-Users] Overheard conversation. Comments please !   Damon Estep
• [Asterisk-Users] Overheard conversation. Comments please !   Wiley Siler
• [Asterisk-Users] PRI Advice...   Michael Crozier
• [Asterisk-Users] Overheard conversation. Comments please !   Mike Benoit
• [Asterisk-Users] making an action based on the status of multiple extensions   Jon Gabrielson
• [Asterisk-Users] Overheard conversation. Comments please !   Asterisk
• [Asterisk-Users] Overheard conversation. Comments please !   Asterisk
• [Asterisk-Users] Fritz Card going Crazy to make it compile   Robson Ribeiro
• [Asterisk-Users] Voicemail name (greet.wav) is not played   Asterisk
• [Asterisk-Users] Overheard conversation. Comments please !   Wiley Siler
• [Asterisk-Users] Overheard conversation. Comments please !   Patrick May
• [Asterisk-Users] Voicemail delivery to pbx or mobile/panasonic dbs   Gregory Wiktor - ADCom Corp.
• [Asterisk-Users] Telephone line installation.   Gregory Wiktor - ADCom Corp.
• [Asterisk-Users] Line Presence:   Gregory Wiktor - ADCom Corp.
• [Asterisk-Users] ZAP channel hangs up with no apparent reason   Ezabi
• [Asterisk-Users] sip phones make connection but no-sound is heared   Giovanni Powell
• [Asterisk-Users] Overheard conversation. Comments please !   Damon Estep
• [Asterisk-Users] making an action based on the status of multiple extensions   C F
• [Asterisk-Users] Overheard conversation. Comments please !   Damon Estep
• [Asterisk-Users] Invalid extension handling   Kanuri, Seshu (Company IT)
• [Asterisk-Users] Overheard conversation. Comments please !   Wiley Siler
• [Asterisk-Users] Call Parking timming out to the wrong extension   Eric Wieling
• [Asterisk-Users] Overheard conversation. Comments please !   Andrew Kohlsmith
• [Asterisk-Users] Verizon ISDN   Brian G
• [Asterisk-Users] Custom/Vanity DIDs   * KAPIL *
• [Asterisk-Users] Polycom IP500 phones do not update time from time server   Dylan VanHerpen
• [Asterisk-Users] MFCR2 compile requirements   Titux
• [Asterisk-Users] Toshiba CTX100 integration with PABX for two site   Brian Leyton
• [Asterisk-Users] Re: MFCR2 compile requirements   Titux
• [Asterisk-Users] Routing on called number via SIP   Kevin Wormington
• [Asterisk-Users] Bizarre - VM just stopped for one user   Wiley Siler
• [Asterisk-Users] Problem with Livevoip incoming context   Chris Mason (Lists)
• [Asterisk-Users] How to reduce asterisk CPU-LOAD?   niels at wxn.nl
• [Asterisk-Users] trying the xc-ast queue_log analyzer   lenz
• [Asterisk-Users] Overheard conversation. Comments please !   Asterisk
• [Asterisk-Users] trying the xc-ast queue_log analyzer   lenz
• [Asterisk-Users] Overheard conversation. Comments please !   Asterisk
• [Asterisk-Users] Toshiba CTX100 integration with PABX for two site   Brian Leyton
• [Asterisk-Users] dial plan   Michael Di Martino
• [Asterisk-Users] Line Presence:   Brian Leyton
• [Asterisk-Users] Problem with Livevoip incoming context   Wiley Siler
• [Asterisk-Users] Overheard conversation. Comments please !   Wiley Siler
• [Asterisk-Users] asterisk + OH323 + NAT + gnomemeeting   Jesse Guardiani
• [Asterisk-Users] dial plan   Wiley Siler
• [Asterisk-Users] dial plan   Andrew Kohlsmith
• [Asterisk-Users] I dont want to hear the FXS port ring - TDM400?   Scott Wolfe
• [Asterisk-Users] codec introducing huge latency   chawki hammoud
• [Asterisk-Users] Realtime Friends   Rod Bacon
• [Asterisk-Users] Dial Macro Arguments   Shaun Tierney
• [Asterisk-Users] Telephone line installation.   Gregory Junker
• [Asterisk-Users] ISDN BRI and signalling   Bob van der Moezel
• [Asterisk-Users] Hylafax and Asterisk   Andrew Yager
• [Asterisk-Users] codec introducing huge latency   Andrew Kohlsmith
• [Asterisk-Users] Dial Macro Arguments   C F
• [Asterisk-Users] Toshiba CTX100 integration with PABX for two site   Henry Devito
• [Asterisk-Users] Toshiba CTX100 integration with PABX for two site   Henry Devito
• [Asterisk-Users] Asterisk@home first experience   Andy Hamilton
• [Asterisk-Users] Toshiba CTX100 integration with PABX for two site   Henry Devito
• [Asterisk-Users] Fax questions   Ronald Wiplinger
• [Asterisk-Users] Stop this I'm trying to help you.(Fwd: Please confirm your message)   C F
• [Asterisk-Users] I dont want to hear the FXS port ring - TDM400?   Henry Devito
• [Asterisk-Users] Toshiba CTX100 integration with PABX for two site   Brian Leyton
• [Asterisk-Users] Cant respond to prompts from SPA1001   Theodore Cekan
• [Asterisk-Users] DTMF does not work with g729 and AGI   ht at phonitel.com
• [Asterisk-Users] Wall Mount PC Case   Paul Hales
• [Asterisk-Users] New Zealand Telco (TelstraClear) query   Matt Riddell
• [Asterisk-Users] Polycom IP500 phones do not update time from time server   Russ Beaupre, P.E.
• [Asterisk-Users] Toshiba CTX100 integration with PABX for two site   Henry Devito
• [Asterisk-Users] Stop this I'm trying to help you.(Fwd: Please confirm your message)   Jon Gabrielson
• [Asterisk-Users] cisco 7960 SIP setup   mk111
• [Asterisk-Users] Mark Spencer and John "Maddog" Hall visiting Toronto - come and join us   Jim Van Meggelen
• [Asterisk-Users] How do I make a call thru *PBX   Kumara Jayaweera
• [Asterisk-Users] How do I make a call thru *PBX   Kerry Garrison
• [Asterisk-Users] About Audio Latency from PSTN to SIP   Qiao Yuansong
• [Asterisk-Users] DID reseller structures   snacktime
• [Asterisk-Users] Problem with Livevoip incoming context   Chris Mason
• [Asterisk-Users] Asterisk became berserk when Internet connection is down and can't register to SIP server.   Kong
• [Asterisk-Users] About Audio Latency from PSTN to SIP   Andrew Kohlsmith
• [Asterisk-Users] Stop this I'm trying to help you.(Fwd: Please confirm your message)   Ronald Wiplinger
• [Asterisk-Users] DID reseller structures   trixter http://www.0xdecafbad.com
• [Asterisk-Users] Stop this I'm trying to help you.(Fwd: Please confirm your message)   Ronald Wiplinger
• [Asterisk-Users] DID reseller structures   snacktime
• [Asterisk-Users] cisco 7960 SIP setup   Andy Hamilton
• [Asterisk-Users] Voicemail Email   Tzafrir Cohen
• [Asterisk-Users] Stop this I'm trying to help you.(Fwd: Please confirm your message)   Daniel Bruce Lynes
• [Asterisk-Users] Asterisk became berserk when Internet connection is down and can't register to SIP server.   snacktime
• [Asterisk-Users] dropping inbound calls from certain regions   snacktime
• [Asterisk-Users] cannot dial two phones using zap   Eddie
• [Asterisk-Users] Asterisk became berserk when Internet connection is down and can't register to SIP server.   Kong
• [Asterisk-Users] DID reseller structures   trixter http://www.0xdecafbad.com
• [Asterisk-Users] Polycom IP500 phones do not update time from time server   Eric Wieling
• [Asterisk-Users] Invalid extension handling   Eric Wieling
• [Asterisk-Users] PRI Advice...   Eric Wieling
• [Asterisk-Users] Segregating a test version of asterisk - libpri/zaptel locations   Tzafrir Cohen
• [Asterisk-Users] distribute outbound calls   snacktime
• [Asterisk-Users] DID reseller structures   Michael D Schelin
• [Asterisk-Users] distribute outbound calls   Damian Funnell
• [Asterisk-Users] Stop this I'm trying to help you.(Fwd: Please confirm your message)   Ronald Wiplinger
• [Asterisk-Users] distribute outbound calls   William Suffill
• [Asterisk-Users] distribute outbound calls   jltaylor
• [Asterisk-Users] does meetme need ztdummy   Tzafrir Cohen
• [Asterisk-Users] S100I - competitive price?   Kevin P. Fleming
• [Asterisk-Users] Stop this I'm trying to help you.(Fwd: Please confirm your message)   William Suffill
• [Asterisk-Users] DID reseller structures   trixter http://www.0xdecafbad.com
• [Asterisk-Users] Asterisk behind NAT   Oswaldo Arratia
• [Asterisk-Users] Line Presence:   Paradise Dove
• [Asterisk-Users] Overheard conversation. Comments please !   lenz
• [Asterisk-Users] Has anyone had problems with Digium TDM400P and hyperthreading?   Damian Funnell
• [Asterisk-Users] Asterisk and CAS   Paradise Dove
• [Asterisk-Users] Grandstream BT Volume   Anton Krall
• [Asterisk-Users] ISDN BRI and signalling   Peter Svensson
• [Asterisk-Users] OctoBRI - unable to specify channel 1   Terry Wade
• [Asterisk-Users] Has anyone had problems with Digium TDM400P and hyperthreading?   Paul Hewlett
• [Asterisk-Users] Pre-install questions   Cameron Beattie
• [Asterisk-Users] Siemens optiPoint 420 phone and Asterisk   richard Coco
• [Asterisk-Users] A-Z Termination   Mehdi Chouikh
• [Asterisk-Users] cisco 7960 SIP setup   Simone Cittadini
• [Asterisk-Users] Soekris net4801 usb isdn avm fritz   asterisk
• [Asterisk-Users] About Audio Latency from PSTN to SIP   Qiao Yuansong
• [Asterisk-Users] codec introducing huge latency   chawki hammoud
• [Asterisk-Users] Asterisk PBX with X100P in India   Min Hwan Chang
• [Asterisk-Users] OH323 and outgoing calls problem.   Adam Rybak
• [Asterisk-Users] ISDN BRI and signalling   Eugenio De Vena
• [Asterisk-Users] E1 PRI: Unable to set channel to linear mode?   Derek Conniffe
• [Asterisk-Users] UDP Sip Data: GS Grandstream - remote office   Etienne Pretorius
• [Asterisk-Users] *8 nor *8# works for me!   Ronald Wiplinger
• [Asterisk-Users] Urgent .... Asterisk <-> Cisco CCM SIP TRUNK   João Amaro
• [Asterisk-Users] qos test   Altus Snyman
• [Asterisk-Users] LiveVoip incoming, no ringback still   David Josephson
• [Asterisk-Users] Empty voicemail attachments?   Andrew C. Brown
• [Asterisk-Users] cannot dial two phones using zap   Michael George
• [Asterisk-Users] Analogue phone transfering   David Wilson
• [Asterisk-Users] Asterisk live chat problem   stefano carlini
• [Asterisk-Users] Asterisk live chat problem   stefano carlini
• [Asterisk-Users] *8 nor *8# works for me!   Chris Mason (Lists)
• [Asterisk-Users] IAX2 to IAX2 - one way audio   Andrew A. Kochetkoff
• [Asterisk-Users] SIP through firewall is intermittent   Chris Mason (Lists)
• [Asterisk-Users] VIC2BRI and J4BRI   Eugenio De Vena
• [Asterisk-Users] SIP stack pluggable?   Vyom A
• [Asterisk-Users] Empty voicemail attachments?   Henry Devito
• [Asterisk-Users] Asterisk live chat problem   betix smart computing - Thomas Novotny
• [Asterisk-Users] Slack 10 install - THANK YOU - & Cisco Reseller Help   Gary Guthary
• [Asterisk-Users] g729 not work with DTMF and AGI   ht at phonitel.com
• [Asterisk-Users] About Audio Latency from PSTN to SIP   Andrew Kohlsmith
• [Asterisk-Users] Problems with a SMS-capable Phone on a ZAP Channel / Question about native bridging on digium cards   Carsten Bock
• [Asterisk-Users] Slack 10 install - THANK YOU - & Cisco ResellerHelp   Henry Devito
• [Asterisk-Users] howto forward UAC codec capabilities to the PSTN gw   Pavel Siderov
• [Asterisk-Users] TDM400P Revision question.   Rich Adamson
• [Asterisk-Users] MoH stopped working with cisco 7912/7960   Simone Cittadini
• [Asterisk-Users] Re: Cisco 7960s and skinny   Sergio
• [Asterisk-Users] Asterisk PBX with X100P in India   Julio Saura
• [Asterisk-Users] Slack 10 install - THANK YOU - & Cisco ResellerHelp   Guillermo Salas M.
• [Asterisk-Users] DTMF does not work with g729 and AGI   Alex Vishnev
• [Asterisk-Users] Excessive re-registration of Broadvoice account in Asterisk@Home 0.8   Wolf N. Paul
• [Asterisk-Users] About Audio Latency from PSTN to SIP   Guillermo Salas M.
• [Asterisk-Users] Changing IRQ's on TDM   Bryan Boatright
• [Asterisk-Users] SIP Message Waiting Notification   Michael B. Murdock
• [Asterisk-Users] Asterisk became berserk when Internet connection is down and can't register to SIP server.   Alex Vishnev
• [Asterisk-Users] Empty voicemail attachments?   Kevin Kiely
• [Asterisk-Users] Asterisk behind NAT   Alex Vishnev
• [Asterisk-Users] TDM400P Revision question.   Eric Wieling
• [Asterisk-Users] Analogue phone transfering   Eric Wieling
• [Asterisk-Users] Debugging zaphfc + PBX integration   Gavin Hamill
• [Asterisk-Users] *8 nor *8# works for me!   Eric Wieling
• [Asterisk-Users] Question on Asterisk CDR / "In-Network Calling" / MySQL CDR   Matt
• [Asterisk-Users] Asterisk working on FC3+X100P+France Telecom line   Bertrand
• [Asterisk-Users] TDM400P Revision question.   Eric Wieling
• [Asterisk-Users] Asterisk behind NAT   Oswaldo Arratia
• [Asterisk-Users] Analogue phone transfering   David Wilson
• [Asterisk-Users] Polycom IP500 phones do not update time fromtime server   Kanuri, Seshu (Company IT)
• [Asterisk-Users] Who is a QUALITY IAX Termination Provider for 800 DID's?   Michael Graves
• [Asterisk-Users] Call Files to Terminate a call to the dialplan not directly to a channel   Julian J. M.
• [Asterisk-Users] Grandstream BT Volume   Anton Krall
• [Asterisk-Users] Lost DTMF digits   David Farrant
• [Asterisk-Users] Asterisk@Home 0.9 released {Scanned}   David Shaw
• [Asterisk-Users] TDM400P Revision question.   Rich Adamson
• [Asterisk-Users] ilbc codec in Asterisk   Stephen
• [Asterisk-Users] Realtime Friends   Matthew Boehm
• [Asterisk-Users] ilbc codec in Asterisk   Gavin Hamill
• [Asterisk-Users] *8 nor *8# works for me!   Ronald Wiplinger
• [Asterisk-Users] Dial Macro Arguments   Shaun Tierney
• [Asterisk-Users] Slack 10 install - THANK YOU - & Cisco Reseller Help   Max W Blackmer Jr
• [Asterisk-Users] sipXphone   Kerry Garrison
• [Asterisk-Users] Ring requested on unconfigured channel 0/31 span 1?   Derek Conniffe
• [Asterisk-Users] No Hope with Fax and X100P?   Chris
• [Asterisk-Users] No Hope with Fax and X100P?   Wojciech Tryc
• [Asterisk-Users] Maybe not worded right, Answering a call   Giovanni Powell
• [Asterisk-Users] Re: Loop Detection   Doug Meredith
• [Asterisk-Users] Bridging 2 Zap channels   Paul Hewlett
• [Asterisk-Users] Re: Fax questions   Jesse Guardiani
• [Asterisk-Users] about volume in Playback() files   Moises Silva
• [Asterisk-Users] Outgoing PRI Call Early Media Detection   Charlie Watts
• [Asterisk-Users] How to avoid CTL file request for Cisco 7970   Parfenenok Sergey
• [Asterisk-Users] cisco 7960 SIP setup   mk111
• [Asterisk-Users] cisco 7960 SIP setup   Steve Blair
• [Asterisk-Users] large analog to asterisk   shane fowler
• [Asterisk-Users] Debugging zaphfc + PBX integration   Tim Robinson
• [Asterisk-Users] Re: Loop Detection   Daniel Corbe
• [Asterisk-Users] Asterisk behind NAT   Alex Vishnev
• [Asterisk-Users] H323 Large Scale   list
• [Asterisk-Users] Debugging zaphfc + PBX integration   Rob Scott
• [Asterisk-Users] Asterisk behind NAT   Jim Sturtevant
• [Asterisk-Users] H323 Large Scale   BJ Weschke
• [Asterisk-Users] large analog to asterisk   Rusty Shackleford
• [Asterisk-Users] Asterisk behind NAT   Oswaldo Arratia
• [Asterisk-Users] Hylafax and Asterisk   Michiel van Baak
• [Asterisk-Users] Asterisk behind NAT   Oswaldo Arratia
• [Asterisk-Users] OT: google groups Asterisk-test and now Asterisk-Users marked as spam on Gmail   Sig Lange
• [Asterisk-Users] H323 Large Scale   Kanuri, Seshu (Company IT)
• [Asterisk-Users] Asterisk behind NAT   Michiel van Baak
• [Asterisk-Users] OT: google groups Asterisk-test and now Asterisk-Users marked as spam on Gmail   Sig Lange
• [Asterisk-Users] OT: USB handsets / softphones   Vahan Yerkanian
• [Asterisk-Users] Has anyone had problems with Digium TDM400P and hyperthreading?   tmassey at obscorp.com
• [Asterisk-Users] large analog to asterisk   William Boehlke
• [Asterisk-Users] OT: USB handsets / softphones   Kerry Garrison
• [Asterisk-Users] OT: google groups Asterisk-test and now Asterisk-Users marked as spam on Gmail   trixter http://www.0xdecafbad.com
• [Asterisk-Users] Asterisk became berserk when Internet connectionis down and can't register to SIP server.   Andre Normandin
• [Asterisk-Users] Re: Asterisk PBX with X100P in India   Min Hwan Chang
• [Asterisk-Users] OT: USB handsets / softphones   Kanuri, Seshu (Company IT)
• [Asterisk-Users] Asterisk PBX with X100P in India   Min Hwan Chang
• [Asterisk-Users] How do I connect two Asterisk in different domains   Mark Bernardi
• [Asterisk-Users] Keypad disabled on AriaVoice SIP phone -- Fixed   Ian Pattison
• [Asterisk-Users] Asterisk behind NAT   Oswaldo Arratia
• [Asterisk-Users] Losing CallerName info if no CID sent   Cameron Beattie
• [Asterisk-Users] OT: google groups Asterisk-test and now Asterisk-Users marked as spam on Gmail   Dalon Westergreen
• [Asterisk-Users] OT: google groups Asterisk-test and now Asterisk-Users marked as spam on Gmail   izo
• [Asterisk-Users] UK PSTN Calling From OH323 Problem   Iain Young
• [Asterisk-Users] OT: google groups Asterisk-test and now Asterisk-Users marked as spam on Gmail   BJ Weschke
• [Asterisk-Users] new install   Michael Di Martino
• [Asterisk-Users] OT: google groups Asterisk-test and now Asterisk-Users marked as spam on Gmail   Time Bandit
• [Asterisk-Users] large analog to asterisk   Harry McGregor
• [Asterisk-Users] new install   Michael Di Martino
• [Asterisk-Users] OT: USB handsets / softphones   Vahan Yerkanian
• [Asterisk-Users] Asterisk became berserk when Internet connectionis down and can't register to SIP server.   Eric Wieling
• [Asterisk-Users] Segregating a test version of asterisk - libpri/zaptel locations   Julien Levi
• [Asterisk-Users] Keypad disabled on AriaVoice SIP phone -- Fixed   Kanuri, Seshu (Company IT)
• [Asterisk-Users] OT: google groups Asterisk-test and now Asterisk-Users marked as spam on Gmail   Gary Reuter
• [Asterisk-Users] OT: google groups Asterisk-test and now Asterisk-Users marked as spam on Gmail   Ben Hencke
• [Asterisk-Users] large analog to asterisk   Michael D Schelin
• [Asterisk-Users] OT: google groups Asterisk-test and now Asterisk-Users marked as spam on Gmail   Andy Hamilton
• [Asterisk-Users] Empty voicemail attachments?   Andrew C. Brown
• [Asterisk-Users] large analog to asterisk   Michael D Schelin
• [Asterisk-Users] USB Controller ztdummy   Michael Di Martino
• [Asterisk-Users] Re: OT: google groups Asterisk-test and now Asterisk-Users marked as spam on Gmail   Bruno Hertz
• [Asterisk-Users] Debugging zaphfc + PBX integration   Gavin Hamill
• [Asterisk-Users] Anyone already pionered outing calling with user selcted background noise?   magnus
• [Asterisk-Users] OT: google groups Asterisk-test and nowAsterisk-Users marked as spam on Gmail   jltaylor
• [Asterisk-Users] fax detect/transfer problem solved   Jim Meehan
• [Asterisk-Users] sipura ATA, VMI, and Realtime   pbx at itsngroup.com
• [Asterisk-Users] cisco 7960 SIP setup   Michael West
• [Asterisk-Users] Anyone already pionered outing calling with user selcted background noise?   BJ Weschke
• [Asterisk-Users] Anyone already pionered outing calling with userselcted background noise?   jltaylor
• [Asterisk-Users] OT: google groups Asterisk-test and now Asterisk-Users marked as spam on Gmail   Rich Adamson
• FW: [Asterisk-Users] USB Controller ztdummy   Michael Di Martino
• [Asterisk-Users] Can't Modprobe ztcfg   chawki hammoud
• [Asterisk-Users] IAX softphone   Pablo Alsina
• [Asterisk-Users] large analog to asterisk   Andy Hamilton
• [Asterisk-Users] delay problem in asterisk   Shahar Livne
• [Asterisk-Users] Debugging zaphfc + PBX integration   Olivier MONNET
• [Asterisk-Users] large analog to asterisk   John Novack
• [Asterisk-Users] USB Controller ztdummy   Daniel Bruce Lynes
• [Asterisk-Users] sipura ATA, VMI, and Realtime   Matthew Boehm
• [Asterisk-Users] Debugging zaphfc + PBX integration   Gavin Hamill
• [Asterisk-Users] How do I make Extention in my Asterisk PBX   Kumara Jayaweera
• [Asterisk-Users] How do I connect my Asterisk PBX to a service Provider   Kumara Jayaweera
• [Asterisk-Users] How do I connect PC clients to my Asterisk PBX   Kumara Jayaweera
• [Asterisk-Users] What is the good client softphone for windows?   Kumara Jayaweera
• [Asterisk-Users] What is the good client softphone for windows?   Kerry Garrison
• [Asterisk-Users] Can't Modprobe ztcfg   Ian Pattison
• [Asterisk-Users] How do I connect my Asterisk PBX to a serviceProvider   Kanuri, Seshu (Company IT)
• [Asterisk-Users] How do I connect my Asterisk PBX to a service Provider   Andy Hamilton
• [Asterisk-Users] VOIP Regulations in INDIA   Julio Arruda
• [Asterisk-Users] Problem with Livevoip incoming context   Moody
• [Asterisk-Users] Sipura 3000 FXO with Asterisk   Razza
• [Asterisk-Users] IAXTEL Passord   Jean-Yves Landry
• [Asterisk-Users] Bridging 2 Zap channels   Matthew Drobnak
• [Asterisk-Users] Bridging 2 Zap channels   jltaylor
• [Asterisk-Users] a simple question .   Weiming Jiang
• [Asterisk-Users] Problem with Livevoip incoming context   Chris Mason
• [Asterisk-Users] xPL Setup   Chuck Smith
• [Asterisk-Users] Problem with Livevoip incoming context   Moody
• [Asterisk-Users] Cisco 7940 issue.   Manny A. Wise
• [Asterisk-Users] CISCO 7970   Dan Levine
• [Asterisk-Users] CISCO 7970   Dan Levine
• [Asterisk-Users] What is the good client softphone for windows?   Dan Levine
• [Asterisk-Users] How do I make Extention in my Asterisk PBX   Dan Levine
• [Asterisk-Users] a simple question .   Steve Totaro
• [Asterisk-Users] AMP/Asterisk   Daniel Dziubanski
• [Asterisk-Users] large analog to asterisk   Joseph Gutowski
• [Asterisk-Users] Dialplan help needed   snacktime
• [Asterisk-Users] Re: Dialplan help needed   snacktime
• [Asterisk-Users] CISCO 7970   Henry Devito
• [Asterisk-Users] Cisco 7940 issue.   Henry Devito
• [Asterisk-Users] large analog to asterisk   Richard Lyman
• [Asterisk-Users] Help needed to configure Asterisk and SJPhone   Amit
• [Asterisk-Users] CISCO 7970   Bernardino Campos
• [Asterisk-Users] CISCO 7970   Henry Devito
• [Asterisk-Users] CISCO 7970   Henry Devito
• [Asterisk-Users] E & M signalling with WCTE11XP - not all calls go through   Jason Walker
• [Asterisk-Users] Re: Dialplan help needed   tgj
• [Asterisk-Users] Re: Asterisk PBX with X100P in India   Vikram Rangnekar
• [Asterisk-Users] Re: Asterisk PBX with X100P in India   Vikram Rangnekar
• [Asterisk-Users] Asterisk connect to Asterisk   lee siang fong
• [Asterisk-Users] Can't Modprobe ztcfg   chawki hammoud
• [Asterisk-Users] acd+transfer+asterisk-1.0.7   voip technocrat
• [Asterisk-Users] Can't Modprobe ztcfg   Brancaleoni Matteo
• [Asterisk-Users] Debugging zaphfc + PBX integration   Peer Oliver Schmidt
• [Asterisk-Users] VAD/DTX implementation through zaptel cards   parijat
• [Asterisk-Users] Debugging zaphfc + PBX integration   Tim Robinson
• [Asterisk-Users] Extensions busy queue   Gregory Wiktor - ADCom Corp.
• [Asterisk-Users] Asterisk and network problems   Wilson Pickett
• [Asterisk-Users] Can't Modprobe ztcfg   Tzafrir Cohen
• [Asterisk-Users] FXO GW Dial in/out syntax   Robson Ribeiro
• [Asterisk-Users] Dialing rules   Shahar Livne
• [Asterisk-Users] Asterisk and network problems   Steve Blair
• [Asterisk-Users] Asterisk@Home & ISDN BRI   Henry Owens
• [Asterisk-Users] Installing Asterisk@Home on VMware Workstation 4.5.2- build 8848   SCollins
• [Asterisk-Users] Can't Modprobe ztcfg   Steve Totaro
• [Asterisk-Users] Can't Modprobe ztcfg   chawki hammoud
• [Asterisk-Users] Sipura 3000 FXO with Asterisk   Razza
• [Asterisk-Users] Debugging zaphfc + PBX integration   Gavin Hamill
• [Asterisk-Users] Segregating a test version of asterisk - libpri/zaptel locations   Julien Levi
• [Asterisk-Users] Asterisk@Home & ISDN BRI   Robson Ribeiro
• [Asterisk-Users] Bridging 2 Zap channels   steve at daviesfam.org
• [Asterisk-Users] IPswitch: How to use speed dialing?   Ronald Wiplinger
• [Asterisk-Users] Codec Linux Bandwidth Reading   chawki hammoud
• [Asterisk-Users] VAD/DTX implementation through zaptel cards   Steve Kann
• [Asterisk-Users] Asterisk and network problems   Chris Mason (Lists)
• [Asterisk-Users] Asterisk and network problems   Andrew Latham
• [Asterisk-Users] Asterisk@Home & ISDN BRI   Brancaleoni Matteo
• [Asterisk-Users] Losing CallerName info if no CID sent   steve at daviesfam.org
• [Asterisk-Users] Asterisk@Home & ISDN BRI   Michiel van Baak
• [Asterisk-Users] IPSwitchBoard now has Zap Support   Thorben Jensen
• [Asterisk-Users] Re: IPswitch: How to use speed dialing?   tgj
• [Asterisk-Users] Extensions busy queue   Henry Devito
• [Asterisk-Users] IPSwitchBoard now has Zap Support   Chris Mason (Lists)
• [Asterisk-Users] OT: Interview With Kevin Fleming   Matt Riddell
• [Asterisk-Users] Asterisk@Home & ISDN BRI   asterisk at frameweb.it
• [Asterisk-Users] Sipura 3000 FXO with Asterisk   Ed Greenberg
• [Asterisk-Users] Asterisk@Home & ISDN BRI   Henry Owens
• [Fwd: Re: [Asterisk-Users] Debugging zaphfc + PBX integration]   Tim Robinson
• [Asterisk-Users] OT: Sourcing Equipment at the HK Electronics Fair   Glen Browley
• [Asterisk-Users] Re: IPSwitchBoard now has Zap Support   tgj
• [Asterisk-Users] Re: Asterisk and network problems   barry at osvoip.net
• [Asterisk-Users] Lots of RTP checksum error   Paradise Dove
• [Asterisk-Users] VAD/DTX implementation through zaptel cards   Steve Kann
• [Asterisk-Users] VOIP to PTSN provider   Ilija Poznic
• [Asterisk-Users] Installing Asterisk@Home on VMware Workstation 4.5.2- build 8848   kibeki at gmx.net
• [Asterisk-Users] Is this normal - Long time to make call - What is your average with your Hardware?   mr. barker
• [Asterisk-Users] Asterisk as Media Gateway (was: ATT CallVantage & Asterisk)   Edwin Horton
• [Asterisk-Users] Can't Modprobe ztcfg   goldhorse
• [Asterisk-Users] How do I make Extention in my Asterisk PBX   Robert Goodyear
• [Asterisk-Users] Cisco/Asterisk codec negotiation problems   Alistair Cunningham
• [Asterisk-Users] Asterisk connect to Asterisk   Moises Silva
• [Asterisk-Users] Is this normal - Long time to make call - What is your average with your Hardware?   John Millican
• [Asterisk-Users] Re: Asterisk and network problems   Wilson Pickett
• [Asterisk-Users] Mitel 5055 dead after wrong flash, any tips appreciated   Loek Gijben
• [Asterisk-Users] Asterisk and openbrick   Guillermo Salas M.
• [Asterisk-Users] Can't Modprobe ztcfg   Derek Whitten
• [Asterisk-Users] 2 Questions   Ben Price
• [Asterisk-Users] zap device detects hangup when phone switches from answer machine announcement to recording   Martin Renschler
• [Asterisk-Users] Problem with wipphone   Rafael Diaz
• [Asterisk-Users] BT100 wrong NAT detection   Tomas Florian
• [Asterisk-Users] Changing IRQ's on TDM   Damian Funnell
• [Asterisk-Users] BT100 wrong NAT detection   Alex Vishnev
• [Asterisk-Users] BT100 wrong NAT detection   Tomas Florian
• [Asterisk-Users] BT100 wrong NAT detection   Tomas Florian
• [Asterisk-Users] Extensions busy queue   Gregory Wiktor - ADCom Corp.
• [Asterisk-Users] Sipura 3000 FXO with Asterisk   Gregory Wiktor - ADCom Corp.
• [Asterisk-Users] BT100 wrong NAT detection   Alex Vishnev
• [Asterisk-Users] 2 Questions   Ronald Wiplinger
• [Asterisk-Users] Sipura SPA-1001 Setup/Review   kerryg at techdatapros.com
• [Asterisk-Users] Dialing rules   Robert P. McKenzie
• [Asterisk-Users] large analog to asterisk   Michael D Schelin
• [Asterisk-Users] Dialing rules   Robert P. McKenzie
• [Asterisk-Users] Sipura SPA-2000 correct settings for Fax in The Netherlands/Europe   Remco Barende
• [Asterisk-Users] VOIP to PTSN provider   Andy Hamilton
• [Asterisk-Users] large analog to asterisk   Andy Hamilton
• [Asterisk-Users] Re: Asterisk PBX with X100P in India   Min Hwan Chang
• [Asterisk-Users] Dialing rules   Robert P. McKenzie
• [Asterisk-Users] can't use 2 port gws simultaneosuly   M. Ehsanul Karim
• [Asterisk-Users] Line Noise HELP!   Damian Funnell
• [Asterisk-Users] VOIP to PTSN provider   Chris Hills
• [Asterisk-Users] first few seconds of outgoing calls cut off   snacktime
• [Asterisk-Users] VOIP to PTSN provider   Peter Corlett
• [Asterisk-Users] Weird issues with Asterisk@home 0.9   Sascha Ferley
• [Asterisk-Users] Weird issues with Asterisk@home 0.9   Henry Devito
• [Asterisk-Users] Slightly [OT] Asterisk Backends   Andrew Niemantsverdriet
• [Asterisk-Users] Slightly [OT] Asterisk Backends   Nabeel Jafferali
• [Asterisk-Users] Slightly [OT] Asterisk Backends   Mailinglists Address
• [Asterisk-Users] Weird issues with Asterisk@home 0.9   Sascha Ferley
• [Asterisk-Users] VOIP to PTSN provider   Ed Greenberg
• [Asterisk-Users] What is the good client softphone for windows?   Time Bandit
• [Asterisk-Users] problem connecting multiple boxes via IAX2   MobilPete
• [Asterisk-Users] SIP/iax devices in Russia   snacktime
• [Asterisk-Users] Receptionist Module   Manjit Riat
• [Asterisk-Users] IAX softphone   Time Bandit
• [Asterisk-Users] problem connecting multiple boxes via IAX2   Robert Webb
• [Asterisk-Users] SIP/iax devices in Russia   Ian Pattison
• [Asterisk-Users] problem connecting multiple boxes via IAX2   MobilPete
• [Asterisk-Users] recommandation for four (4) port FXS ATA   Joseph
• [Asterisk-Users] large analog to asterisk   C F
• [Asterisk-Users] Can't Native Bridge Any More   Robert DeVries
• [Asterisk-Users] recommandation for four (4) port FXS ATA   Joseph
• [Asterisk-Users] Asterisk behind NAT   Mark Phillips
• [Asterisk-Users] OT: USB handsets / softphones   Paul Mahler
• [Asterisk-Users] cisco 7960 SIP setup   Paul Mahler
• [Asterisk-Users] Phone Compatibility   Chris
• [Asterisk-Users] recommandation for four (4) port FXS ATA   Sascha Ferley
• [Asterisk-Users] recommandation for four (4) port FXS ATA   Arnaud PIGNARD
• [Asterisk-Users] Hitachi WIP-5000/IP-5000 firmware   Jim Meehan
• [Asterisk-Users] Is there a SIP protocol stack inside asterisk?   Race Vanderdecken
• [Asterisk-Users] snom and "hint" priority   Lance Grover
• [Asterisk-Users] SIP stack pluggable?   Race Vanderdecken
• [Asterisk-Users] Hitachi WIP-5000/IP-5000 firmware   Roman Volf
• [Asterisk-Users] Park a call then hunt for a *willing* person   Philip Warner
• [Asterisk-Users] asterisk + OH323 + NAT + gnomemeeting   Karl J. Vesterling
• [Asterisk-Users] Park a call then hunt for a *willing* person   C F
• [Asterisk-Users] Park a call then hunt for a *willing* person   Philip Warner
• [Asterisk-Users] Park a call then hunt for a *willing* person   C F
• [Asterisk-Users] OT VoIP related jobs in Eu   Wilson Pickett
• [Asterisk-Users] SNOM 190: Unknown SIP command 'PUBLISH'   Ronald Wiplinger
• [Asterisk-Users] Warning, flexible rate not heavily tested!   Ronald Wiplinger
• [Asterisk-Users] IPswitch: How to use speed dialing?   Ronald Wiplinger
• [Asterisk-Users] Detecting shorter hangup tone (UK)   Vassilis Konstantinou
• [Asterisk-Users] Point-to-Point Asterisk Link to Reduce Bandwidth   chawki hammoud
• [Asterisk-Users] Line name same as user name   Eng. Emitrax
• [Asterisk-Users] Re: IPswitch: How to use speed dialing?   tgj
• [Asterisk-Users] Line name same as user name   Eng. Emitrax
• [Asterisk-Users] Illegal instruction (core dumped)   Tom Fanning
• [Asterisk-Users] Re: IPswitch: How to use speed dialing?   Ronald Wiplinger
• [Asterisk-Users] IPSwitchBoard Version 0.91 Released   Thorben Jensen
• [Asterisk-Users] Re: IPswitch: How to use speed dialing?   tgj
• [Asterisk-Users] Cisco/Asterisk codec negotiation problems   Alistair Cunningham
• [Asterisk-Users] Re: IPswitch: How to use speed dialing?   Ronald Wiplinger
• [Asterisk-Users] Zaptel fxo & late distinctive ring   Philip Warner
• [Asterisk-Users] snom and "hint" priority   Eugenio De Vena
• [Asterisk-Users] IPSwitchBoard Version 0.91 Released   Ronald Wiplinger
• [Asterisk-Users] Re: IPswitch: How to use speed dialing?   tgj
• [Asterisk-Users] Re: IPSwitchBoard Version 0.91 Released   tgj
• [Asterisk-Users] Bridging 2 Zap channels   Paul Hewlett
• [Asterisk-Users] cisco mgcp and CARD.XML   Sergio
• [Asterisk-Users] first few seconds of outgoing calls cut off   Adam Greenbaum
• [Asterisk-Users] res_perl compile problem   mohammad
• [Asterisk-Users] *8 nor *8# works for me!   Walt Reed
• [Asterisk-Users] Park a call then hunt for a *willing* person   Philip Warner
• [Asterisk-Users] Illegal instruction (core dumped)   Andrew Kohlsmith
• [Asterisk-Users] app_dtmftotext.c   Ezabi
• [Asterisk-Users] Park a call then hunt for a *willing* person   Philip Warner
• [Asterisk-Users] new install   Michael George
• [Asterisk-Users] SNOM 190: Unknown SIP command 'PUBLISH'   Maik Schmitt
• [Asterisk-Users] Sipura 3000 FXO with Asterisk   Razza
• [Asterisk-Users] AMP + POLYCOM   Daniel Dziubanski
• [Asterisk-Users] res_perl compile problem   Brancaleoni Matteo
• [Asterisk-Users] Bandwidth Reduction using Compressed RTP   chawki hammoud
• [Asterisk-Users] IPP g729 & x86_64   Ermakov Sergey
• [Asterisk-Users] Illegal instruction (core dumped)   tmassey at obscorp.com
• [Asterisk-Users] problem connecting multiple boxes via IAX2   jltaylor
• [Asterisk-Users] Sipura 3000 FXO with Asterisk   Razza
• [Asterisk-Users] Re: asterisk + OH323 + NAT + gnomemeeting   Jesse Guardiani
• [Asterisk-Users] Loop Detection   Cameron Beattie
• [Asterisk-Users] RE: Asterisk-Users Digest, Vol 9, Issue 152   Tom Fanning
• [Asterisk-Users] Re: asterisk + OH323 + NAT + gnomemeeting   Bruno Hertz
• [Asterisk-Users] Loop Detection   Daniel Corbe
• [Asterisk-Users] ISDN BRI vs. VOIP DID's, is it worth it?   Gregory Wiktor - ADCom Corp.
• [Asterisk-Users] VOIP to PTSN provider   Gregory Wiktor - ADCom Corp.
• [Asterisk-Users] High Availability - Again   ottodurr
• [Asterisk-Users] VOIP to PTSN provider   Chris Hills
• [Asterisk-Users] Who is a QUALITY IAX Termination Provider for 800DID's?   Cameron Beattie
• [Asterisk-Users] VOIP to PTSN provider   Andy Hamilton
• [Asterisk-Users] spandsp and cvs head   Anton Krall
• [Asterisk-Users] Installing Asterisk@Home on VMware Workstation 4.5.2- build 8848   SCollins
• [Asterisk-Users] VOIP to PTSN provider   Andy Hamilton
• [Asterisk-Users] RE: Illegal instruction (core dumped)   Tom Fanning
• [Asterisk-Users] ISDN BRI vs. VOIP DID's, is it worth it?   snacktime
• [Asterisk-Users] Hitachi WIP-5000/IP-5000 firmware   Jim Meehan
• [Asterisk-Users] E & M signalling with WCTE11XP - not all calls go through   Jason Walker
• [Asterisk-Users] Unbelievable...   snacktime
• [Asterisk-Users] Park a call then hunt for a *willing* person   Andrew Kohlsmith
• [Asterisk-Users] Unbelievable...   Brian Capouch
• [Asterisk-Users] Re: iaxcomm   Michael Van Donselaar
• [Asterisk-Users] Unbelievable...   Rusty Shackleford
• [Asterisk-Users] IPP g729 & x86_64   denon
• [Asterisk-Users] G.729A codec amd64/intel x86-64 optimisation?   Roy Sigurd Karlsbakk
• [Asterisk-Users] extension dialing resistivity   Joseph
• [Asterisk-Users] extension dialing resistivity   Damian Funnell
• [Asterisk-Users] extension dialing resistivity   Joseph
• [Asterisk-Users] MeetMe   Matt Schwartz
• [Asterisk-Users] IPP g723 and getting error when starting asterisk   CM Rahman Jr.
• [Asterisk-Users] Re: asterisk + OH323 + NAT + gnomemeeting   Joel Newkirk
• [Asterisk-Users] Register two account at Broadvoice with one asterisk box   John Millican
• [Asterisk-Users] Can anyone send me sample config files for asterisk and X-Lite?   Abraham WEI
• [Asterisk-Users] ZyXEL Router Terrible Voice Quality   Greg Boehnlein
• [Asterisk-Users] S100I - competitive price?   Greg Boehnlein
• [Asterisk-Users] MeetMe   Eric Wieling aka ManxPower
• [Asterisk-Users] TDM400P Revision question.   Ian Pattison
• [Asterisk-Users] Register two account at Broadvoice with one asterisk box   trixter http://www.0xdecafbad.com
• [Asterisk-Users] ZyXEL Router Terrible Voice Quality   Dave Weis
• [Asterisk-Users] cannot dial two phones using zap   Eddie
• [Asterisk-Users] Re: asterisk + OH323 + NAT + gnomemeeting   Jesse Guardiani
• [Asterisk-Users] Register two account at Broadvoice with one asterisk box   John Millican
• [Asterisk-Users] MeetMe   Vamsi Pottangi
• [Asterisk-Users] Can anyone send me sample config files for asterisk and X-Lite?   Vamsi Pottangi
• [Asterisk-Users] Can anyone send me sample config files for asterisk and X-Lite?   Vaniah Voip
• [Asterisk-Users] Digium G.729 vs. IPP G.729   Boris Bakchiev
• [Asterisk-Users] MeetMe   Vaniah Voip
• [Asterisk-Users] (FIXED) extension dialing responsivity   Joseph
• [Asterisk-Users] Register two account at Broadvoice with one asterisk box   trixter http://www.0xdecafbad.com
• [Asterisk-Users] hangs pc   Altus Snyman
• [Asterisk-Users] Unbelievable...   Robert Goodyear
• [Asterisk-Users] dynamic callrouting and billing?   maka
• [Asterisk-Users] Dynamic Dialplan - Turn VM on/off?   Rod Bacon
• [Asterisk-Users] Digium G.729 vs. IPP G.729   Rod Bacon
• [Asterisk-Users] Dynamic Dialplan - Turn VM on/off?   Boris Bakchiev
• [Asterisk-Users] dynamic callrouting and billing?   Rod Bacon
• [Asterisk-Users] hangs pc   Rod Bacon
• [Asterisk-Users] Dynamic Dialplan - Turn VM on/off?   Rod Bacon
• [Asterisk-Users] Re: asterisk + OH323 + NAT + gnomemeeting   Bruno Hertz
• [Asterisk-Users] Re: asterisk + OH323 + NAT + gnomemeeting   Brian Capouch
• [Asterisk-Users] dynamic callrouting and billing?   Adam Goryachev
• [Asterisk-Users] App_Conference   E rikje
• [Asterisk-Users] Re: asterisk + OH323 + NAT + gnomemeeting   Bruno Hertz
• [Asterisk-Users] Re: Asterisk PBX with X100P in India   Vikram Rangnekar
• [Asterisk-Users] SIP/iax devices in Russia   Vahan Yerkanian
• [Asterisk-Users] Re: IPswitch: How to use speed dialing?   Ronald Wiplinger
• [Asterisk-Users] Re: IPSwitchBoard Version 0.91 Released   Ronald Wiplinger
• [Asterisk-Users] analog gsm router   Altus Snyman
• [Asterisk-Users] Got SIP response 302 "Moved Temporarily" back....   etiennep at kingsley.co.za
• [Asterisk-Users] App_Conference   Vladyslav
• [Asterisk-Users] analog gsm router   Matteo Brancaleoni
• [Asterisk-Users] Got SIP response 302 "Moved Temporarily" back....   etiennep at kingsley.co.za
• [Asterisk-Users] Got SIP response 302 "Moved Temporarily" back....   etiennep at kingsley.co.za
• [Asterisk-Users] Distributed organizations - large scale public sector rollout   Eivind Trondsen
• [Asterisk-Users] Error on install of AMP   Remco Barende
• [Asterisk-Users] Can anyone send me sample config files for asterisk and X-Lite?   Zoa
• [Asterisk-Users] Re: Siemens optiPoint 420 phone and Asterisk   Franz Knipp
• [Asterisk-Users] Re: Siemens optiPoint 420 phone and Asterisk   richard Coco
• [Asterisk-Users] Still having broadvoice issues   Mark Phillips
• [Asterisk-Users] cannot dial two phones using zap   Michael George
• [Asterisk-Users] Unbelievable...   Rich Adamson
• [Asterisk-Users] Cisco External Directory   Marshall, Ed
• [Asterisk-Users] Dynamic Dialplan - Turn VM on/off?   Rich Adamson
• [Asterisk-Users] Changing Codecs when dialing out...   etiennep at kingsley.co.za
• [Asterisk-Users] ISDN BRI vs. VOIP DID's, is it worth it?   Walt Reed
• [Asterisk-Users] Distributed organizations - large scale public sector rollout   Alex Vishnev
• [Asterisk-Users] Installing Asterisk@Home on VMware Workstation 4.5.2- build 8848   SCollins
• [Asterisk-Users] queue - transfer calls   Dov Bigio
• [Asterisk-Users] Problem with Livevoip incoming context   Wiley Siler
• Fw: [Asterisk-Users] Analogue phone transfering   David Wilson
• [Asterisk-Users] wcte11xp digium card   Nathaniel Angelo A. Torres (247talk)
• [Asterisk-Users] Changing Codecs when dialing out...   etiennep at kingsley.co.za
• [Asterisk-Users] Problem with Livevoip incoming context   Chris Mason (Lists)
• [Asterisk-Users] OT: USB handsets / softphones   Kanuri, Seshu (Company IT)
• [Asterisk-Users] 99% CPU - CVS 03.28.05   Moody
• [Asterisk-Users] Follow-me script - user changeable options   Chris Mason (Lists)
• [Asterisk-Users] wcte11xp digium card   Matteo Brancaleoni
• [Asterisk-Users] Cisco 7940   Thomas RULMONT
• [Asterisk-Users] TDM card periodic buzz   Trent Tuggle
• [Asterisk-Users] queue - transfer calls   Ariel Batista
• [Asterisk-Users] zap device detects hangup when phone switches from answer machine announcement to recording   Moises Silva
• [Asterisk-Users] Indicating when other party has answered   Daniel Nyström
• [Asterisk-Users] Re: Fax questions   Ronald Wiplinger
• [Asterisk-Users] Help compiling zaptel in Debian   Manuel Casal
• [Asterisk-Users] TDM card periodic buzz   Andrew Kohlsmith
• [Asterisk-Users] Motherboard failure with 2 Digium TE405P cards   mattf
• [Asterisk-Users] TDM400P Revision question.   Ian Pattison
• [Asterisk-Users] Re: Unbelievable...   Bruno Hertz
• [Asterisk-Users] wcte11xp digium card   Nathaniel Angelo A. Torres (247talk)
• [Asterisk-Users] 99% CPU - CVS 03.28.05   Dan Levine
• [Asterisk-Users] 99% CPU - CVS 03.28.05   Mohit Muthanna
• [Asterisk-Users] Help compiling zaptel in Debian   Andres Paglayan
• [Asterisk-Users] Motherboard failure with 2 Digium TE405P cards   Andrew Latham
• [Asterisk-Users] Re:queue - transfer calls   Dov Bigio
• [Asterisk-Users] 99% CPU - CVS 03.28.05   Moody
• [Asterisk-Users] Motherboard failure with 2 Digium TE405P car ds   mattf
• [Asterisk-Users] RE: queue - transfer calls   Dov Bigio
• [Asterisk-Users] Re: Unbelievable...   Chris Mason (Lists)
• [Asterisk-Users] Snom subscribe/notify problem   Michael George
• [Asterisk-Users] Calling Card   Huddleston, Robert
• [Asterisk-Users] Can I use Asterisk for a modified Hoot and Holler?   Machen, Matthew T.
• [Asterisk-Users] Snom subscribe/notify problem   Michael George
• [Asterisk-Users] One-way audio   Andrejus Stavickis
• [Asterisk-Users] snom and "hint" priority   Josh Dady
• [Asterisk-Users] Asterix Manager Proxy in Java/EJB?   Colin Stefani
• [Asterisk-Users] Re: Unbelievable...   Rich Adamson
• [Asterisk-Users] Can I use Asterisk for a modified Hoot and Holler?   BJ Weschke
• [Asterisk-Users] Cisco 7940   Andy Hamilton
• [Asterisk-Users] Lots of RTP checksum errors   Paradise Dove
• [Asterisk-Users] Strange tones when placing a PSTN call.   Michael Martin
• [Asterisk-Users] Asterix Manager Proxy in Java/EJB?   Colin Stefani
• [Asterisk-Users] Calling Card   Jean-Michel Hiver
• [Asterisk-Users] Re: IPswitch: How to use speed dialing?   tgj
• [Asterisk-Users] Cisco 7970 startup problem   Parfenenok Sergey
• [Asterisk-Users] Help compiling zaptel in Debian   Manuel Casal
• [Asterisk-Users] Calling Card   Race Vanderdecken
• [Asterisk-Users] Unable to specify channel 1: No such device   Gregorio Toscano
• [Asterisk-Users] Calling Card   jltaylor
• [Asterisk-Users] Unable to specify channel 1: No such device   Wiley Siler
• [Asterisk-Users] 99% CPU - CVS 03.28.05   Luki
• [Asterisk-Users] Still having broadvoice issues   Luki
• [Asterisk-Users] Hitachi WIP-5000/IP-5000 firmware   Russ Beaupre
• [Asterisk-Users] Cisco/Asterisk codec negotiation problems   Alistair Cunningham
• [Asterisk-Users] Only one PRI out of four working on TE405p?   Derek Conniffe
• [Asterisk-Users] Help compiling zaptel in Debian   Matt Roth
• [Asterisk-Users] Can I use Asterisk for a modified Hoot and Holler?   Rich Adamson
• [Asterisk-Users] Voicemail not working...   Wiley Siler
• [Asterisk-Users] TDM400P Revision question.   David Brodbeck
• [Asterisk-Users] 99% CPU - CVS 03.28.05   Zoa
• [Asterisk-Users] Re: IPswitch: How to use speed dialing?   Ronald Wiplinger
• [Asterisk-Users] Only one PRI out of four working on TE405p?   Andres
• [Asterisk-Users] Re: Can I use Asterisk for a modified Hoot and Holler?   Noah Miller
• [Asterisk-Users] Can I use Asterisk for a modified Hoot andHoller?   Race Vanderdecken
• [Asterisk-Users] Calling Card   Dylan VanHerpen
• [Asterisk-Users] Toshiba CTX100 integration with PABX for two site   David Brodbeck
• [Asterisk-Users] Unbelievable...   Rusty Shackleford
• [Asterisk-Users] High Availability - Again   Race Vanderdecken
• [Asterisk-Users] Unable to specify channel 1: No such device   Rich Adamson
• [Asterisk-Users] SNOM 190: Unknown SIP command 'PUBLISH'   Shanon Swafford
• [Asterisk-Users] Hitachi WIP-5000/IP-5000 firmware   Jim Meehan
• [Asterisk-Users] TDM400P Revision question.   Eric Wieling aka ManxPower
• [Asterisk-Users] Unbelievable...   geek at mnwifi.net
• [Asterisk-Users] maximum value for LEN(x)   Benjamin Lawetz
• [Fwd: Re: [Asterisk-Users] Only one PRI out of four working on TE405p?]   Tim Robinson
• [Asterisk-Users] Calling Card   Nabeel Jafferali
• [Asterisk-Users] Motherboard failure with 2 Digium TE405P cards   Race Vanderdecken
• [Asterisk-Users] Can I use Asterisk for a modified Hoot andHoller?   Steve Totaro
• [Asterisk-Users] Help compiling zaptel in Debian   Manuel Casal
• [Asterisk-Users] spandsp and cvs head   Anton Krall
• [Asterisk-Users] Grandstream BT Volume   Anton Krall
• [Asterisk-Users] show translation   Anton Krall
• [Asterisk-Users] Hold on outbound calls and the SNOM 190   Clint Miller
• [Asterisk-Users] Motherboard failure with 2 Digium TE405P car ds   mattf
• [Asterisk-Users] Random SIP Phone Problem   Shaun Tierney
• [Asterisk-Users] TDM400P Revision question.   Andrew Kohlsmith
• [Asterisk-Users] show translation   Steve Totaro
• [Asterisk-Users] show translation   Luki
• [Asterisk-Users] ackcall with AddQueueMember   Ryan Courtnage
• [Asterisk-Users] Motherboard failure with 2 Digium TE405P car ds   jltaylor
• [Asterisk-Users] spandsp and cvs head   Rich Adamson
• [Asterisk-Users] TDM400P Revision question.   Robert Webb
• [Asterisk-Users] Unable to specify channel 1: No such device   Robert Webb
• [Asterisk-Users] ni1 (ppp) and national isdn on te110p   Jason McAffee
• [Asterisk-Users] TDM400P Revision question.   Ian Pattison
• [Asterisk-Users] TDM400P Revision question.   Julian J. M.
• [Asterisk-Users] TDM400P Revision question.   Rich Adamson
• [Asterisk-Users] callback broken?   CM Rahman Jr.
• [Asterisk-Users] Calling Card   Huddleston, Robert
• [Asterisk-Users] 99% CPU - CVS 03.28.05   Brian Capouch
• [Asterisk-Users] Fedore CORE 2   Michael Di Martino
• [Asterisk-Users] Hold on outbound calls and the SNOM 190   Clint Miller
• [Asterisk-Users] Unable to specify channel 1: No such device   pbx at itsngroup.com
• [Asterisk-Users] callback broken?   Kanuri, Seshu (Company IT)
• [Asterisk-Users] Fedore CORE 2   Kanuri, Seshu (Company IT)
• [Asterisk-Users] Fedore CORE 2   Colin Anderson
• [Asterisk-Users] show translation   Anton Krall
• [Asterisk-Users] show translation   Anton Krall
• [Fwd: Re: [Asterisk-Users] Only one PRI out of four working onTE405p?]   Derek Conniffe
• [Asterisk-Users] Only one PRI out of four working on TE405p?   Derek Conniffe
• [Asterisk-Users] Junghans QuadBRI and fax detection   Marc Storck
• [Asterisk-Users] DIAL FROM CONSOLE   Dan Levine
• [Asterisk-Users] Only one PRI out of four working on TE405p?   Derek Conniffe
• [Asterisk-Users] DIAL FROM CONSOLE   Josiah Bryan
• [Asterisk-Users] Only one PRI out of four working on TE405p?   Andres
• [Asterisk-Users] DIAL FROM CONSOLE   Ronald Wiplinger
• [Asterisk-Users] CLI Numbers   Ronald Wiplinger
• [Asterisk-Users] Looking for ATAs   Anton Krall
• [Asterisk-Users] Problems with Cisco ATA 186/MGCP   navarrocarlos
• [Asterisk-Users] snom 220 hints lost after reload   Eric Lawman
• [Asterisk-Users] system wide speed dialing   Ronald Wiplinger
• [Asterisk-Users] system wide speed dialing   Paul Crick
• [Asterisk-Users] system wide speed dialing   Joel Newkirk
• [Asterisk-Users] asterisk on MIPS   Wang Xiangzhou
• [Asterisk-Users] zombie channels & missed transfer   Dov Bigio
• [Asterisk-Users] IPP g723 and getting error when starting asterisk   Rod Bacon
• [Asterisk-Users] TDM400P Revision question.   Neal Walton
• [Asterisk-Users] system wide speed dialing   Ronald Wiplinger
• [Asterisk-Users] Upgraded now Asterisk won't start   Paul A Brown
• [Asterisk-Users] Upgraded now Asterisk won't start   Andy Hamilton
• [Asterisk-Users] [Announcement] Updated Web-MeetMe   Dan Austin
• [Asterisk-Users] DIAL FROM CONSOLE   Dan Levine
• [Asterisk-Users] RealTime Vs. AGI and PHP or MySQL calls within extensions.conf   Me
• [Asterisk-Users] Why *3* entries?   Matt
• [Asterisk-Users] INVALID Extension   Dan Levine
• [Asterisk-Users] DIAL FROM CONSOLE   Henry Devito
• [Asterisk-Users] wcte11xp digium card   Nathaniel Angelo A. Torres (247talk)
• [Asterisk-Users] system wide speed dialing   Paul Crick
• [Asterisk-Users] ISDN BRI vs. VOIP DID's, is it worth it?   Gregory Wiktor - ADCom Corp.
• [Asterisk-Users] wcte11xp digium card   Nathaniel Angelo A. Torres (247talk)
• [Asterisk-Users] Problems with incoming calls on a E1 ISDN PRI   Roberto Reiner Uhry
• [Asterisk-Users] Re: Problems with incoming calls on a E1 ISDN PRI   Roberto Reiner Uhry
• [Asterisk-Users] Calling Card   Dylan VanHerpen
• [Asterisk-Users] Citrix   Javier Godinez
• [Asterisk-Users] DTMF intermittently stops working   Pedro
• [Asterisk-Users] Blind Transfers - any ideas?   Jason Brown
• [Asterisk-Users] Extension busy issue on TDM01A   ajeo at hungryforlife.org
• [Asterisk-Users] Blind Transfers - any ideas?   Roman Volf
• [Asterisk-Users] SIP calls being lost "frame from cahnnel" error   Michael George
• [Asterisk-Users] Still having broadvoice issues   Mark Phillips
• [Asterisk-Users] Want to use Asterisk instead of existing Meridian Norstar system ... need some help   Imran
• [Asterisk-Users] Want to use Asterisk instead of existing MeridianNorstar system ... need some help   Joe Dennick
• [Asterisk-Users] G729 Key Registration Problem   chawki hammoud
• [Asterisk-Users] callback broken?   CM Rahman Jr.
• [Asterisk-Users] Help compiling zaptel in Debian   Tzafrir Cohen
• [Asterisk-Users] Help compiling zaptel in Debian   Tzafrir Cohen
• [Asterisk-Users] DTMF intermittently stops working   steve szmidt
• [Asterisk-Users] G729 Key Registration Problem   Colin Anderson
• [Asterisk-Users] HELP: How to detect a hangup tone?   Charles Wang
• [Asterisk-Users] Asterisk & POE   Callum McGillivray
• [Asterisk-Users] Remapping Woes in features.conf   John D. Lewis
• [Asterisk-Users] Vici Dialer   TOBY
• [Asterisk-Users] TDM400P and SCSI/SATA = * noise problems???   Damian Funnell
• [Asterisk-Users] Call Progress Detection   TOBY
• [Asterisk-Users] Citrix   C F
• [Asterisk-Users] Asterisk & POE   Paul Hales
• [Asterisk-Users] DTMF in outbound calls   Rod Bacon
• [Asterisk-Users] Queues-Agents Problem   Bharat M. Sarvan
• [Asterisk-Users] Citrix   David Phelan
• [Asterisk-Users] Asterisk timer on Digium's TDM cards?   Boris Bakchiev
• [Asterisk-Users] Citrix   Rod Bacon
• [Asterisk-Users] CLI Numbers   Kanishka Somaratne
• [Asterisk-Users] Asterisk & POE   Kristof Hardy
• [Asterisk-Users] Junghans QuadBRI and fax detection   Kristof Hardy
• [Asterisk-Users] Help compiling zaptel in Debian   Scott Henderson
• [Asterisk-Users] Asterisk timer on Digium's TDM cards?   Scott Henderson
• [Asterisk-Users] Asterisk & POE   Paul Hales
• [Asterisk-Users] Codec/Phone negociation(s)   etiennep at kingsley.co.za
• [Asterisk-Users] Billing   Rizwan Chaudhry
• [Asterisk-Users] codec negotiation with CISCO 7960 and Firefly softphone   raymond
• [Asterisk-Users] wcfxo problem   VoIP Newbie
• [Asterisk-Users] Astrisk + Cisco 5350   Rene Pavelko
• [Asterisk-Users] asterisk on MIPS   Vladyslav
• [Asterisk-Users] Upgraded now Asterisk won't start   Vladyslav
• [Asterisk-Users] Sipura SPA-841 Phone Review   Gareth Blades
• [Asterisk-Users] Sipura SPA-841 distinctive ring   Gareth Blades
• [Asterisk-Users] *8 nor *8# works for me!   Vladyslav
• [Asterisk-Users] DID ~ Extension   Nathaniel Angelo A. Torres (247talk)
• [Asterisk-Users] Upgraded now Asterisk won't start   Paul A Brown
• [Asterisk-Users] Re: V92 modem with asterisk   VoIP Newbie
• [Asterisk-Users] Junghans QuadBRI and fax detection   Eugenio De Vena
• [Asterisk-Users] Asterisk with Softswitch   Sharath Chandra
• [Asterisk-Users] Billing   Areski
• [Asterisk-Users] Installed ztdummy, Asterisk doesnt work anymore   Michel Bachofen
• [Asterisk-Users] *8 nor *8# works for me!   Ronald Wiplinger
• [Asterisk-Users] Voice quality problem from calls from * to * thru IPtel SER server.   Kong
• [Asterisk-Users] Help compiling zaptel in Debian   Manuel Casal
• [Asterisk-Users] TE405p PRI ISDN [E1] RED Recovering ?   Derek Conniffe
• [Asterisk-Users] Asterisk and T.38.   Jairo Buendia
• [Asterisk-Users] Junghans QuadBRI and fax detection   Kristof Hardy
• [Asterisk-Users] Billing   Maxim Litnitsky
• [Asterisk-Users] Newbie - VoIP route SIP calls to provider   iMRAN
• [Asterisk-Users] Junghans QuadBRI and fax detection   Marc Storck
• [Asterisk-Users] Billing   Eric Wieling aka ManxPower
• [Asterisk-Users] Newbie - VoIP route SIP calls to provider   iMRAN
• [Asterisk-Users] TE405p PRI ISDN [E1] RED Recovering ?   Peter Svensson
• [Asterisk-Users] TDM400P and SCSI/SATA = * noise problems???   Eric Wieling aka ManxPower
• [Asterisk-Users] VoIP route SIP calls to provider   iMRAN
• [Asterisk-Users] Re: Problems with incoming calls on a E1 ISDN PRI   Roberto Reiner Uhry
• [Asterisk-Users] Cisco External Directory   Florian Overkamp
• [Asterisk-Users] DID ~ Extension   Yair Hakak
• [Asterisk-Users] VPN/Asterisk combo   Chris Mason (Lists)
• [Asterisk-Users] Asterisk & POE   SCollins
• [Asterisk-Users] Bridging 2 Zap channels - SOLUTION   Paul Hewlett
• [Asterisk-Users] AT-320 phones with IAX2   Gavin Hamill
• [Asterisk-Users] Looking for some real basic doccos...   Richard Malcolm-Smith
• [Asterisk-Users] TE405p PRI ISDN [E1] RED Recovering ?   Derek Conniffe
• [Asterisk-Users] SIP users, OH323 to provider, g729 - high level of echo   Shaoul Jacobson - TELLINK
• [Asterisk-Users] VPN/Asterisk combo   Jean-Michel Hiver
• [Asterisk-Users] DID ~ Extension   Nathaniel Angelo A. Torres (247talk)
• [Asterisk-Users] VPN/Asterisk combo   Henry Devito
• [Asterisk-Users] IPTables   Michael Sanders
• [Asterisk-Users] Snom subscribe/notify problem   Michael George
• [Asterisk-Users] answered time   Rizwan Chaudhry
• [Asterisk-Users] Re: Starting with Asterisk-SIP   ruben cuevas rumin
• [Asterisk-Users] VPN/Asterisk combo   Chris Mason (Lists)
• [Asterisk-Users] VPN/Asterisk combo   Chris Mason (Lists)
• [Asterisk-Users] Asterisk with Softswitch   Michael B. Murdock
• [Asterisk-Users] Cisco External Directory   Joseph
• [Asterisk-Users] DTMF intermittently stops working   Joseph
• [Asterisk-Users] Snom NOTIFY on IAX2 channel   Michael George
• [Asterisk-Users] VPN/Asterisk combo   Giles Coochey
• [Asterisk-Users] Voicemail email text:   Tim Connolly
• [Asterisk-Users] MF instead of DTMF   Michael B. Murdock
• [Asterisk-Users] Citrix   Kanuri, Seshu (Company IT)
• [Asterisk-Users] VPN/Asterisk combo   Chris Hills
• [Asterisk-Users] 802.1p , precedence and TOS   Eugenio De Vena
• [Asterisk-Users] Vici Dialer   mattf
• [Asterisk-Users] Re: [Serusers] Ser + Asterisk   Iqbal
• [Asterisk-Users] VPN/Asterisk combo   Chris Hills
• [Asterisk-Users] Voicemail email text:   JD
• [Asterisk-Users] Voicemail email text:   Peter Bowyer
• [Asterisk-Users] Any work around for ISPs that block port 5060 and 69   Joel Jn-Francois
• [Asterisk-Users] VPN/Asterisk combo   Chris Mason (Lists)
• [Asterisk-Users] VPN/Asterisk combo   Henry Devito
• [Asterisk-Users] Sipura SPA-841 Phone Review   Me
• [Asterisk-Users] Cisco External Directory   Brent DeShazer
• [Asterisk-Users] Soft Video phone for Windows XP   Ronald Wiplinger
• [Asterisk-Users] Any work around for ISPs that block port 5060 and 69   Remco Barende
• [Asterisk-Users] IPv6 possible?   Ronald Wiplinger
• [Asterisk-Users] Firefly w/*?   Me
• [Asterisk-Users] VPN/Asterisk combo   Chris Mason (Lists)
• [Asterisk-Users] Voicemail email text:   Chris Mason (Lists)
• [Asterisk-Users] Any work around for ISPs that block port 5060 and69   Me
• [Asterisk-Users] Any work around for ISPs that block port 5060 and69   Kanuri, Seshu (Company IT)
• [Asterisk-Users] IPTables   Joel Newkirk
• [Asterisk-Users] Re: Starting with Asterisk-SIP   Moises Silva
• [Asterisk-Users] Sipura SPA-841 distinctive ring   Joel Newkirk
• [Asterisk-Users] IPTables   Moises Silva
• [Asterisk-Users] Sipura PSA-841 -suitable headset   Chris Mason (Lists)
• [Asterisk-Users] VPN/Asterisk combo   Chris Mason (Lists)
• [Asterisk-Users] VoIP PSTN numbers in Australia?   Daniel Pocock
• [Asterisk-Users] VPN/Asterisk combo   Giles Coochey
• [Asterisk-Users] RealTime Vs. AGI and PHP or MySQL calls within extensions.conf   Moises Silva
• [Asterisk-Users] Firefly w/*?   Peter Bowyer
• [Asterisk-Users] MF instead of DTMF   jltaylor
• [Asterisk-Users] IPTables   Vladyslav
• [Asterisk-Users] VPN/Asterisk combo   Colin Anderson
• [Asterisk-Users] Looking for ATAs   jltaylor
• [Asterisk-Users] Firefly w/*?   John D. Lewis
• [Asterisk-Users] Soft Video phone for Windows XP   Sascha Ferley
• [Asterisk-Users] VPN/Asterisk combo   Jean-Michel Hiver
• [Asterisk-Users] VPN/Asterisk combo   Joseph
• [Asterisk-Users] Firefly w/*?   Jean-Michel Hiver
• [Asterisk-Users] Voicemail email text:   Anton Krall
• [Asterisk-Users] Testing my TDM01A   Aaron O'Hara
• [Asterisk-Users] Sipura SPA-841 Phone Review   Thibault Lamy
• [Asterisk-Users] Cisco 7960 directory.xlm   Davin O'Neill
• [Asterisk-Users] Looking for ATAs   Anton Krall
• [Asterisk-Users] FXO lines on TDM04B not responding   Goutam Shaw
• [Asterisk-Users] Firefly w/*?   Dmitriy Yermakov
• [Asterisk-Users] Firefly w/*?   Derek Whitten
• [Asterisk-Users] Re: Citrix   Noah Miller
• [Asterisk-Users] Voicemail email text:   BJ Weschke
• [Asterisk-Users] 99% CPU - CVS 03.28.05   Moody
• [Asterisk-Users] IPv6 possible?   Chris Hills
• [Asterisk-Users] VPN/Asterisk combo   Chris Hills
• [Asterisk-Users] Cisco 7970   Paul A Brown
• [Asterisk-Users] Using voicemail independently from Asterisk PBX   John Riek
• [Asterisk-Users] Asterisk and Request Tracker, RT.?   Mike Dent
• [Asterisk-Users] Firefly w/*?   Kanuri, Seshu (Company IT)
• [Asterisk-Users] VPN/Asterisk combo   jltaylor
• [Asterisk-Users] Voicemail email text:   Time Bandit
• [Asterisk-Users] Re: Starting with Asterisk-SIP   Dana Olson
• [Asterisk-Users] Want to use Asterisk instead of existingMeridianNorstar system ... need some help   Mike Robinson
• [Asterisk-Users] MF instead of DTMF   Michael B. Murdock
• [Asterisk-Users] Billing   Sathya Weerasooriya
• [Asterisk-Users] Asterisk & POE   Mike Robinson
• [Asterisk-Users] Re: Any work around for ISPs that block port ....   barry at osvoip.net
• [Asterisk-Users] Firefly w/*?   Remco Barende
• [Asterisk-Users] Testing the TDM01A   Aaron O'Hara
• [Asterisk-Users] Extensions unavailable after to sucessfull call (Registration lose)   kritikus Araklidas
• [Asterisk-Users] Sipura SPA-841 Phone Review   Paul Dugas
• [Asterisk-Users] Firefly w/*?   Mojo with Horan & Company, LLC
• [Asterisk-Users] VPN/Asterisk combo   Chris Hills
• [Asterisk-Users] Citrix   Javier Godinez
• [Asterisk-Users] Any work around for ISPs that block port 5060 and 69   Asterisk guy
• [Asterisk-Users] PRI - T1 feasibility   Ronald Hartmann
• [Asterisk-Users] Re: Cisco 7960 directory.xlm   Sergio
• [Asterisk-Users] VPN/Asterisk combo   Chris Mason (Lists)
• [Asterisk-Users] Re: [Serusers] Ser + Asterisk   Iqbal
• [Asterisk-Users] Conference solution for 100+ users   Sergio Veltri
• [Asterisk-Users] OutBOund Dial problem   kurt x
• [Asterisk-Users] TDM400P and SCSI/SATA = * noise problems???   Ben Hencke
• [Asterisk-Users] Any work around for ISPs that block port 5060 and69   Michael D Schelin
• [Asterisk-Users] Predictive Dialer   Jesus Mogollon
• [Asterisk-Users] Asterisk Business Case - Who is using it!?   Denis Galvão - iSolve
• [Asterisk-Users] Server failing to Boot with   Brian Watters
• [Asterisk-Users] OutBOund Dial problem   Eric Wieling aka ManxPower
• [Asterisk-Users] ASterisk OH323.CONF Gateway & Gatekeeper   Jorge Alayon
• [Asterisk-Users] mysql from dialplan   Jason Kawakami
• [Asterisk-Users] VPN/Asterisk combo   Colin Anderson
• [Asterisk-Users] Firefly w/*?   Steve Totaro
• [Asterisk-Users] Using voicemail independently from Asterisk PBX   Steve Totaro
• [Asterisk-Users] 802.1p , precedence and TOS   SCollins
• [Asterisk-Users] ATA - PBX   Maximiliano J. Goldsmid
• [Asterisk-Users] OutBOund Dial problem   Goutam Shaw
• [Asterisk-Users] Firefly w/*?   Bashir Ullah - www.Lamsre.Com
• [Asterisk-Users] Asterisk & POE   Jeff Pratt
• [Asterisk-Users] VPN/Asterisk combo   Colin Anderson
• [Asterisk-Users] VoIP PSTN numbers in Australia?   Craig
• [Asterisk-Users] Citrix   Loek Gijben
• [Asterisk-Users] TDM400P and SCSI/SATA = * noise problems???   Damian Funnell
• [Asterisk-Users] Server failing to Boot with   Brian Watters
• [Asterisk-Users] Re: Any work around for ISPs that block port 5060 and 69   Doug Meredith
• [Asterisk-Users] TDM400P and SCSI/SATA = * noise problems???   tmassey at obscorp.com
• [Asterisk-Users] Dutch callerid: sending on FXS?   Mark Wormgoor
• [Asterisk-Users] Re: Starting with Asterisk-SIP   ruben cuevas rumin
• [Asterisk-Users] Asterisk Business Case - Who is using it!?   Colin Anderson
• [Asterisk-Users] Latest CVS breaks voicemail app   Malcolm Bader
• [Asterisk-Users] VPN/Asterisk combo   Chris Mason (Lists)
• [Asterisk-Users] Billing   trixter http://www.0xdecafbad.com
• [Asterisk-Users] Unix softphone   Javier Godinez
• [Asterisk-Users] spandsp and cvs head   Matt Roth
• [Asterisk-Users] Server failing to Boot with   Colin Anderson
• [Asterisk-Users] Sipura SPA-841 Phone Review   Craig
• [Asterisk-Users] Asterisk Netgear FSM7326P and Cisco 7960 on VLAN   Ron Wellsted
• [Asterisk-Users] Show accountcode in both directions?   Matthew Boehm
• [Asterisk-Users] Server failing to Boot with   Chris Mason (Lists)
• [Asterisk-Users] Installed ztdummy, Asterisk doesnt work anymore   Matt Roth
• [Asterisk-Users] Cisco External Directory   Mike Holloway
• [Asterisk-Users] Citrix   tim panton
• [Asterisk-Users] BRI channels not answering   Terry Wade
• [Asterisk-Users] Fax and spandsp   Julian J. M.
• [Asterisk-Users] Asterisk Business Case - Who is using it!?   Manjit Riat
• [Asterisk-Users] SIP Video Support   Nir Simionovich
• [Asterisk-Users] Any work around for ISPs that block port 5060 and 69   Mike Holloway
• [Asterisk-Users] New AstManProxy Manager Proxy v0.98   David Troy
• [Asterisk-Users] Upgraded now Asterisk won't start   Matt Roth
• [Asterisk-Users] PRI - T1 feasibility   jltaylor
• [Asterisk-Users] Looking for ATAs   jltaylor
• [Asterisk-Users] MF instead of DTMF   jltaylor
• [Asterisk-Users] ATA - PBX   Nir Simionovich
• [Asterisk-Users] Using voicemail independently from Asterisk PBX   C F
• [Asterisk-Users] snom 220 hints lost after reload   ewr at erols.com
• [Asterisk-Users] Cisco 7970   C F
• [Asterisk-Users] Any work around for ISPs that block port 5060   Iassen Hristov
• [Asterisk-Users] Which free calling card app most suited for commercial use?   snacktime
• [Asterisk-Users] Server failing to Boot with   Eric Wieling aka ManxPower
• [Asterisk-Users] Using voicemail independently from Asterisk PBX   John Riek
• [Asterisk-Users] Asterisk Business Case - Who is using it!?   Jeff Heath
• [Asterisk-Users] Help compiling zaptel in Debian   Matt Roth
• [Asterisk-Users] Help needed on Utstarcom F1000 Wifi Handset.   Satchid
• [Asterisk-Users] US$200 bounty for * paging feature   Mike
• [Asterisk-Users] TDM400P and SCSI/SATA = * noise problems???   Scott Laird
• [Asterisk-Users] OT: Sourcing Equipment at the HK Electronics Fair   Cameron Beattie
• [Asterisk-Users] Asterisk Business Case - Who is using it!?   C F
• [Asterisk-Users] US$200 bounty for * paging feature   Richard
• [Asterisk-Users] Asterisk Netgear FSM7326P and Cisco 7960 on VLAN   SCollins
• [Asterisk-Users] snom 220 hints lost after reload   Maik Schmitt
• [Asterisk-Users] Re: Problems with incoming calls on a E1 ISDN PRI   Roberto Reiner Uhry
• [Asterisk-Users] VPN/Asterisk combo   Rich Adamson
• [Asterisk-Users] Asterisk & POE   Paul Hales
• [Asterisk-Users] DTMF in outbound calls   Rod Bacon
• [Asterisk-Users] Citrix   Javier Godinez
• [Asterisk-Users] Industrial Cordless handsets analog or voip based ??   TC
• [Asterisk-Users] TDM400P and SCSI/SATA = * noise problems???   Boris Bakchiev
• [Asterisk-Users] Attended transfer on sipura ATA/Phone?   Tim Pushor
• [Asterisk-Users] Using voicemail independently from Asterisk PBX   BJ Weschke
• [Asterisk-Users] US$200 bounty for * paging feature   Matthew Walster
• [Asterisk-Users] Sipura SPA-841 Phone Review   Master Abi
• [Asterisk-Users] US$200 bounty for * paging feature   William Boehlke
• [Asterisk-Users] Cisco 7970   Ing CIP Alejandro Celi Mariátegui
• [Asterisk-Users] US$200 bounty for * paging feature   trixter http://www.0xdecafbad.com
• [Asterisk-Users] Looking for a softswitch   Didier Hess
• [Asterisk-Users] Problems with Cisco ATA 186/MGCP   Carlos Navarro
• licensing *sigh* (was Re: [Asterisk-Users] US$200 bounty for * paging feature)   Andrew Kohlsmith
• [Asterisk-Users] NuFone problems to non-na numbers   Matthew Asham
• [Asterisk-Users] Asterisk@Home v.0.9 and Digum   Nathaniel Angelo A. Torres (247talk)
• [Asterisk-Users] FXO lines on TDM04B not responding   Rich Adamson
• [Asterisk-Users] Re: Any work around for ISPs that block port ....   Rich Adamson
• [Asterisk-Users] Cisco 7960/7960G   Craig
• [Asterisk-Users] Cisco 7960/7960G   Jeffrey C. Ollie
• [Asterisk-Users] US$200 bounty for * paging feature   Matt Darnell
• licensing *sigh* (was Re: [Asterisk-Users] US$200 bounty for * paging feature)   snacktime
• [Asterisk-Users] Cisco 7920 - chan_sccp - asterisk@home .9   Chuck Smith
• [Asterisk-Users] Cisco 7920 - chan_sccp - asterisk@home .9   Chuck Smith
• [Asterisk-Users] FW: Cisco 7920 - chan_sccp - asterisk@home .9   Chuck Smith
• [Asterisk-Users] Citrix   Rod Bacon
• [Asterisk-Users] Libunicall   Nathaniel Angelo A. Torres (247talk)
• licensing *sigh* (was Re: [Asterisk-Users] US$200 bounty for * paging feature)   trixter http://www.0xdecafbad.com
• [Asterisk-Users] US$200 bounty for * paging feature   Henry Devito
• [Asterisk-Users] 99% CPU - CVS 03.28.05   Adam Goryachev
• [Asterisk-Users] Cisco 7970   C F
• [Asterisk-Users] Conference solution for 100+ users   Vamsi Pottangi
• [Asterisk-Users] US$200 bounty for * paging feature   Richard
• [Asterisk-Users] US$200 bounty for * paging feature   Tim Litwiller
• [Asterisk-Users] SIP Phone Compatability   Daniel Salama
• [Asterisk-Users] VoiceMail Config Questions   David
• [Asterisk-Users] CVS-HEAD and CheckGroup/SetGroup   Sean A. Newton
• licensing *sigh* (was Re: [Asterisk-Users] US$200 bounty for *paging feature)   Paul Fielding
• [Asterisk-Users] Sample AGI Scripts in C needed.   Bharat M. Sarvan
• [Asterisk-Users] US$200 bounty for * paging feature   Shane Burrell
• [Asterisk-Users] help needed for sound device setup   someshwarak at tataelxsi.co.in
• [Asterisk-Users] Conference solution for 100+ users   Paul Hales
• [Asterisk-Users] RealTime ignoring switch => Realtime/context@realtime_ext   Me
• [Asterisk-Users] NAT and only been able to have 1 SIP phone behind   Anton Krall
• [Asterisk-Users] Text Messages   Anton Krall
• [Asterisk-Users] Using voicemail independently from Asterisk PBX   Anton Krall
• [Asterisk-Users] IAX realtime HELP   Paul Dracevich
• [Asterisk-Users] IPv6 possible?   Ronald.vanderPol at rvdp.org
• [Asterisk-Users] Where to post my impovements to ASTCC?   Jason Williams
• [Asterisk-Users] Fax and spandsp   Kristof Hardy
• [Asterisk-Users] OH323 incoming audio stutter   Tony Mountifield
• [Asterisk-Users] Setting SIP username for CallerID   Gavin Hamill
• [Asterisk-Users] "friendly networks" via **   Ronald Wiplinger
• [Asterisk-Users] Fax detected, but no fax extension   Ronald Wiplinger
• [Asterisk-Users] Setting SIP username for CallerID   Ronald Wiplinger
• [Asterisk-Users] Cisco ATA Help   Sahil Gupta
• [Asterisk-Users] Setting SIP username for CallerID   Gavin Hamill
• [Asterisk-Users] RealTime ignoring switch => Realtime/context@realtime_ext   Ronald Wiplinger
• [Asterisk-Users] Fax detected, but no fax extension   Vladyslav
• [Asterisk-Users] help needed for sound device setup   Vladyslav
• [Asterisk-Users] Using voicemail independently from Asterisk PBX   João Amaro
• [Asterisk-Users] NAT issues   Steven Langley
• [Asterisk-Users] Setting SIP username for CallerID   Arunachala
• [Asterisk-Users] Fax detected, but no fax extension   Ronald Wiplinger
• [Asterisk-Users] TE410P PCI-slot   Tais M. Hansen
• [Asterisk-Users] Asterisk and VAD   Pavel Siderov
• [Asterisk-Users] Snom 360s and Asterisk   Colin E. McDonald
• [Asterisk-Users] Setting SIP username for CallerID   Gavin Hamill
• [Asterisk-Users] NAT issues   Eric Wieling aka ManxPower
• [Asterisk-Users] SIP Phone Compatability   Chris Mason (Lists)
• [Asterisk-Users] Asterisk and VAD   Eric Wieling aka ManxPower
• [Asterisk-Users] SPAM SPAM SPAM SAM SPAM SPAM SPAM   Al
• [Asterisk-Users] IAX realtime HELP   Paul Dracevich
• [Asterisk-Users] Cisco 2800 with Asterisk   Sharath Chandra
• [Asterisk-Users] NAT and only been able to have 1 SIP phone behind   Eric Wieling aka ManxPower
• [Asterisk-Users] SPAM SPAM SPAM SAM SPAM SPAM SPAM   Jean-Michel Hiver
• [Asterisk-Users] Setting SIP username for CallerID   Gavin Hamill
• [Asterisk-Users] SPAM SPAM SPAM SAM SPAM SPAM SPAM   Ronald Wiplinger
• [Asterisk-Users] Billing   David John Walsh
• [Asterisk-Users] Re: Fax detect/transfer problem?   Andrew C. Brown
• [Norton AntiSpam] [Asterisk-Users] IAX realtime HELP   Ronald Wiplinger
• [Asterisk-Users] Issues of reliability, hardware, platforms   Chris Mason (Lists)
• [Asterisk-Users] Sipura SPA-841 Phone Review   Gareth Blades
• [Asterisk-Users] Testing my TDM01A   Wilson Pickett
• [Asterisk-Users] TE410P PCI-slot   Domjan Attila
• [Asterisk-Users] Asterisk + Adit 600 questions   Daniel Nyström
• [Asterisk-Users] Conference solution for 100+ users   Stefan Märkle
• [Asterisk-Users] Rhino Channel Bank   Dan Goscomb
• [Asterisk-Users] Fax detected, but no fax extension   Vladyslav
• [Asterisk-Users] A question about queues   Brett, Gary
• [Asterisk-Users] IPSwitchBoard connects to CDR   Thorben Jensen
• [Asterisk-Users] capi segfault when incoming call is answered   Jason Williams
• [Asterisk-Users] Rhino Channel Bank   BJ Weschke
• [Asterisk-Users] Rhino Channel Bank   Dan Goscomb
• [Asterisk-Users] NuFone problems to non-na numbers   Pedro
• [Asterisk-Users] Help with [codec_g729.c:196 g729tolin_framein: Invalid data]   Doug Reid - Stormcorp
• [Asterisk-Users] RealTime ignoring switch=> Realtime/context@realtime_ext   Me
• [Asterisk-Users] capi segfault when incoming call is answered   Ivan Meic (Vox Mundi)
• [Asterisk-Users] US$200 bounty for * paging feature   Jeb Campbell
• [Asterisk-Users] US$200 bounty for * paging feature   Walt Reed
• [Asterisk-Users] A question about queues   Henry Devito
• [Asterisk-Users] FXO lines on TDM04B not responding   David Brodbeck
• [Asterisk-Users] Which free calling card app most suited forcommercial use?   Kanuri, Seshu (Company IT)
• [Asterisk-Users] NuFone problems to non-na numbers   steve at daviesfam.org
• [Asterisk-Users] General voip mailing list   Gerard Marcel
• [Asterisk-Users] A question about queues   C F
• [Asterisk-Users] CVS-HEAD and CheckGroup/SetGroup   Jason Williams
• [Asterisk-Users] A question about queues   David John Walsh
• [Asterisk-Users] G723.1 and G729 on Athlon 64   Ronald Wiplinger
• [Asterisk-Users] Conference solution for 100+ users   Sergio Veltri
• [Asterisk-Users] SPAM SPAM SPAM SAM SPAM SPAM SPAM   C F
• [Asterisk-Users] Anyone have a GXP-2000 working with Asterisk yet?   Andre Normandin
• [Asterisk-Users] OH323 incoming audio stutter   Nardis Dome
• [Asterisk-Users] TE410P PCI-slot   Tais M. Hansen
• [Asterisk-Users] G723.1 and G729 on Athlon 64   Kanuri, Seshu (Company IT)
• [Asterisk-Users] Wait in Dial String   David Choo
• [Asterisk-Users] Conference solution for 100+ users   Sergio Veltri
• [Asterisk-Users] G723.1 and G729 on Athlon 64   Marcin Kwiatkowski
• [Asterisk-Users] Can I do something with Caller-ID?   Ronald Wiplinger
• [Asterisk-Users] Sample AGI Scripts in C needed.   Moises Silva
• [Asterisk-Users] CVS Head and SetLanguage   Carlos Chavez
• [Asterisk-Users] Re: CVS-HEAD and CheckGroup/SetGroup   Noah Miller
• [Asterisk-Users] RxFax not hanging up...   Carlos Chavez
• [Asterisk-Users] Issues of reliability, hardware, platforms   Steve Kann
• [Asterisk-Users] A question about queues   Joseph Gutowski
• [Asterisk-Users] signate.com webcall   Moody
• [Asterisk-Users] SIP Phone Compatability   Kerry Garrison
• [Asterisk-Users] Cisco 7960 SIP registration???   List Receiver
• [Asterisk-Users] Which free calling card app most suitedforcommercial use?   Alex Vishnev
• [Asterisk-Users] UIP200   Daniel Salama
• [Asterisk-Users] Wait in Dial String   Josiah Bryan
• [Asterisk-Users] Transfer of incoming call from external to internal number   Paul Goodyear
• [Asterisk-Users] MF instead of DTMF   Michael B. Murdock
• [Asterisk-Users] Wait in Dial String   Robert Keller
• [Asterisk-Users] Which free calling card app most suitedforcommercial use?   Dave Kettmann
• [Asterisk-Users] Want to use Asterisk instead of existingMeridianNorstar system ... need some help   Robert Goodyear
• [Asterisk-Users] Can I do something with Caller-ID?   Marty Mastera
• [Asterisk-Users] Asterisk and T.38.   Rafael Gonzalez Lomeña
• [Asterisk-Users] Monitor via Manager question   Dana Olson
• [Asterisk-Users] Wait in Dial String   Robert Webb
• [Asterisk-Users] iaxtel.com ???   Ronald Wiplinger
• [Asterisk-Users] Cisco 7960 SIP registration???   end1r
• [Asterisk-Users] RE: Re: a simple question   Weiming Jiang
• [Asterisk-Users] US$200 bounty for * paging feature   trixter http://www.0xdecafbad.com
• [Asterisk-Users] NAT and only been able to have 1 SIP phone behind   Anton Krall
• [Asterisk-Users] CVS Head and SetLanguage   Alexander Lopez
• [Asterisk-Users] Transfer of incoming call from external to internal number   Paul Goodyear
• [Asterisk-Users] ADSI phones in the UK   Dan Goscomb
• [Asterisk-Users] signate.com webcall   Moody
• [Asterisk-Users] trying to figure out a few error messages in *   Doug Reid - Stormcorp
• [Asterisk-Users] Monitor via Manager question   mattf
• [Asterisk-Users] SIP Phone Compatability   Daniel Salama
• [Asterisk-Users] Rhino Channel Bank   Chuck Smith
• [Asterisk-Users] Monitor via Manager question   Alexander Lopez
• [Asterisk-Users] UIP200   Nathan C. Smith
• [Asterisk-Users] Monitor via Manager question   Dana Olson
• [Asterisk-Users] Transfer of incoming call from external tointernal number   Tim Thompson
• [Asterisk-Users] Route SIP calls to provider   iMRAN
• [Asterisk-Users] Line Noise UPDATE - If you've got line noise, read this   Paul
• [Asterisk-Users] Active calls not responding to entries   Tim Touhsaent
• [Asterisk-Users] Route SIP calls to provider   Wiley Siler
• [Asterisk-Users] signate.com webcall   William Boehlke
• [Asterisk-Users] Want to use Asterisk instead of existing MeridianNorstar system ... need some help   Jon Lewis
• [Asterisk-Users] SIP Phone Compatability   Kerry Garrison
• [Asterisk-Users] Zap Extensions unavailable after a call   Roberto Reiner Uhry
• [Asterisk-Users] Cisco 7960 SIP registration???   List Receiver
• [Asterisk-Users] Dialplan not showing up.   Michael Di Martino
• [Asterisk-Users] G723.1 and G729 on Athlon 64   Michael Bielicki
• [Asterisk-Users] FXS --> FXO Converter   Stephen
• [Asterisk-Users] ZyXEL Router Terrible Voice Quality   Greg Boehnlein
• [Asterisk-Users] Recommendations for IAX/SIP ATA   Ian Pattison
• [Asterisk-Users] Zap Extensions unavailable after a call   Tim Touhsaent
• [Asterisk-Users] Want to use Asterisk instead of existingMeridianNorstar system ... need some help   Wiley Siler
• [Asterisk-Users] Annoying SIP registration problem behind ?Linksys?   Tomas Florian
• [Asterisk-Users] BYOD provider other than broadvoice   Michael Lyszczek
• [Asterisk-Users] US$200 bounty for * paging feature   Walt Reed
• [Asterisk-Users] Grandstream GXP-2000   Daniel Salama
• [Asterisk-Users] US$200 bounty for * paging feature   Michael D Schelin
• [Asterisk-Users] signate.com webcall   Kanuri, Seshu (Company IT)
• [Asterisk-Users] Want to use Asterisk instead of existing MeridianNorstar system ... need some help   Jon Gabrielson
• [Asterisk-Users] Unable to create channel of type 'Zap'   Jaime Blanco
• [Asterisk-Users] SIP Phone Compatability   John Novack
• [Asterisk-Users] SIP Phone Compatability   Greg Boehnlein
• [Asterisk-Users] Want to use Asterisk instead of existing MeridianNorstar system ... need some help   John Novack
• [Asterisk-Users] choose audio codec with chan_sccp driver and 7920 wireless?   Mojo with Horan & Company, LLC
• [Asterisk-Users] RE: Re: a simple question   Matt Roth
• [Asterisk-Users] signate.com webcall   VOIP Consultant
• [Asterisk-Users] ZyXEL Router Terrible Voice Quality   Aza
• [Asterisk-Users] Line Noise UPDATE - If you've got line noise, read this   Paul
• [Asterisk-Users] spandsp   Daniel Salama
• [Asterisk-Users] Unable to create channel of type 'Zap'   Jaime Blanco
• [Asterisk-Users] Call waiting   Sascha Ferley
• [Asterisk-Users] 802.1p , precedence and TOS   Eugenio De Vena
• [Asterisk-Users] US$200 bounty for * paging feature   John D. Lewis
• licensing *sigh* (was Re: [Asterisk-Users] US$200 bounty for*paging feature)   Race Vanderdecken
• [Asterisk-Users] G723.1 and G729 on Athlon 64   CM Rahman Jr.
• [Asterisk-Users] PSTN->5300->asterisk(sip)   CM Rahman Jr.
• [Asterisk-Users] FW: Cisco 7920 - chan_sccp - asterisk@home .9   Mojo with Horan & Company, LLC
• [Asterisk-Users] Unable to create channel of type 'Zap'   Robert Webb
• [Asterisk-Users] RealTime ignoring switch=>Realtime/context@realtime_ext   Matthew Boehm
• [Asterisk-Users] US$200 bounty for * paging feature   Race Vanderdecken
• [Asterisk-Users] Unable to create channel of type 'Zap'   Tim Touhsaent
• [Asterisk-Users] signate.com webcall   VOIP Consultant
• [Asterisk-Users] Monitor via Manager question   Dana Olson
• [Asterisk-Users] Voicemail 2 Email   list
• [Asterisk-Users] US$200 bounty for * paging feature   trixter http://www.0xdecafbad.com
• [Asterisk-Users] BYOD provider other than broadvoice   pbx at itsngroup.com
• [Asterisk-Users] One-Way NO audio (and sometimes both ways)   Richard Lyman
• [Asterisk-Users] GotoIf in Stable 1.0.4   Mark Halverson
• [Asterisk-Users] Anyone have a GXP-2000 working with Asterisk yet?   Ariel Batista
• [Asterisk-Users] Large Asterisk Setup (~500 Concurrent Calls + Scalability)   Matt Roth
• [Asterisk-Users] Line Noise UPDATE - If you've got line noise, read this   Andre Normandin
• [Asterisk-Users] Grandstream GXP-2000   Andre Normandin
• [Asterisk-Users] Adit 3104 - user experiences?   Peter Hoppe
• [Asterisk-Users] Lucent EMRS PRI Card   Huddleston, Robert
• [Asterisk-Users] FXS --> FXO Converter   Peter Hoppe
• [Asterisk-Users] Help with [codec_g729.c:196 g729tolin_framein:Invalid data]   Brian Chrystal
• [Asterisk-Users] Anyone have a GXP-2000 working with Asterisk yet?   Brian Chrystal
• [Asterisk-Users] ZyXEL Router Terrible Voice Quality   snacktime
• [Asterisk-Users] Snom 360s and Asterisk   Mike Roelofs
• [Asterisk-Users] signate.com webcall   trixter http://www.0xdecafbad.com
• [Asterisk-Users] Cisco 7920 - chan_sccp - asterisk@home .9   Andy Hamilton
• [Asterisk-Users] TE110P card installation errors   Michael D Schelin
• [Asterisk-Users] Wait in Dial String   David Choo
• [Asterisk-Users] spa 3000 pstn with amp   Sruly
• [Asterisk-Users] General voip mailing list   James H. Thompson
• [Asterisk-Users] BYOD provider other than broadvoice   Sean Kennedy
• [Asterisk-Users] What do Digium use for tracking support tickets?   Mike Dent
• [Asterisk-Users] chan_unicall.c compile error   Fabio Vasco
• [Asterisk-Users] Unable to create channel of type 'Zap'   Rod Bacon
• [Asterisk-Users] Grandstream GXP-2000   Rod Bacon
• [Asterisk-Users] BYOD provider other than broadvoice   Dan Perik
• [Asterisk-Users] Route SIP calls to provider   Alexander Scheerschmidt
• [Asterisk-Users] Call waiting   Sascha Ferley
• [Asterisk-Users] TE110P   Michael D Schelin
• [Asterisk-Users] SIP Phone Compatability   Daniel Dziubanski
• [Asterisk-Users] Setting SIP username for CallerID   Joel Newkirk
• [Asterisk-Users] error in asterisk and LOTS OF log files generated   William Zhang
• [Asterisk-Users] chan_unicall.c compile error   Titux
• [Asterisk-Users] Line Noise UPDATE - If you've got line noise, read this   Henry Devito
• [Asterisk-Users] US$200 bounty for * paging feature   trixter http://www.0xdecafbad.com
• [Asterisk-Users] wcte11xp digium card   Titux
• [Asterisk-Users] RealTime ignoringswitch=>Realtime/context@realtime_ext   Me
• [Asterisk-Users] Call waiting   Henry Devito
• [Asterisk-Users] Line Noise UPDATE - If you've got line noise, read this   Henry Devito
• Ringing problems was [Asterisk-Users] TDM400P Revision question.   David Josephson
• [Asterisk-Users] signate.com webcall   William Boehlke
• [Asterisk-Users] Line Noise UPDATE - If you've got line noise, read this   Andrew Kohlsmith
• [Asterisk-Users] 5300 to asterisk   CM Rahman Jr.
• [Asterisk-Users] GotoIf in Stable 1.0.4   MF Hulber
• [Asterisk-Users] What do I need to get started?   jonr
• [Asterisk-Users] Large Asterisk Setup (~500 Concurrent Calls +Scalability)   Wiley Siler
• [Asterisk-Users] RealTime ignoringswitch=>Realtime/context@realtime_ext   Me
• [Asterisk-Users] simple question   Weiming Jiang
• [Asterisk-Users] US$200 bounty for * paging feature   Matt Klein
• [Asterisk-Users] mpg123 won't compile, arch x86_64   Michael Welter
• [Asterisk-Users] Volume of call waiting beeps   Michael Welter
• licensing *sigh* (was Re: [Asterisk-Users] US$200 bounty for*paging feature)   Andrew Kohlsmith
• [Asterisk-Users] Queuing with busy detect   Gregory Wiktor - ADCom Corp.
• [Asterisk-Users] Unable to create channel of type 'Zap'   Jaime Blanco
• [Asterisk-Users] US$200 bounty for * paging feature   trixter http://www.0xdecafbad.com
• [Asterisk-Users] asterisk home wiring question   snacktime
• [Asterisk-Users] TE110P card installation errors   Michael D Schelin
• [Asterisk-Users] TE110P   Matthew Boehm
• [Asterisk-Users] US$200 bounty for * paging feature   Dan Perik
• [Asterisk-Users] Recommended Linux Dist. for Asterisk   Paul Shiflet
• [Asterisk-Users] One-way audio   Cameron Beattie
• [Asterisk-Users] Motherboard failure with 2 Digium TE405P car ds   Gregory Wiktor - ADCom Corp.
• [Asterisk-Users] US$200 bounty for * paging feature   trixter http://www.0xdecafbad.com
• [Asterisk-Users] ZyXEL Router Terrible Voice Quality   Greg Boehnlein
• [Asterisk-Users] ZyXEL Router Terrible Voice Quality   Greg Boehnlein
• [Asterisk-Users] spandsp   Michael Bielicki
• [Asterisk-Users] signate.com webcall   Moody
• [Asterisk-Users] SIP Phone Compatability   Greg Boehnlein
• [Asterisk-Users] Anyone have a GXP-2000 working with Asterisk yet?   The VoIP Connection
• [Asterisk-Users] GotoIf in Stable 1.0.4   Moody
• [Asterisk-Users] Re: CVS-HEAD and CheckGroup/SetGroup   Sean A. Newton
• [Asterisk-Users] Large Asterisk Setup (~500 Concurrent Calls + Scalability)   Callum McGillivray
• [Asterisk-Users] Zap channels busy. Have to soft hangup.   Thomas
• [Asterisk-Users] Motherboard failure with 2 Digium TE405P car ds   Peter A. Ericksen
• [Asterisk-Users] Asterisk@home 0.9 and Cisco Callmanager   Dinesh
• [Asterisk-Users] Motherboard failure with 2 Digium TE405P car ds   mattf
• [Asterisk-Users] US$200 bounty for * paging feature   Damian Funnell
• [Asterisk-Users] Tonelist questions   David Josephson
• [Asterisk-Users] Fax Problems   Anton Krall
• [Asterisk-Users] Line Noise UPDATE - If you've got line noise, read this   Andre Normandin
• [Asterisk-Users] Anyone have a GXP-2000 working with Asterisk yet?   Anton Krall
• [Asterisk-Users] US$200 bounty for * paging feature   Matt Klein
• [Asterisk-Users] Fax Problems   Michael Bielicki
• [Asterisk-Users] US$200 bounty for * paging feature   Matt Klein
• [Asterisk-Users] Large Asterisk Setup (~500 Concurrent Calls + Scalability)   Michael Bielicki
• [Asterisk-Users] Fax Problems   trixter http://www.0xdecafbad.com
• [Asterisk-Users] Unable to create channel of type 'SIP'   Ronald Wiplinger
• [Asterisk-Users] Fax Problems   Anton Krall
• [Asterisk-Users] Unable to create channel of type 'SIP'   Tais M. Hansen
• [Asterisk-Users] Re: HELP: How to detect a hangup tone?   Charles Wang
• [Asterisk-Users] Asterisk Cisco Conection   Obihuan
• [Asterisk-Users] RE: Large Asterisk Setup (~500 Concurrent Calls + Scalability)   Stefan Märkle
• [Asterisk-Users] Annoying SIP registration problem behind ?Linksys?   Wilson Pickett
• [Asterisk-Users] What do I need to get started?   Wilson Pickett
• [Asterisk-Users] Asterisk@home 0.9 and Cisco Callmanager   Edgar de Leon at SESCAM
• [Asterisk-Users] Asterisk Cisco Connection   igil at europesip.com
• [Asterisk-Users] Problem using ztdummy kernelmodul with Kernel 2.6.8   harald.rinker at cocox.de
• [Asterisk-Users] Attended Transfer   harald.rinker at cocox.de
• [Asterisk-Users] Re: IPswitch: How to use speed dialing?   Ronald Wiplinger
• [Asterisk-Users] Dial W option usage   Master Abi
• [Asterisk-Users] DID ~ Extension   Nathaniel Angelo A. Torres (247talk)
• [Asterisk-Users] Transfer of incoming call from external tointernal number   Paul Goodyear
• [Asterisk-Users] capi problem with dialout   Paweł Staszewski
• [Asterisk-Users] GotoIf in Stable 1.0.4   oguer
• [Asterisk-Users] Problems with soundcards   Markus Hakansson
• [Asterisk-Users] capi problem with dialout   Michael Bielicki
• [Asterisk-Users] Annoying SIP registration problem behind ?Linksys?   Michael Bielicki
• [Asterisk-Users] 802.1p , precedence and TOS   SCollins
• [Asterisk-Users] asterisk home wiring question   Rich Adamson
• [Asterisk-Users] Line Noise UPDATE - If you've got line noise, read this   Andrew Kohlsmith
• [Asterisk-Users] How suppress echo   Michaël Delvoye
• [Asterisk-Users] Deny certain extension   Ashraf Salah
• [Asterisk-Users] Asterisk and T.38.   Jairo Buendia
• [Asterisk-Users] Deny certain extension   Andrew Kohlsmith
• [Asterisk-Users] How suppress echo   Andrew Kohlsmith
• [Asterisk-Users] How suppress echo   Gavin Hamill
• [Asterisk-Users] Anyone have a GXP-2000 working with Asterisk yet?   The VoIP Connection
• [Asterisk-Users] BYOD provider other than broadvoice   Adam Robins
• [Asterisk-Users] capi problem with dialout   Dave Cotton
• [Asterisk-Users] capi problem with dialout   Robert Webb
• [Asterisk-Users] Recommended Linux Dist. for Asterisk   Andrew Latham
• [Asterisk-Users] Issues of reliability, hardware, platforms   Chris Mason (Lists)
• [Asterisk-Users] i like my colors, thanks..   Matthew Boehm
• [Asterisk-Users] Recommended Linux Dist. for Asterisk   John Novack
• [Asterisk-Users] Large Asterisk Setup (~500 Concurrent Calls + Scalability)   Daniel Salama
• [Asterisk-Users] How suppress echo   Gareth Blades
• [Asterisk-Users] Bristuff and Belgium   David Masure
• [Asterisk-Users] Recommended Linux Dist. for Asterisk   Ariel Batista
• Odp: Re: [Asterisk-Users] capi problem with dialout   Paweł Staszewski
• [Asterisk-Users] hint priority and realtime in asterisk cvs-head   Eric Lawman
• [Asterisk-Users] Re: Fax Problems   Doug Meredith
• [Asterisk-Users] Queues configuration   Daniel Salama
• [Asterisk-Users] Recommended Linux Dist. for Asterisk   Michael George
• [Asterisk-Users] Cisco 7920 - chan_sccp - asterisk@home .9   Chuck Smith
• [Asterisk-Users] PBX replacement   igil at europesip.com
• [Asterisk-Users] Basic Setup Question   Terry Bomersbach
• [Asterisk-Users] 503 Error   doug at simflex.com
• [Asterisk-Users] * not send SIP Notify for IAX2 channel   Michael George
• [Asterisk-Users] dialup internet via Asterisk   Michael 'Moose' Dinn
• [Asterisk-Users] Re: CVS-HEAD and CheckGroup/SetGroup   Sean A. Newton
• [Asterisk-Users] Voicemail 2 Email   info at beprojects.com
• Odp: Re: [Asterisk-Users] capi problem with dialout   Peer Oliver Schmidt
• [Asterisk-Users] i like my colors, thanks..   Jeffrey C. Ollie
• [Asterisk-Users] Voicemail 2 Email   Peter Bowyer
• [Asterisk-Users] capi problem with dialout   Paweł Staszewski
• [Asterisk-Users] Recommended Linux Dist. for Asterisk   David Choo
• [Asterisk-Users] security   Altus Snyman
• [Asterisk-Users] Voicemail 2 Email   Gavin Hamill
• [Asterisk-Users] BYOD provider other than broadvoice   Trevor Harrison
• [Asterisk-Users] FXS --> FXO Converter   Stephen
• [Asterisk-Users] Max concurrent faxes using SpanDSP?   Bob Goddard
• [Asterisk-Users] Error in starting asterisk   Nathaniel Angelo A. Torres (247talk)
• [Asterisk-Users] Large Asterisk Setup (~500 Concurrent Calls + Scalability)   Matt Roth
• [Asterisk-Users] asterisk home wiring question   Dylan VanHerpen
• [Asterisk-Users] Large Asterisk Setup (~500 Concurrent Calls + Scalability)   Matt Roth
• [Asterisk-Users] One-way audio   Andrejus Stavickis
• [Asterisk-Users] BYOD provider other than broadvoice   Wiley Siler
• [Asterisk-Users] Error in starting asterisk   Matt Roth
• [Asterisk-Users] 802.1p , precedence and TOS   Eugenio De Vena
• [Asterisk-Users] Basic Setup Question   Andy Hamilton
• [Asterisk-Users] Re: Basic Setup Question   Noah Miller
• [Asterisk-Users] 802.1p , precedence and TOS   Steve Blair
• [Asterisk-Users] ZAP - outgoing call using different D-Channel each time ?   Walter Klomp
• [Asterisk-Users] Voicemail 2 Email   Andrew Niemantsverdriet
• [Asterisk-Users] TE406P & TE411P - Channelized voice and data T1/E1 PCI cards with Echo Cancellation   Joel Vandal
• [Asterisk-Users] asterisk home wiring question   Wiley Siler
• [Asterisk-Users] Recommended Linux Dist. for Asterisk   snacktime
• [Asterisk-Users] Libunicall Make Error   Nathaniel Angelo A. Torres (247talk)
• [Asterisk-Users] asterisk home wiring question   Brian Leyton
• [Asterisk-Users] Error in starting asterisk   Nathaniel Angelo A. Torres (247talk)
• [Asterisk-Users] asterisk home wiring question   Jon Pounder
• [Asterisk-Users] BYOD provider other than broadvoice   Adam Robins
• [Asterisk-Users] Recommended Linux Dist. for Asterisk   Mojo with Horan & Company, LLC
• [Asterisk-Users] capi problem with dialout   Peer Oliver Schmidt
• [Asterisk-Users] Error in starting asterisk   Nathaniel Angelo A. Torres (247talk)
• [Asterisk-Users] Error in starting asterisk   Robert Webb
• [Asterisk-Users] AgentCallbackLogin AckCall Problems (Current CVS-HEAD)   patrick-lists at csma.biz
• [Asterisk-Users] Max concurrent faxes using SpanDSP?   Anton Krall
• [Asterisk-Users] Recommended Linux Dist. for Asterisk   Dana Olson
• [Asterisk-Users] Recommended Linux Dist. for Asterisk   Paul
• [Asterisk-Users] ASTCC   Chris
• [Asterisk-Users] dialup internet via Asterisk   William M. Sandiford
• [Asterisk-Users] Queues configuration   Henry Devito
• [Asterisk-Users] 802.1p , precedence and TOS   Eugenio De Vena
• [Asterisk-Users] BYOD provider other than broadvoice   Daniel Dziubanski
• [Asterisk-Users] Re: chan_unicall.c compile error   Fabio Vasco
• [Asterisk-Users] Bristuff and Belgium   Kristof Hardy
• [Asterisk-Users] Queues configuration   Daniel Salama
• [Asterisk-Users] adding a thrid asterisk server   MDM
• [Asterisk-Users] Problems with app_dbodbc.c   Matt
• [Asterisk-Users] Large Asterisk Setup (~500 Concurrent Calls + Scalability)   Daniel Salama
• [Asterisk-Users] queue - transfer calls   Juan Manuel Coronado Zúñiga
• [Asterisk-Users] leastrecent queue option   Dana Olson
• [Asterisk-Users] BYOD provider other than broadvoice   Andrew Kohlsmith
• [Asterisk-Users] BYOD provider other than broadvoice   Wiley Siler
• [Asterisk-Users] ZAP - outgoing call using different D-Channel eachtime ?   Matthew Boehm
• [Asterisk-Users] One touch voicemail on Cisco 7940/60   Morris, Simon
• [Asterisk-Users] Line Noise UPDATE - If you've got line noise, read this   Paul
• [Asterisk-Users] Re: Starting with Asterisk-SIP   ruben cuevas rumin
• [Asterisk-Users] BYOD provider other than broadvoice   Brian Capouch
• [Asterisk-Users] Broadvoice gateways!   Gerard Marcel
• [Asterisk-Users] Asterisk and SER   Daniel Salama
• [Asterisk-Users] 802.1p , precedence and TOS   Steve Blair
• [Asterisk-Users] One touch voicemail on Cisco 7940/60   Alexander Lopez
• [Asterisk-Users] BYOD provider other than broadvoice   Mark Musone
• [Asterisk-Users] Multiple Line config help   kurt x
• [Asterisk-Users] BYOD provider other than broadvoice   Curren C. Calhoun
• [Asterisk-Users] Re: Libunicall Make Error   Fabio Vasco
• [Asterisk-Users] Asterisk and SER   Alexander Lopez
• [Asterisk-Users] Zap channels busy. Have to soft hangup.   Gregory Wiktor - ADCom Corp.
• [Asterisk-Users] Queues configuration   Henry Devito
• [Asterisk-Users] Broadvoice gateways!   trixter http://www.0xdecafbad.com
• [Asterisk-Users] One touch voicemail on Cisco 7940/60   Henry Devito
• [Asterisk-Users] Re: Starting with Asterisk-SIP   jltaylor
• [Asterisk-Users] Multiple Line config help   Henry Devito
• [Asterisk-Users] One touch voicemail on Cisco 7940/60   Ron Wellsted
• [Asterisk-Users] BYOD provider other than broadvoice   Luki
• [Asterisk-Users] Call waiting   Asterisk at Home
• [Asterisk-Users] Multiple Line config help   kurt x
• [Asterisk-Users] Broadvoice gateways!   Darren Wright
• [Asterisk-Users] BYOD provider other than broadvoice   Mark Musone
• [Asterisk-Users] using * for Internet call waiting   Gary Carr
• [Asterisk-Users] Re: Large Asterisk Setup (~500 Concurrent Calls +Scalability)   Iassen Hristov
• [Asterisk-Users] Problems with app_dbodbc.c   Matthew Boehm
• [Asterisk-Users] What do I need to get started?   jonr
• [Asterisk-Users] Queue member persistent stats   Matthew Boehm
• [Asterisk-Users] Broadvoice gateways!   trixter http://www.0xdecafbad.com
• [Asterisk-Users] BYOD provider other than broadvoice   Wiley Siler
• [Asterisk-Users] CVS-HEAD: Sip not paying attention to context   Matthew Boehm
• [Asterisk-Users] using * for Internet call waiting   Steve Prior
• [Asterisk-Users] using * for Internet call waiting   Dan Levine
• [Asterisk-Users] Provisioning lines 5 and 6 via TFTP   Robert Goodyear
• [Asterisk-Users] Large Asterisk Setup (~500 Concurrent Calls + Scalability)   Matt Roth
• [Asterisk-Users] forwarding Sip call to IAX and vice-versa   Daniel HAIDUC
• [Asterisk-Users] using * for Internet call waiting   jltaylor
• [Asterisk-Users] Recording Queue agents   David John Walsh
• [Asterisk-Users] using * for Internet call waiting   Mojo with Horan & Company, LLC
• [Asterisk-Users] Error in starting asterisk   Nathaniel Angelo A. Torres (247talk)
• [Asterisk-Users] wcte11xp digium card   Nathaniel Angelo A. Torres (247talk)
• [Asterisk-Users] One touch voicemail on Cisco 7940/60   William Suffill
• [Asterisk-Users] Problems with app_dbodbc.c   Matt Roth
• [Asterisk-Users] does ast_app_getdata() reset timeout?   Xu Wang
• [Asterisk-Users] Cisco 7920 - chan_sccp - asterisk@home .9   Paul A Brown
• [Asterisk-Users] Recording Queue agents   kyle Hagan
• [Asterisk-Users] Playing mp3's while recording voicemail   Rafal Kaniewski
• [Asterisk-Users] using * for Internet call waiting   jltaylor
• [Asterisk-Users] Queues configuration   Anton Krall
• [Asterisk-Users] Cisco 7920 - chan_sccp - asterisk@home .9   Mojo with Horan & Company, LLC
• [Asterisk-Users] Problems with app_dbodbc.c   Matt
• [Asterisk-Users] Demo phones with advertisement announcements   Ronald Wiplinger
• [Asterisk-Users] IAX peer   MDM
• [Asterisk-Users] dynamic port problem !!   CM Rahman Jr.
• [Asterisk-Users] FAX   Anton Krall
• [Asterisk-Users] FAX   Ronald Wiplinger
• [Asterisk-Users] FAX   Anton Krall
• [Asterisk-Users] GS ATA 286 goes deaf.   Chuck Keeter
• [Asterisk-Users] Looking for an IAX(2) or SIP DID provider for LA, Orlando and Chicago areas.   Carmi Weinzweig
• [Asterisk-Users] Music Onhold Problem   kritikus Araklidas
• [Asterisk-Users] asterisk@home 0.9 zap problems   Dinesh
• [Asterisk-Users] Email to Fax   Anton Krall
• [Asterisk-Users] Bug?   Kevin Bockman
• [Asterisk-Users] Bug?   Kevin Bockman
• [Asterisk-Users] 100 & AAH .9   Daniel Dziubanski
• [Asterisk-Users] Bug?   Ronald Wiplinger
• [Asterisk-Users] Asterisk Netgear FSM7326P and Cisco 7960 on VLAN   Allen Niven
• [Asterisk-Users] Bug?   Kevin Bockman
• [Asterisk-Users] Cisco 7920 - chan_sccp - asterisk@home .9   Chuck Smith
• [Asterisk-Users] Bug?   Ronald Wiplinger
• [Asterisk-Users] asterisk@home 0.9 zap problems   Henry Devito
• [Asterisk-Users] TE110p - universal voltage?   Craig Guy
• [Asterisk-Users] Re: Email to Fax   Justin Newman
• [Asterisk-Users] Playing mp3's while recording voicemail   Rod Bacon
• [Asterisk-Users] Asterisk Restart after crash   Chuck Smith
• [Asterisk-Users] Playing mp3's while recording voicemail   trixter http://www.0xdecafbad.com
• [Asterisk-Users] BYOD provider other than broadvoice   Luki
• [Asterisk-Users] Digium Card Issues   Sahil Gupta
• [Asterisk-Users] Asterisk Restart after crash   David Phelan
• [Asterisk-Users] Provider offering IAX and T.38 origination+termination?   Adam Megacz
• [Asterisk-Users] 503 Error   doug at simflex.com
• [Asterisk-Users] Re: Email to Fax   Anton Krall
• [Asterisk-Users] What do I need to get started?   Wilson Pickett
• [Asterisk-Users] LiveVoip status report   David Josephson
• [Asterisk-Users] Asterisk increased memory   Asterisk
• [Asterisk-Users] Echo cancelling with Adit 600   Daniel Nyström
• [Asterisk-Users] Line Noise UPDATE - If you've got line noise, read this   Paul
• [Asterisk-Users] How to attended/supervisor transfer   varadhan m
• [Asterisk-Users] dialling problem with astcc   wassim darwish
• [Asterisk-Users] One touch voicemail on Cisco 7940/60   bam
• [Asterisk-Users] Spandsp 0.0.2Pre15 with bristuff-0.2.0-RC8 Problem - Hangup   Peter De Schrijver
• [Asterisk-Users] Re: TE110p - universal voltage?   Tony Mountifield
• [Asterisk-Users] Provisioning lines 5 and 6 via TFTP   Ron Wellsted
• [Asterisk-Users] Provisioning lines 5 and 6 via TFTP   C F
• [Asterisk-Users] mpg123 won't compile, arch x86_64   C F
• [Asterisk-Users] X100P delayed ring on incoming calls?   bam
• [Asterisk-Users] Adit 3104 - user experiences?   C F
• [Asterisk-Users] X100P delayed ring on incoming calls?   Dave Cotton
• [Asterisk-Users] X100P delayed ring on incoming calls?   Gavin Hamill
• [Asterisk-Users] X100P delayed ring on incoming calls?   Joseph Gutowski
• [Asterisk-Users] X100P delayed ring on incoming calls?   bam
• [Asterisk-Users] One touch voicemail on Cisco 7940/60   Simon Morris
• [Asterisk-Users] DTFM tones almost completly muted.   Ian Hailey
• [Asterisk-Users] Dell PowerEdge SC1425 w/ TE405P?   Greg Boehnlein
• [Asterisk-Users] livevoip callerid   MF Hulber
• [Asterisk-Users] Echo cancelling with Adit 600   Peter Svensson
• [Asterisk-Users] DTFM tones almost completly muted.   Peter Bowyer
• [Asterisk-Users] Line Noise UPDATE - If you've got line noise, read this   Walt Reed
• [Asterisk-Users] asterisk@home 0.9 zap problems   Time Bandit
• [Asterisk-Users] X100P delayed ring on incoming calls?   Peter Corlett
• [Asterisk-Users] X100P delayed ring on incoming calls?   Gavin Hamill
• [Asterisk-Users] X100P delayed ring on incoming calls?   Adrian Chapman
• [Asterisk-Users] X100P delayed ring on incoming calls?   Peter Corlett
• [Asterisk-Users] Line Noise UPDATE - If you've got line noise, read this   Henry Devito
• [Asterisk-Users] Asterisk Restart after crash   Craig Guy
• [Asterisk-Users] Re: Email to Fax   Craig Guy
• [Asterisk-Users] Line Noise UPDATE - If you've got line noise, read this   Eric Wieling
• [Asterisk-Users] Asterisk Restart after crash   Guido Hecken
• [Asterisk-Users] IAX2 Error   Andrew Kohlsmith
• [Asterisk-Users] Asterisk Restart after crash   Chuck Smith
• [Asterisk-Users] Line Noise UPDATE - If you've got line noise, read this   Andrew Kohlsmith
• [Asterisk-Users] Asterisk transcoding   Georg Natsikos
• [Asterisk-Users] X100P delayed ring on incoming calls?   Rich Adamson
• [Asterisk-Users] callto: URL (URI) tag for dialing   Mark Elkins
• [Asterisk-Users] Error in starting asterisk   Rich Adamson
• [Asterisk-Users] IAX2 Error   Robson Ribeiro
• [Asterisk-Users] Re: chan_unicall.c compile error   Titux
• [Asterisk-Users] Mysql using Sip and voicemail   Ben Johnson
• [Asterisk-Users] using * for Internet call waiting   Gary Carr
• [Asterisk-Users] ASTCC   Dave Kettmann
• [Asterisk-Users] using * for Internet call waiting   Gary Carr
• [Asterisk-Users] using * for Internet call waiting   Gary Carr
• [Asterisk-Users] ASTCC   Dave Kettmann
• [Asterisk-Users] Asterisk acting as PBX + SIP Proxy ... possible?   Tomas Florian
• [Asterisk-Users] One touch voicemail on Cisco 7940/60   Simon Morris
• [Asterisk-Users] No such context/extension   MDM
• [Asterisk-Users] Alcaterl IP-touch phones   aref.cheikhrouhou at free.fr
• [Asterisk-Users] No such context/extension   Peter Bowyer
• [Asterisk-Users] Debugging zaphfc + PBX integration   Frank Sautter
• [Asterisk-Users] Dynamic queue member behaviour   Dana Olson
• [Asterisk-Users] One touch voicemail on Cisco 7940/60   Paul Dugas
• [Asterisk-Users] One touch voicemail on Cisco 7940/60   Henry Devito
• [Asterisk-Users] Alcaterl IP-touch phones   Hervé Mabille
• [Asterisk-Users] Debugging zaphfc + PBX integration   Gavin Hamill
• [Asterisk-Users] Asterisk Restart after crash   Craig Guy
• [Asterisk-Users] can't make my PRI dial out   Mark Phillips
• [Asterisk-Users] RE:Qwest opens 911 infrastructure to Vonage   Norm Zimon
• [Asterisk-Users] can't make my PRI dial out   Robert Webb
• [Asterisk-Users] Asterisk Restart after crash   Dave Cotton
• [Asterisk-Users] Error loading zaptel on RHEL4   Remco Barende
• [Asterisk-Users] MozPhone   Mark Elkins
• [Asterisk-Users] Asterisk and QSIG   Dpto. Técnico (Softec).
• [Asterisk-Users] can't make my PRI dial out   Mark Phillips
• [Asterisk-Users] can't make my PRI dial out   Andrew Kohlsmith
• [Asterisk-Users] Asterisk Restart after crash   Guido Hecken
• [Asterisk-Users] No sound with voicemail and musiconhold?!?   Antoine Courouble
• [Asterisk-Users] Asterisk Restart after crash   Alessio Focardi
• [Asterisk-Users] can't make my PRI dial out   Jerimiah Cole
• [Asterisk-Users] chan capi: Long incomingmsn line in capi.conf?   Stefan Helbing
• [Asterisk-Users] RE:Qwest opens 911 infrastructure to Vonage   Me
• [Asterisk-Users] can't make my PRI dial out   Mark Phillips
• [Asterisk-Users] can't make my PRI dial out   Dana Olson
• [Asterisk-Users] system wide speed dialing   Ronald Wiplinger
• [Asterisk-Users] changing port on IAX2 protocol   mak kwak
• [Asterisk-Users] TE11OP -> Mitel 200Sx??   Scott Wolfe
• [Asterisk-Users] routing in extensions.conf   Joao Pereira
• [Asterisk-Users] Asterisk Restart after crash   Dave Cotton
• [Asterisk-Users] can't make my PRI dial out   Marc Storck
• [Asterisk-Users] Playing mp3's while recording voicemail   Race Vanderdecken
• [Asterisk-Users] can't make my PRI dial out   Andrew Kohlsmith
• [Asterisk-Users] Mysql using Sip and voicemail   Race Vanderdecken
• [Asterisk-Users] ASTCC Database Creation   Mark Halverson
• [Asterisk-Users] can't make my PRI dial out   Robert Webb
• [Asterisk-Users] IAX2 Error   Andrew Kohlsmith
• [Asterisk-Users] IAXTel problems   Ronald Wiplinger
• [Asterisk-Users] using * for Internet call waiting   Nicolás Gudiño
• [Asterisk-Users] Demo phones with advertisement announcements   Race Vanderdecken
• [Asterisk-Users] OH323 incoming audio stutter   Michael Manousos
• [Asterisk-Users] ASterisk OH323.CONF Gateway & Gatekeeper   Michael Manousos
• [Asterisk-Users] Asterisk Restart after crash   Marc Storck
• [Asterisk-Users] voice pulse connect - no dtmf   Doug Harris
• [Asterisk-Users] livevoip callerid   Paul Fielding
• [Asterisk-Users] Re: TE110p - universal voltage?   Michael D Schelin
• [Asterisk-Users] BYOD provider other than broadvoice   Michael D Schelin
• [Asterisk-Users] Realtime with PostGreSQL ?   oguer
• [Asterisk-Users] QOS Routers   Max Clark
• [Asterisk-Users] Unable to create channel of type 'SIP'   Giovanni Powell
• [Asterisk-Users] IAX2 Error   Robson Ribeiro
• [Asterisk-Users] Line Noise UPDATE - If you've got line noise, read this   Paul
• [Asterisk-Users] using * for Internet call waiting   jltaylor
• [Asterisk-Users] Line Noise UPDATE - If you've got line noise, read this   Paul
• [Asterisk-Users] Quadbri & bristuff: can * respond only to 1 MSN and leave 1 number to other ISDN phones ?   Robert Rozman
• [Asterisk-Users] QOS Routers   Jared Watkins
• [Asterisk-Users] No sound with voicemail and musiconhold?!?   Ron Wellsted
• [Asterisk-Users] IAX channel   MDM
• [Asterisk-Users] QOS Routers   Andrew Pyles
• [Asterisk-Users] Realtime with PostGreSQL ?   G.Marshall
• [Asterisk-Users] QOS Routers   Andrew Pyles
• [Asterisk-Users] Digium Hardware Problem   Robert Bungener
• [Asterisk-Users] Dell PowerEdge SC1425 w/ TE405P?   Brian Leyton
• [Asterisk-Users] One touch voicemail on Cisco 7940/60   Ron Wellsted
• Re: [Asterisk-Users] routing in extensions.conf   Stefan Helbing
• [Asterisk-Users] QOS Routers   Jay Milk
• [Asterisk-Users] DTFM tones almost completly muted.   steve at daviesfam.org
• [Asterisk-Users] DTFM tones almost completly muted.   Peter Bowyer
• [Asterisk-Users] Quadbri & bristuff: can * respond only to 1 MSN and leave 1 number to other ISDN phones ?   Brancaleoni Matteo
• [Asterisk-Users] Mysql using Sip and voicemail   Ben Johnson
• [Asterisk-Users] Upgrade Cisco 7940/7960 firmware   Michael Welter
• [Asterisk-Users] Dell PowerEdge SC1425 w/ TE405P?   William Boehlke
• [Asterisk-Users] forwarding Sip call to IAX and vice-versa   Mikael Magnusson
• [Asterisk-Users] DID ~ Extension   Cameron Beattie
• [Asterisk-Users] Route SIP calls to provider   Cameron Beattie
• [Asterisk-Users] can't make my PRI dial out   Peter Svensson
• [Asterisk-Users] Re: Error loading zaptel on RHEL4   Remco Barende
• [Asterisk-Users] can't make my PRI dial out   Peter Svensson
• [Asterisk-Users] can't make my PRI dial out   tim panton
• [Asterisk-Users] Demo phones with advertisement announcements   Craig
• [Asterisk-Users] TE11OP -> Mitel 200Sx??   Henry Devito
• [Asterisk-Users] can't make my PRI dial out   Andrew Kohlsmith
• [Asterisk-Users] Re: can't make my PRI dial out   Edwin Groothuis
• [Asterisk-Users] Demo phones with advertisement announcements   Dean Collins
• [Asterisk-Users] Demo phones with advertisement announcements   trixter http://www.0xdecafbad.com
• [Asterisk-Users] Rejected connect attempt   G.Marshall
• [Asterisk-Users] Questions about a 7960 and images   Gregory Wiktor - ADCom Corp.
• [Asterisk-Users] QOS Routers   Gregory Wiktor - ADCom Corp.
• [Asterisk-Users] Recommendations for Spanish Voice Talent   George Pajari
• [Asterisk-Users] TE11OP -> Mitel 200Sx??   Dennis Walker
• [Asterisk-Users] Grandstream : low bandwidth codec (ilbc doesn't work, any other ? )   Robert Rozman
• [Asterisk-Users] Questions about a 7960 and images   Ron Wellsted
• [Asterisk-Users] can't make my PRI dial out   Mark Phillips
• [Asterisk-Users] Demo phones with advertisement announcements   Dean Collins
• [Asterisk-Users] Recommendations for Spanish Voice Talent   Gustavo Russo
• [Asterisk-Users] Asterisk + Cisco 2620   Christopher Barmonde
• [Asterisk-Users] TDM-fxo card and zttest - logic probem?   Rich Adamson
• [Asterisk-Users] QOS Routers   Jay Milk
• [Asterisk-Users] SS7 for *   Luciano Ramos
• [Asterisk-Users] Dell PowerEdge SC1425 w/ TE405P?   Greg Boehnlein
• [Asterisk-Users] voice pulse connect - no dtmf   Me
• [Asterisk-Users] Re: QOS Routers   Iassen Hristov
• [Asterisk-Users] can't make my PRI dial out   Andres
• [Asterisk-Users] Dell PowerEdge SC1425 w/ TE405P?   William Boehlke
• [Asterisk-Users] can't make my PRI dial out   Julian J. M.
• [Asterisk-Users] Dell PowerEdge SC1425 w/ TE405P?   Ben Hencke
• [Asterisk-Users] can't make my PRI dial out   Ken Godee
• [Asterisk-Users] Dell PowerEdge SC1425 w/ TE405P?   Ariel Batista
• [Asterisk-Users] Demo phones with advertisement announcements   Ronald Wiplinger
• [Asterisk-Users] Line Noise UPDATE - If you've got line noise, read this   Rich Adamson
• [Asterisk-Users] TE11OP -> Mitel 200Sx??   Michael D Schelin
• [Asterisk-Users] Re: QOS Routers   Chris Mason (Lists)
• [Asterisk-Users] TE11OP -> Mitel 200Sx??   Scott Wolfe
• [Asterisk-Users] Demo phones with advertisement announcements   trixter http://www.0xdecafbad.com
• [Asterisk-Users] IAX help   Michael DiMartino
• [Asterisk-Users] Libunicall Compile Error   Nathaniel Angelo A. Torres (247talk)
• [Asterisk-Users] console /distinctive ring   William C. Lohr Jr.
• [Asterisk-Users] Questions about a 7960 and images   Gregory Wiktor - ADCom Corp.
• [Asterisk-Users] TE11OP -> Mitel 200Sx??   Henry Devito
• [Asterisk-Users] console /distinctive ring   William C. Lohr Jr.
• [Asterisk-Users] TE11OP -> Mitel 200Sx??   Henry Devito
• [Asterisk-Users] wIPPhone with Asterisk   Josh Alberts
• [Asterisk-Users] Large Asterisk Setup (~500 Concurrent Calls + Scalability)   Brian Roy
• [Asterisk-Users] New Asterisk installation problem - 7290 calling problem   Paul A Brown
• [Asterisk-Users] can't make my PRI dial out   Andrew Kohlsmith
• [Asterisk-Users] TE11OP -> Mitel 200Sx??   Dennis Walker
• [Asterisk-Users] if outgoing call fails with provider 1 then auto try provider 2   Thomas Miller
• [Asterisk-Users] if outgoing call fails with provider 1 then auto try provider 2   Andrew Kohlsmith
• [Asterisk-Users] Mitel Ip phone ?   Pedro
• [Asterisk-Users] Large Asterisk Setup (~500 Concurrent Calls + Scalability)   Daniel Salama
• [Asterisk-Users] Questions about a 7960 and images   Henry Devito
• [Asterisk-Users] TE11OP -> Mitel 200Sx??   Henry Devito
• [Asterisk-Users] Cisco 7960 won't register as SIP device   List Receiver
• [Asterisk-Users] X100P delayed ring on incoming calls?   Joseph Gutowski
• [Asterisk-Users] Cisco 7960 won't register as SIP device   Henry Devito
• [Asterisk-Users] Asterisk Restart after crash   Craig Guy
• [Asterisk-Users] Cisco 7960 won't register as SIP device   List Receiver
• [Asterisk-Users] IP Phones and firewalls ...   Brian Watters
• [Asterisk-Users] Best of the best of IP Phones   Chris Coulthurst
• [Asterisk-Users] voice pulse connect - no dtmf   Justin Richards
• [Asterisk-Users] Connecting Elmeg CS100 ISDN system phones to Asterisk   Taco Scargo
• [Asterisk-Users] Most affordable 8-port NT-capable ISDN card   Taco Scargo
• [Asterisk-Users] voice pulse connect - no dtmf   Me
• [Asterisk-Users] Provisioning lines 5 and 6 via TFTP   Robert Goodyear
• [Asterisk-Users] QOS Routers   Robert Goodyear
• [Asterisk-Users] Hotel billing in IPSwitchBoard   Thorben Jensen
• Fwd: [Asterisk-Users] Asterisk transcoding   Georg Natsikos
• [Asterisk-Users] IAX help   Peter Bowyer
• [Asterisk-Users] Quadbri & bristuff: can * respond only to 1 MSN and leave 1 number to other ISDN phones ?   Michiel van Baak
• [Asterisk-Users] Hotel billing in IPSwitchBoard   Guido Hecken
• [Asterisk-Users] Failed to authenticate   lie ka
• [Asterisk-Users] ast_expr.y:243 to_integer:Overflow   lie ka
• [Asterisk-Users] chan_sip.c:7174 handle_request : Failed to authenticate user   lie ka
• [Asterisk-Users] usb phone(AU-100) and usb phone adapter(TJ560B)   Zen Kato
• [Asterisk-Users] Dial While on IVR   Robson Ribeiro
• [Asterisk-Users] Hotel billing in IPSwitchBoard   Chris Mason (Lists)
• [Asterisk-Users] Re: QOS Routers   MF Hulber
• [Asterisk-Users] can't make my PRI dial out   Mark Phillips
• [Asterisk-Users] can't make my PRI dial out   Mark Phillips
• [Asterisk-Users] Re: Hotel billing in IPSwitchBoard   tgj
• [Asterisk-Users] No sound with voicemail and musiconhold?!?   Antoine Courouble
• [Asterisk-Users] can't make my PRI dial out   Mark Phillips
• [Asterisk-Users] Re: Hotel billing in IPSwitchBoard   Chris Mason (Lists)
• [Asterisk-Users] Re: can't make my PRI dial out   Noah Miller
• [Asterisk-Users] Cisco 7960 won't register as SIP device   Henry Devito
• [Asterisk-Users] IP Phones and firewalls ...   Henry Devito
• [Asterisk-Users] PA168 ip phone setup iax2 to LiveVoip   C W Nel
• [Asterisk-Users] Re: IPswitch: How to use speed dialing?   Ronald Wiplinger
• [Asterisk-Users] if outgoing call fails with provider 1 then auto try provider 2   Rich Adamson
• [Asterisk-Users] OctoBRI and 2.6kernel   Terry Wade
• [Asterisk-Users] Cisco 7960 won't register as SIP device   List Receiver
• [Asterisk-Users] if outgoing call fails with provider 1 then auto try provider 2   Thomas Miller
• [Asterisk-Users] if outgoing call fails with provider 1 then auto try provider 2   Thomas Miller
• [Asterisk-Users] Cisco 7960 won't register as SIP device   Robert Webb
• [Asterisk-Users] Hotel billing in IPSwitchBoard   Steve Rawlings
• [Asterisk-Users] Re: Hotel billing in IPSwitchBoard   Mathew McKernan
• [Asterisk-Users] Cisco 7960 won't register as SIP device   Robert Webb
• [Asterisk-Users] Cisco 7960 won't register as SIP device   List Receiver
• [Asterisk-Users] Quadbri & bristuff: can * respond only to 1 MSN and leave 1 number to other ISDN phones ?   Michiel van Baak
• [Asterisk-Users] Re: Hotel billing in IPSwitchBoard   tgj
• [Asterisk-Users] Re: Re: Hotel billing in IPSwitchBoard   tgj
• [Asterisk-Users] Hotel billing in IPSwitchBoard   Chris Mason (Lists)
• [Fwd: FW: [Asterisk-Users] IAX help]   Michael DiMartino
• [Asterisk-Users] OctoBRI and 2.6kernel   Michael Bielicki
• [Fwd: FW: [Asterisk-Users] IAX help]   Peter Bowyer
• [Asterisk-Users] Re: Hotel billing in IPSwitchBoard   David John Walsh
• [Fwd: FW: [Asterisk-Users] IAX help]   Michael DiMartino
• [Fwd: FW: [Asterisk-Users] IAX help]   Andrew Kohlsmith
• [Fwd: FW: [Asterisk-Users] IAX help]   Michael DiMartino
• [Fwd: FW: [Asterisk-Users] IAX help]   Andrew Kohlsmith
• [Asterisk-Users] Re: Hotel billing in IPSwitchBoard   Chris Mason (Lists)
• [Fwd: FW: [Asterisk-Users] IAX help]   Michael DiMartino
• [Asterisk-Users] Dialing problem - Cisco 7290 to anything   Paul A Brown
• [Asterisk-Users] ASTERISK PROGRAMER   Franz
• [Asterisk-Users] ztcfg doesn't do anything from /etc/rc.d/rc.local   Remco Barende
• [Asterisk-Users] ASTERISK PROGRAMER   Kerry Garrison
• [Asterisk-Users] ztcfg doesn't do anything from /etc/rc.d/rc.local   Chris
• [Asterisk-Users] ASTERISK PROGRAMER   Matt Klein
• [Asterisk-Users] ASTERISK PROGRAMER   Bob Goddard
• [Asterisk-Users] IP Phones and firewalls ...   Brian Watters
• [Asterisk-Users] Best of the best of IP Phones   Mark Phillips
• [Asterisk-Users] How to replace VM busy.gsm and unavail.gsm messages with custom files   Chris Coulthurst
• [Asterisk-Users] Recommended Linux Dist. for Asterisk   Umair Bari
• [Asterisk-Users] Provisioning Lines   Manjit Riat
• [Asterisk-Users] ASTERISK PROGRAMER   Gary Stimson
• [Asterisk-Users] ASTERISK PROGRAMER   Matt Klein
• [Asterisk-Users] Best of the best of IP Phones   Ron Wellsted
• [Asterisk-Users] Re: Hotel billing in IPSwitchBoard   Henry Devito
• [Asterisk-Users] Provisioning Lines   Ron Wellsted
• [Asterisk-Users] How to replace VM busy.gsm and unavail.gsmmessages with custom files   Chris Coulthurst
• [Asterisk-Users] ignorepat doesn't work   Jaime Blanco
• [Asterisk-Users] ignorepat doesn't work   Alexander Lopez
• [Asterisk-Users] Re: ztcfg doesn't do anything from /etc/rc.d/rc.local   Tony Mountifield
• [Asterisk-Users] Provisioning Lines   Tom
• [Asterisk-Users] Provisioning Lines   Chris Mason (Lists)
• [Asterisk-Users] Best of the best of IP Phones   Gregory Wiktor - ADCom Corp.
• [Asterisk-Users] ignorepat doesn't work   Eric Wieling aka ManxPower
• [Asterisk-Users] ztcfg doesn't do anything from /etc/rc.d/rc.local   Eric Wieling aka ManxPower
• [Asterisk-Users] callto: URL (URI) tag for dialing   Gregory Wiktor - ADCom Corp.
• [Asterisk-Users] ztcfg doesn't do anything from /etc/rc.d/rc.local   Chris
• [Asterisk-Users] ignorepat doesn't work   Jerry
• [Asterisk-Users] SIP registration behind Linksys WRT54G   Tomas Florian
• [Asterisk-Users] SIP registration behind Linksys WRT54G   Scott Henderson
• [Asterisk-Users] SIP registration behind Linksys WRT54G   Tomas Florian
• [Asterisk-Users] ADSI Input from 480 keypad?   Chris Coulthurst
• [Asterisk-Users] TE11OP -> Mitel 200Sx??   Scott Wolfe
• [Asterisk-Users] SIP registration behind Linksys WRT54G   Scott Henderson
• [Asterisk-Users] SIP registration behind Linksys WRT54G   Luki
• [Asterisk-Users] SIP registration behind Linksys WRT54G   Mojo-Jojo
• [Asterisk-Users] SIP registration behind Linksys WRT54G   Mojo-Jojo
• [Asterisk-Users] SIP registration behind Linksys WRT54G   Tomas Florian
• [Asterisk-Users] SIP registration behind Linksys WRT54G   Rich Adamson
• [Asterisk-Users] SIP registration behind Linksys WRT54G   Pedro
• [Asterisk-Users] ztcfg doesn't do anything from /etc/rc.d/rc.local   Remco Barende
• [Asterisk-Users] ignorepat doesn't work   Jaime Blanco
• [Asterisk-Users] SIP registration behind Linksys WRT54G   Tomas Florian
• [Asterisk-Users] Best of the best of IP Phones   Eric Wieling aka ManxPower
• [Asterisk-Users] SIP registration behind Linksys WRT54G - URI?   Tomas Florian
• [Asterisk-Users] TE11OP -> Mitel 200Sx??   Henry Devito
• [Asterisk-Users] Quadbri & bristuff: can * respond only to 1 MSN and leave 1 number to other ISDN phones ?   Michiel van Baak
• [Asterisk-Users] Re: IPswitch: How to use speed dialing?   Ronald Wiplinger
• [Asterisk-Users] help:Memory Consumption   Yusuf Iqbal
• [Asterisk-Users] ztdummy and Debian   Thore
• [Asterisk-Users] Netjet/Linux/Asterisk issue   Bob Purdon
• AW: [Asterisk-Users] ztdummy and Debian   Manuel Schroeder
• [Asterisk-Users] ztdummy and Debian   Samuel T. Cossette
• [Asterisk-Users] Meetme Announcement   Mohamed Farid
• [Asterisk-Users] QSIG.   Dpto. Técnico (Softec).
• [Asterisk-Users] ztdummy and Debian   Thore
• AW: [Asterisk-Users] ztdummy and Debian   Manuel Schroeder
• [Asterisk-Users] inband DTMF with IAX   Alex Brett
• [Asterisk-Users] Asterisk2mp3   Stiffe
• [Asterisk-Users] VSAT and Asterisk   Chris Mason (Lists)
• [Asterisk-Users] Asterisk management GUI   Ezabi
• [Asterisk-Users] Re: Asterisk2mp3   Stiffe
• [Asterisk-Users] Astcc Working but Can't Make The Call   chawki hammoud
• [Asterisk-Users] Re: Meetme Announcement   Tony Mountifield
• [Asterisk-Users] Registerport 5060 or 1720?   Ronald Wiplinger
• [Asterisk-Users] T1 E&M wink issues - bad int'l dial-out and occasional dropped calls   bill black
• [Asterisk-Users] QOS Routers   Michael Graves
• [Asterisk-Users] TE11OP -> Mitel 200Sx??   Scott Wolfe
• [Asterisk-Users] Re: QOS Routers   Michael Graves
• [Asterisk-Users] Feedback on Junction Networks conferences?   Moody
• [Asterisk-Users] Registerport 5060 or 1720?   Charles Wang
• [Asterisk-Users] cidsignailling mode question   Alejandro G
• [Asterisk-Users] How can several Asterisk boxes working together?   Ronald Wiplinger
• [Asterisk-Users] DTFM tones almost completly muted.   Ian Hailey
• [Asterisk-Users] DTFM tones almost completly muted.   Ian Hailey
• [Asterisk-Users] NEED HELP PROGRAMING ASTERISK VoIP NETWORK   Franz
• [Asterisk-Users] Fritz+chan_misdn - any working example ?   Robert Rozman
• [Asterisk-Users] Re: Asterisk-Users Digest, Vol 9, Issue 215   Kumara Jayaweera
• [Asterisk-Users] What is the best client's protocol for my softphones   Kumara Jayaweera
• [Asterisk-Users] Re: Asterisk-Users Digest, Vol 9, Issue 215   Brian Watters
• [Asterisk-Users] ignorepat doesn't work   Jerry
• [Asterisk-Users] sm bounty validate length of e164/e212 number for all countries   Thomas Miller
• [Asterisk-Users] ignorepat doesn't work   Eric Wieling aka ManxPower
• [Asterisk-Users] What is the best client's protocol for my softphones   Time Bandit
• [Asterisk-Users] How to prevent native bridging between SIP channels   Wolf N. Paul
• [Asterisk-Users] How to prevent native bridging between SIP channels   Marc Storck
• [Asterisk-Users] Astcc Working but Can't Make The Call   chawki hammoud
• [Asterisk-Users] Can Asterisk do the following for me ?   asterisk at top-consulting.net
• [Asterisk-Users] Best of the best of IP Phones   Greg Boehnlein
• [Asterisk-Users] Quantum A800 (SIP) <-> Asterisk Config   Bashir Ullah - www.Lamsre.Com
• [Asterisk-Users] Can Asterisk do the following for me ?   Kerry Garrison
• [Asterisk-Users] Need info on necessary config of new T1/PRIs   Jess Coburn
• [Asterisk-Users] T1 E&M wink issues - bad int'l dial-out andoccasional dropped calls   bill black
• [Asterisk-Users] SIP registration behind Linksys WRT54G - FIXED :-)   Tomas Florian
• Which protocol? was Re: [Asterisk-Users] Re: Asterisk-Users Digest, Vol 9, Issue 215   tim panton
• AW: [Asterisk-Users] Can Asterisk do the following for me ?   Manuel Schroeder
• [Asterisk-Users] Can Asterisk do the following for me ?   asterisk at top-consulting.net
• [Asterisk-Users] Can Asterisk do the following for me ?   Dean Collins
• [Asterisk-Users] T1 E&M wink issues - bad int'l dial-out and occasional dropped calls   Richard Lyman
• [Asterisk-Users] TE11OP -> Mitel 200Sx??   Henry Devito
• [Asterisk-Users] DTMF intermittently stops working   Kevin P. Fleming
• [Asterisk-Users] g729 passthrough?   Brian Capouch
• [Asterisk-Users] T1 E&M wink issues - bad int'l dial-out and occasional dropped calls   Richard Lyman
• [Asterisk-Users] Best of the best of IP Phones   Eric Wieling aka ManxPower
• [Asterisk-Users] g729 passthrough?   Brian Capouch
• [Asterisk-Users] T1 E&M false busy after dial   John Ackley
• [Asterisk-Users] g729 passthrough?   jltaylor
• [Asterisk-Users] Zaphfc problem   Michał Mosiewicz
• [Asterisk-Users] signate.com webcall   adriavidal at gmail.com
• [Asterisk-Users] ztdummy and Debian   Tzafrir Cohen
• [Asterisk-Users] T1 E&M false busy after dial   jltaylor
• [Asterisk-Users] ztcfg doesn't do anything from /etc/rc.d/rc.local   Tzafrir Cohen
• [Asterisk-Users] Transfers fails, even after upgrade to 1.0.7   Pablo Alsina
• [Asterisk-Users] signate.com webcall   Ronald Wiplinger
• [Asterisk-Users] AGI problem on Zaptel channel   YANG TAO
• [Asterisk-Users] signate.com webcall   Brian Watters
• [Asterisk-Users] Static and echo on PRI   Mark Johnson
• [Asterisk-Users] signate.com webcall   Joseph Gutowski
• [Asterisk-Users] Static and echo on PRI   Michael Welter
• [Asterisk-Users] T1 E&M false busy after dial   John Ackley
• [Asterisk-Users] Asterisk best practices   Craig Simon
• [Asterisk-Users] Static and echo on PRI   Mark Johnson
• [Asterisk-Users] T1 E&M false busy after dial   jltaylor
• [Asterisk-Users] g729 passthrough?   Asterisk guy
• [Asterisk-Users] Problems with gotoiftime and cvs head   Anton Krall
• [Asterisk-Users] Asterisk best practices   Brian Watters
• [Asterisk-Users] What software and types of connections are used by VOIP providers   Bartosz Wegrzyn - asterisk
• [Asterisk-Users] Problems with gotoiftime and cvs head   Anton Krall
• [Asterisk-Users] Static and echo on PRI   Michael Welter
• [Asterisk-Users] Why can't I hear audio?   Michael D Schelin
• [Asterisk-Users] Asterisk best practices   Sig Lange
• [Asterisk-Users] Can Asterisk do the following for me ?   Lee Howard
• [Asterisk-Users] Can Asterisk do the following for me ?   Dean Collins
• [Asterisk-Users] Can Asterisk do the following for me ?   Kerry Garrison
• [Asterisk-Users] Failed to authenticate   lie ka
• [Asterisk-Users] Trouble with call parking/transfer   Tim Pushor
• [Asterisk-Users] Can Asterisk do the following for me ?   Juergen K. Zick
• [Asterisk-Users] NEED HELP PROGRAMING ASTERISK VoIP NETWORK   Jean-Michel Hiver
• [Asterisk-Users] matching sip connection under sip.conf   CM Rahman Jr.
• [Asterisk-Users] g729 passthrough?   Brian Capouch
• [Asterisk-Users] PA168 ip phone setup iax2 to LiveVoip   Jessie Mabanglo
• [Asterisk-Users] Quantum A800 (SIP) <-> Asterisk Config   Jessie Mabanglo
• [Asterisk-Users] Playing mp3's while recording voicemail   Rafal Kaniewski
• [Asterisk-Users] No busy tone when dialing out over ISDN with Polycom 500 IP   Kib Eki
• [Asterisk-Users] NEED HELP PROGRAMING ASTERISK VoIP NETWORK   Thierry Wehr
• [Asterisk-Users] Re: How to prevent native bridging between SIP channels   Wolf N. Paul
• [Asterisk-Users] Zap event On hook(1) handling problem   Vincent
• [Asterisk-Users] signaling during a call   Tulika Pradhan
• [Asterisk-Users] [ANNOUNCEMENT] Amatix InstantPBX   Amatisoft SRL
• [Asterisk-Users] i like my colors, thanks..   Andreas Sikkema
• [Asterisk-Users] Can Asterisk do the following for me ?   asterisk at top-consulting.net
• [Asterisk-Users] asteriks without h/w   chaitanya kiran
• [Asterisk-Users] NEED HELP PROGRAMING ASTERISK VoIP NETWORK   Jean-Michel Hiver
• [Asterisk-Users] Re: Can Asterisk do the following for me ?   Tony Mountifield
• [Asterisk-Users] Cisco's description of echo   Chris Mason
• [Asterisk-Users] each 64K channel's ABCD bits for E100P Digium Cards.   Frank Lin
• [Asterisk-Users] UK (english) sound files   Alex Barnes
• [Asterisk-Users] UK (english) sound files   Alex Barnes
• [Asterisk-Users] Can Asterisk do the following for me ?   Andrew Kohlsmith
• [Asterisk-Users] Can Asterisk do the following for me ?   Andrew Kohlsmith
• [Asterisk-Users] Large Asterisk Setup (~500 Concurrent Calls + Scalability)   Adam Goryachev
• [Asterisk-Users] UK (english) sound files   David John Walsh
• [Asterisk-Users] X100P delayed ring on incoming calls?   Peter Corlett
• [Asterisk-Users] Trouble with call parking/transfer   David John Walsh
• [Asterisk-Users] chan_capi: no dialstatus, no causes, no branches   Jason Williams
• [Asterisk-Users] What small PC can take 8 FXS + 8 FXO cards   ht at phonitel.com
• [Asterisk-Users] What small PC can take 8 FXS + 8 FXO cards   David John Walsh
• [Asterisk-Users] Re: Best of the best of IP Phones   Sergio
• [Asterisk-Users] UK (english) sound files   Bicom Systems
• [Asterisk-Users] Trouble with call parking/transfer   Julian J. M.
• [Asterisk-Users] What small PC can take 8 FXS + 8 FXO cards   Andrew Kohlsmith
• [Asterisk-Users] What small PC can take 8 FXS + 8 FXO cards   ht at phonitel.com
• [Asterisk-Users] UK (english) sound files   Ian Pattison
• [Asterisk-Users] UK (english) sound files   Ron Wellsted
• [Asterisk-Users] UK (english) sound files   Alex Barnes
• [Asterisk-Users] UK (english) sound files   David John Walsh
• [Asterisk-Users] need resources to include iax softphone functionality in vb6 app   Steven Langley
• [Asterisk-Users] Cisco's description of echo   Rich Adamson
• [Asterisk-Users] What small PC can take 8 FXS + 8 FXO cards   Karl H. Putz
• [Asterisk-Users] Call Recording via monitor   Chris Mason (Lists)
• [Asterisk-Users] Cisco's description of echo   Steve Underwood
• [Asterisk-Users] Cisco ATA 186   Serge Matveev
• [Asterisk-Users] UK (english) sound files   Bicom Systems
• [Asterisk-Users] What small PC can take 8 FXS + 8 FXO cards   Andrew Kohlsmith
• [Asterisk-Users] Cisco's description of echo   Andrew Kohlsmith
• [Asterisk-Users] Cisco's description of echo   Chris Mason (Lists)
• [Asterisk-Users] asttapi and identapop pro   Tavis Patterson
• [Asterisk-Users] i like my colors, thanks..   Matthew Boehm
• [Asterisk-Users] Basic telephony hardware questions   Sudhakar Chandra
• [Asterisk-Users] Trouble with call parking/transfer   Tim Pushor
• [Asterisk-Users] using * for Internet call waiting   Gary Carr
• [Asterisk-Users] Repost: Dialing problem - Cisco 7290 to anything   Paul A Brown
• [Asterisk-Users] sm bounty validate length of e164/e212 number for all countries   Adam Goryachev
• [Asterisk-Users] Cisco's description of echo   Rich Adamson
• [Asterisk-Users] Call Recording via monitor   Eric Alexander
• [Asterisk-Users] Cisco's description of echo   Rich Adamson
• [Asterisk-Users] TDM card periodic buzz   Trent Tuggle
• [Asterisk-Users] Call Recording via monitor   Greg Eaton
• [Asterisk-Users] TDM card periodic buzz   Andrew Kohlsmith
• [Asterisk-Users] BYOD provider other than broadvoice   Paul
• [Asterisk-Users] astrecipes v2.0   lenz
• [Asterisk-Users] BYOD provider other than broadvoice   Kerry Garrison
• [Asterisk-Users] OH323 incoming audio stutter   Rafael J. Risco G.V.
• [Asterisk-Users] QoS Help and survey   Noah Miller
• [Asterisk-Users] Re: astrecipes v2.0   Bruno Hertz
• [Asterisk-Users] Re: OH323 incoming audio stutter   Tony Mountifield
• [Asterisk-Users] Re: astrecipes v2.0   lenz
• [Asterisk-Users] Alternatives to SpanDSP??   Jeremy Melanson
• [Asterisk-Users] Re: astrecipes v2.0   Tony Mountifield
• [Asterisk-Users] Re: astrecipes v2.0   lenz
• [Asterisk-Users] Realtime & voicemail   Edwin Horton
• [Asterisk-Users] Grandstream ATA 286 problems   Anton Krall
• [Asterisk-Users] Re: astrecipes v2.0   David John Walsh
• [Asterisk-Users] Static and echo on PRI   Mark Johnson
• [Asterisk-Users] Large Asterisk Setup (~500 Concurrent Calls + Scalability)   Matt Roth
• [Asterisk-Users] Random SIP Phone Problem   Asterisk List
• [Asterisk-Users] Realtime & voicemail   Matthew Boehm
• [Asterisk-Users] Alternatives to SpanDSP??   Eric Wieling aka ManxPower
• [Asterisk-Users] Broadvoice Down?   Max Clark
• [Asterisk-Users] Re: astrecipes v2.0   lenz
• [Asterisk-Users] No busy tone when dialing out over ISDN with Polycom 500 IP   Eric Wieling aka ManxPower
• [Asterisk-Users] Trouble with call parking/transfer   Eric Wieling aka ManxPower
• [Asterisk-Users] Broadvoice Down?   Jerry Geis
• [Asterisk-Users] chan capi: Long incomingmsn line in capi.conf?   Peter Braidwood
• [Asterisk-Users] UK (english) sound files   Mark Phillips
• [Asterisk-Users] Broadvoice Down?   Kerry Garrison
• [Asterisk-Users] Realtime & voicemail   Joe Dennick
• [Asterisk-Users] Why can't I hear audio?   Michael D Schelin
• [Asterisk-Users] Broadvoice Down?   JD Austin
• [Asterisk-Users] Trouble with call parking/transfer   Tim Pushor
• [Asterisk-Users] Broadvoice Down?   List Receiver
• [Asterisk-Users] Alternatives to SpanDSP??   Jeremy Melanson
• [Asterisk-Users] Broadvoice Down?   Sean Milheim (iDREUS Corporation)
• [Asterisk-Users] Broadvoice Down?   Mark Phillips
• [Asterisk-Users] Broadvoice Down?   JD Austin
• [Asterisk-Users] Broadvoice Down?   Sean Milheim (iDREUS Corporation)
• [Asterisk-Users] Alternatives to SpanDSP??   Anton Krall
• [Asterisk-Users] Recommendations for Spanish Voice Talent   Wilson Pickett
• [Asterisk-Users] Does ztmonitor record the audio channel?   Alejandro G
• [Asterisk-Users] IAX help   Kanuri, Seshu (Company IT)
• [Asterisk-Users] Grandstream ATA 286 problems   Dana Olson
• [Asterisk-Users] DTMF intermittently stops working   Pedro
• [Asterisk-Users] Realtime & voicemail   Matthew Boehm
• [Asterisk-Users] Phone Recommendation.   Sean A. Newton
• [Asterisk-Users] Broadvoice Down?   Sean Kennedy
• [Asterisk-Users] Dial Plan - How to prepend a digit   Daniel Salama
• [Asterisk-Users] Broadvoice Down?   trixter http://www.0xdecafbad.com
• [Asterisk-Users] Dial Plan - How to prepend a digit   Josiah Bryan
• [Asterisk-Users] Broadvoice Down?   Chuck Smith
• [Asterisk-Users] Broadvoice Down?   JD Austin
• [Asterisk-Users] Phone Recommendation.   Wiley Siler
• [Asterisk-Users] Broadvoice Down?   JD Austin
• [Asterisk-Users] Dial Plan - How to prepend a digit   Franco Bellagamba
• [Asterisk-Users] Has anyone used Libretel DIDs with Asterisk?   beonice
• [Asterisk-Users] Broadvoice Down?   Gerard Marcel
• [Asterisk-Users] Re: Dial Plan - How to prepend a digit   Tony Mountifield
• [Asterisk-Users] voip problems   Gerard Marcel
• [Asterisk-Users] Broadvoice Down?   Darren Wright
• [Asterisk-Users] Broadvoice Down?   Mailing List
• [Asterisk-Users] Broadvoice Down?   JD Austin
• [Asterisk-Users] Broadvoice Down?   Andre Normandin
• [Asterisk-Users] Broadvoice Down?   Kerry Garrison
• [Asterisk-Users] Broadvoice Down?   Andre Normandin
• [Asterisk-Users] Cisco's description of echo   Keith O'Brien
• [Asterisk-Users] Broadvoice Down?   Derek Whitten
• [Asterisk-Users] Broadvoice Down?   Andre Normandin
• [Asterisk-Users] Polycom IP4000 Conference Phone   Wiley Siler
• [Asterisk-Users] voip problems   Wiley Siler
• [Asterisk-Users] Broadvoice Down?   Luki
• [Asterisk-Users] voip problems   Kerry Garrison
• [Asterisk-Users] voip problems   Gerard Marcel
• [Asterisk-Users] Asterisk replacing CCM using Catalyst 6608   Aaron Daniel
• [Asterisk-Users] Playing mp3's while recording voicemail   Rafal Kaniewski
• [Asterisk-Users] Alternatives to SpanDSP??   Jeremy Melanson
• [Asterisk-Users] Broadvoice Down?   JD Austin
• [Asterisk-Users] Dial Plan - How to prepend a digit   Daniel Salama
• [Asterisk-Users] Cannot make outgoing calls on Mediatrix 1204 from Asterisk   bam
• [Asterisk-Users] Broadvoice Down?   Wiley Siler
• [Asterisk-Users] Asterisk integration with Alcatel 4400   asterisk at frameweb.it
• [Asterisk-Users] Broadvoice Down?   JD Austin
• [Asterisk-Users] Dialing to a remote extension   Robert G. Ristroph
• [Asterisk-Users] voip problems   Wiley Siler
• [Asterisk-Users] Broadvoice Down?   Rich Adamson
• [Asterisk-Users] Dial Plan - How to prepend a digit   Stiffe
• [Asterisk-Users] voip problems   Rich Adamson
• [Asterisk-Users] voip problems   JD Austin
• [Asterisk-Users] astrecipes v2.0   Daniel Bruce Lynes
• [Asterisk-Users] Has anyone used Libretel DIDs with Asterisk?   Mark Charlton
• [Asterisk-Users] Transfers tend to fail after upgrade to 1.0.7   Pablo Alsina
• [Asterisk-Users] Broadvoice Down?   Wiley Siler
• [Asterisk-Users] Has anyone used Libretel DIDs with Asterisk?   William Suffill
• [Asterisk-Users] Broadvoice Down?   William Suffill
• [Asterisk-Users] Broadvoice Down?   trixter http://www.0xdecafbad.com
• [Asterisk-Users] using goto to do selective dialing   David John Walsh
• [Asterisk-Users] Broadvoice Down?   Gregory Wiktor - ADCom Corp.
• [Asterisk-Users] Grandstream ATA 286 problems   Gregory Wiktor - ADCom Corp.
• [Asterisk-Users] Polycom ip500 (Not-Registered)   Dan Morin
• [Asterisk-Users] Siemens SX66 wi-fi handset released   Dean Collins
• [Asterisk-Users] T1 E&M wink issues - bad int'l dial-outandoccasional dropped calls   bill black
• [Asterisk-Users] Alternatives to SpanDSP??   Adam Goryachev
• [Asterisk-Users] voip problems   William Boehlke
• [Asterisk-Users] stanaphone now terminating fax   Dean Collins
• [Asterisk-Users] Cisco's description of echo   Alexander Lopez
• [Asterisk-Users] Asterisk ADSI   John Bittner
• [Asterisk-Users] Phone Recommendation.   Anton Krall
• [Asterisk-Users] Playback dosen't play Playtone(Congestion) does ?   Michael D Schelin
• [Asterisk-Users] Citel Handset Gateways   Jeff Heath
• [Asterisk-Users] Citel Handset Gateways   William Boehlke
• [Asterisk-Users] Playback dosen't play Playtone(Congestion) does ?   Tim Connolly
• [Asterisk-Users] Call Recording via monitor   Chris Mason
• [Asterisk-Users] Grandstream ATA 286 problems   seong
• [Asterisk-Users] Siemens SX66 wi-fi handset released   Kevin Ragsdale
• [Asterisk-Users] Siemens SX66 wi-fi handset released   Dean Collins
• [Asterisk-Users] Polycom IP4000 Conference Phone   Paul Hales
• [Asterisk-Users] Polycom ip500 (Not-Registered)   Sean Kennedy
• [Asterisk-Users] Error on the Mysql, realtime database HELP so close so far; .   Paul Dracevich
• [Asterisk-Users] Large Asterisk Setup (~500 Concurrent Calls + Scalability)   Callum McGillivray
• [Norton AntiSpam] [Asterisk-Users] Error on the Mysql, realtime database HELP so close so far; .   Ronald Wiplinger
• [Asterisk-Users] Asterisk integration with Alcatel 4400   Tim Connolly
• [Norton AntiSpam] [Asterisk-Users] Error on the Mysql, realtimedatabase HELP so close so far; .   Paul Dracevich
• [Asterisk-Users] Polycom ip500 (Not-Registered)   Eric Alexander
• [Asterisk-Users] Digium Quad Span Cards   Callum McGillivray
• [Asterisk-Users] Broadvoice Down?   Tim Connolly
• [Asterisk-Users] Only want softphone account from Vonage:   Tim Connolly
• [Asterisk-Users] voip problems   Tim Connolly
• [Asterisk-Users] Broadvoice Down?   Kerry Garrison
• SV: [Asterisk-Users] Re: IPswitch: How to use speed dialing?   Thorben Jensen
• [Asterisk-Users] Error on the Mysql, realtime database HELP so close so far; .   Matthew Boehm
• [Asterisk-Users] Broadvoice Down?   trixter http://www.0xdecafbad.com
• [Asterisk-Users] Digium Quad Span Cards   Matthew Boehm
• [Asterisk-Users] Digium Card Issues   Alex
• SV: [Asterisk-Users] Re: IPswitch: How to use speed dialing?   Ronald Wiplinger
• SV: [Asterisk-Users] Re: IPswitch: How to use speed dialing?   Robert Goodyear
• [Asterisk-Users] astrecipes v2.0   Tzafrir Cohen
• [Asterisk-Users] Asterisk integration with Alcatel 4400   asterisk at frameweb.it
• [Asterisk-Users] No busy tone when dialing out over ISDN with Polycom 500 IP   Kib Eki
• [Asterisk-Users] Distinctive ring on BT100   Tomas Florian
• [Asterisk-Users] astrecipes v2.0   lenz
• SV: [Asterisk-Users] Re: IPswitch: How to use speed dialing?   Ronald Wiplinger
• [Asterisk-Users] NO ringback tone for VOIP call to another SIP server   raymond
• [Asterisk-Users] VOIP Gateways & Asterisk   Callum McGillivray
• [Asterisk-Users] Problem with long delay. VPN ?   Matthew Oulton
• [Asterisk-Users] IAX help   Brian Capouch
• [Asterisk-Users] Re: NO ringback tone for VOIP call to another SIP server   raymond
• [Asterisk-Users] Error on the Mysql, realtime database HELP soclose so far; .   Shaoul Jacobson - TELLINK
• [Asterisk-Users] Re: NO ringback tone for VOIP call to another SIP server   Peter Svensson
• [Asterisk-Users] VOIP Gateways & Asterisk   Dpto. Técnico (Softec).
• [Asterisk-Users] VOIP Gateways & Asterisk   Callum McGillivray
• [Asterisk-Users] help to configure sip server asterisk   serge perreard
• [Asterisk-Users] SIP/NetMeeting   William C. Lohr Jr.
• [Asterisk-Users] Distinctive ring on BT100   Henry Devito
• [Asterisk-Users] IP Softphone Recommendations   William C. Lohr Jr.
• [Asterisk-Users] how to use dialparties.agi   Christian Wengel
• SV: [Asterisk-Users] Re: IPswitch: How to use speed dialing?   David John Walsh
• [Asterisk-Users] ACD in Asterisk   Mamadou Lamine KA
• [Asterisk-Users] IP Softphone Recommendations   Dinesh
• [Asterisk-Users] routing in extensions.conf   Joao Pereira
• [Asterisk-Users] Unexpected control subclass 17   Peter De Schrijver
• [Asterisk-Users] Asterisk best practices   Mohamed Farid
• [Asterisk-Users] Group/Broadcast Voicemail   Eric Wieling aka ManxPower
• [Asterisk-Users] No busy tone when dialing out over ISDN with Polycom 500 IP   Eric Wieling aka ManxPower
• [Asterisk-Users] VOIP Gateways & Asterisk   Jason Williams
• [Asterisk-Users] chan capi: Long incomingmsn line in capi.conf?   Jason Williams
• [Asterisk-Users] bri cli error   Altus Snyman
• [Asterisk-Users] SIP/NetMeeting   Jason Williams
• [Asterisk-Users] Asterisk and Cisco Call Manager   Alessio Focardi
• [Asterisk-Users] i need Asterisk free Billing systems   Yousri Farouk
• [Asterisk-Users] Realtime & voicemail   Joe Dennick
• [Asterisk-Users] ForkCDR question   Roy Sigurd Karlsbakk
• [Asterisk-Users] How to set jitter buffer for SIP   Asterisk guy
• [Asterisk-Users] Good FXO for UK use.   Razza
• [Asterisk-Users] How to set jitter buffer for SIP   Asterisk guy
• [Asterisk-Users] Good FXO for UK use.   Ian D. Willoughby
• [Asterisk-Users] Good FXO for UK use.   Patrick Lidstone (Personal E-mail)
• [Asterisk-Users] VOIP Gateways & Asterisk   john at gxc24.com
• [Asterisk-Users] pridialplan/TON question   Klaus Darilion
• [Asterisk-Users] Good FXO for UK use.   Ian D. Willoughby
• [Asterisk-Users] Shanghai or Bangalore DIDs   Marc Storck
• [Asterisk-Users] pridialplan/TON question   Eric Wieling aka ManxPower
• [Asterisk-Users] Alternatives to SpanDSP??   Steve Underwood
• [Asterisk-Users] Cisco Systems to Acquire Sipura Technology   Eric Wieling aka ManxPower
• [Asterisk-Users] pridialplan/TON question   Klaus Darilion
• [Asterisk-Users] IP Softphone Recommendations   Time Bandit
• [Asterisk-Users] Asterisk and Cisco Call Manager   Edgar de Leon at SESCAM
• [Asterisk-Users] VOIP Gateways & Asterisk   Michael Baird
• [Asterisk-Users] Digium for ETSI ISDN   Nathaniel Angelo A. Torres (247talk)
• [Asterisk-Users] Shanghai or Bangalore DIDs   Marc Storck
• [Asterisk-Users] Digium for ETSI ISDN   Steve Underwood
• [Asterisk-Users] Dial CLI Command   flavio patria
• [Asterisk-Users] Alternatives to SpanDSP??   Andrew Kohlsmith
• [Asterisk-Users] Digium for ETSI ISDN   Nathaniel Angelo A. Torres (247talk)
• [Asterisk-Users] pridialplan/TON question   Peter Svensson
• [Asterisk-Users] Alternatives to SpanDSP??   Julian J. M.
• [Asterisk-Users] Dial CLI Command   flavio patria
• [Asterisk-Users] Alternatives to SpanDSP??   W. Kevin Hunt
• [Asterisk-Users] Good FXO for UK use.   Razza
• [Asterisk-Users] Phone Recommendation.   Asterisk at Home
• [Asterisk-Users] pridialplan/TON question   Marc Storck
• [Asterisk-Users] Alternatives to SpanDSP??   Steve Underwood
• [Asterisk-Users] Alternatives to SpanDSP??   Steve Underwood
• [Asterisk-Users] Phone Recommendation.   Wiley Siler
• [Asterisk-Users] Polycom IP4000 Conference Phone   Wiley Siler
• [Asterisk-Users] pridialplan/TON question   Klaus Darilion
• [Asterisk-Users] return a value from dial macro   Steve Dolloff
• [Asterisk-Users] Alternatives to SpanDSP??   Adam Goryachev
• [Asterisk-Users] Alternatives to SpanDSP??   Adam Goryachev
• [Asterisk-Users] Incoming Not Answering   David Sampson
• [Asterisk-Users] Phone Recommendation.   Anton Krall
• [Asterisk-Users] SIP behind IPTables/NAT   Ian Pattison
• [Asterisk-Users] YAC and IPs   Anton Krall
• [Asterisk-Users] pridialplan/TON question   Klaus Darilion
• [Asterisk-Users] Zap/PRI: received AOC-E charging   Matthew Boehm
• [Asterisk-Users]I wanted to understand   Deborah MALKA
• [Asterisk-Users] Cisco to buy Sipura   Cory Andrews
• [Asterisk-Users] X100P + spandsp locks machine with zaptel & asterisk 1.0.7   Jesse Guardiani
• [Asterisk-Users] VOIP Gateways & Asterisk   Jason Williams
• [Asterisk-Users] Digium for ETSI ISDN   Jason Williams
• [Asterisk-Users] CLI dial command   flavio patria
• [Asterisk-Users] call an ldap search result   Guy Boehm
• [Asterisk-Users] Alternatives to SpanDSP??   Rich Adamson
• [Asterisk-Users] IP Softphone Recommendations   Derek Whitten
• [Asterisk-Users] Good FXO for UK use.   Johan Akerstrom
• [Asterisk-Users] pridialplan/TON question   Peter Svensson
• [Asterisk-Users] Alternatives to SpanDSP??   Colin Anderson
• [Asterisk-Users] IP Softphone Recommendations   Guillermo Salas M
• [Asterisk-Users] pridialplan/TON question   Peter Svensson
• [Asterisk-Users] Alternatives to SpanDSP??   Andrew Kohlsmith
• [Asterisk-Users] Incoming Not Answering   Chad Osmond
• [Asterisk-Users] Fail over solutions   Sean Kennedy
• [Asterisk-Users] Phone Recommendation.   Sean A. Newton
• [Asterisk-Users] Alternatives to SpanDSP??   Eric Wieling aka ManxPower
• [Asterisk-Users] YAC and IPs   Josiah Bryan
• [Asterisk-Users] japanese voice files   Isamar Maia
• [Asterisk-Users] Playback dosen't play Playtone(Congestion) does ?   Michael D Schelin
• [Asterisk-Users] Alternatives to SpanDSP??   Adam Goryachev
• [Asterisk-Users] asterisk xlite nat problem   Mostafa
• [Re] Re: [Asterisk-Users] Alternatives to SpanDSP??   Cyril VELTER
• [Asterisk-Users] Re: Polycom IP4000 Conference Phone   Noah Miller
• [Asterisk-Users] Shanghai or Bangalore DIDs   ht at phonitel.com
• [Asterisk-Users] Polycom SIP 1.5.0 Firmware   Wiley Siler
• [Asterisk-Users] YAC and IPs   Adam Greenbaum
• [Asterisk-Users] *HOWTO* : using mime-construct with outlook - send fax to email recipient   Brian Dingman
• [Asterisk-Users] Cisco 7290 calling problems :-(   Paul A Brown
• [Asterisk-Users] Polycom IP4000 Conference Phone   Max W Blackmer Jr
• [Asterisk-Users] QUICK QUESTION   Dan Levine
• [Asterisk-Users] IP Softphone Recommendations   Anton Krall
• [Asterisk-Users]I wanted to understand   Julius Kidubuka
• [Asterisk-Users] Good FXO for UK use.   Chris Stenton
• [Asterisk-Users] Good FXO for UK use.   Razza
• [Asterisk-Users] Polycom Config - SIP 1.4.1   Wiley Siler
• [Asterisk-Users] Re: Asterisk-Users Digest, Vol 9, Issue 224   Iassen Hristov
• [Asterisk-Users] pri_dchannel: PRI got event: HDLC Abort   Doug Reid - Stormcorp
• [Re] Re: [Asterisk-Users] Alternatives to SpanDSP??   Andrew Kohlsmith
• [Asterisk-Users] Alternatives to SpanDSP??   Andrew Kohlsmith
• [Asterisk-Users] SIP behind IPTables/NAT   Johan Akerstrom
• [Asterisk-Users] Digium Quad Span Cards   Matt Roth
• [Asterisk-Users] Large Asterisk Setup (~500 Concurrent Calls + Scalability)   Matt Roth
• [Asterisk-Users] YAC and IPs   Anton Krall
• [Asterisk-Users] pridialplan/TON question   Klaus Darilion
• [Asterisk-Users] Cisco 7290 calling problems :-( - Sorry if this comes through twice   Paul A Brown
• [Asterisk-Users] Using Asterisk to dial a number and then wait to dial the extension   Andrew Elchuk
• [Asterisk-Users] Mitel Ip phone ?   Pedro
• [Asterisk-Users] Is There Media Accelerator For Better Asterisk Calls   chawki hammoud
• [Asterisk-Users] Strange queue agent issue - Agent busy problem   Daniel W. Halverson
• [Asterisk-Users] QUICK QUESTION   Mark Phillips
• [Asterisk-Users] rxfax over IAX ulaw   Darrell Collins
• [Asterisk-Users] Re: *HOWTO* : using mime-construct with outlook - send fax to email recipient   Jesse Guardiani
• [Asterisk-Users] Polycom IP4000 Conference Phone   Wiley Siler
• [Asterisk-Users] SIP behind IPTables/NAT   David John Walsh
• [Asterisk-Users] Checking for a sound file   Marc
• [Asterisk-Users] Voicemails stopping   Chris Stinson
• [Asterisk-Users] Re: Re: Fax questions   Jesse Guardiani
• [Asterisk-Users] Provisioning Lines   Manjit Riat
• [Asterisk-Users] Grandstream ATA 286 problems   Andrejus Stavickis
• [Asterisk-Users] Establish a ppp connection with a Pipeline 50   Jason McAffee
• [Asterisk-Users] IP Softphone Recommendations   Ing CIP Alejandro Celi Mariátegui
• [Asterisk-Users] the CLI dial command   flavio patria
• [Asterisk-Users] YAC and IPs   Peter Bowyer
• [Asterisk-Users] Cisco 7290 calling problems :-(   Mojo with Horan & Company, LLC
• [Asterisk-Users] VOIP Gateways & Asterisk   Dana Olson
• [Asterisk-Users] Extensions / Contexts   Sebastian Silva
• [Asterisk-Users] CID signalling for DTMF   Alejandro G
• [Asterisk-Users] Remote Phones - No Audio In Either Direction   Paul Tyreman
• [Asterisk-Users] Re: IP Softphone Recommendations   Bruno Hertz
• [Asterisk-Users] Cisco 7960 earpiece speaker echo question   Jeremy Koski
• [Asterisk-Users] Fail over solutions   snacktime
• [Asterisk-Users] Extensions / Contexts   Wiley Siler
• [Asterisk-Users] Large Asterisk Setup (~500 Concurrent Calls + Scalability)   Sean Milheim (iDREUS Corporation)
• [Asterisk-Users] Stable Asterisk Version ??? / SIP problem ???   Joseph
• [Asterisk-Users] Digium Quad Span Cards   Alexander Lopez
• [Asterisk-Users] Phone Recommendation.   Dana Olson
• [Asterisk-Users] YAC and IPs   Florian Overkamp
• [Asterisk-Users] YAC and IPs   Matthew Marlowe
• [Asterisk-Users] IP Softphone Recommendations   Guillermo Salas M
• [Asterisk-Users] Good FXO for UK use.   Adrian Chapman
• [Asterisk-Users] Shanghai or Bangalore DIDs   Matt Riddell
• [Asterisk-Users] pri_dchannel: PRI got event: HDLC Abort   Andrew Kohlsmith
• [Asterisk-Users] Checking for a sound file   Wiley Siler
• [Asterisk-Users] Alternatives to SpanDSP??   Trevor Peirce
• [Asterisk-Users] Turn off Music on Hold   Ed Greenberg
• [Asterisk-Users] Alternatives to SpanDSP??   Julian J. M.
• [Asterisk-Users] Good FXO for UK use.   Razza
• [Asterisk-Users] TE405P w/ Intel SE7210TP1_E Motherboard   Greg Boehnlein
• [Asterisk-Users] Polycom Config - SIP 1.4.1   Sean Kennedy
• [Asterisk-Users] AAH 0.9 - SIP DTMF negotiation problem   Marty Mastera
• [Asterisk-Users] Remote Phones - No Audio In Either Direction   Sean Kennedy
• [Asterisk-Users] Is There Media Accelerator For Better Asterisk Calls   Matt Riddell
• [Asterisk-Users] IAX ATA's   Garrett Smith
• [Asterisk-Users] Is There Media Accelerator For Better AsteriskCalls   Kanuri, Seshu (Company IT)
• [Asterisk-Users] Good FXO for UK use.   Razza
• [Asterisk-Users] YAC and IPs   Josiah Bryan
• [Asterisk-Users] amaflags=Documentation   Kumara Jayaweera
• [Asterisk-Users] Extensions / Contexts   Alexander Lopez
• [Asterisk-Users] Extensions / Contexts   Eric Wieling aka ManxPower
• [Asterisk-Users] Remote Phones - No Audio In Either Direction   Alexander Lopez
• [Asterisk-Users] Extensions / Contexts   Chris Wade
• [Asterisk-Users] Extensions / Contexts   Matt Riddell
• [Asterisk-Users] Extensions / Contexts   Daniel Salama
• [Asterisk-Users] Cisco 7290 calling problems :-(   Paul A Brown
• [Asterisk-Users] the CLI dial command   Eric Wieling aka ManxPower
• [Asterisk-Users] Extensions / Contexts   David John Walsh
• [Asterisk-Users] Static and echo on PRI   Mark Johnson
• [Asterisk-Users] Variable names in dial plans   Daniel Salama
• [Asterisk-Users] Controlling extentions through Management Interface   Sascha Ferley
• [Asterisk-Users] Controlling extentions through Management Interface (fwd)   Sascha Ferley
• [Asterisk-Users] chan capi: Long incomingmsn line in capi.conf?   Peter Braidwood
• [Asterisk-Users] Alternatives to SpanDSP??   Colin Anderson
• [Asterisk-Users] Turn off Music on Hold   David Phelan
• [Asterisk-Users] Checking for a sound file   Andy Hamilton
• [Asterisk-Users] Queue Management and Command Execution   Daniel Salama
• [Asterisk-Users] IAX ATA's   Joseph
• [Asterisk-Users] Grandstream ATA 286 problems   The VoIP Connection
• [Asterisk-Users] VOIP Gateways & Asterisk   Paul Hales
• [Asterisk-Users] Polycom IP4000 Conference Phone   Paul Hales
• [Asterisk-Users] Alternatives to SpanDSP??   Craig Guy
• [Asterisk-Users] voice pulse connect - no dtmf   Spencer Nassar
• [Asterisk-Users] YAC and IPs   William Suffill
• [Asterisk-Users] Warm standby boxes - keeping config syncronised?   David John Walsh
• [Asterisk-Users] Polycom Images   Manjit Riat
• [Asterisk-Users] Static and echo on PRI   Andrew Kohlsmith
• [Asterisk-Users] Warm standby boxes - keeping config syncronised?   Ed Greenberg
• [Asterisk-Users] US$100 bounty for two features in voicemail   Richard
• [Asterisk-Users] Grandstream ATA 286 problems   Michael D Schelin
• [Asterisk-Users] Polycom Images   Cory Andrews
• [Asterisk-Users] No Audio sent using playback cmd   Michael D Schelin
• [Asterisk-Users] The TCP in Asterisk   maxsu
• [Asterisk-Users] TE405P w/ Intel SE7210TP1_E Motherboard   john at gxc24.com
• [Asterisk-Users] US$100 bounty for two features in voicemail   TC
• [Asterisk-Users] SIP, Asterisk and NAT   Irakli Natsvlishvili
• [Asterisk-Users] Verizon ISDN   Gregory Wiktor - ADCom Corp.
• [Asterisk-Users] Auto Fax transfer problem?   Andrew C. Brown
• [Asterisk-Users] US$100 bounty for two features in voicemail   Richard
• [Asterisk-Users] IAX ATA's   Rod Bacon
• [Asterisk-Users] Auto Fax transfer problem?   Andrew C. Brown
• [Asterisk-Users] No Audio sent using playback cmd   Rod Bacon
• [Asterisk-Users] Grandstream ATA 286 problems   Rod Bacon
• [Asterisk-Users] Warm standby boxes - keeping config syncronised?   Rod Bacon
• [Asterisk-Users] Basic telephony hardware questions   Min Hwan Chang
• [Asterisk-Users] rxfax over IAX ulaw   Steve Underwood
• [Asterisk-Users] IAX ATA's   VOIP Consultant
• [Asterisk-Users] IAX ATA's   Joseph
• [Asterisk-Users] SIP, Asterisk and NAT   Michael D Schelin
• [Asterisk-Users] Grandstream ATA 286 problems   Brandon Patterson
• [Asterisk-Users] Digium Quad Span Cards   Adam Goryachev
• [Asterisk-Users] VOIP Gateways & Asterisk   Rusty Shackleford
• [Asterisk-Users] No Audio sent using playback cmd   Michael D Schelin
• [Asterisk-Users] Is There Media Accelerator For Better Asterisk Calls   Adam Goryachev
• [Asterisk-Users] Cisco 7960 earpiece speaker echo question   Henry Devito
• [Asterisk-Users] YAC and IPs   Josiah Bryan
• [Asterisk-Users] Strange queue agent issue - Agent busy problem   Adam Goryachev
• [Asterisk-Users] US$100 bounty for two features in voicemail   Josiah Bryan
• SV: [Asterisk-Users] Re: IPswitch: How to use speed dialing?   Race Vanderdecken
• [Asterisk-Users] Checking for a sound file   Adam Goryachev
• [Asterisk-Users] Grandstream 100 power problem   Michael Cobb
• [Asterisk-Users] Basic telephony hardware questions   Adam Goryachev
• [Asterisk-Users] Recommend IP Phones???   Paul
• [Asterisk-Users] Digium Quad Span Cards   Daniel Salama
• [Asterisk-Users] removing monitor IN/OUT wav file   Joseph
• [Asterisk-Users] Static and echo on PRI   Mark Johnson
• SV: [Asterisk-Users] Re: IPswitch: How to use speed dialing?   Ronald Wiplinger
• [Asterisk-Users] Re: *HOWTO* : using mime-construct with outlook - send fax to email recipient   Jesse Guardiani
• [Asterisk-Users] Variable names in dial plans   Jason Walker
• [Asterisk-Users] Cisco to buy Sipura   Leo Ann Boon
• [Asterisk-Users] Alternatives to SpanDSP??   Matthew Boehm
• [Asterisk-Users] Static and echo on PRI   Andrew Kohlsmith
• [Asterisk-Users] Static and echo on PRI   Matt Klein
• [Asterisk-Users] Cisco 7290 calling problems :-( - Sorry if this comes through twice   Andrew A. Kochetkoff
• [Asterisk-Users] The TCP in Asterisk   Michael Bielicki
• [Asterisk-Users] Variable names in dial plans   Daniel Salama
• [Asterisk-Users] Static and echo on PRI   Mark Johnson
• [Asterisk-Users] Static and echo on PRI   Matt Klein
• [Asterisk-Users] Static and echo on PRI   Mark Johnson
• [Asterisk-Users] Static and echo on PRI   Michael Welter
• [Asterisk-Users] Static and echo on PRI   Matt Klein
• [Asterisk-Users] Static and echo on PRI   Mark Johnson
• [Asterisk-Users] Static and echo on PRI   Adam Goryachev
• [Asterisk-Users] Alternatives to SpanDSP??   Adam Goryachev
• [Asterisk-Users] Cisco 7960 earpiece speaker echo question   Jeremy Koski
• [Asterisk-Users] Zaptel FXO crashing.   Jason Leach
• [Asterisk-Users] unsubscribe   MSL
• [Asterisk-Users] Cisco 7290 calling problems :-( - Sorry if thiscomes through twice   Paul A Brown
• [Asterisk-Users] unsubscribe   Luki
• [Asterisk-Users] QUICK QUESTION   Wilson Pickett
• [Asterisk-Users] Digium Quad Span Cards   Peter Svensson
• [Asterisk-Users] call a peer over the asterisk manager with a php script   Guy Boehm
• [Asterisk-Users] RTP vs cRTP vs IAX   Jean-Michel Hiver
• [Asterisk-Users] Zap/PRI: received AOC-E charging   Jason Williams
• [Asterisk-Users] SIP, Asterisk and NAT   Olle E. Johansson
• [Asterisk-Users] Fail over solutions   Jason Williams
• [Asterisk-Users] call a ldap result via my x-lite   Guy Boehm
• [Asterisk-Users] Queue Management and Command Execution   Anton Krall
• [Asterisk-Users] YAC and IPs   Anton Krall
• [Asterisk-Users] RTP vs cRTP vs IAX   Zoa
• [Asterisk-Users] SIP, Asterisk and NAT   Zoa
• [Asterisk-Users] Fail over solutions   trixter http://www.0xdecafbad.com
• [Asterisk-Users] Fail over solutions   trixter http://www.0xdecafbad.com
• [Asterisk-Users] Fail over solutions   Zoa
• [Asterisk-Users] (no subject)   Sina
• [Asterisk-Users] Fail over solutions   trixter http://www.0xdecafbad.com
• [Asterisk-Users] No music on hold when transferring call   Bam
• [Asterisk-Users] do I configure ISDN in zapata.conf?   Tomasz Chmielewski
• [Asterisk-Users] Re: Using Asterisk to dial a number and then wait to dial the extension   Tony Mountifield
• [Asterisk-Users] b0rked hfc config   Thomas Andrews
• [Asterisk-Users] Cisco 7.4 SIP firmware   Betül Gözlükoğlu
• [Asterisk-Users] No music on hold when transferring call   Kristof Hardy
• [Asterisk-Users] b0rked hfc config   Thomas Andrews
• [Asterisk-Users] Supervised transfer problem.   Cesar Garcia
• [Asterisk-Users] Cisco 7.4 SIP firmware   Shaoul Jacobson - TELLINK
• [Asterisk-Users] Static and echo on PRI   Eric Wieling aka ManxPower
• [Asterisk-Users] Variable names in dial plans   Eric Wieling aka ManxPower
• [Asterisk-Users] Asterisk doesn't disconnect when I hang up SIP (SIP -> PSTN call)   Tomasz Chmielewski
• [Asterisk-Users] Alternatives to SpanDSP??   Craig Guy
• [Asterisk-Users] Good FXO for UK use.   Johan Akerstrom
• [Asterisk-Users] Asterisk doesn't disconnect when I hang up SIP (SIP -> PSTN call)   Umair Bari
• [Asterisk-Users] Cisco 7960 earpiece speaker echo question   Henry Devito
• [Asterisk-Users] Dial Tone   Henry Jensen
• [Asterisk-Users] agent monitor filename   Asterisk
• [Asterisk-Users] Asterisk doesn't disconnect when I hang up SIP (SIP -> PSTN call)   Tomasz Chmielewski
• [Asterisk-Users] noload res_musiconhold.so breaksa IAX   Ed Greenberg
• [Asterisk-Users] Dial Tone   Gavin Hamill
• [Asterisk-Users] Queue Management and Command Execution   Daniel Salama
• [Asterisk-Users] noload res_musiconhold.so breaksa IAX   Eric Wieling aka ManxPower
• [Asterisk-Users] Static and echo on PRI   Andrew Kohlsmith
• [Asterisk-Users] Static and echo on PRI   Andrew Kohlsmith
• [Asterisk-Users] All lines are busy   Colin E. McDonald
• [Asterisk-Users] Cisco 7960 earpiece speaker echo question   Jason Williams
• [Asterisk-Users] Busy Tone   Dennie Verstrepen
• [Asterisk-Users] (no subject)   Andre Normandin
• [Asterisk-Users] noload res_musiconhold.so breaksa IAX   Ed Greenberg
• [Asterisk-Users] (no subject)   Jason Williams
• [Re] Re: [Asterisk-Users] Alternatives to SpanDSP??   Steve Underwood
• [Asterisk-Users] Dialing out from remote.   Daniel Dziubanski
• [Asterisk-Users] TE405P w/ Intel SE7210TP1_E Motherboard   Rob Lith
• [Asterisk-Users] Good FXO for UK use.   Patrick Lidstone (Personal e-mail)
• [Asterisk-Users] Alternatives to SpanDSP??   Rich Adamson
• [Asterisk-Users] Alternatives to SpanDSP??   Andrew Kohlsmith
• [Asterisk-Users] b0rked hfc config   Julian J. M.
• [Asterisk-Users] return a value from dial macro   Steve Dolloff
• [Asterisk-Users] Checking for a sound file   Eric Wieling aka ManxPower
• [Asterisk-Users] SetGroup on dialed calls?   mike castleman
• [Asterisk-Users] Cisco SIP Firmware Price Increase   Dan Levine
• [Asterisk-Users] Extensions / Contexts   Wiley Siler
• [Asterisk-Users] Polycom IP4000 Conference Phone   Wiley Siler
• [Asterisk-Users] Cisco SIP Firmware Price Increase   Henry Devito
• [Asterisk-Users] Checking for a sound file   John Novack
• [Asterisk-Users] Checking for a sound file   Rich Adamson
• [Asterisk-Users] All lines are busy   Henry Devito
• [Asterisk-Users] No audio playback after upgrade from 1.0.1   Robert Derr
• [Asterisk-Users] Checking for a sound file   Tom Fanning
• [Asterisk-Users] Grandstream BT101 Firmware   Adam Goryachev
• [Asterisk-Users] Queue Management and Command Execution   Henry Devito
• [Asterisk-Users] Determinating Phone status   Elmar Haneke
• [Asterisk-Users] how to use dialparties.agi   David John Walsh
• [Asterisk-Users] Redirect two channels to each other?   Tony Mountifield
• [Asterisk-Users] Grandstream BT101 Firmware   Dean Collins
• [Asterisk-Users] Dialing out from remote.   Dean Collins
• [Re] Re: [Re] Re: [Asterisk-Users] Alternatives to SpanDSP??   Steve Underwood
• [Re] Re: [Re] Re: [Asterisk-Users] Alternatives to SpanDSP??   Cyril VELTER
• [Asterisk-Users] Determinating SIP Phone status   Olle E. Johansson
• [Asterisk-Users] Extensions / Contexts   Sebastian Silva
• [Asterisk-Users] Connection Timeout problem with SIP phones from Gnet   Jean-Francois Theroux
• [Asterisk-Users] Checking for a sound file   Rich Adamson
• [Asterisk-Users] Cisco SIP Firmware Price Increase   Rich Adamson
• [Asterisk-Users] Re: Using Asterisk to dial a number and then wait to dial the extension   Andrew Elchuk
• [Asterisk-Users] Redirect two channels to each other?   Josiah Bryan
• [Asterisk-Users] Determinating Phone status   Alexander Lopez
• [Asterisk-Users] Extensions / Contexts   Sebastian Silva
• [Asterisk-Users] Alternatives to SpanDSP??   Rich Adamson
• [Asterisk-Users] Redirect two channels to each other?   Alexander Lopez
• [Asterisk-Users] Determinating SIP Phone status   Alexander Lopez
• [Asterisk-Users] Phone Recommendation.   Sean A. Newton
• [Asterisk-Users] Re: Using Asterisk to dial a number and then wait to dial the extension   Andrew Elchuk
• [Asterisk-Users] Redirect two channels to each other?   mattf
• [Asterisk-Users] Cisco SIP Firmware Price Increase   Rich Adamson
• [Asterisk-Users] IP Softphone Recommendations   Christian Stredicke
• [Asterisk-Users] QuadBRI card on Suse 9.2 Unable to load qozap.ko   Massimo
• [Asterisk-Users] Remote Phones - No Audio In Either Direction   Paul Tyreman
• [Asterisk-Users] Confused on G723 and G729   Matt
• [Asterisk-Users] Re: Redirect two channels to each other?   Tony Mountifield
• [Asterisk-Users] Phone Recommendation.   Dana Olson
• [Asterisk-Users] call a peer over the asterisk manager with a php script   Richard Lyman
• [Asterisk-Users] Re: Using Asterisk to dial a number and thenwait to dial the extension   Andrew Elchuk
• [Asterisk-Users] [Fwd: Supervised transfer problem.]   Cesar Garcia
• [Asterisk-Users] Checking for a sound file   Matthew Marlowe
• [Asterisk-Users] tonezone in tunisia   Samuel T. Cossette
• [Asterisk-Users] oh323 Zone   Sebastian Atala
• [Asterisk-Users] Re: Redirect two channels to each other?   mattf
• [Asterisk-Users] Transcoding Capacity   Daniel Salama
• [Asterisk-Users] Receiving Incoming Calls not working properly on TDM400P with 4 FXO modules   Andrew Elchuk
• [Asterisk-Users] tonezone in tunisia   mattf
• [Asterisk-Users] Receiving Incoming Calls not working properly on TDM400P with 4 FXO modules   Robert Webb
• [Asterisk-Users] Static and echo on PRI   Mark Johnson
• [Asterisk-Users] tonezone in tunisia -- I got it.   Samuel T. Cossette
• [Asterisk-Users] Redirect two channels to each other?   Josiah Bryan
• [Asterisk-Users] j'ai un probleme de connexion   youssef ouadou
• [Asterisk-Users] RJ45 to RJ11?   Paul Shiflet
• [Asterisk-Users] Busy Tone   Peter Svensson
• [Asterisk-Users] Static and echo on PRI   Jorge Mendoza
• [Asterisk-Users] Confused on G723 and G729   Rusty Shackleford
• [Asterisk-Users] Phone Recommendation.   Steve Clark
• [Asterisk-Users] Re: Cisco to buy Sipura   Tom Ivar Helbekkmo
• [Asterisk-Users] RJ45 to RJ11?   Jeremy Melanson
• [Asterisk-Users] RJ45 to RJ11?   Josiah Bryan
• [Asterisk-Users] RJ45 to RJ11?   Gregory Junker
• [Asterisk-Users] RJ45 to RJ11?   Robert Webb
• [Asterisk-Users] RJ45 to RJ11?   Mojo with Horan & Company, LLC
• [Asterisk-Users] Remote Phones - No Audio In Either   Sruly
• [Asterisk-Users] Connection Timeout problem with SIP phones from Gnet   Time Bandit
• [Asterisk-Users] Public IP for SIP and NAT   Aaron O'Hara
• [Asterisk-Users] RJ45 to RJ11?   Jon Pounder
• [Asterisk-Users] RJ45 to RJ11?   John Novack
• [Asterisk-Users] Phone Recommendation.   Wiley Siler
• [Asterisk-Users] RJ45 to RJ11?   Ian Pattison
• [Asterisk-Users] Receiving Incoming Calls not working properly on TDM400P with 4 FXO modules   Andrew Elchuk
• [Asterisk-Users] Call Recording via monitor   Time Bandit
• [Asterisk-Users] No Audio sent using playback cmd   Michael D Schelin
• [Fwd: [Asterisk-Users] Voicemails stopping]   Chris Stinson
• [Asterisk-Users] Panasonic KX-TD1232 Signaling   Dan Morin
• [Asterisk-Users] RJ45 to RJ11?   Paul Shiflet
• [Asterisk-Users] Static and echo on PRI   Mark Johnson
• [Asterisk-Users] wip 5000 in 12 hour time mode - anyone?   Jerry Geis
• [Asterisk-Users] call a peer over the asterisk manager with a php script   Dana Olson
• [Asterisk-Users] re: RJ45 to RJ11?   Justin Newman
• [Asterisk-Users] Re: Cisco to buy Sipura   Julio Arruda
• [Asterisk-Users] RJ45 to RJ11?   Dana Olson
• [Asterisk-Users] Sip Registrations   jamesm
• [Asterisk-Users] Don't know what to do if second ROSE component is of type 0x6   Jess Coburn
• [Asterisk-Users] RJ45 to RJ11?   Chuck Smith
• [Asterisk-Users] Phone Recommendation.   Sean A. Newton
• [Fwd: [Asterisk-Users] Voicemails stopping]   Michael D Schelin
• [Asterisk-Users] Panasonic KX-TD1232 Signaling   Brian Leyton
• [Asterisk-Users] No Audio sent using playback cmd   Michael D Schelin
• [Asterisk-Users] Asterisk on Solaris 10 x86   Srinivasa Cherukuri
• [Asterisk-Users] UK (english) sound files (Paul R)   Paul Redstone
• [Asterisk-Users] Call Type = Data   Jason McAffee
• [Asterisk-Users] Anyting special needed for fax on a ATA186?   Tim Connolly
• [Asterisk-Users] CDR Billing Question.   Paul Dracevich
• [Asterisk-Users] Panasonic KX-TD1232 Signaling   John Novack
• [Asterisk-Users] UK (english) sound files (Paul R   Paul Redstone
• [Asterisk-Users] CDR Billing Question.   Radovan.Mihalik
• [Asterisk-Users] SIP -> capi problem (no sound)   Cyrille Demaret
• [Asterisk-Users] CDR Billing Question.   Paul Dracevich
• [Asterisk-Users] CDR Billing Question.   Paul Dracevich
• [Asterisk-Users] CDR Billing Question.   jltaylor
• [Asterisk-Users] Panasonic KX-TD1232 Signaling   Dan Morin
• [Asterisk-Users] j'ai un probleme de connexion   Steve Kann
• [Asterisk-Users] ser rtpproxy asterisk problems....   G.Marshall
• [Asterisk-Users] * and Sipgate (UK)   Robert P. McKenzie
• [Asterisk-Users] QUICK QUESTION   Time Bandit
• [Asterisk-Users] Cisco 7290 calling problems :-(   Paul A Brown
• [Asterisk-Users] RJ45 to RJ11?   David Phelan
• [Asterisk-Users] CDR Billing Question.   Paul Dracevich
• [Asterisk-Users] RJ45 to RJ11?   Mojo with Horan & Company, LLC
• [Asterisk-Users] CDR Billing Question.   jltaylor
• [Asterisk-Users] RJ45 to RJ11?   David Phelan
• [Asterisk-Users] Cisco 7960s and skinny   Paul
• [Asterisk-Users] Cisco 7940G SIP Conversion   Paul
• [Asterisk-Users] IAX aproprietary protocol   Joseph
• [Asterisk-Users] Is There Media Accelerator For Better AsteriskCalls   chawki hammoud
• [Asterisk-Users] Panasonic KX-TD1232 Signaling   Brian Leyton
• [Asterisk-Users] IAX aproprietary protocol   Stefan de Konink
• [Asterisk-Users] IAX aproprietary protocol   Marc Storck
• [Asterisk-Users] IAX aproprietary protocol   Steve Underwood
• [Asterisk-Users] IAX aproprietary protocol   Steve Kann
• [Asterisk-Users] ATA 186 MGCP Firmware   Ken Bowman
• [Asterisk-Users] Dialing out...   Manny A. Wise
• [Asterisk-Users] Asterisk on a media stream vs. direct RTP communication between endpoints   Irakli Natsvlishvili
• [Asterisk-Users] Any other MoH source except *   Irakli Natsvlishvili
• [Asterisk-Users] Linksys/Cisco buys Sipura   MF Hulber
• [Asterisk-Users] SIP, Asterisk and NAT   Irakli Natsvlishvili
• [Asterisk-Users] IAX aproprietary protocol   Joseph
• [Asterisk-Users] IAX aproprietary protocol   Jonathan
• [Asterisk-Users] IAX aproprietary protocol   Marc Storck
• [Asterisk-Users] Linksys/Cisco buys Sipura   Isamar Maia
• [Asterisk-Users] voice pulse connect - no dtmf   Justin Richards
• [Asterisk-Users] IAX aproprietary protocol   Joseph
• [Asterisk-Users] Transcoding times   Irakli Natsvlishvili
• [Asterisk-Users] MOH   Nabeel Jafferali
• [Asterisk-Users] Linksys/Cisco buys Sipura   Joseph
• [Asterisk-Users] Transcoding times   Boris Bakchiev
• [Asterisk-Users] Dialing out...   Time Bandit
• [Asterisk-Users] IAX aproprietary protocol   Adam Hart
• [Asterisk-Users] Automatic Follow-Me Forwarding Based on Cell GPS   Joseph Gutowski
• [Asterisk-Users] UK (english) sound files (Paul R)   Mark Phillips
• [Asterisk-Users] Automatic Follow-Me Forwarding Based on Cell GPS   Alexander Lopez
• [Asterisk-Users] UK (english) sound files (Paul R)   Alexander Lopez
• [Asterisk-Users] * and Sipgate (UK)   Luki
• [Asterisk-Users] Linksys/Cisco buys Sipura   Henry Devito
• [Asterisk-Users] Linksys/Cisco buys Sipura   Steven Kalcevich
• [Asterisk-Users] TDM400 doesn't know the hangup signal in china   lanfei chen
• [Asterisk-Users] Questions about ongoing calls   Irakli Natsvlishvili
• [Asterisk-Users] Supervised transfer problem.   Arunachala
• [Asterisk-Users] j'ai un probleme de connexion   Leo Ann Boon
• [Asterisk-Users] ATA 186 MGCP Firmware   Stewart Nelson
• [Asterisk-Users] RTP vs cRTP vs IAX   Brian Capouch
• [Asterisk-Users] Asterisk@home questions   Sascha Ferley
• [Asterisk-Users] X100P Clone any hints for recognizing RINGing ?   Christoph Rothe
• [Asterisk-Users] IAX aproprietary protocol   Jean-Michel Hiver
• [Asterisk-Users] X100P Clone any hints for recognizing RINGing ?   Paul
• [Asterisk-Users] Linux SoftPhone with Sound Daemon Support   Rod Bacon
• [Asterisk-Users] pridialplan/TON question   Klaus Darilion
• [Asterisk-Users] SIP, Asterisk and NAT   Zoa
• [Asterisk-Users] Eicon DIVA PCI ISDN cards (not server) work withasterisk!   Gregory Wiktor - ADCom Corp.
• [Asterisk-Users] Panasonic KX-TD1232 Signaling   Peter Svensson
• [Asterisk-Users] Re: UK (english) sound files (Paul R)   Paul Redstone
• [Asterisk-Users] ATA 186 MGCP Firmware   Ken Bowman
• [Asterisk-Users] IAX ATA's   clive at sentechsa.com
• [Asterisk-Users] Eicon DIVA PCI ISDN cards (not server) work withasterisk!   Tomasz Chmielewski
• [Asterisk-Users] call recording problem   Joseph Shi
• [Asterisk-Users] Eicon DIVA PCI ISDN cards (not server) workwithasterisk!   Gregory Wiktor - ADCom Corp.
• [Asterisk-Users] QuadBRI card on Suse 9.2 Unable to load qozap.ko   Kristof Hardy
• [Asterisk-Users] Eicon DIVA PCI ISDN cards (not server) workwithasterisk!   Tomasz Chmielewski
• [Asterisk-Users] Zaptel FXO crashing.   Richard Scobie
• [Asterisk-Users] Eicon DIVA PCI ISDN cards (notserver) workwithasterisk!   Gregory Wiktor - ADCom Corp.
• [Asterisk-Users] Eicon DIVA PCI ISDN cards (notserver) workwithasterisk!   Tomasz Chmielewski
• [Asterisk-Users] Incoming calls and CAPI   igil at europesip.com
• [Asterisk-Users] * and Sipgate (UK)   Robert P. McKenzie
• [Asterisk-Users] Monitoring B chans and G.729 High Water Marks   George Pajari
• [Asterisk-Users] oh323 Zone   Michael Manousos
• [Asterisk-Users] SIP -> capi problem (no sound)   Cyrille Demaret
• [Asterisk-Users] Problem with X101P(Red Alarm)   Yusuf Iqbal
• [Asterisk-Users] (no subject)   Claude- Gaelle ONGBIL
• [Asterisk-Users] sip and analog   Claude- Gaelle ONGBIL
• [Asterisk-Users] H323 FAX   Mahmoud Badran
• [Asterisk-Users] Cisco 7960s and skinny   Derek Conniffe
• [Asterisk-Users] problem with skinny   Yusuf Iqbal
• [Asterisk-Users] Cisco 7960s and skinny   Mahmoud Badran
• [Asterisk-Users] Is There Media Accelerator For Better Asterisk Calls   chawki hammoud
• [Asterisk-Users] RSS feed Asterisk-Users   Sjaak Nabuurs
• [Asterisk-Users] Zaptel FXO crashing.   Andrew Kohlsmith
• [Asterisk-Users] Confused on G723 and G729   Matt
• [Asterisk-Users] Confused on G723 and G729   Matt
• [Asterisk-Users] Connection Timeout problem with SIP phones from Gnet   Jean-Francois Theroux
• [Asterisk-Users] Connection Timeout problem with SIP phones from Gnet   administrator tootai
• [Asterisk-Users] Newer Dell Servers + TDM card   Matt Schulte
• [Asterisk-Users] Connection Timeout problem with SIP phones from Gnet   Eric Wieling aka ManxPower
• [Asterisk-Users] IAX aproprietary protocol   Joseph
• [Asterisk-Users] Asterisk Agents   Kashif Anwar
• [Asterisk-Users] Connection Timeout problem with SIP phones from Gnet   Jean-Francois Theroux
• [Asterisk-Users] 800 number provider suggestions   Sean Kennedy
• [Asterisk-Users] IAX aproprietary protocol   Julio Arruda
• [Asterisk-Users] how to use dialparties.agi   Christian Wengel
• [Asterisk-Users] Advice on Adtran 600 setup   Chris Mason (Lists)
• [Asterisk-Users] Agents CallBackLogin and HangUp to calling party on pick-up   Peer Oliver Schmidt
• [Asterisk-Users] MGCP and CISCO 7960?   Sergio
• [Asterisk-Users] H323 FAX   Mahmoud Badran
• [Asterisk-Users] Realtime & voicemail   Edwin Horton
• [Asterisk-Users] asterisk-h.323   gale81 at virgilio.it
• [Asterisk-Users] Confused on G723 and G729   Pedro
• [Asterisk-Users] IAX aproprietary protocol   Eric Wieling aka ManxPower
• [Asterisk-Users] Cisco 7960s and skinny   Eric Wieling aka ManxPower
• [Asterisk-Users] RJ45 to RJ11?   Rich Adamson
• [Asterisk-Users] asterisk-h.323   Shaoul Jacobson - TELLINK
• [Asterisk-Users] Linksys/Cisco buys Sipura   Rich Adamson
• [Asterisk-Users] Confused on G723 and G729   Matt
• [Asterisk-Users] TDM400 doesn't know the hangup signal in china   Rich Adamson
• [Asterisk-Users] proper 2-card ISDN modem.conf configuration?   Tomasz Chmielewski
• [Asterisk-Users] Prefix to CALLING Number ?   barney
• [Asterisk-Users] Confused on G723 and G729   Adam Goryachev
• [Asterisk-Users] Cisco SIP Firmware Price Increase   Rich Adamson
• [Asterisk-Users] Experienced Asterisk Consultant in Chicago, IL   Jon Dahl
• [Asterisk-Users] Realtime & voicemail   Matthew Boehm
• [Asterisk-Users] Confused on G723 and G729   Matt
• [Asterisk-Users] Experienced Asterisk Consultant in Chicago, IL   Kerry Garrison
• [Asterisk-Users] IAX attempt -> Segmentation fault   Victor Alvarez
• [Asterisk-Users] Console Warning Message   Daniel Salama
• [Asterisk-Users] Queue member persistent stats   Kevin P. Fleming
• [Asterisk-Users] Queues configuration   Kevin P. Fleming
• [Asterisk-Users] start asterisk   Luz Lopez
• [Asterisk-Users] IAX attempt -> Segmentation fault   Eric Wieling aka ManxPower
• [Asterisk-Users] start asterisk   Jerry Geis
• [Asterisk-Users] Number of production asterisk systems   Christopher Jacob
• [Asterisk-Users] start asterisk   Robert Webb
• [Asterisk-Users] Call transfer   Henry Devito
• [Asterisk-Users] Delete voicemail   Damian Funnell
• [Asterisk-Users] start asterisk   Henry Devito
• [Asterisk-Users] Confused on G723 and G729   Rusty Shackleford
• [Asterisk-Users] BIND VoIP anyone?   Andres Paglayan
• [Asterisk-Users] Confused on G723 and G729   Dana Olson
• [Asterisk-Users] Delete voicemail   Wiley Siler
• [Asterisk-Users] asterisk-h.323   Neal Walton
• [Asterisk-Users] SIP calling Error from MP108 please help - confs included   iMRAN
• [Asterisk-Users] T1 Technology and VoIP Gateway Primer   Matt Roth
• [Asterisk-Users] Redirect two channels to each other?   Nicolás Gudiño
• [Asterisk-Users] asterisk call generator   Sam Njenga
• [Asterisk-Users] BIND VoIP anyone?   barney
• [Asterisk-Users] start asterisk   Luz Lopez
• [Asterisk-Users] VoicpulseConnect problems?   beonice
• [Asterisk-Users] Call transfer   Eric Wieling aka ManxPower
• [Asterisk-Users] SIP calling Error from MP108 please help - confs included   iMRAN
• [Asterisk-Users] T1 Technology and VoIP Gateway Primer   Michael Welter
• [Asterisk-Users] BIND VoIP anyone?   Max W Blackmer Jr
• [Asterisk-Users] Redirect two channels to each other?   Josiah Bryan
• [Asterisk-Users] ATA 186 MGCP Firmware   Stewart Nelson
• [Asterisk-Users] Web interface Suggestions   jamesm
• [Asterisk-Users] RE: Number of production asterisk systems (Christopher Jacob)   Greg Eaton
• [Asterisk-Users] Music on Hold can' t hear it!   Wiley Siler
• [Asterisk-Users] Music on Hold can' t hear it!   Eric Wieling aka ManxPower
• [Asterisk-Users] Polycom IP500 - Phone TIme   Wiley Siler
• [Asterisk-Users] Redirect two channels to each other?   mattf
• [Asterisk-Users] Prefix to CALLING Number ?   Josiah Bryan
• [Asterisk-Users] Polycom IP500 - Phone TIme   Eric Wieling aka ManxPower
• [Asterisk-Users] Polycom IP500 - Phone TIme   Doug Millsaps
• [Asterisk-Users] No audio playback   Robert Derr
• [Asterisk-Users] ftp.digium.com HTTP mirror, Digium's FTP server   Kristian Kielhofner
• [Asterisk-Users] Can no one help me with my Cisco 7920 problem :-(   Paul A Brown
• [Asterisk-Users] Polycom IP500 - Phone TIme   Wiley Siler
• [Asterisk-Users] help / advice needed on a project   Patrick Zwahlen
• [Asterisk-Users] Polycom IP500 - Phone TIme   Wiley Siler
• [Asterisk-Users] Polycom IP500 - Phone TIme   Marty Mastera
• [Asterisk-Users] Number of production asterisk systems   Chris Mason (Lists)
• [Asterisk-Users] Asterisk CVS and bristuff-0.2.0-RC8a-CVS: no callerid   td
• [Asterisk-Users] Music on Hold can' t hear it!   Robson Ribeiro
• [Asterisk-Users] Re: T1/DS1/ISDN PRI   David Josephson
• [Asterisk-Users] Web interface Suggestions   G.Marshall
• [Asterisk-Users] Zap/PRI: received AOC-E charging   Matt Fredrickson
• [Asterisk-Users] ftp.digium.com HTTP mirror, Digium's FTP server   Sean Milheim (iDREUS Corporation)
• [Asterisk-Users] ftp.digium.com HTTP mirror, Digium's FTP server   Bob Goddard
• [Asterisk-Users] ftp.digium.com HTTP mirror, Digium's FTP server   gARetH baBB
• [Asterisk-Users] CLI dial command   Martijn van Oosterhout
• [Asterisk-Users] Install Asterisk on CCM MCS-7835 Server   Walid Azab
• [Asterisk-Users] Polycom IP500 - Phone TIme   Sean Kennedy
• [Asterisk-Users] Call transfer   Henry Devito
• [Asterisk-Users] PRI ISDN NFAS configuration needed   Steve Edwards
• [Asterisk-Users] can asterisk send AT commands to a modem?   Tomasz Chmielewski
• [Asterisk-Users] RE: Number of production asterisk systems (Christopher Jacob)   Josiah Bryan
• [Asterisk-Users] Help to configure asterisk to dial to an PSTN line   Amit Singla
• [Asterisk-Users] Hints: What are they? How do they work?   Matthew Boehm
• [Asterisk-Users] Good FXO for UK use.   Razza
• [Asterisk-Users] Asterisk Home .9 with TDM11B   Kevin Lu
• [Asterisk-Users] Spandsp compile error   administrator tootai
• [Asterisk-Users] Help to configure asterisk to dial to an PSTN line   Tomasz Chmielewski
• [Asterisk-Users] Polycom IP500 - Phone TIme   Wiley Siler
• [Asterisk-Users] Install Asterisk on CCM MCS-7835 Server   Trey Scarborough
• [Asterisk-Users] Newer Dell Servers + TDM card   Duane Cox
• [Asterisk-Users] Prompts and MoH not working - AAH .09   Nuno Viegas
• [Asterisk-Users] Asterisk SIP sound issue   Jeff Ramsey
• [Asterisk-Users] Linksys/Cisco buys Sipura   David Boyd
• [Asterisk-Users] T1 Technology and VoIP Gateway Primer   Race Vanderdecken
• [Asterisk-Users] Console Warning Message   Andy Hamilton
• [Asterisk-Users] Console Warning Message   Daniel Salama
• [Asterisk-Users] music on hold on R key not working.   Eugenio De Vena
• [Asterisk-Users] Linksys/Cisco buys Sipura   Andres
• [Asterisk-Users] Linksys/Cisco buys Sipura   Guillermo Salas M
• [Asterisk-Users] Asterisk Home .9 with TDM11B   Sascha Ferley
• [Asterisk-Users] Asterisk Hardware Recommendation   Daniel Salama
• [Asterisk-Users] Linksys/Cisco buys Sipura   Brian Leyton
• [Asterisk-Users] Asterisk not paying attention to NAT Setting   Matthew Boehm
• [Asterisk-Users] Linksys/Cisco buys Sipura   Eric Wieling aka ManxPower
• [Asterisk-Users] Sipura SPA-841 and firewall   Chris Mason (Lists)
• [Asterisk-Users] Asterisk SIP sound issue   Jeff Ramsey
• [Asterisk-Users] Video Conferencing   Matthew Boehm
• [Asterisk-Users] Asterisk Hardware Recommendation   Michael D Schelin
• [Asterisk-Users] Asterisk Hardware Recommendation   mattf
• [Asterisk-Users] Gabled voice problem on Asterisk for two remote users   Donald Holloway
• [Asterisk-Users] T1 Technology and VoIP Gateway Primer   Matt Roth
• [Asterisk-Users] Asterisk Hardware Recommendation   Daniel Salama
• [Asterisk-Users] E1 legacy multi PBX integration?   magnus
• [Asterisk-Users] Asterisk Hardware Recommendation   Daniel Salama
• [Asterisk-Users] Asterisk Hardware Recommendation   Andres
• [Asterisk-Users] VoicpulseConnect problems?   Tim Burt
• [Asterisk-Users] Asterisk Hardware Recommendation   Matt Roth
• [Asterisk-Users] Monitoring B chans and G.729 High Water Marks   Matt Riddell
• [Asterisk-Users] ftp.digium.com HTTP mirror, Digium's FTP server   marek cervenka
• [Asterisk-Users] chan_capi crashes asterisk   Sebastian Voitzsch
• [Asterisk-Users] Monitoring B chans and G.729 High Water Marks   Andres
• [Asterisk-Users] Re: Re: T1/DS1/ISDN PRI   David Josephson
• [Asterisk-Users] Problems with TDM400P card   Anton Krall
• [Asterisk-Users] Re: Re: T1/DS1/ISDN PRI   Matt Klein
• [Asterisk-Users] Traffic Testing   mmiranda at americatel.com.sv
• [Asterisk-Users] Traffic Testing   Anton Krall
• [Asterisk-Users] Assigning DID and Extension with similar value   Nathaniel Angelo A. Torres (247talk)
• [Asterisk-Users] Traffic Testing   Anton Krall
• [Asterisk-Users] VoicpulseConnect problems?   beonice
• [Asterisk-Users] Re: Re: T1/DS1/ISDN PRI   Andrew Kohlsmith
• [Asterisk-Users] Traffic Testing   Anton Krall
• [Asterisk-Users] Asterisk Hardware Recommendation   mattf
• [Asterisk-Users] 800 number provider suggestions   Gregory Wiktor - ADCom Corp.
• [Asterisk-Users] voip connection problems   trixter http://www.0xdecafbad.com
• [Asterisk-Users] missing first digit when dial extension / dtmf problem ???   Joseph
• [Asterisk-Users] How do I add an IP to an Exten   Michael D Schelin
• [Asterisk-Users] Help to configure asterisk to dial to an PSTNline   Gregory Wiktor - ADCom Corp.
• [Asterisk-Users] Asterisk Hardware Recommendation   Daniel Salama
• [Asterisk-Users] missing first digit when dial extension / dtmfproblem ???   David Phelan
• [Asterisk-Users] Polycom IP500 - Phone TIme   Dan Morin
• [Asterisk-Users] missing first digit when dial extension / dtmfproblem ???   Joseph
• [Asterisk-Users] dtmfmode problem   raymond
• [Asterisk-Users] Polycom IP500 - Phone TIme   Dan Morin
• [Asterisk-Users] Assigning DID and Extension with similar value   Henry Devito
• [Asterisk-Users] missing first digit when dial extension / dtmf problem ???   snacktime
• [Asterisk-Users] UK (english) sound files (Paul R   Adam Goryachev
• [Asterisk-Users] How to prevent number of agents   Kashif Anwar
• [Asterisk-Users] Call Recording via monitor   Leo Ann Boon
• [Fwd: [Asterisk-Users] Voicemails stopping]   Chris Stinson
• [Asterisk-Users] Re: Asterisk-Users Digest, Vol 9, Issue 255   Justin Newman
• [Fwd: Re: [Fwd: [Asterisk-Users] Voicemails stopping]]   Chris Stinson
• [Asterisk-Users] How to prevent number of agents   Kashif Anwar
• [Asterisk-Users] How to Restrict Number of Lines   Daniel Salama
• [Asterisk-Users] vmail.cgi: -rwsr-sr-x as root *still* won't read the files   Brian Capouch
• [Asterisk-Users] vmail.cgi: -rwsr-sr-x as root *still* won't read the files   mike castleman
• [Asterisk-Users] Polycom IP500 - Phone TIme   Rick Baranowski
• [Fwd: [Asterisk-Users] Voicemails stopping]   Eric Wieling aka ManxPower
• [Asterisk-Users] missing first digit when dial extension / dtmf problem ???   Joseph
• [Asterisk-Users] codec introducing huge latency   Dinesh Nair
• [Asterisk-Users] Web interface Suggestions   Andrew Niemantsverdriet
• [Asterisk-Users] T1 Technology and VoIP Gateway Primer   Callum McGillivray
• [Asterisk-Users] vmail.cgi: -rwsr-sr-x as root *still* won't read the files   Brian Capouch
• [Asterisk-Users] voip connection problems   Brian Capouch
• [Asterisk-Users] Polycom IP500 - Phone TIme   Paul Hales
• [Asterisk-Users] Voicemail Broadcasts   Chris Stinson
• [Asterisk-Users] call a peer over the asterisk manager with a php script   Guy Boehm
• [Asterisk-Users] call a peer over the asterisk manager with a php script   Guy Boehm
• [Asterisk-Users] Pattern Matching   Mojo Jojo
• [Asterisk-Users] Re: Prefix to CALLING Number ?   Tony Mountifield
• [Asterisk-Users] first few seconds of call is lost   snacktime
• [Asterisk-Users] Asterisk Hardware Recommendation   Michael Welter
• [Asterisk-Users] Re: Prefix to CALLING Number ?   Tony Mountifield
• [Asterisk-Users] Some * scripts: Pull asterisk config from LDAP and authenticate() against voicemail passwords   Simon Morris
• [Asterisk-Users] Prefix to CALLING Number ?   barney
• [Asterisk-Users] Polycom IP500 - Phone TIme   Richard Scobie
• [Asterisk-Users] Re: Prefix to CALLING Number ?   Tony Mountifield
• [Asterisk-Users] (no subject)   deepak.dhiman at uf4.net
• [Asterisk-Users] how to share asterisk load with ser   deepak.dhiman at uf4.net
• [Asterisk-Users] how to share asterisk load with ser server   deepak.dhiman at uf4.net
• [Asterisk-Users] how to configure ser and asterisk together to share the load   deepak.dhiman at uf4.net
• [Asterisk-Users] Re: Prefix to CALLING Number ?   barney
• [Asterisk-Users] Major problems with TDM400 and specifictelephones: suggestions?   Soner Tari
• [Asterisk-Users] Barge In With Queues   usman at user.iphonica.net
• [Asterisk-Users] bri error   Altus Snyman
• [Asterisk-Users] bri error   David Masure
• [Asterisk-Users] bri error   David Masure
• [Asterisk-Users] bri error   Altus Snyman
• [Asterisk-Users] bri error   Altus Snyman
• [Asterisk-Users] IPSwitchBoard Version 0.110 Released   Thorben Jensen
• [Asterisk-Users] asterisk-oh323   gale81 at virgilio.it
• [Asterisk-Users] SIP Errors from MP108 please help - confs included   iMRAN
• [Asterisk-Users] Sipura SPA-841 and firewall   Chris Mason (Lists)
• [Asterisk-Users] IAX attempt -> Segmentation fault   Victor Alvarez
• [Asterisk-Users] Music on Hold can' t hear it!   Robson Ribeiro
• [Asterisk-Users] CALEA compliance (was voip connection problems)   Henry Devito
• [Asterisk-Users] DNID empty on incoming calls   Thomas Andrews
• [Asterisk-Users] Re: asterisk-oh323   Tony Mountifield
• [Asterisk-Users] Problems with TDM400P card   Anton Krall
• [Asterisk-Users] Cost field in Call Detail Records (cdr)   Brett, Gary
• [Asterisk-Users] recomended phone   Cesar Garcia
• [Asterisk-Users] Traffic Testing   Nils Ohlmeier
• [Asterisk-Users] missing first digit when dial extension / dtmf problem ???   Rich Adamson
• [Asterisk-Users] Realtime feature   Rodrigo P. Telles
• [Asterisk-Users] Problems with TDM400P card   Klaus Darilion
• [Asterisk-Users] T1 Technology and VoIP Gateway Primer   Adam Robins
• [Asterisk-Users] need help   Tim Touhsaent
• [Asterisk-Users] first few seconds of call is lost   Rich Adamson
• [Asterisk-Users] Queue Monitor Filename Problem   Dana Olson
• [Asterisk-Users] EuroISDN bearer capability pass thru from (fax) a/b adapter on OctoBRI to TE410P   Bruno.Voigt at ic3s.de
• [Asterisk-Users] Problems with TDM400P card   Robert Webb
• [Asterisk-Users] Fail over solutions   Nicolás Gudiño
• [Asterisk-Users] Major problems with TDM400 and specifictelephones: suggestions?   Rich Adamson
• [Asterisk-Users] Problems with TDM400P card   Anton Krall
• [Asterisk-Users] need help   igil at europesip.com
• [Asterisk-Users] need help   Ian Pattison
• [Asterisk-Users] Problems with TDM400P card   Robert Webb
• [Asterisk-Users] Problems with TDM400P card   Rich Adamson
• [Asterisk-Users] Asterisk on VMWare ESX/blade servers   Daryl G. Jurbala
• [Asterisk-Users] Problems with TDM400P card   John Novack
• [Asterisk-Users] Asterisk Hardware Recommendation   mattf
• [Asterisk-Users] Experienced Asterisk Consultant in Chicago, IL   Jon Dahl
• [Asterisk-Users] Major problems with TDM400 and specifictelephones: suggestions?   John Novack
• [Asterisk-Users] need help   Tim Touhsaent
• [Asterisk-Users] Recording in a call center   Steve Totaro
• [Asterisk-Users] Firefly Qualify Problem   David Choo
• [Asterisk-Users] vmail.cgi: -rwsr-sr-x as root *still* won't read the files   mike castleman
• [Asterisk-Users] quadbri bristuff ztcfg fail   Sander
• [Asterisk-Users] Recording in a call center   Dana Olson
• [Asterisk-Users] Recording in a call center   sjaak imap
• [Asterisk-Users] Queue Monitor Filename Problem   usman at user.iphonica.net
• [Asterisk-Users] voip connection problems   Mailing List
• [Asterisk-Users] quadbri bristuff ztcfg fail   Michael Bielicki
• [Asterisk-Users] chan_zap graceful failure   Jeb Campbell
• [Asterisk-Users] Web interface Suggestions   Dean Collins
• [Asterisk-Users] Asterisk Manager interface, setting global vars   Umar Sear
• [Asterisk-Users] Channel bank of E1s? (one E1 input --> 2 x E1 output)   ht at phonitel.com
• [Asterisk-Users] quadbri bristuff ztcfg fail   Sander
• [Asterisk-Users] Recording in a call center   Steve Totaro
• [Asterisk-Users] need help   Tim Touhsaent
• [Asterisk-Users] chan_zap graceful failure   Andrew Kohlsmith
• [Asterisk-Users] *@home 0.9 and AVM B1 Card   Jorge Marin
• [Asterisk-Users] Realtime feature   Matthew Boehm
• [Asterisk-Users] Problems with TDM400P card   Rich Adamson
• [Asterisk-Users] Asterisk on VMWare ESX/blade servers   itamar
• [Asterisk-Users] Channel bank of E1s? (one E1 input --> 2 x E1 output)   Matteo Brancaleoni
• [Asterisk-Users] the beginning of voice menu is cutted   Kib Eki
• [Asterisk-Users] the beginning of voice menu is cutted   Josiah Bryan
• [Asterisk-Users] Recording in a call center   Dana Olson
• [Asterisk-Users] Asterisk Hardware Recommendation   Daniel Salama
• [Asterisk-Users] chan_zap graceful failure   Jeb Campbell
• [Asterisk-Users] Asterisk Hardware Recommendation   Daniel Salama
• [Asterisk-Users] Problems with MusicOnHold   Nathan Bowyer
• [Asterisk-Users] Sip endpoints that support re-invite??   Hamza Moore
• [Asterisk-Users] chan_zap graceful failure   Andrew Kohlsmith
• [Asterisk-Users] IAX2 one way audio   geek at mnwifi.net
• [Asterisk-Users] User events - a dumb question   Asterisk
• [Asterisk-Users] Asterisk Manager interface, setting global vars   Johann
• [Asterisk-Users] Asterisk Hardware Recommendation   Steve Totaro
• [Asterisk-Users] IAX2 one way audio   Duane Cox
• [Asterisk-Users] call a peer over the asterisk manager with a php script   Richard Lyman
• [Asterisk-Users] Re: Re: T1/DS1/ISDN PRI   Matt Roth
• [Asterisk-Users] Traffic Testing   René Mayorga
• [Asterisk-Users] first few seconds of call is lost   snacktime
• [Asterisk-Users] IAX2 one way audio   Richard Lyman
• [Asterisk-Users] Group/Broadcast Voicemail   Chris Stinson
• [Asterisk-Users] Realtime feature   Rodrigo P. Telles
• [Asterisk-Users] IAX2 one way audio   Ken N. March
• [Asterisk-Users] Problems with TDM400P card   Anton Krall
• [Asterisk-Users] Asterisk Hardware Recommendation   Matt Roth
• [Asterisk-Users] Re: Sip endpoints that support re-invite??   Gene Willingham
• [Asterisk-Users] Asterisk Hardware Recommendation   Matt Roth
• [Asterisk-Users] Realtime feature   Matthew Boehm
• [Asterisk-Users] Adtran 600   Steve Totaro
• [Asterisk-Users] T1 Technology and VoIP Gateway Primer   Matt Roth
• [Asterisk-Users] SIP/IAX softphone with g729/723   M. Ehsanul Karim
• [Asterisk-Users] Asterisk Hardware Recommendation   Steve Totaro
• [Asterisk-Users] quadbri bristuff ztcfg fail   Kristof Hardy
• [Asterisk-Users] Channel bank of E1s? (one E1 input --> 2 x E1 output)   Julio Arruda
• [Asterisk-Users] Asterisk Hardware Recommendation   Ken N. March
• [Asterisk-Users] GR-303 zaptel and zapata configurations   Chris A. Icide
• [Asterisk-Users] Asterisk Hardware Architecture Group   Matt Roth
• [Asterisk-Users] chan_zap graceful failure   Jeb Campbell
• [Asterisk-Users] Problems with TDM400P card   Anton Krall
• [Asterisk-Users] Traffic Testing   Mailing List
• [Asterisk-Users] Traffic Testing   Anton Krall
• [Asterisk-Users] Channel bank of E1s? (one E1 input --> 2 x E1 output)   ht at phonitel.com
• [Asterisk-Users] Asterisk Hardware Recommendation   Daniel Salama
• [Asterisk-Users] Asterisk Hardware Architecture Group   Daniel Salama
• [Asterisk-Users] ISPCON: SIP CPE experts wanted for panel   John Todd
• [Asterisk-Users] Traffic Testing   Anton Krall
• [Asterisk-Users] Detecting DeadLocks   Paradise Dove
• [Asterisk-Users] User events - a dumb question   Thomas Miller
• [Asterisk-Users] Channel bank of E1s? (one E1 input --> 2 x E1 output)   Neal Walton
• [Asterisk-Users] Curious behaviour for pound (#) key with SIP X-lite SoftPhones   Jeff Stokoe
• [Asterisk-Users] Asterisk@Home 1.0 released   Asterisk at Home
• [Asterisk-Users] Problems with TDM400P card   Robert Webb
• [Asterisk-Users] Bouncing DTMF?   Jan Johansson
• [Asterisk-Users] chan_zap graceful failure   Andrew Kohlsmith
• [Asterisk-Users] Asterisk Hardware Architecture Group   snacktime
• [Asterisk-Users] More TDM questions....   Robert Webb
• [Asterisk-Users] txfax and Ghostscript 8.51   Me
• [Asterisk-Users] Asterisk Hardware Architecture Group   Daniel Salama
• [Asterisk-Users] Problems with TDM400P card   GEOFFREY SACHS
• [Asterisk-Users] Bouncing DTMF?   Brian McSpadden
• [Asterisk-Users] UTSTARCOM Wifi handset?   Michael Graves
• [Asterisk-Users] Paging and intercom   Jacob Cazzell
• [Asterisk-Users] Paging and intercom   Dean Collins
• [Asterisk-Users] Paging and intercom   Alexander Lopez
• [Asterisk-Users] Bouncing DTMF?   Henry Devito
• [Asterisk-Users] * and Sipgate (UK)   Rafal Kaniewski
• [Asterisk-Users] Paging and intercom   Henry Devito
• [Asterisk-Users] Caller-ID Block   Daniel Salama
• [Asterisk-Users] Problems with TDM400P card   John Novack
• [Asterisk-Users] Any workaround for long DISA timeout before it actually dials ?   Robert Rozman
• [Asterisk-Users] * and Sipgate (UK)   Paul Tyreman
• [Asterisk-Users] Asterisk Hardware Architecture Group   John Todd
• [Asterisk-Users] Asterisk Hardware Architecture Group   Race Vanderdecken
• [Asterisk-Users] Need info : lspci   Marco Supino
• [Asterisk-Users] Need info : lspci   Derek Whitten
• [Asterisk-Users] Need info : lspci   Mojo with Horan & Company, LLC
• [Asterisk-Users] UTSTARCOM Wifi handset?   Henry Devito
• [Asterisk-Users] Polycom IP500 Ringer Volume   John Harrison
• [Asterisk-Users] Caller-ID Block   Tim Litwiller
• [Asterisk-Users] Need info : lspci   Dan Perik
• [Asterisk-Users] Pattern Matching   Michael D Schelin
• [Asterisk-Users] Asterisk and sendmail   Chuck Keeter
• [Asterisk-Users] Can't get incoming calls with IAX trunks (FWD & Teliax)   Patrick Gray, Jr.
• [Asterisk-Users] Can't get incoming calls with IAX trunks (FWD &Teliax)   Henry Devito
• [Asterisk-Users] Asterisk and sendmail   Bill Ford
• [Asterisk-Users] Can't get incoming calls with IAX trunks (FWD &Teliax)   Dan Levine
• [Asterisk-Users] Caller-ID Block   Daniel Salama
• [Asterisk-Users] CallerID on cell phone   Lee Lee
• [Asterisk-Users] Call routing   Daniel Salama
• [Asterisk-Users] SER + Asterisk   Deon
• [Asterisk-Users] Caller-ID Block   Daniel Salama
• [Asterisk-Users] Asterisk Hardware Architecture Group   Dinesh Nair
• [Asterisk-Users] Asterisk and sendmail   Roger Hanson
• [Asterisk-Users] Caller-ID Block   lenz
• [Asterisk-Users] CID Number problem   Ma Zhiyong
• [Asterisk-Users] Re: Can't get incoming calls with IAX trunks (FWD & Teliax)   Iassen Hristov
• [Asterisk-Users] CID Number problem   Peter Svensson
• [Asterisk-Users] sipp example   Anton Krall
• [Asterisk-Users] Asterisk and sendmail   Craig Guy
• [Asterisk-Users] quadbri bristuff ztcfg fail   Florian Overkamp
• [Asterisk-Users] txfax and Ghostscript 8.51   Steve Underwood
• [Asterisk-Users] Asterisk Hardware Architecture Group   Nir Simionovich
• [Asterisk-Users] Caller-ID Block   Time Bandit
• [Asterisk-Users] Avaya 4610SW IP phone?   Walt Reed
• [Asterisk-Users] Dynamic phone groups.   Joris Vandalon
• [Asterisk-Users] Problems with TDM400P card   Kim Culhan
• [Asterisk-Users] Can't get incoming calls with IAX trunks (FWD&Teliax)   Patrick M. Gray, Jr.
• [Asterisk-Users] Can't get incoming calls with IAX trunks(FWD&Teliax)   Joshua Colp
• [Asterisk-Users] Problems with TDM400P card   Kim Culhan
• [Asterisk-Users] Can't get incoming calls with IAXtrunks(FWD&Teliax)   Patrick M. Gray, Jr.
• [Asterisk-Users] Dynamic phone groups.   Robert P. McKenzie
• [Asterisk-Users] Dynamic phone groups.   Joel Duffield
• [Asterisk-Users] Caller-ID Block   Tim Litwiller
• [Asterisk-Users] Problems with TDM400P card   Anton Krall
• [Asterisk-Users] Polycom IP500 Forward problem codec issue   Scott Herrick
• [Asterisk-Users] Problems with TDM400P card   Rich Adamson
• [Asterisk-Users] Cisco 7960s and skinny   Anton Krall
• [Asterisk-Users] Asterisk and sendmail   Derek Whitten
• [Asterisk-Users] Dynamic phone groups.   Peter Svensson
• [Asterisk-Users] Queues configuration   Anton Krall
• [Asterisk-Users] Queues configuration   Anton Krall
• [Asterisk-Users] Can't get incoming calls with IAX trunks (FWD&Teliax)   Rich Adamson
• [Asterisk-Users] Dynamic phone groups.   Joris Vandalon
• [Asterisk-Users] Problems with TDM400P card   Anton Krall
• [Asterisk-Users] IPSwitchBoard version 0.111 released   Thorben Jensen
• [Asterisk-Users] 7910 and Skinny   Mark Johnson
• [Asterisk-Users] Bouncing DTMF?   Jan Johansson
• [Asterisk-Users] Problems with TDM400P card   Rich Adamson
• [Asterisk-Users] IPSwitchBoard version 0.111 released   TC
• [Asterisk-Users] Re: chan_capi crashes asterisk   Stefan Tichy
• [Asterisk-Users] Problems with TDM400P card   Rich Adamson
• [Asterisk-Users] Re: IPSwitchBoard version 0.111 released   tgj
• [Asterisk-Users] Grandstream Budgetone 100 Caller ID shows extension, not incoming Caller ID   Amit Sharma
• [Asterisk-Users] Chan_modem_*   Marco Supino
• [Asterisk-Users] Intel 536EP   Jeff
• [Asterisk-Users] transfer pstn call to voip line, thus freeing up pstn line   Thomas Miller
• [Asterisk-Users] transfer pstn call to voip line, thus freeing up pstn line   Time Bandit
• [Asterisk-Users] help with compiling addons for cdr   forums
• [Asterisk-Users] Amp extensions script   Michael D Schelin
• [Asterisk-Users] Amp extensions script   Robert Webb
• [Asterisk-Users] transfer pstn call to voip line, thus freeing up pstn line   Robert Webb
• [Asterisk-Users] Cisco 7960s and skinny   Andy Hamilton
• [Asterisk-Users] Group/Broadcast Voicemail   Eric Wieling aka ManxPower
• [Asterisk-Users] transfer pstn call to voip line, thus freeing up pstn line   Joseph
• [Asterisk-Users] help with compiling addons for cdr   Matthew Boehm
• [Asterisk-Users] ANNOUNCEMENT: Asterisk-java 0.1 released   Stefan Reuter
• [Asterisk-Users] transfer pstn call to voip line, thus freeing up pstn line   Eric Wieling aka ManxPower
• [Asterisk-Users] Grandstream Budgetone 100 Caller ID shows extension, not incoming Caller ID   Brian Capouch
• [Asterisk-Users] Zaptel and Boostringer   Ian Pattison
• [Asterisk-Users] X-lite and * behind Firewalls   Julio Zavala
• [Asterisk-Users] A good SIP receptionist phone   Jason Brown
• [Asterisk-Users] A good SIP receptionist phone   Eric Wieling aka ManxPower
• [Asterisk-Users] A good SIP receptionist phone   Mike Dent
• [Asterisk-Users] A good SIP receptionist phone   Ron Wellsted
• [Asterisk-Users] A good SIP receptionist phone   Mike Clark
• [Asterisk-Users] A good SIP receptionist phone   Michael Welter
• [Asterisk-Users] A good SIP receptionist phone   Florian Overkamp
• [Asterisk-Users] A good SIP receptionist phone   Florian Overkamp
• [Asterisk-Users] Pattern Matching   Mojo Jojo
• [Asterisk-Users] Send DTMF *AFTER* channels are bridged   Shady
• [Asterisk-Users] Cisco 7960s and skinny   Anton Krall
• [Asterisk-Users] help with compiling addons for cdr   Jim Sturtevant
• [Asterisk-Users] Install Asterisk on CCM MCS-7835 Server   Walid Azab
• [Asterisk-Users] Hotel CDR Software   Anton Krall
• [Asterisk-Users] Call-park timeouts..   Tim Connolly
• [Asterisk-Users] A good SIP receptionist phone   Tim Connolly
• [Asterisk-Users] Pattern Matching   Tim Connolly
• [Asterisk-Users] A good SIP receptionist phone   Chris Mason (Lists)
• [Asterisk-Users] Asterisk on Radio Tonight   Kerry Garrison
• [Asterisk-Users] Problem with PSTN   Salina Jain
• [Asterisk-Users] Problem with Sangoma/Adtran 600 installation   Chris Mason (Lists)
• [Asterisk-Users] A good SIP receptionist phone   Rusty Shackleford
• [Asterisk-Users] SIP over IAX2   Daniel Salama
• [Asterisk-Users] Kernel 2.4 or 2.6   Daniel Salama
• [Asterisk-Users] budgetphone   Bert Haverkamp
• [Asterisk-Users] budgetphone   Michiel van Baak
• [Asterisk-Users] Kernel 2.4 or 2.6   Ian Pattison
• [Asterisk-Users] How to bridge 2 calls   Anton Krall
• [Asterisk-Users] SIP over IAX2   Daniel Salama
• [Asterisk-Users] Asterisk@Home bug   Mike
• [Asterisk-Users] Zaptel and Boostringer   Andrew Kohlsmith
• [Asterisk-Users] Problems with TDM400P card   Andrew Kohlsmith
• [Asterisk-Users] Programing a call forward feature to cel phones   Anton Krall
• [Asterisk-Users] SIP over IAX2   Tim Connolly
• [Asterisk-Users] Send DTMF *AFTER* channels are bridged   Matt Riddell
• [Asterisk-Users] Programing a call forward feature to cel phones   Matt Riddell
• [Asterisk-Users] Programing a call forward feature to cel phones   Anton Krall
• [Asterisk-Users] A good SIP receptionist phone   Sean Kennedy
• [Asterisk-Users] Programing a call forward feature to cel phones   Matt Riddell
• [Asterisk-Users] A good SIP receptionist phone   Gregory Wiktor - ADCom Corp.
• [Asterisk-Users] Programing a call forward feature to cel phones   Gregory Wiktor - ADCom Corp.
• [Asterisk-Users] How to bridge 2 calls   Gregory Wiktor - ADCom Corp.
• [Asterisk-Users] Programing a call forward feature to cel phones   Race Vanderdecken
• [Asterisk-Users] Programing a call forward feature to cel phones   Anton Krall
• [Asterisk-Users] Programing a call forward feature to cel phones   Anton Krall
• [Asterisk-Users] Programing a call forward feature to cel phones   Tim Connolly
• [Asterisk-Users] Programing a call forward feature to cel phones   Anton Krall
• [Asterisk-Users] Programing a call forward feature to cel phones   Anton Krall
• [Asterisk-Users] Programing a call forward feature to cel phones   Tim Connolly
• [Asterisk-Users] Programing a call forward feature to cel phones   Tim Connolly
• [Asterisk-Users] Problem with Sangoma/Adtran 600 installation   mattf
• [Asterisk-Users] Asterisk@Home bug   Manny A. Wise
• [Asterisk-Users] Programing a call forward feature to cel phones   Anton Krall
• [Asterisk-Users] Programing a call forward feature to cel phones   Anton Krall
• [Asterisk-Users] Programing a call forward feature to cel phones   Anton Krall
• [Asterisk-Users] Cisco 7960s and skinny   Julien Goodwin
• [Asterisk-Users] Asterisk on Radio Tonight   Jim Sturtevant
• [Asterisk-Users] SIP over IAX2   Daniel Salama
• [Asterisk-Users] Asterisk on Radio Tonight   Anton Krall
• [Asterisk-Users] Asterisk on Radio Tonight   Jim Sturtevant
• [Asterisk-Users] Re: Hotel CDR Software   tgj
• [Asterisk-Users] Asterisk on Radio Tonight   Anton Krall
• [Asterisk-Users] Asterisk@Home bug   Steve Prior
• [Asterisk-Users] Asterisk on Radio Tonight   Kerry Garrison
• [Asterisk-Users] Broadvoice limits???   Tim Connolly
• [Asterisk-Users] Send DTMF *AFTER* channels are bridged   Peter Svensson
• [Asterisk-Users] Broadvoice limits???   Kerry Garrison
• [Asterisk-Users] Asterisk on Radio Tonight   Anton Krall
• [Asterisk-Users] T1 Technology and VoIP Gateway Primer   Callum McGillivray
• [Asterisk-Users] Broadvoice limits???   trixter http://www.0xdecafbad.com
• [Asterisk-Users] Broadvoice limits???   Tim Connolly
• [Asterisk-Users] Broadvoice limits???   Tim Connolly